diff options
| author | Jocelyn Turcotte <jocelyn.turcotte@digia.com> | 2014-08-08 14:30:41 +0200 |
|---|---|---|
| committer | Jocelyn Turcotte <jocelyn.turcotte@digia.com> | 2014-08-12 13:49:54 +0200 |
| commit | ab0a50979b9eb4dfa3320eff7e187e41efedf7a9 (patch) | |
| tree | 498dfb8a97ff3361a9f7486863a52bb4e26bb898 /chromium/third_party/webrtc/modules | |
| parent | 4ce69f7403811819800e7c5ae1318b2647e778d1 (diff) | |
Update Chromium to beta version 37.0.2062.68
Change-Id: I188e3b5aff1bec75566014291b654eb19f5bc8ca
Reviewed-by: Andras Becsi <andras.becsi@digia.com>
Diffstat (limited to 'chromium/third_party/webrtc/modules')
720 files changed, 25353 insertions, 46891 deletions
diff --git a/chromium/third_party/webrtc/modules/OWNERS b/chromium/third_party/webrtc/modules/OWNERS new file mode 100644 index 00000000000..bbffda7e492 --- /dev/null +++ b/chromium/third_party/webrtc/modules/OWNERS @@ -0,0 +1,6 @@ +per-file *.isolate=kjellander@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c index 5c23f7ab71e..c66be2e484a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c @@ -67,9 +67,9 @@ int WebRtcIsacfix_EncHistMulti(Bitstr_enc *streamData, W_upper_LSB = W_upper & 0x0000FFFF; W_upper_MSB = WEBRTC_SPL_RSHIFT_W32(W_upper, 16); W_lower = WEBRTC_SPL_UMUL(W_upper_MSB, cdfLo); - W_lower += WEBRTC_SPL_UMUL_RSFT16(W_upper_LSB, cdfLo); + W_lower += ((W_upper_LSB * cdfLo) >> 16); W_upper = WEBRTC_SPL_UMUL(W_upper_MSB, cdfHi); - W_upper += WEBRTC_SPL_UMUL_RSFT16(W_upper_LSB, cdfHi); + W_upper += ((W_upper_LSB * cdfHi) >> 16); /* shift interval such that it begins at zero */ W_upper -= ++W_lower; diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c index b540ed5eef2..9391fb3c1d0 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c @@ -17,7 +17,6 @@ #include "arith_routins.h" - /* Tables for piecewise linear cdf functions: y = k*x */ /* x Points for function piecewise() in Q15 */ @@ -248,7 +247,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, int16_t envCount; uint16_t tmpARSpecQ8 = 0; int k, i; - + int offset = 0; /* point to beginning of stream buffer */ streamPtr = streamData->stream + streamData->stream_index; @@ -304,7 +303,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB); - W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB); + W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; if (streamVal > W_tmp) { @@ -313,7 +312,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB); - W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB); + W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; while (streamVal > W_tmp) { @@ -323,7 +322,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB); - W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB); + W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; /* error check */ if (W_lower == W_tmp) { @@ -342,7 +341,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB); - W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB); + W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; while ( !(streamVal > W_tmp) ) { @@ -352,7 +351,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB); - W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB); + W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; /* error check */ if (W_upper == W_tmp){ @@ -377,14 +376,27 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, * W_upper < 2^24 */ while ( !(W_upper & 0xFF000000) ) { - /* read next byte from stream */ - if (streamData->full == 0) { - streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | (*streamPtr++ & 0x00FF); - streamData->full = 1; + if (streamPtr < streamData->stream + streamData->stream_size) { + /* read next byte from stream */ + if (streamData->full == 0) { + streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | (*streamPtr++ & 0x00FF); + streamData->full = 1; + } else { + streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | + ((*streamPtr) >> 8); + streamData->full = 0; + } } else { - streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | - WEBRTC_SPL_RSHIFT_U16(*streamPtr, 8); - streamData->full = 0; + /* Intending to read outside the stream. This can happen for the last + * two or three bytes. It is how the algorithm is implemented. Do + * not read from the bit stream and insert zeros instead. */ + streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8); + if (streamData->full == 0) { + offset++; // We would have incremented the pointer in this case. + streamData->full = 1; + } else { + streamData->full = 0; + } } W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8); } @@ -392,7 +404,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, envCount++; } - streamData->stream_index = streamPtr - streamData->stream; + streamData->stream_index = streamPtr + offset - streamData->stream; streamData->W_upper = W_upper; streamData->streamval = streamVal; diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h index 88c7e1abe84..2f649324e77 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h @@ -179,6 +179,21 @@ void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0, int32_t* ptr2); #endif +#if defined(MIPS32_LE) +int WebRtcIsacfix_AutocorrMIPS(int32_t* __restrict r, + const int16_t* __restrict x, + int16_t N, + int16_t order, + int16_t* __restrict scale); + +void WebRtcIsacfix_FilterMaLoopMIPS(int16_t input0, + int16_t input1, + int32_t input2, + int32_t* ptr0, + int32_t* ptr1, + int32_t* ptr2); +#endif + /* Function pointers associated with the above functions. */ typedef int (*AutocorrFix)(int32_t* __restrict r, diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c index 6bccb8c8393..6bd5843cd97 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c @@ -326,7 +326,7 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16, for( k = 0; k < lag0; k++ ) { corr = WEBRTC_SPL_ADD_SAT_W32( corr, WEBRTC_SPL_ABS_W32( - WEBRTC_SPL_SUB_SAT_W16( + WebRtcSpl_SubSatW16( (ISACdec_obj->plcstr_obj).lastPitchLP[k], (ISACdec_obj->plcstr_obj).prevPitchInvIn[ FRAMESAMPLES_HALF - 2*lag0 - 10 + i + k ] ) ) ); diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c index e209c0ee59a..daf0d629993 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c @@ -15,18 +15,21 @@ * */ -#include "arith_routins.h" -#include "bandwidth_estimator.h" -#include "codec.h" -#include "pitch_gain_tables.h" -#include "pitch_lag_tables.h" -#include "entropy_coding.h" -#include "lpc_tables.h" -#include "lpc_masking_model.h" -#include "pitch_estimator.h" -#include "structs.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h" + +#include <assert.h> #include <stdio.h> +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h" + int WebRtcIsacfix_EncodeImpl(int16_t *in, ISACFIX_EncInst_t *ISACenc_obj, @@ -450,12 +453,14 @@ int WebRtcIsacfix_EncodeImpl(int16_t *in, while (stream_length < MinBytes) { + assert(stream_length >= 0); if (stream_length & 0x0001){ ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] |= (uint16_t)(ISACenc_obj->bitstr_seed & 0xFF); } else { ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); - ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] = WEBRTC_SPL_LSHIFT_U16(ISACenc_obj->bitstr_seed, 8); + ISACenc_obj->bitstr_obj.stream[stream_length / 2] = + ((uint16_t)ISACenc_obj->bitstr_seed << 8); } stream_length++; } @@ -467,7 +472,8 @@ int WebRtcIsacfix_EncodeImpl(int16_t *in, } else { ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF; - ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += WEBRTC_SPL_LSHIFT_U16((MinBytes - usefulstr_len) & 0x00FF, 8); + ISACenc_obj->bitstr_obj.stream[usefulstr_len >> 1] += + ((uint16_t)((MinBytes - usefulstr_len) & 0x00FF) << 8); } } else diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h index 28d10357245..3fefc1a5dcc 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h @@ -58,6 +58,17 @@ void WebRtcIsacfix_AllpassFilter2FixDec16Neon( int32_t *filter_state_ch2); #endif +#if defined(MIPS_DSP_R1_LE) +void WebRtcIsacfix_AllpassFilter2FixDec16MIPS( + int16_t *data_ch1, + int16_t *data_ch2, + const int16_t *factor_ch1, + const int16_t *factor_ch2, + const int length, + int32_t *filter_state_ch1, + int32_t *filter_state_ch2); +#endif + #if defined(__cplusplus) || defined(c_plusplus) } #endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c index 9c9d098aeef..64557e132d0 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c @@ -102,8 +102,8 @@ void WebRtcIsacfix_HighpassFilterFixDec32(int16_t *io, #ifdef WEBRTC_ARCH_ARM_V7 { - int tmp_coeff0 = 0; - int tmp_coeff1 = 0; + register int tmp_coeff0; + register int tmp_coeff1; __asm __volatile( "ldr %[tmp_coeff0], [%[coeff]]\n\t" "ldr %[tmp_coeff1], [%[coeff], #4]\n\t" @@ -113,12 +113,12 @@ void WebRtcIsacfix_HighpassFilterFixDec32(int16_t *io, "ldr %[tmp_coeff1], [%[coeff], #12]\n\t" "smmulr %[a1], %[tmp_coeff0], %[state0]\n\t" "smmulr %[b1], %[tmp_coeff1], %[state1]\n\t" - :[a2]"+r"(a2), - [b2]"+r"(b2), - [a1]"+r"(a1), - [b1]"+r"(b1), - [tmp_coeff0]"+r"(tmp_coeff0), - [tmp_coeff1]"+r"(tmp_coeff1) + :[a2]"=&r"(a2), + [b2]"=&r"(b2), + [a1]"=&r"(a1), + [b1]"=r"(b1), + [tmp_coeff0]"=&r"(tmp_coeff0), + [tmp_coeff1]"=&r"(tmp_coeff1) :[coeff]"r"(coefficient), [state0]"r"(state0), [state1]"r"(state1) diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c new file mode 100644 index 00000000000..1887745b7c0 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c @@ -0,0 +1,102 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h" + +// WebRtcIsacfix_AllpassFilter2FixDec16 function optimized for MIPSDSP platform +// Bit-exact with WebRtcIsacfix_AllpassFilter2FixDec16C from filterbanks.c +void WebRtcIsacfix_AllpassFilter2FixDec16MIPS( + int16_t *data_ch1, // Input and output in channel 1, in Q0 + int16_t *data_ch2, // Input and output in channel 2, in Q0 + const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15 + const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15 + const int length, // Length of the data buffers + int32_t *filter_state_ch1, // Filter state for channel 1, in Q16 + int32_t *filter_state_ch2) { // Filter state for channel 2, in Q16 + + int32_t st0_ch1, st1_ch1; // channel1 state variables + int32_t st0_ch2, st1_ch2; // channel2 state variables + int32_t f_ch10, f_ch11, f_ch20, f_ch21; // factor variables + int32_t r0, r1, r2, r3, r4, r5; // temporary ragister variables + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + // Load all the state and factor variables + "lh %[f_ch10], 0(%[factor_ch1]) \n\t" + "lh %[f_ch20], 0(%[factor_ch2]) \n\t" + "lh %[f_ch11], 2(%[factor_ch1]) \n\t" + "lh %[f_ch21], 2(%[factor_ch2]) \n\t" + "lw %[st0_ch1], 0(%[filter_state_ch1]) \n\t" + "lw %[st1_ch1], 4(%[filter_state_ch1]) \n\t" + "lw %[st0_ch2], 0(%[filter_state_ch2]) \n\t" + "lw %[st1_ch2], 4(%[filter_state_ch2]) \n\t" + // Allpass filtering loop + "1: \n\t" + "lh %[r0], 0(%[data_ch1]) \n\t" + "lh %[r1], 0(%[data_ch2]) \n\t" + "addiu %[length], %[length], -1 \n\t" + "mul %[r2], %[r0], %[f_ch10] \n\t" + "mul %[r3], %[r1], %[f_ch20] \n\t" + "sll %[r0], %[r0], 16 \n\t" + "sll %[r1], %[r1], 16 \n\t" + "sll %[r2], %[r2], 1 \n\t" + "addq_s.w %[r2], %[r2], %[st0_ch1] \n\t" + "sll %[r3], %[r3], 1 \n\t" + "addq_s.w %[r3], %[r3], %[st0_ch2] \n\t" + "sra %[r2], %[r2], 16 \n\t" + "mul %[st0_ch1], %[f_ch10], %[r2] \n\t" + "sra %[r3], %[r3], 16 \n\t" + "mul %[st0_ch2], %[f_ch20], %[r3] \n\t" + "mul %[r4], %[r2], %[f_ch11] \n\t" + "mul %[r5], %[r3], %[f_ch21] \n\t" + "sll %[st0_ch1], %[st0_ch1], 1 \n\t" + "subq_s.w %[st0_ch1], %[r0], %[st0_ch1] \n\t" + "sll %[st0_ch2], %[st0_ch2], 1 \n\t" + "subq_s.w %[st0_ch2], %[r1], %[st0_ch2] \n\t" + "sll %[r4], %[r4], 1 \n\t" + "addq_s.w %[r4], %[r4], %[st1_ch1] \n\t" + "sll %[r5], %[r5], 1 \n\t" + "addq_s.w %[r5], %[r5], %[st1_ch2] \n\t" + "sra %[r4], %[r4], 16 \n\t" + "mul %[r0], %[r4], %[f_ch11] \n\t" + "sra %[r5], %[r5], 16 \n\t" + "mul %[r1], %[r5], %[f_ch21] \n\t" + "sh %[r4], 0(%[data_ch1]) \n\t" + "sh %[r5], 0(%[data_ch2]) \n\t" + "addiu %[data_ch1], %[data_ch1], 2 \n\t" + "sll %[r2], %[r2], 16 \n\t" + "sll %[r0], %[r0], 1 \n\t" + "subq_s.w %[st1_ch1], %[r2], %[r0] \n\t" + "sll %[r3], %[r3], 16 \n\t" + "sll %[r1], %[r1], 1 \n\t" + "subq_s.w %[st1_ch2], %[r3], %[r1] \n\t" + "bgtz %[length], 1b \n\t" + " addiu %[data_ch2], %[data_ch2], 2 \n\t" + // Store channel states + "sw %[st0_ch1], 0(%[filter_state_ch1]) \n\t" + "sw %[st1_ch1], 4(%[filter_state_ch1]) \n\t" + "sw %[st0_ch2], 0(%[filter_state_ch2]) \n\t" + "sw %[st1_ch2], 4(%[filter_state_ch2]) \n\t" + ".set pop \n\t" + : [f_ch10] "=&r" (f_ch10), [f_ch20] "=&r" (f_ch20), + [f_ch11] "=&r" (f_ch11), [f_ch21] "=&r" (f_ch21), + [st0_ch1] "=&r" (st0_ch1), [st1_ch1] "=&r" (st1_ch1), + [st0_ch2] "=&r" (st0_ch2), [st1_ch2] "=&r" (st1_ch2), + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), + [r3] "=&r" (r3), [r4] "=&r" (r4), [r5] "=&r" (r5) + : [factor_ch1] "r" (factor_ch1), [factor_ch2] "r" (factor_ch2), + [filter_state_ch1] "r" (filter_state_ch1), + [filter_state_ch2] "r" (filter_state_ch2), + [data_ch1] "r" (data_ch1), [data_ch2] "r" (data_ch2), + [length] "r" (length) + : "memory", "hi", "lo" + ); +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_mips.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_mips.c new file mode 100644 index 00000000000..056dc275d39 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_mips.c @@ -0,0 +1,365 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h" + +// MIPS optimized implementation of the Autocorrelation function in fixed point. +// NOTE! Different from SPLIB-version in how it scales the signal. +int WebRtcIsacfix_AutocorrMIPS(int32_t* __restrict r, + const int16_t* __restrict x, + int16_t N, + int16_t order, + int16_t* __restrict scale) { + int i = 0; + int16_t scaling = 0; + int16_t* in = (int16_t*)x; + int loop_size = (int)(N >> 3); + int count = (int)(N & 7); + // Declare temporary variables used as registry values. + int32_t r0, r1, r2, r3; +#if !defined(MIPS_DSP_R2_LE) + // For non-DSPR2 optimizations 4 more registers are used. + int32_t r4, r5, r6, r7; +#endif + + // Calculate r[0] and scaling needed. + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "mult $0, $0 \n\t" + // Loop is unrolled 8 times, set accumulator to zero in branch delay slot. + "beqz %[loop_size], 2f \n\t" + " mult $0, $0 \n\t" + "1: \n\t" + // Load 8 samples per loop iteration. +#if defined(MIPS_DSP_R2_LE) + "ulw %[r0], 0(%[in]) \n\t" + "ulw %[r1], 4(%[in]) \n\t" + "ulw %[r2], 8(%[in]) \n\t" + "ulw %[r3], 12(%[in]) \n\t" +#else + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 2(%[in]) \n\t" + "lh %[r2], 4(%[in]) \n\t" + "lh %[r3], 6(%[in]) \n\t" + "lh %[r4], 8(%[in]) \n\t" + "lh %[r5], 10(%[in]) \n\t" + "lh %[r6], 12(%[in]) \n\t" + "lh %[r7], 14(%[in]) \n\t" +#endif + "addiu %[loop_size], %[loop_size], -1 \n\t" + // Multiply and accumulate. +#if defined(MIPS_DSP_R2_LE) + "dpa.w.ph $ac0, %[r0], %[r0] \n\t" + "dpa.w.ph $ac0, %[r1], %[r1] \n\t" + "dpa.w.ph $ac0, %[r2], %[r2] \n\t" + "dpa.w.ph $ac0, %[r3], %[r3] \n\t" +#else + "madd %[r0], %[r0] \n\t" + "madd %[r1], %[r1] \n\t" + "madd %[r2], %[r2] \n\t" + "madd %[r3], %[r3] \n\t" + "madd %[r4], %[r4] \n\t" + "madd %[r5], %[r5] \n\t" + "madd %[r6], %[r6] \n\t" + "madd %[r7], %[r7] \n\t" +#endif + "bnez %[loop_size], 1b \n\t" + " addiu %[in], %[in], 16 \n\t" + "2: \n\t" + "beqz %[count], 4f \n\t" +#if defined(MIPS_DSP_R1_LE) + " extr.w %[r0], $ac0, 31 \n\t" +#else + " mfhi %[r2] \n\t" +#endif + // Process remaining samples (if any). + "3: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "addiu %[count], %[count], -1 \n\t" + "madd %[r0], %[r0] \n\t" + "bnez %[count], 3b \n\t" + " addiu %[in], %[in], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "extr.w %[r0], $ac0, 31 \n\t" +#else + "mfhi %[r2] \n\t" +#endif + "4: \n\t" +#if !defined(MIPS_DSP_R1_LE) + "mflo %[r3] \n\t" + "sll %[r0], %[r2], 1 \n\t" + "srl %[r1], %[r3], 31 \n\t" + "addu %[r0], %[r0], %[r1] \n\t" +#endif + // Calculate scaling (the value of shifting). + "clz %[r1], %[r0] \n\t" + "addiu %[r1], %[r1], -32 \n\t" + "subu %[scaling], $0, %[r1] \n\t" + "slti %[r1], %[r0], 0x1 \n\t" + "movn %[scaling], $0, %[r1] \n\t" +#if defined(MIPS_DSP_R1_LE) + "extrv.w %[r0], $ac0, %[scaling] \n\t" + "mfhi %[r2], $ac0 \n\t" +#else + "addiu %[r1], %[scaling], -32 \n\t" + "subu %[r1], $0, %[r1] \n\t" + "sllv %[r1], %[r2], %[r1] \n\t" + "srlv %[r0], %[r3], %[scaling] \n\t" + "addu %[r0], %[r0], %[r1] \n\t" +#endif + "slti %[r1], %[scaling], 32 \n\t" + "movz %[r0], %[r2], %[r1] \n\t" + ".set pop \n\t" + : [loop_size] "+r" (loop_size), [in] "+r" (in), [r0] "=&r" (r0), + [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), +#if !defined(MIPS_DSP_R2_LE) + [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), +#endif + [count] "+r" (count), [scaling] "=r" (scaling) + : [N] "r" (N) + : "memory", "hi", "lo" + ); + r[0] = r0; + + // Correlation calculation is divided in 3 cases depending on the scaling + // value (different accumulator manipulation needed). Three slightly different + // loops are written in order to avoid branches inside the loop. + if (scaling == 0) { + // In this case, the result will be in low part of the accumulator. + for (i = 1; i < order + 1; i++) { + in = (int16_t*)x; + int16_t* in1 = (int16_t*)x + i; + count = N - i; + loop_size = (count) >> 2; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "mult $0, $0 \n\t" + "beqz %[loop_size], 2f \n\t" + " andi %[count], %[count], 0x3 \n\t" + // Loop processing 4 pairs of samples per iteration. + "1: \n\t" +#if defined(MIPS_DSP_R2_LE) + "ulw %[r0], 0(%[in]) \n\t" + "ulw %[r1], 0(%[in1]) \n\t" + "ulw %[r2], 4(%[in]) \n\t" + "ulw %[r3], 4(%[in1]) \n\t" +#else + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 0(%[in1]) \n\t" + "lh %[r2], 2(%[in]) \n\t" + "lh %[r3], 2(%[in1]) \n\t" + "lh %[r4], 4(%[in]) \n\t" + "lh %[r5], 4(%[in1]) \n\t" + "lh %[r6], 6(%[in]) \n\t" + "lh %[r7], 6(%[in1]) \n\t" +#endif + "addiu %[loop_size], %[loop_size], -1 \n\t" +#if defined(MIPS_DSP_R2_LE) + "dpa.w.ph $ac0, %[r0], %[r1] \n\t" + "dpa.w.ph $ac0, %[r2], %[r3] \n\t" +#else + "madd %[r0], %[r1] \n\t" + "madd %[r2], %[r3] \n\t" + "madd %[r4], %[r5] \n\t" + "madd %[r6], %[r7] \n\t" +#endif + "addiu %[in], %[in], 8 \n\t" + "bnez %[loop_size], 1b \n\t" + " addiu %[in1], %[in1], 8 \n\t" + "2: \n\t" + "beqz %[count], 4f \n\t" + " mflo %[r0] \n\t" + // Process remaining samples (if any). + "3: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 0(%[in1]) \n\t" + "addiu %[count], %[count], -1 \n\t" + "addiu %[in], %[in], 2 \n\t" + "madd %[r0], %[r1] \n\t" + "bnez %[count], 3b \n\t" + " addiu %[in1], %[in1], 2 \n\t" + "mflo %[r0] \n\t" + "4: \n\t" + ".set pop \n\t" + : [loop_size] "+r" (loop_size), [in] "+r" (in), [in1] "+r" (in1), +#if !defined(MIPS_DSP_R2_LE) + [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), +#endif + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), + [count] "+r" (count) + : + : "memory", "hi", "lo" + ); + r[i] = r0; + } + } else if (scaling == 32) { + // In this case, the result will be high part of the accumulator. + for (i = 1; i < order + 1; i++) { + in = (int16_t*)x; + int16_t* in1 = (int16_t*)x + i; + count = N - i; + loop_size = (count) >> 2; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "mult $0, $0 \n\t" + "beqz %[loop_size], 2f \n\t" + " andi %[count], %[count], 0x3 \n\t" + // Loop processing 4 pairs of samples per iteration. + "1: \n\t" +#if defined(MIPS_DSP_R2_LE) + "ulw %[r0], 0(%[in]) \n\t" + "ulw %[r1], 0(%[in1]) \n\t" + "ulw %[r2], 4(%[in]) \n\t" + "ulw %[r3], 4(%[in1]) \n\t" +#else + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 0(%[in1]) \n\t" + "lh %[r2], 2(%[in]) \n\t" + "lh %[r3], 2(%[in1]) \n\t" + "lh %[r4], 4(%[in]) \n\t" + "lh %[r5], 4(%[in1]) \n\t" + "lh %[r6], 6(%[in]) \n\t" + "lh %[r7], 6(%[in1]) \n\t" +#endif + "addiu %[loop_size], %[loop_size], -1 \n\t" +#if defined(MIPS_DSP_R2_LE) + "dpa.w.ph $ac0, %[r0], %[r1] \n\t" + "dpa.w.ph $ac0, %[r2], %[r3] \n\t" +#else + "madd %[r0], %[r1] \n\t" + "madd %[r2], %[r3] \n\t" + "madd %[r4], %[r5] \n\t" + "madd %[r6], %[r7] \n\t" +#endif + "addiu %[in], %[in], 8 \n\t" + "bnez %[loop_size], 1b \n\t" + " addiu %[in1], %[in1], 8 \n\t" + "2: \n\t" + "beqz %[count], 4f \n\t" + " mfhi %[r0] \n\t" + // Process remaining samples (if any). + "3: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 0(%[in1]) \n\t" + "addiu %[count], %[count], -1 \n\t" + "addiu %[in], %[in], 2 \n\t" + "madd %[r0], %[r1] \n\t" + "bnez %[count], 3b \n\t" + " addiu %[in1], %[in1], 2 \n\t" + "mfhi %[r0] \n\t" + "4: \n\t" + ".set pop \n\t" + : [loop_size] "+r" (loop_size), [in] "+r" (in), [in1] "+r" (in1), +#if !defined(MIPS_DSP_R2_LE) + [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), +#endif + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), + [count] "+r" (count) + : + : "memory", "hi", "lo" + ); + r[i] = r0; + } + } else { + // In this case, the result is obtained by combining low and high parts + // of the accumulator. +#if !defined(MIPS_DSP_R1_LE) + int32_t tmp_shift = 32 - scaling; +#endif + for (i = 1; i < order + 1; i++) { + in = (int16_t*)x; + int16_t* in1 = (int16_t*)x + i; + count = N - i; + loop_size = (count) >> 2; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "mult $0, $0 \n\t" + "beqz %[loop_size], 2f \n\t" + " andi %[count], %[count], 0x3 \n\t" + "1: \n\t" +#if defined(MIPS_DSP_R2_LE) + "ulw %[r0], 0(%[in]) \n\t" + "ulw %[r1], 0(%[in1]) \n\t" + "ulw %[r2], 4(%[in]) \n\t" + "ulw %[r3], 4(%[in1]) \n\t" +#else + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 0(%[in1]) \n\t" + "lh %[r2], 2(%[in]) \n\t" + "lh %[r3], 2(%[in1]) \n\t" + "lh %[r4], 4(%[in]) \n\t" + "lh %[r5], 4(%[in1]) \n\t" + "lh %[r6], 6(%[in]) \n\t" + "lh %[r7], 6(%[in1]) \n\t" +#endif + "addiu %[loop_size], %[loop_size], -1 \n\t" +#if defined(MIPS_DSP_R2_LE) + "dpa.w.ph $ac0, %[r0], %[r1] \n\t" + "dpa.w.ph $ac0, %[r2], %[r3] \n\t" +#else + "madd %[r0], %[r1] \n\t" + "madd %[r2], %[r3] \n\t" + "madd %[r4], %[r5] \n\t" + "madd %[r6], %[r7] \n\t" +#endif + "addiu %[in], %[in], 8 \n\t" + "bnez %[loop_size], 1b \n\t" + " addiu %[in1], %[in1], 8 \n\t" + "2: \n\t" + "beqz %[count], 4f \n\t" +#if defined(MIPS_DSP_R1_LE) + " extrv.w %[r0], $ac0, %[scaling] \n\t" +#else + " mfhi %[r0] \n\t" +#endif + "3: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 0(%[in1]) \n\t" + "addiu %[count], %[count], -1 \n\t" + "addiu %[in], %[in], 2 \n\t" + "madd %[r0], %[r1] \n\t" + "bnez %[count], 3b \n\t" + " addiu %[in1], %[in1], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "extrv.w %[r0], $ac0, %[scaling] \n\t" +#else + "mfhi %[r0] \n\t" +#endif + "4: \n\t" +#if !defined(MIPS_DSP_R1_LE) + "mflo %[r1] \n\t" + "sllv %[r0], %[r0], %[tmp_shift] \n\t" + "srlv %[r1], %[r1], %[scaling] \n\t" + "addu %[r0], %[r0], %[r1] \n\t" +#endif + ".set pop \n\t" + : [loop_size] "+r" (loop_size), [in] "+r" (in), [in1] "+r" (in1), +#if !defined(MIPS_DSP_R2_LE) + [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), +#endif + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), + [count] "+r" (count) + : [scaling] "r" (scaling) +#if !defined(MIPS_DSP_R1_LE) + , [tmp_shift] "r" (tmp_shift) +#endif + : "memory", "hi", "lo" + ); + r[i] = r0; + } + } + *scale = scaling; + + return (order + 1); +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c index 8baa30738f6..7635908094e 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c @@ -179,7 +179,7 @@ int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst) } /**************************************************************************** - * WebRtcAecm_InitNeon(...) + * WebRtcIsacfix_InitNeon(...) * * This function initializes function pointers for ARM Neon platform. */ @@ -200,6 +200,23 @@ static void WebRtcIsacfix_InitNeon(void) { #endif /**************************************************************************** + * WebRtcIsacfix_InitMIPS(...) + * + * This function initializes function pointers for MIPS platform. + */ + +#if defined(MIPS32_LE) +static void WebRtcIsacfix_InitMIPS(void) { + WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrMIPS; + WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopMIPS; +#if defined(MIPS_DSP_R1_LE) + WebRtcIsacfix_AllpassFilter2FixDec16 = + WebRtcIsacfix_AllpassFilter2FixDec16MIPS; +#endif +} +#endif + +/**************************************************************************** * WebRtcIsacfix_EncoderInit(...) * * This function initializes a ISAC instance prior to the encoder calls. @@ -296,6 +313,10 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst, WebRtcIsacfix_InitNeon(); #endif +#if defined(MIPS32_LE) + WebRtcIsacfix_InitMIPS(); +#endif + return statusInit; } @@ -587,15 +608,11 @@ int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst, { ISACFIX_SubStruct *ISAC_inst; Bitstr_dec streamdata; - uint16_t partOfStream[5]; #ifndef WEBRTC_ARCH_BIG_ENDIAN int k; #endif int16_t err; - /* Set stream pointer to point at partOfStream */ - streamdata.stream = (uint16_t *)partOfStream; - /* typecast pointer to real structure */ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; @@ -675,15 +692,11 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst, { ISACFIX_SubStruct *ISAC_inst; Bitstr_dec streamdata; - uint16_t partOfStream[5]; #ifndef WEBRTC_ARCH_BIG_ENDIAN int k; #endif int16_t err; - /* Set stream pointer to point at partOfStream */ - streamdata.stream = (uint16_t *)partOfStream; - /* typecast pointer to real structure */ ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; @@ -790,7 +803,7 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst, return -1; } - (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (uint16_t *)encoded; + ISAC_inst->ISACdec_obj.bitstr_obj.stream_size = (len + 1) >> 1; /* convert bitstream from int16_t to bytes */ #ifndef WEBRTC_ARCH_BIG_ENDIAN @@ -891,7 +904,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst, return -1; } - (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (uint16_t *)encoded; + ISAC_inst->ISACdec_obj.bitstr_obj.stream_size = (len + 1) >> 1; /* convert bitstream from int16_t to bytes */ #ifndef WEBRTC_ARCH_BIG_ENDIAN @@ -1266,15 +1279,11 @@ int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded, int16_t* frameLength) { Bitstr_dec streamdata; - uint16_t partOfStream[5]; #ifndef WEBRTC_ARCH_BIG_ENDIAN int k; #endif int16_t err; - /* Set stream pointer to point at partOfStream */ - streamdata.stream = (uint16_t *)partOfStream; - streamdata.W_upper = 0xFFFFFFFF; streamdata.streamval = 0; streamdata.stream_index = 0; @@ -1315,15 +1324,11 @@ int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded, int16_t* rateIndex) { Bitstr_dec streamdata; - uint16_t partOfStream[5]; #ifndef WEBRTC_ARCH_BIG_ENDIAN int k; #endif int16_t err; - /* Set stream pointer to point at partOfStream */ - streamdata.stream = (uint16_t *)partOfStream; - streamdata.W_upper = 0xFFFFFFFF; streamdata.streamval = 0; streamdata.stream_index = 0; diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi index 87c98606a11..a18a803d659 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi @@ -85,6 +85,30 @@ 'pitch_filter_c.c', ], }], + ['target_arch=="mipsel"', { + 'sources': [ + 'filters_mips.c', + 'lattice_mips.c', + ], + 'sources!': [ + 'lattice_c.c', + ], + 'conditions': [ + ['mips_dsp_rev>0', { + 'sources': [ + 'filterbanks_mips.c', + ], + }], + ['mips_dsp_rev>1', { + 'sources': [ + 'pitch_filter_mips.c', + ], + 'sources!': [ + 'pitch_filter_c.c', + ], + }], + ], + }], ], }, ], diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c new file mode 100644 index 00000000000..c596922168e --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c @@ -0,0 +1,327 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h" +#include "webrtc/typedefs.h" + +// Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients +// cth_Q15[] and sth_Q15[]. +void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples + int16_t* ar_f_Q0, // Input samples + int16_t* cth_Q15, // Filter coefficients + int16_t* sth_Q15, // Filter coefficients + int16_t order_coef) { // order of the filter + int n = 0; + + for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) { + int count = order_coef - 1; + int offset; +#if !defined(MIPS_DSP_R1_LE) + int16_t* tmp_cth; + int16_t* tmp_sth; + int16_t* tmp_arg; + int32_t max_q16 = 0x7fff; + int32_t min_q16 = 0xffff8000; +#endif + // Declare variables used as temporary registers. + int32_t r0, r1, r2, t0, t1, t2, t_ar; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "bltz %[count], 2f \n\t" + " lh %[t_ar], 0(%[tmp]) \n\t" + // Inner loop + "1: \n\t" + "sll %[offset], %[count], 1 \n\t" +#if defined(MIPS_DSP_R1_LE) + "lhx %[r0], %[offset](%[cth_Q15]) \n\t" + "lhx %[r1], %[offset](%[sth_Q15]) \n\t" + "lhx %[r2], %[offset](%[ar_g_Q0]) \n\t" +#else + "addu %[tmp_cth], %[cth_Q15], %[offset] \n\t" + "addu %[tmp_sth], %[sth_Q15], %[offset] \n\t" + "addu %[tmp_arg], %[ar_g_Q0], %[offset] \n\t" + "lh %[r0], 0(%[tmp_cth]) \n\t" + "lh %[r1], 0(%[tmp_sth]) \n\t" + "lh %[r2], 0(%[tmp_arg]) \n\t" +#endif + "mul %[t0], %[r0], %[t_ar] \n\t" + "mul %[t1], %[r1], %[t_ar] \n\t" + "mul %[t2], %[r1], %[r2] \n\t" + "mul %[r0], %[r0], %[r2] \n\t" + "subu %[t0], %[t0], %[t2] \n\t" + "addu %[t1], %[t1], %[r0] \n\t" +#if defined(MIPS_DSP_R1_LE) + "shra_r.w %[t1], %[t1], 15 \n\t" + "shra_r.w %[t0], %[t0], 15 \n\t" +#else + "addiu %[t1], %[t1], 0x4000 \n\t" + "sra %[t1], %[t1], 15 \n\t" + "addiu %[t0], %[t0], 0x4000 \n\t" + "sra %[t0], %[t0], 15 \n\t" +#endif + "addiu %[offset], %[offset], 2 \n\t" +#if defined(MIPS_DSP_R1_LE) + "shll_s.w %[t1], %[t1], 16 \n\t" + "shll_s.w %[t_ar], %[t0], 16 \n\t" +#else + "slt %[r0], %[t1], %[max_q16] \n\t" + "slt %[r1], %[t0], %[max_q16] \n\t" + "movz %[t1], %[max_q16], %[r0] \n\t" + "movz %[t0], %[max_q16], %[r1] \n\t" +#endif + "addu %[offset], %[offset], %[ar_g_Q0] \n\t" +#if defined(MIPS_DSP_R1_LE) + "sra %[t1], %[t1], 16 \n\t" + "sra %[t_ar], %[t_ar], 16 \n\t" +#else + "slt %[r0], %[t1], %[min_q16] \n\t" + "slt %[r1], %[t0], %[min_q16] \n\t" + "movn %[t1], %[min_q16], %[r0] \n\t" + "movn %[t0], %[min_q16], %[r1] \n\t" + "addu %[t_ar], $zero, %[t0] \n\t" +#endif + "sh %[t1], 0(%[offset]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[count], %[count], -1 \n\t" + "2: \n\t" + "sh %[t_ar], 0(%[tmp]) \n\t" + "sh %[t_ar], 0(%[ar_g_Q0]) \n\t" + ".set pop \n\t" + : [t_ar] "=&r" (t_ar), [count] "+r" (count), [offset] "=&r" (offset), + [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [t0] "=&r" (t0), +#if !defined(MIPS_DSP_R1_LE) + [tmp_cth] "=&r" (tmp_cth), [tmp_sth] "=&r" (tmp_sth), + [tmp_arg] "=&r" (tmp_arg), +#endif + [t1] "=&r" (t1), [t2] "=&r" (t2) + : [tmp] "r" (&ar_f_Q0[n+1]), [cth_Q15] "r" (cth_Q15), +#if !defined(MIPS_DSP_R1_LE) + [max_q16] "r" (max_q16), [min_q16] "r" (min_q16), +#endif + [sth_Q15] "r" (sth_Q15), [ar_g_Q0] "r" (ar_g_Q0) + : "memory", "hi", "lo" + ); + } +} + +// MIPS optimization of the inner loop used for function +// WebRtcIsacfix_NormLatticeFilterMa(). It does: +// +// for 0 <= n < HALF_SUBFRAMELEN - 1: +// *ptr2 = input2 * (*ptr2) + input0 * (*ptr0)); +// *ptr1 = input1 * (*ptr0) + input0 * (*ptr2); +// +// Note, function WebRtcIsacfix_FilterMaLoopMIPS and WebRtcIsacfix_FilterMaLoopC +// are not bit-exact. The accuracy of the MIPS function is same or better. +void WebRtcIsacfix_FilterMaLoopMIPS(int16_t input0, // Filter coefficient + int16_t input1, // Filter coefficient + int32_t input2, // Inverse coeff (1/input1) + int32_t* ptr0, // Sample buffer + int32_t* ptr1, // Sample buffer + int32_t* ptr2) { // Sample buffer +#if defined(MIPS_DSP_R2_LE) + // MIPS DSPR2 version. 4 available accumulators allows loop unrolling 4 times. + // This variant is not bit-exact with WebRtcIsacfix_FilterMaLoopC, since we + // are exploiting 64-bit accumulators. The accuracy of the MIPS DSPR2 function + // is same or better. + int n = (HALF_SUBFRAMELEN - 1) >> 2; + int m = (HALF_SUBFRAMELEN - 1) & 3; + + int r0, r1, r2, r3; + int t0, t1, t2, t3; + int s0, s1, s2, s3; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "lw %[r0], 0(%[ptr0]) \n\t" + "lw %[r1], 4(%[ptr0]) \n\t" + "lw %[r2], 8(%[ptr0]) \n\t" + "lw %[r3], 12(%[ptr0]) \n\t" + "mult $ac0, %[r0], %[input0] \n\t" + "mult $ac1, %[r1], %[input0] \n\t" + "mult $ac2, %[r2], %[input0] \n\t" + "mult $ac3, %[r3], %[input0] \n\t" + "lw %[t0], 0(%[ptr2]) \n\t" + "extr_rs.w %[s0], $ac0, 15 \n\t" + "extr_rs.w %[s1], $ac1, 15 \n\t" + "extr_rs.w %[s2], $ac2, 15 \n\t" + "extr_rs.w %[s3], $ac3, 15 \n\t" + "lw %[t1], 4(%[ptr2]) \n\t" + "lw %[t2], 8(%[ptr2]) \n\t" + "lw %[t3], 12(%[ptr2]) \n\t" + "addu %[t0], %[t0], %[s0] \n\t" + "addu %[t1], %[t1], %[s1] \n\t" + "addu %[t2], %[t2], %[s2] \n\t" + "addu %[t3], %[t3], %[s3] \n\t" + "mult $ac0, %[t0], %[input2] \n\t" + "mult $ac1, %[t1], %[input2] \n\t" + "mult $ac2, %[t2], %[input2] \n\t" + "mult $ac3, %[t3], %[input2] \n\t" + "addiu %[ptr0], %[ptr0], 16 \n\t" + "extr_rs.w %[t0], $ac0, 16 \n\t" + "extr_rs.w %[t1], $ac1, 16 \n\t" + "extr_rs.w %[t2], $ac2, 16 \n\t" + "extr_rs.w %[t3], $ac3, 16 \n\t" + "addiu %[n], %[n], -1 \n\t" + "mult $ac0, %[r0], %[input1] \n\t" + "mult $ac1, %[r1], %[input1] \n\t" + "mult $ac2, %[r2], %[input1] \n\t" + "mult $ac3, %[r3], %[input1] \n\t" + "sw %[t0], 0(%[ptr2]) \n\t" + "extr_rs.w %[s0], $ac0, 15 \n\t" + "extr_rs.w %[s1], $ac1, 15 \n\t" + "extr_rs.w %[s2], $ac2, 15 \n\t" + "extr_rs.w %[s3], $ac3, 15 \n\t" + "sw %[t1], 4(%[ptr2]) \n\t" + "sw %[t2], 8(%[ptr2]) \n\t" + "sw %[t3], 12(%[ptr2]) \n\t" + "mult $ac0, %[t0], %[input0] \n\t" + "mult $ac1, %[t1], %[input0] \n\t" + "mult $ac2, %[t2], %[input0] \n\t" + "mult $ac3, %[t3], %[input0] \n\t" + "addiu %[ptr2], %[ptr2], 16 \n\t" + "extr_rs.w %[t0], $ac0, 15 \n\t" + "extr_rs.w %[t1], $ac1, 15 \n\t" + "extr_rs.w %[t2], $ac2, 15 \n\t" + "extr_rs.w %[t3], $ac3, 15 \n\t" + "addu %[t0], %[t0], %[s0] \n\t" + "addu %[t1], %[t1], %[s1] \n\t" + "addu %[t2], %[t2], %[s2] \n\t" + "addu %[t3], %[t3], %[s3] \n\t" + "sw %[t0], 0(%[ptr1]) \n\t" + "sw %[t1], 4(%[ptr1]) \n\t" + "sw %[t2], 8(%[ptr1]) \n\t" + "sw %[t3], 12(%[ptr1]) \n\t" + "bgtz %[n], 1b \n\t" + " addiu %[ptr1], %[ptr1], 16 \n\t" + "beq %[m], %0, 3f \n\t" + " nop \n\t" + "2: \n\t" + "lw %[r0], 0(%[ptr0]) \n\t" + "lw %[t0], 0(%[ptr2]) \n\t" + "addiu %[ptr0], %[ptr0], 4 \n\t" + "mult $ac0, %[r0], %[input0] \n\t" + "mult $ac1, %[r0], %[input1] \n\t" + "extr_rs.w %[r1], $ac0, 15 \n\t" + "extr_rs.w %[t1], $ac1, 15 \n\t" + "addu %[t0], %[t0], %[r1] \n\t" + "mult $ac0, %[t0], %[input2] \n\t" + "extr_rs.w %[t0], $ac0, 16 \n\t" + "sw %[t0], 0(%[ptr2]) \n\t" + "mult $ac0, %[t0], %[input0] \n\t" + "addiu %[ptr2], %[ptr2], 4 \n\t" + "addiu %[m], %[m], -1 \n\t" + "extr_rs.w %[t0], $ac0, 15 \n\t" + "addu %[t0], %[t0], %[t1] \n\t" + "sw %[t0], 0(%[ptr1]) \n\t" + "bgtz %[m], 2b \n\t" + " addiu %[ptr1], %[ptr1], 4 \n\t" + "3: \n\t" + ".set pop \n\t" + : [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), + [r3] "=&r" (r3), [t0] "=&r" (t0), [t1] "=&r" (t1), + [t2] "=&r" (t2), [t3] "=&r" (t3), [s0] "=&r" (s0), + [s1] "=&r" (s1), [s2] "=&r" (s2), [s3] "=&r" (s3), + [ptr0] "+r" (ptr0), [ptr1] "+r" (ptr1), [m] "+r" (m), + [ptr2] "+r" (ptr2), [n] "+r" (n) + : [input0] "r" (input0), [input1] "r" (input1), + [input2] "r" (input2) + : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", + "$ac2lo", "$ac3hi", "$ac3lo" + ); +#else + // Non-DSPR2 version of the function. Avoiding the accumulator usage due to + // large latencies. This variant is bit-exact with C code. + int n = HALF_SUBFRAMELEN - 1; + int32_t t16a, t16b; + int32_t r0, r1, r2, r3, r4; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "sra %[t16a], %[input2], 16 \n\t" + "andi %[t16b], %[input2], 0xFFFF \n\t" +#if defined(MIPS32R2_LE) + "seh %[t16b], %[t16b] \n\t" + "seh %[input0], %[input0] \n\t" + "seh %[input1], %[input1] \n\t" +#else + "sll %[t16b], %[t16b], 16 \n\t" + "sra %[t16b], %[t16b], 16 \n\t" + "sll %[input0], %[input0], 16 \n\t" + "sra %[input0], %[input0], 16 \n\t" + "sll %[input1], %[input1], 16 \n\t" + "sra %[input1], %[input1], 16 \n\t" +#endif + "addiu %[r0], %[t16a], 1 \n\t" + "slt %[r1], %[t16b], $zero \n\t" + "movn %[t16a], %[r0], %[r1] \n\t" + "1: \n\t" + "lw %[r0], 0(%[ptr0]) \n\t" + "lw %[r1], 0(%[ptr2]) \n\t" + "addiu %[ptr0], %[ptr0], 4 \n\t" + "sra %[r2], %[r0], 16 \n\t" + "andi %[r0], %[r0], 0xFFFF \n\t" + "mul %[r3], %[r2], %[input0] \n\t" + "mul %[r4], %[r0], %[input0] \n\t" + "mul %[r2], %[r2], %[input1] \n\t" + "mul %[r0], %[r0], %[input1] \n\t" + "addiu %[ptr2], %[ptr2], 4 \n\t" + "sll %[r3], %[r3], 1 \n\t" + "sra %[r4], %[r4], 1 \n\t" + "addiu %[r4], %[r4], 0x2000 \n\t" + "sra %[r4], %[r4], 14 \n\t" + "addu %[r3], %[r3], %[r4] \n\t" + "addu %[r1], %[r1], %[r3] \n\t" + "sra %[r3], %[r1], 16 \n\t" + "andi %[r4], %[r1], 0xFFFF \n\t" + "sra %[r4], %[r4], 1 \n\t" + "mul %[r1], %[r1], %[t16a] \n\t" + "mul %[r3], %[r3], %[t16b] \n\t" + "mul %[r4], %[r4], %[t16b] \n\t" + "sll %[r2], %[r2], 1 \n\t" + "sra %[r0], %[r0], 1 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "addu %[r0], %[r0], %[r2] \n\t" + "addiu %[n], %[n], -1 \n\t" + "addu %[r1], %[r1], %[r3] \n\t" + "addiu %[r4], %[r4], 0x4000 \n\t" + "sra %[r4], %[r4], 15 \n\t" + "addu %[r1], %[r1], %[r4] \n\t" + "sra %[r2], %[r1], 16 \n\t" + "andi %[r3], %[r1], 0xFFFF \n\t" + "mul %[r3], %[r3], %[input0] \n\t" + "mul %[r2], %[r2], %[input0] \n\t" + "sw %[r1], -4(%[ptr2]) \n\t" + "sra %[r3], %[r3], 1 \n\t" + "addiu %[r3], %[r3], 0x2000 \n\t" + "sra %[r3], %[r3], 14 \n\t" + "addu %[r0], %[r0], %[r3] \n\t" + "sll %[r2], %[r2], 1 \n\t" + "addu %[r0], %[r0], %[r2] \n\t" + "sw %[r0], 0(%[ptr1]) \n\t" + "bgtz %[n], 1b \n\t" + " addiu %[ptr1], %[ptr1], 4 \n\t" + ".set pop \n\t" + : [t16a] "=&r" (t16a), [t16b] "=&r" (t16b), [r0] "=&r" (r0), + [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), + [r4] "=&r" (r4), [ptr0] "+r" (ptr0), [ptr1] "+r" (ptr1), + [ptr2] "+r" (ptr2), [n] "+r" (n) + : [input0] "r" (input0), [input1] "r" (input1), + [input2] "r" (input2) + : "hi", "lo", "memory" + ); +#endif +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c index 0dc8174399e..deba0d5e29f 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c @@ -834,13 +834,15 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0, /* bandwidth expansion */ for (n = 1; n <= ORDERLO; n++) { - a_LOQ11[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]); + a_LOQ11[n] = (int16_t) ((WEBRTC_SPL_MUL_16_16( + kPolyVecLo[n-1], a_LOQ11[n]) + ((int32_t) (1 << 14))) >> 15); } polyHI[0] = a_HIQ12[0]; for (n = 1; n <= ORDERHI; n++) { - a_HIQ12[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]); + a_HIQ12[n] = (int16_t) ((WEBRTC_SPL_MUL_16_16( + kPolyVecHi[n-1], a_HIQ12[n]) + ((int32_t) (1 << 14))) >> 15); polyHI[n] = a_HIQ12[n]; } diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c new file mode 100644 index 00000000000..8334f7eb18b --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c @@ -0,0 +1,133 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" + +void WebRtcIsacfix_PitchFilterCore(int loopNumber, + int16_t gain, + int index, + int16_t sign, + int16_t* inputState, + int16_t* outputBuf2, + const int16_t* coefficient, + int16_t* inputBuf, + int16_t* outputBuf, + int* index2) { + int ind2t = *index2; + int i = 0; + int16_t* out2_pos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)] + ind2t; + int32_t w1, w2, w3, w4, w5, gain32, sign32; + int32_t coef1, coef2, coef3, coef4, coef5 = 0; + // Define damp factors as int32_t (pair of int16_t) + int32_t kDampF0 = 0x0000F70A; + int32_t kDampF1 = 0x51EC2000; + int32_t kDampF2 = 0xF70A2000; + int16_t* input1 = inputBuf + ind2t; + int16_t* output1 = outputBuf + ind2t; + int16_t* output2 = outputBuf2 + ind2t + PITCH_BUFFSIZE; + + // Load coefficients outside the loop and sign-extend gain and sign + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lwl %[coef1], 3(%[coefficient]) \n\t" + "lwl %[coef2], 7(%[coefficient]) \n\t" + "lwl %[coef3], 11(%[coefficient]) \n\t" + "lwl %[coef4], 15(%[coefficient]) \n\t" + "lwr %[coef1], 0(%[coefficient]) \n\t" + "lwr %[coef2], 4(%[coefficient]) \n\t" + "lwr %[coef3], 8(%[coefficient]) \n\t" + "lwr %[coef4], 12(%[coefficient]) \n\t" + "lhu %[coef5], 16(%[coefficient]) \n\t" + "seh %[gain32], %[gain] \n\t" + "seh %[sign32], %[sign] \n\t" + ".set pop \n\t" + : [coef1] "=&r" (coef1), [coef2] "=&r" (coef2), [coef3] "=&r" (coef3), + [coef4] "=&r" (coef4), [coef5] "=&r" (coef5), [gain32] "=&r" (gain32), + [sign32] "=&r" (sign32) + : [coefficient] "r" (coefficient), [gain] "r" (gain), + [sign] "r" (sign) + : "memory" + ); + + for (i = 0; i < loopNumber; i++) { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + // Filter to get fractional pitch + "li %[w1], 8192 \n\t" + "mtlo %[w1] \n\t" + "mthi $0 \n\t" + "lwl %[w1], 3(%[out2_pos2]) \n\t" + "lwl %[w2], 7(%[out2_pos2]) \n\t" + "lwl %[w3], 11(%[out2_pos2]) \n\t" + "lwl %[w4], 15(%[out2_pos2]) \n\t" + "lwr %[w1], 0(%[out2_pos2]) \n\t" + "lwr %[w2], 4(%[out2_pos2]) \n\t" + "lwr %[w3], 8(%[out2_pos2]) \n\t" + "lwr %[w4], 12(%[out2_pos2]) \n\t" + "lhu %[w5], 16(%[out2_pos2]) \n\t" + "dpa.w.ph $ac0, %[w1], %[coef1] \n\t" + "dpa.w.ph $ac0, %[w2], %[coef2] \n\t" + "dpa.w.ph $ac0, %[w3], %[coef3] \n\t" + "dpa.w.ph $ac0, %[w4], %[coef4] \n\t" + "dpa.w.ph $ac0, %[w5], %[coef5] \n\t" + "addiu %[out2_pos2], %[out2_pos2], 2 \n\t" + "mthi $0, $ac1 \n\t" + "lwl %[w2], 3(%[inputState]) \n\t" + "lwl %[w3], 7(%[inputState]) \n\t" + // Fractional pitch shift & saturation + "extr_s.h %[w1], $ac0, 14 \n\t" + "li %[w4], 16384 \n\t" + "lwr %[w2], 0(%[inputState]) \n\t" + "lwr %[w3], 4(%[inputState]) \n\t" + "mtlo %[w4], $ac1 \n\t" + // Shift low pass filter state + "swl %[w2], 5(%[inputState]) \n\t" + "swl %[w3], 9(%[inputState]) \n\t" + "mul %[w1], %[gain32], %[w1] \n\t" + "swr %[w2], 2(%[inputState]) \n\t" + "swr %[w3], 6(%[inputState]) \n\t" + // Low pass filter accumulation + "dpa.w.ph $ac1, %[kDampF1], %[w2] \n\t" + "dpa.w.ph $ac1, %[kDampF2], %[w3] \n\t" + "lh %[w4], 0(%[input1]) \n\t" + "addiu %[input1], %[input1], 2 \n\t" + "shra_r.w %[w1], %[w1], 12 \n\t" + "sh %[w1], 0(%[inputState]) \n\t" + "dpa.w.ph $ac1, %[kDampF0], %[w1] \n\t" + // Low pass filter shift & saturation + "extr_s.h %[w2], $ac1, 15 \n\t" + "mul %[w2], %[w2], %[sign32] \n\t" + // Buffer update + "subu %[w2], %[w4], %[w2] \n\t" + "shll_s.w %[w2], %[w2], 16 \n\t" + "sra %[w2], %[w2], 16 \n\t" + "sh %[w2], 0(%[output1]) \n\t" + "addu %[w2], %[w2], %[w4] \n\t" + "shll_s.w %[w2], %[w2], 16 \n\t" + "addiu %[output1], %[output1], 2 \n\t" + "sra %[w2], %[w2], 16 \n\t" + "sh %[w2], 0(%[output2]) \n\t" + "addiu %[output2], %[output2], 2 \n\t" + ".set pop \n\t" + : [w1] "=&r" (w1), [w2] "=&r" (w2), [w3] "=&r" (w3), [w4] "=&r" (w4), + [w5] "=&r" (w5), [input1] "+r" (input1), [out2_pos2] "+r" (out2_pos2), + [output1] "+r" (output1), [output2] "+r" (output2) + : [coefficient] "r" (coefficient), [inputState] "r" (inputState), + [gain32] "r" (gain32), [sign32] "r" (sign32), [kDampF0] "r" (kDampF0), + [kDampF1] "r" (kDampF1), [kDampF2] "r" (kDampF2), + [coef1] "r" (coef1), [coef2] "r" (coef2), [coef3] "r" (coef3), + [coef4] "r" (coef4), [coef5] "r" (coef5) + : "hi", "lo", "$ac1hi", "$ac1lo", "memory" + ); + } + (*index2) += loopNumber; +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h index 4d043566369..bd20ba0165a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h @@ -26,13 +26,14 @@ /* Bitstream struct for decoder */ typedef struct Bitstreamstruct_dec { - uint16_t *stream; /* Pointer to bytestream to decode */ + uint16_t stream[STREAM_MAXW16_60MS]; /* Array bytestream to decode */ uint32_t W_upper; /* Upper boundary of interval W */ uint32_t streamval; uint16_t stream_index; /* Index to the current position in bytestream */ int16_t full; /* 0 - first byte in memory filled, second empty*/ /* 1 - both bytes are empty (we just filled the previous memory */ + int stream_size; /* The size of stream. */ } Bitstr_dec; /* Bitstream struct for encoder */ diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S index 46682ac556a..6713b28695c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S @@ -42,7 +42,11 @@ DEFINE_FUNCTION WebRtcIsacfix_Time2SpecNeon add r5, sp, #(16 + FRAMESAMPLES * 2) @ tmpimQ16; adr r9, WebRtcIsacfix_kCosTab1 +#if defined(__APPLE__) + mov r6, #:lower16:(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1) +#else mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1) +#endif add r10, r9, r6 @ WebRtcIsacfix_kSinTab1 vmov.u32 q14, #0 @ Initialize the maximum values for tmpInIm. @@ -455,7 +459,12 @@ TransformAndFindMax: bgt TransformAndFindMax adr r10, WebRtcIsacfix_kSinTab1 +#if defined(__APPLE__) + mov r2, #:lower16:(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1) +#else mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1) +#endif + sub r11, r10, r2 @ WebRtcIsacfix_kCosTab1 @ Find the maximum value in the Neon registers diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h index f937b3453fa..76a61e6d33c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h @@ -14,7 +14,7 @@ /* * Define the fixed-point numeric formats */ -#include "typedefs.h" +#include "webrtc/typedefs.h" typedef struct WebRtcISACStruct ISACStruct; diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c index 66bf06d472b..9ae69a0bbf2 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c @@ -1449,13 +1449,18 @@ void WebRtcIsac_EncodeRc(int16_t* RCQ15, Bitstr* streamdata) { /* quantize reflection coefficients (add noise feedback?) */ for (k = 0; k < AR_ORDER; k++) { index[k] = WebRtcIsac_kQArRcInitIndex[k]; - + // The safe-guards in following while conditions are to suppress gcc 4.8.3 + // warnings, Issue 2888. Otherwise, first and last elements of + // |WebRtcIsac_kQArBoundaryLevels| are such that the following search + // *never* cause an out-of-boundary read. if (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k]]) { - while (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k] + 1]) { + while (index[k] + 1 < NUM_AR_RC_QUANT_BAUNDARY && + RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k] + 1]) { index[k]++; } } else { - while (RCQ15[k] < WebRtcIsac_kQArBoundaryLevels[--index[k]]) ; + while (index[k] > 0 && + RCQ15[k] < WebRtcIsac_kQArBoundaryLevels[--index[k]]) ; } RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]); } diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c index f3f1650b42b..fa54a8d873c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c @@ -15,20 +15,21 @@ * */ -#include "isac.h" -#include "bandwidth_estimator.h" -#include "crc.h" -#include "entropy_coding.h" -#include "codec.h" -#include "structs.h" -#include "signal_processing_library.h" -#include "lpc_shape_swb16_tables.h" -#include "os_specific_inline.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" +#include <math.h> #include <stdio.h> -#include <string.h> #include <stdlib.h> -#include <math.h> +#include <string.h> + +#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/codec.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/crc.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h" #define BIT_MASK_DEC_INIT 0x0001 #define BIT_MASK_ENC_INIT 0x0002 @@ -273,7 +274,7 @@ int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) { ISACMainStruct* instISAC; if (ISAC_main_inst != NULL) { - instISAC = (ISACMainStruct*)WEBRTC_SPL_VNEW(ISACMainStruct, 1); + instISAC = (ISACMainStruct*)malloc(sizeof(ISACMainStruct)); *ISAC_main_inst = (ISACStruct*)instISAC; if (*ISAC_main_inst != NULL) { instISAC->errorCode = 0; @@ -306,7 +307,7 @@ int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) { */ int16_t WebRtcIsac_Free(ISACStruct* ISAC_main_inst) { ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - WEBRTC_SPL_FREE(instISAC); + free(instISAC); return 0; } @@ -552,8 +553,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, } if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) { - WebRtcSpl_AnalysisQMF(speech_in_ptr, speechInLB, speechInUB, - instISAC->analysisFBState1, + WebRtcSpl_AnalysisQMF(speech_in_ptr, SWBFRAMESAMPLES_10ms, speechInLB, + speechInUB, instISAC->analysisFBState1, instISAC->analysisFBState2); /* Convert from fixed to floating point. */ @@ -1314,7 +1315,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst, speechIdx = 0; while (speechIdx < numSamplesLB) { WebRtcSpl_SynthesisQMF(&outFrameLB[speechIdx], &outFrameUB[speechIdx], - &decoded[(speechIdx << 1)], + FRAMESAMPLES_10ms, &decoded[(speechIdx << 1)], instISAC->synthesisFBState1, instISAC->synthesisFBState2); diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c index 9eae0555f65..0f6d889225d 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c @@ -13,68 +13,69 @@ /********************* AR Coefficient Tables ************************/ /* cdf for quantized reflection coefficient 1 */ -const uint16_t WebRtcIsac_kQArRc1Cdf[12] = { +const uint16_t WebRtcIsac_kQArRc1Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531, 65533, 65535}; /* cdf for quantized reflection coefficient 2 */ -const uint16_t WebRtcIsac_kQArRc2Cdf[12] = { +const uint16_t WebRtcIsac_kQArRc2Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531, 65533, 65535}; /* cdf for quantized reflection coefficient 3 */ -const uint16_t WebRtcIsac_kQArRc3Cdf[12] = { +const uint16_t WebRtcIsac_kQArRc3Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531, 65533, 65535}; /* cdf for quantized reflection coefficient 4 */ -const uint16_t WebRtcIsac_kQArRc4Cdf[12] = { +const uint16_t WebRtcIsac_kQArRc4Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531, 65533, 65535}; /* cdf for quantized reflection coefficient 5 */ -const uint16_t WebRtcIsac_kQArRc5Cdf[12] = { +const uint16_t WebRtcIsac_kQArRc5Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531, 65533, 65535}; /* cdf for quantized reflection coefficient 6 */ -const uint16_t WebRtcIsac_kQArRc6Cdf[12] = { +const uint16_t WebRtcIsac_kQArRc6Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531, 65533, 65535}; /* representation levels for quantized reflection coefficient 1 */ -const int16_t WebRtcIsac_kQArRc1Levels[11] = { +const int16_t WebRtcIsac_kQArRc1Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104 }; /* representation levels for quantized reflection coefficient 2 */ -const int16_t WebRtcIsac_kQArRc2Levels[11] = { +const int16_t WebRtcIsac_kQArRc2Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104 }; /* representation levels for quantized reflection coefficient 3 */ -const int16_t WebRtcIsac_kQArRc3Levels[11] = { +const int16_t WebRtcIsac_kQArRc3Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { -32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104 }; /* representation levels for quantized reflection coefficient 4 */ -const int16_t WebRtcIsac_kQArRc4Levels[11] = { +const int16_t WebRtcIsac_kQArRc4Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { -32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104 }; /* representation levels for quantized reflection coefficient 5 */ -const int16_t WebRtcIsac_kQArRc5Levels[11] = { +const int16_t WebRtcIsac_kQArRc5Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { -32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104 }; /* representation levels for quantized reflection coefficient 6 */ -const int16_t WebRtcIsac_kQArRc6Levels[11] = { +const int16_t WebRtcIsac_kQArRc6Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { -32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104 }; /* quantization boundary levels for reflection coefficients */ -const int16_t WebRtcIsac_kQArBoundaryLevels[12] = { --32768, -31441, -27566, -21458, -13612, -4663, 4663, 13612, 21458, 27566, 31441, 32767 +const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY] = { +-32768, -31441, -27566, -21458, -13612, -4663, 4663, 13612, 21458, 27566, 31441, +32767 }; /* initial index for AR reflection coefficient quantizer and cdf table search */ diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h index 22fe6a2102c..989cb367bff 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h @@ -21,27 +21,29 @@ #include "structs.h" +#define NUM_AR_RC_QUANT_BAUNDARY 12 + /********************* AR Coefficient Tables ************************/ /* cdf for quantized reflection coefficient 1 */ -extern const uint16_t WebRtcIsac_kQArRc1Cdf[12]; +extern const uint16_t WebRtcIsac_kQArRc1Cdf[NUM_AR_RC_QUANT_BAUNDARY]; /* cdf for quantized reflection coefficient 2 */ -extern const uint16_t WebRtcIsac_kQArRc2Cdf[12]; +extern const uint16_t WebRtcIsac_kQArRc2Cdf[NUM_AR_RC_QUANT_BAUNDARY]; /* cdf for quantized reflection coefficient 3 */ -extern const uint16_t WebRtcIsac_kQArRc3Cdf[12]; +extern const uint16_t WebRtcIsac_kQArRc3Cdf[NUM_AR_RC_QUANT_BAUNDARY]; /* cdf for quantized reflection coefficient 4 */ -extern const uint16_t WebRtcIsac_kQArRc4Cdf[12]; +extern const uint16_t WebRtcIsac_kQArRc4Cdf[NUM_AR_RC_QUANT_BAUNDARY]; /* cdf for quantized reflection coefficient 5 */ -extern const uint16_t WebRtcIsac_kQArRc5Cdf[12]; +extern const uint16_t WebRtcIsac_kQArRc5Cdf[NUM_AR_RC_QUANT_BAUNDARY]; /* cdf for quantized reflection coefficient 6 */ -extern const uint16_t WebRtcIsac_kQArRc6Cdf[12]; +extern const uint16_t WebRtcIsac_kQArRc6Cdf[NUM_AR_RC_QUANT_BAUNDARY]; /* quantization boundary levels for reflection coefficients */ -extern const int16_t WebRtcIsac_kQArBoundaryLevels[12]; +extern const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY]; /* initial indices for AR reflection coefficient quantizer and cdf table search */ extern const uint16_t WebRtcIsac_kQArRcInitIndex[AR_ORDER]; diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h index 1bd73e75bd0..62c890c84bc 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h @@ -18,10 +18,9 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ - -#include "typedefs.h" -#include "settings.h" -#include "isac.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/source/settings.h" +#include "webrtc/typedefs.h" typedef struct Bitstreamstruct { diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h index 1370aff06a6..7998fdbdebf 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_ #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_ -#include "typedefs.h" +#include "webrtc/typedefs.h" #ifdef __cplusplus extern "C" { @@ -59,6 +59,64 @@ int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples, */ int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate); +/**************************************************************************** + * WebRtcOpus_SetPacketLossRate(...) + * + * This function configures the encoder's expected packet loss percentage. + * + * Input: + * - inst : Encoder context + * - loss_rate : loss percentage in the range 0-100, inclusive. + * Return value : 0 - Success + * -1 - Error + */ +int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate); + +/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate + * is needed. It might not be very useful since there are not many use cases and + * the caller can always maintain the states. */ + +/**************************************************************************** + * WebRtcOpus_EnableFec() + * + * This function enables FEC for encoding. + * + * Input: + * - inst : Encoder context + * + * Return value : 0 - Success + * -1 - Error + */ +int16_t WebRtcOpus_EnableFec(OpusEncInst* inst); + +/**************************************************************************** + * WebRtcOpus_DisableFec() + * + * This function disables FEC for encoding. + * + * Input: + * - inst : Encoder context + * + * Return value : 0 - Success + * -1 - Error + */ +int16_t WebRtcOpus_DisableFec(OpusEncInst* inst); + +/* + * WebRtcOpus_SetComplexity(...) + * + * This function adjusts the computational complexity. The effect is the same as + * calling the complexity setting of Opus as an Opus encoder related CTL. + * + * Input: + * - inst : Encoder context + * - complexity : New target complexity (0-10, inclusive) + * + * Return value : 0 - Success + * -1 - Error + */ +int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity); + int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels); int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst); @@ -113,6 +171,7 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded, int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type); + /**************************************************************************** * WebRtcOpus_DecodePlc(...) * TODO(tlegrand): Remove master and slave functions when NetEq4 is in place. @@ -138,6 +197,28 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames); /**************************************************************************** + * WebRtcOpus_DecodeFec(...) + * + * This function decodes the FEC data from an Opus packet into one or more audio + * frames at the ACM interface's sampling rate (32 kHz). + * + * Input: + * - inst : Decoder context + * - encoded : Encoded data + * - encoded_bytes : Bytes in encoded vector + * + * Output: + * - decoded : The decoded vector (previous frame) + * + * Return value : >0 - Samples per channel in decoded vector + * 0 - No FEC data in the packet + * -1 - Error + */ +int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, + int16_t encoded_bytes, int16_t* decoded, + int16_t* audio_type); + +/**************************************************************************** * WebRtcOpus_DurationEst(...) * * This function calculates the duration of an opus packet. @@ -152,6 +233,40 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst, const uint8_t* payload, int payload_length_bytes); +/* TODO(minyue): Check whether it is needed to add a decoder context to the + * arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells + * the duration. The decoder context in WebRtcOpus_DurationEst(...) is not used. + * So it may be advisable to remove it from WebRtcOpus_DurationEst(...). */ + +/**************************************************************************** + * WebRtcOpus_FecDurationEst(...) + * + * This function calculates the duration of the FEC data within an opus packet. + * Input: + * - payload : Encoded data pointer + * - payload_length_bytes : Bytes of encoded data + * + * Return value : >0 - The duration of the FEC data in the + * packet in samples. + * 0 - No FEC data in the packet. + */ +int WebRtcOpus_FecDurationEst(const uint8_t* payload, + int payload_length_bytes); + +/**************************************************************************** + * WebRtcOpus_PacketHasFec(...) + * + * This function detects if an opus packet has FEC. + * Input: + * - payload : Encoded data pointer + * - payload_length_bytes : Bytes of encoded data + * + * Return value : 0 - the packet does NOT contain FEC. + * 1 - the packet contains FEC. + */ +int WebRtcOpus_PacketHasFec(const uint8_t* payload, + int payload_length_bytes); + #ifdef __cplusplus } // extern "C" #endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus.gypi index 406870232bb..b1dedd7d4a6 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus.gypi +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus.gypi @@ -32,4 +32,26 @@ ], }, ], + 'conditions': [ + ['include_tests==1', { + 'targets': [ + { + 'target_name': 'webrtc_opus_fec_test', + 'type': 'executable', + 'dependencies': [ + 'webrtc_opus', + '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', + '<(webrtc_root)/test/test.gyp:test_support_main', + '<(DEPTH)/testing/gtest.gyp:gtest', + ], + 'include_dirs': [ + '<(webrtc_root)', + ], + 'sources': [ + 'opus_fec_test.cc', + ], + }, + ], + }], + ], } diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc new file mode 100644 index 00000000000..fb4cb04f361 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc @@ -0,0 +1,249 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +using ::std::string; +using ::std::tr1::tuple; +using ::std::tr1::make_tuple; +using ::std::tr1::get; +using ::testing::TestWithParam; +using ::testing::ValuesIn; + +namespace webrtc { + +// Define coding parameter as <channels, bit_rate, filename, extension>. +typedef tuple<int, int, string, string> coding_param; +typedef struct mode mode; + +struct mode { + bool fec; + uint8_t target_packet_loss_rate; +}; + +const int kOpusBlockDurationMs = 20; +const int kOpusInputSamplingKhz = 48; +const int kOpusOutputSamplingKhz = 32; + +class OpusFecTest : public TestWithParam<coding_param> { + protected: + OpusFecTest(); + + virtual void SetUp(); + virtual void TearDown(); + + virtual void EncodeABlock(); + + virtual void DecodeABlock(bool lost_previous, bool lost_current); + + int block_duration_ms_; + int input_sampling_khz_; + int output_sampling_khz_; + + // Number of samples-per-channel in a frame. + int input_length_sample_; + + // Expected output number of samples-per-channel in a frame. + int output_length_sample_; + + int channels_; + int bit_rate_; + + size_t data_pointer_; + size_t loop_length_samples_; + int max_bytes_; + int encoded_bytes_; + + WebRtcOpusEncInst* opus_encoder_; + WebRtcOpusDecInst* opus_decoder_; + + string in_filename_; + + scoped_ptr<int16_t[]> in_data_; + scoped_ptr<int16_t[]> out_data_; + scoped_ptr<uint8_t[]> bit_stream_; +}; + +void OpusFecTest::SetUp() { + channels_ = get<0>(GetParam()); + bit_rate_ = get<1>(GetParam()); + printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_); + + in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam())); + + FILE* fp = fopen(in_filename_.c_str(), "rb"); + ASSERT_FALSE(fp == NULL); + + // Obtain file size. + fseek(fp, 0, SEEK_END); + loop_length_samples_ = ftell(fp) / sizeof(int16_t); + rewind(fp); + + // Allocate memory to contain the whole file. + in_data_.reset(new int16_t[loop_length_samples_ + + input_length_sample_ * channels_]); + + // Copy the file into the buffer. + ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp), + loop_length_samples_); + fclose(fp); + + // The audio will be used in a looped manner. To ease the acquisition of an + // audio frame that crosses the end of the excerpt, we add an extra block + // length of samples to the end of the array, starting over again from the + // beginning of the array. Audio frames cross the end of the excerpt always + // appear as a continuum of memory. + memcpy(&in_data_[loop_length_samples_], &in_data_[0], + input_length_sample_ * channels_ * sizeof(int16_t)); + + // Maximum number of bytes in output bitstream. + max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t); + + out_data_.reset(new int16_t[2 * output_length_sample_ * channels_]); + bit_stream_.reset(new uint8_t[max_bytes_]); + + // Create encoder memory. + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_)); + EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); + // Set bitrate. + EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_)); +} + +void OpusFecTest::TearDown() { + // Free memory. + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); + EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); +} + +OpusFecTest::OpusFecTest() + : block_duration_ms_(kOpusBlockDurationMs), + input_sampling_khz_(kOpusInputSamplingKhz), + output_sampling_khz_(kOpusOutputSamplingKhz), + input_length_sample_(block_duration_ms_ * input_sampling_khz_), + output_length_sample_(block_duration_ms_ * output_sampling_khz_), + data_pointer_(0), + max_bytes_(0), + encoded_bytes_(0), + opus_encoder_(NULL), + opus_decoder_(NULL) { +} + +void OpusFecTest::EncodeABlock() { + int16_t value = WebRtcOpus_Encode(opus_encoder_, + &in_data_[data_pointer_], + input_length_sample_, + max_bytes_, &bit_stream_[0]); + EXPECT_GT(value, 0); + + encoded_bytes_ = value; +} + +void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) { + int16_t audio_type; + int16_t value_1 = 0, value_2 = 0; + + if (lost_previous) { + // Decode previous frame. + if (!lost_current && + WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) { + value_1 = WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], + encoded_bytes_, &out_data_[0], + &audio_type); + } else { + value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1); + } + EXPECT_EQ(output_length_sample_, value_1); + } + + if (!lost_current) { + // Decode current frame. + value_2 = WebRtcOpus_DecodeNew(opus_decoder_, &bit_stream_[0], + encoded_bytes_, + &out_data_[value_1 * channels_], + &audio_type); + EXPECT_EQ(output_length_sample_, value_2); + } +} + +TEST_P(OpusFecTest, RandomPacketLossTest) { + const int kDurationMs = 200000; + int time_now_ms, fec_frames; + int actual_packet_loss_rate; + bool lost_current, lost_previous; + mode mode_set[3] = {{true, 0}, + {false, 0}, + {true, 50}}; + + lost_current = false; + for (int i = 0; i < 3; i++) { + if (mode_set[i].fec) { + EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_)); + EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, + mode_set[i].target_packet_loss_rate)); + printf("FEC is ON, target at packet loss rate %d percent.\n", + mode_set[i].target_packet_loss_rate); + } else { + EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_)); + printf("FEC is OFF.\n"); + } + // In this test, we let the target packet loss rate match the actual rate. + actual_packet_loss_rate = mode_set[i].target_packet_loss_rate; + // Run every mode a certain time. + time_now_ms = 0; + fec_frames = 0; + while (time_now_ms < kDurationMs) { + // Encode & decode. + EncodeABlock(); + + // Check if payload has FEC. + int16_t fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_); + + // If FEC is disabled or the target packet loss rate is set to 0, there + // should be no FEC in the bit stream. + if (!mode_set[i].fec || mode_set[i].target_packet_loss_rate == 0) { + EXPECT_EQ(fec, 0); + } else if (fec == 1) { + fec_frames++; + } + + lost_previous = lost_current; + lost_current = rand() < actual_packet_loss_rate * (RAND_MAX / 100); + DecodeABlock(lost_previous, lost_current); + + time_now_ms += block_duration_ms_; + + // |data_pointer_| is incremented and wrapped across + // |loop_length_samples_|. + data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) % + loop_length_samples_; + } + if (mode_set[i].fec) { + printf("%.2f percent frames has FEC.\n", + static_cast<float>(fec_frames) * block_duration_ms_ / 2000); + } + } +} + +const coding_param param_set[] = + {make_tuple(1, 64000, string("audio_coding/testfile32kHz"), + string("pcm")), + make_tuple(1, 32000, string("audio_coding/testfile32kHz"), + string("pcm")), + make_tuple(2, 64000, string("audio_coding/teststereo32kHz"), + string("pcm"))}; + +// 64 kbps, stereo +INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest, + ValuesIn(param_set)); + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_interface.c index 98b924f219c..24fc4fc405a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_interface.c +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_interface.c @@ -103,7 +103,40 @@ int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples, int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) { if (inst) { - return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate)); + return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate)); + } else { + return -1; + } +} + +int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) { + if (inst) { + return opus_encoder_ctl(inst->encoder, + OPUS_SET_PACKET_LOSS_PERC(loss_rate)); + } else { + return -1; + } +} + +int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) { + if (inst) { + return opus_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(1)); + } else { + return -1; + } +} + +int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) { + if (inst) { + return opus_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(0)); + } else { + return -1; + } +} + +int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) { + if (inst) { + return opus_encoder_ctl(inst->encoder, OPUS_SET_COMPLEXITY(complexity)); } else { return -1; } @@ -217,6 +250,23 @@ static int DecodeNative(OpusDecoder* inst, const int16_t* encoded, return -1; } +static int DecodeFec(OpusDecoder* inst, const int16_t* encoded, + int16_t encoded_bytes, int frame_size, + int16_t* decoded, int16_t* audio_type) { + unsigned char* coded = (unsigned char*) encoded; + opus_int16* audio = (opus_int16*) decoded; + + int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 1); + + /* TODO(tlegrand): set to DTX for zero-length packets? */ + *audio_type = 0; + + if (res > 0) { + return res; + } + return -1; +} + /* Resample from 48 to 32 kHz. Length of state is assumed to be * kWebRtcOpusStateSize (7). */ @@ -542,6 +592,52 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, return resampled_samples; } +int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, + int16_t encoded_bytes, int16_t* decoded, + int16_t* audio_type) { + /* |buffer| is big enough for 120 ms (the largest Opus packet size) of stereo + * audio at 48 kHz. */ + int16_t buffer[kWebRtcOpusMaxFrameSize]; + int16_t* coded = (int16_t*)encoded; + int decoded_samples; + int resampled_samples; + int fec_samples; + + if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) { + return 0; + } + + fec_samples = opus_packet_get_samples_per_frame(encoded, 48000); + + /* Decode to a temporary buffer. */ + decoded_samples = DecodeFec(inst->decoder_left, coded, encoded_bytes, + fec_samples, buffer, audio_type); + if (decoded_samples < 0) { + return -1; + } + + /* If mono case, just do a regular call to the decoder. + * If stereo, we need to de-interleave the stereo output into blocks with + * left and right channel. Each block is resampled to 32 kHz, and then + * interleaved again. */ + if (inst->channels == 2) { + /* De-interleave and resample. */ + resampled_samples = WebRtcOpus_DeInterleaveResample(inst, + buffer, + decoded_samples, + decoded); + } else { + /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is + * used for mono signals. */ + resampled_samples = WebRtcOpus_Resample48to32(buffer, + decoded_samples, + inst->state_48_32_left, + decoded); + } + + return resampled_samples; +} + int WebRtcOpus_DurationEst(OpusDecInst* inst, const uint8_t* payload, int payload_length_bytes) { @@ -562,3 +658,79 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst, samples = samples * 2 / 3; return samples; } + +int WebRtcOpus_FecDurationEst(const uint8_t* payload, + int payload_length_bytes) { + int samples; + if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) { + return 0; + } + + samples = opus_packet_get_samples_per_frame(payload, 48000); + if (samples < 480 || samples > 5760) { + /* Invalid payload duration. */ + return 0; + } + /* Compensate for the down-sampling from 48 kHz to 32 kHz. + * This should be removed when the resampling in WebRtcOpus_Decode is + * removed. */ + samples = samples * 2 / 3; + return samples; +} + +int WebRtcOpus_PacketHasFec(const uint8_t* payload, + int payload_length_bytes) { + int frames, channels, payload_length_ms; + int n; + opus_int16 frame_sizes[48]; + const unsigned char *frame_data[48]; + + if (payload == NULL || payload_length_bytes <= 0) + return 0; + + /* In CELT_ONLY mode, packets should not have FEC. */ + if (payload[0] & 0x80) + return 0; + + payload_length_ms = opus_packet_get_samples_per_frame(payload, 48000) / 48; + if (10 > payload_length_ms) + payload_length_ms = 10; + + channels = opus_packet_get_nb_channels(payload); + + switch (payload_length_ms) { + case 10: + case 20: { + frames = 1; + break; + } + case 40: { + frames = 2; + break; + } + case 60: { + frames = 3; + break; + } + default: { + return 0; // It is actually even an invalid packet. + } + } + + /* The following is to parse the LBRR flags. */ + if (opus_packet_parse(payload, payload_length_bytes, NULL, frame_data, + frame_sizes, NULL) < 0) { + return 0; + } + + if (frame_sizes[0] <= 1) { + return 0; + } + + for (n = 0; n < channels; n++) { + if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1))) + return 1; + } + + return 0; +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc new file mode 100644 index 00000000000..16099c6d93a --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -0,0 +1,119 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" +#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h" + +using ::std::string; +using ::std::tr1::make_tuple; +using ::testing::ValuesIn; + +namespace webrtc { + +static const int kOpusBlockDurationMs = 20; +static const int kOpusInputSamplingKhz = 48; +static const int kOpustOutputSamplingKhz = 32; + +class OpusSpeedTest : public AudioCodecSpeedTest { + protected: + OpusSpeedTest(); + virtual void SetUp() OVERRIDE; + virtual void TearDown() OVERRIDE; + virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, + int max_bytes, int* encoded_bytes); + virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes, + int16_t* out_data); + WebRtcOpusEncInst* opus_encoder_; + WebRtcOpusDecInst* opus_decoder_; +}; + +OpusSpeedTest::OpusSpeedTest() + : AudioCodecSpeedTest(kOpusBlockDurationMs, + kOpusInputSamplingKhz, + kOpustOutputSamplingKhz), + opus_encoder_(NULL), + opus_decoder_(NULL) { +} + +void OpusSpeedTest::SetUp() { + AudioCodecSpeedTest::SetUp(); + /* Create encoder memory. */ + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_)); + EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); + /* Set bitrate. */ + EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_)); +} + +void OpusSpeedTest::TearDown() { + AudioCodecSpeedTest::TearDown(); + /* Free memory. */ + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); + EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); +} + +float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream, + int max_bytes, int* encoded_bytes) { + clock_t clocks = clock(); + int value = WebRtcOpus_Encode(opus_encoder_, in_data, + input_length_sample_, max_bytes, + bit_stream); + clocks = clock() - clocks; + EXPECT_GT(value, 0); + *encoded_bytes = value; + return 1000.0 * clocks / CLOCKS_PER_SEC; +} + +float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream, + int encoded_bytes, int16_t* out_data) { + int value; + int16_t audio_type; + clock_t clocks = clock(); + value = WebRtcOpus_DecodeNew(opus_decoder_, bit_stream, encoded_bytes, + out_data, &audio_type); + clocks = clock() - clocks; + EXPECT_EQ(output_length_sample_, value); + return 1000.0 * clocks / CLOCKS_PER_SEC; +} + +#define ADD_TEST(complexity) \ +TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \ + /* Test audio length in second. */ \ + size_t kDurationSec = 400; \ + /* Set complexity. */ \ + printf("Setting complexity to %d ...\n", complexity); \ + EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \ + EncodeDecode(kDurationSec); \ +} + +ADD_TEST(10); +ADD_TEST(9); +ADD_TEST(8); +ADD_TEST(7); +ADD_TEST(6); +ADD_TEST(5); +ADD_TEST(4); +ADD_TEST(3); +ADD_TEST(2); +ADD_TEST(1); +ADD_TEST(0); + +// List all test cases: (channel, bit rat, filename, extension). +const coding_param param_set[] = + {make_tuple(1, 64000, string("audio_coding/speech_mono_32_48kHz"), + string("pcm"), true), + make_tuple(1, 32000, string("audio_coding/speech_mono_32_48kHz"), + string("pcm"), true), + make_tuple(2, 64000, string("audio_coding/music_stereo_48kHz"), + string("pcm"), true)}; + +INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest, + ValuesIn(param_set)); + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc index b699cf9df5e..ed876cd1050 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -202,6 +202,27 @@ TEST_F(OpusTest, OpusSetBitRate) { EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); } +TEST_F(OpusTest, OpusSetComplexity) { + // Test without creating encoder memory. + EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_mono_encoder_, 9)); + EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_stereo_encoder_, 9)); + + // Create encoder memory, try with different complexities. + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1)); + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); + + EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, 0)); + EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_, 0)); + EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, 10)); + EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_, 10)); + EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_mono_encoder_, 11)); + EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_stereo_encoder_, 11)); + + // Free memory. + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_)); + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); +} + // Encode and decode one frame (stereo), initialize the decoder and // decode once more. TEST_F(OpusTest, OpusDecodeInit) { @@ -265,6 +286,47 @@ TEST_F(OpusTest, OpusDecodeInit) { EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_new_)); } +TEST_F(OpusTest, OpusEnableDisableFec) { + // Test without creating encoder memory. + EXPECT_EQ(-1, WebRtcOpus_EnableFec(opus_mono_encoder_)); + EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_stereo_encoder_)); + + // Create encoder memory, try with different bitrates. + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1)); + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); + + EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_mono_encoder_)); + EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_stereo_encoder_)); + EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_mono_encoder_)); + EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_stereo_encoder_)); + + // Free memory. + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_)); + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); +} + +TEST_F(OpusTest, OpusSetPacketLossRate) { + // Test without creating encoder memory. + EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_mono_encoder_, 50)); + EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_stereo_encoder_, 50)); + + // Create encoder memory, try with different bitrates. + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1)); + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); + + EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_mono_encoder_, 50)); + EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_stereo_encoder_, 50)); + EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_mono_encoder_, -1)); + EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_stereo_encoder_, -1)); + EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_mono_encoder_, 101)); + EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_stereo_encoder_, 101)); + + // Free memory. + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_)); + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); +} + + // PLC in mono mode. TEST_F(OpusTest, OpusDecodePlcMono) { // Create encoder memory. diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/OWNERS new file mode 100644 index 00000000000..bbffda7e492 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/OWNERS @@ -0,0 +1,6 @@ +per-file *.isolate=kjellander@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc new file mode 100644 index 00000000000..c7cafdff9ba --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc @@ -0,0 +1,124 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h" + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/test/testsupport/fileutils.h" + +using ::std::tr1::get; + +namespace webrtc { + +AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms, + int input_sampling_khz, + int output_sampling_khz) + : block_duration_ms_(block_duration_ms), + input_sampling_khz_(input_sampling_khz), + output_sampling_khz_(output_sampling_khz), + input_length_sample_(block_duration_ms_ * input_sampling_khz_), + output_length_sample_(block_duration_ms_ * output_sampling_khz_), + data_pointer_(0), + loop_length_samples_(0), + max_bytes_(0), + encoded_bytes_(0), + encoding_time_ms_(0.0), + decoding_time_ms_(0.0), + out_file_(NULL) { +} + +void AudioCodecSpeedTest::SetUp() { + channels_ = get<0>(GetParam()); + bit_rate_ = get<1>(GetParam()); + in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam())); + save_out_data_ = get<4>(GetParam()); + + FILE* fp = fopen(in_filename_.c_str(), "rb"); + assert(fp != NULL); + + // Obtain file size. + fseek(fp, 0, SEEK_END); + loop_length_samples_ = ftell(fp) / sizeof(int16_t); + rewind(fp); + + // Allocate memory to contain the whole file. + in_data_.reset(new int16_t[loop_length_samples_ + + input_length_sample_ * channels_]); + + data_pointer_ = 0; + + // Copy the file into the buffer. + ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp), + loop_length_samples_); + fclose(fp); + + // Add an extra block length of samples to the end of the array, starting + // over again from the beginning of the array. This is done to simplify + // the reading process when reading over the end of the loop. + memcpy(&in_data_[loop_length_samples_], &in_data_[0], + input_length_sample_ * channels_ * sizeof(int16_t)); + + max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t); + out_data_.reset(new int16_t[output_length_sample_ * channels_]); + bit_stream_.reset(new uint8_t[max_bytes_]); + + if (save_out_data_) { + std::string out_filename = + ::testing::UnitTest::GetInstance()->current_test_info()->name(); + + // Erase '/' + size_t found; + while ((found = out_filename.find('/')) != std::string::npos) + out_filename.replace(found, 1, "_"); + + out_filename = test::OutputPath() + out_filename + ".pcm"; + + out_file_ = fopen(out_filename.c_str(), "wb"); + assert(out_file_ != NULL); + + printf("Output to be saved in %s.\n", out_filename.c_str()); + } +} + +void AudioCodecSpeedTest::TearDown() { + if (save_out_data_) { + fclose(out_file_); + } +} + +void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) { + size_t time_now_ms = 0; + float time_ms; + + printf("Coding %d kHz-sampled %d-channel audio at %d bps ...\n", + input_sampling_khz_, channels_, bit_rate_); + + while (time_now_ms < audio_duration_sec * 1000) { + // Encode & decode. + time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0], + max_bytes_, &encoded_bytes_); + encoding_time_ms_ += time_ms; + time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]); + decoding_time_ms_ += time_ms; + if (save_out_data_) { + fwrite(&out_data_[0], sizeof(int16_t), + output_length_sample_ * channels_, out_file_); + } + data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) % + loop_length_samples_; + time_now_ms += block_duration_ms_; + } + + printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n", + (encoding_time_ms_ / audio_duration_sec) / 10.0, + (decoding_time_ms_ / audio_duration_sec) / 10.0); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h new file mode 100644 index 00000000000..2c9b45e4f86 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h @@ -0,0 +1,90 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ + +#include <string> +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +// Define coding parameter as +// <channels, bit_rate, file_name, extension, if_save_output>. +typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param; + +class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> { + protected: + AudioCodecSpeedTest(int block_duration_ms, + int input_sampling_khz, + int output_sampling_khz); + virtual void SetUp(); + virtual void TearDown(); + + // EncodeABlock(...) does the following: + // 1. encodes a block of audio, saved in |in_data|, + // 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size, + // 3. assign |encoded_bytes| with the length of the bit stream (in bytes), + // 4. return the cost of time (in millisecond) spent on actual encoding. + virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, + int max_bytes, int* encoded_bytes) = 0; + + // DecodeABlock(...) does the following: + // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes| + // (in bytes), + // 2. save the decoded audio in |out_data|, + // 3. return the cost of time (in millisecond) spent on actual decoding. + virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes, + int16_t* out_data) = 0; + + // Encoding and decode an audio of |audio_duration| (in seconds) and + // record the runtime for encoding and decoding separately. + void EncodeDecode(size_t audio_duration); + + int block_duration_ms_; + int input_sampling_khz_; + int output_sampling_khz_; + + // Number of samples-per-channel in a frame. + int input_length_sample_; + + // Expected output number of samples-per-channel in a frame. + int output_length_sample_; + + scoped_ptr<int16_t[]> in_data_; + scoped_ptr<int16_t[]> out_data_; + size_t data_pointer_; + size_t loop_length_samples_; + scoped_ptr<uint8_t[]> bit_stream_; + + // Maximum number of bytes in output bitstream for a frame of audio. + int max_bytes_; + + int encoded_bytes_; + float encoding_time_ms_; + float decoding_time_ms_; + FILE* out_file_; + + int channels_; + + // Bit rate is in bit-per-second. + int bit_rate_; + + std::string in_filename_; + + // Determines whether to save the output to file. + bool save_out_data_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi new file mode 100644 index 00000000000..4d675e10cfa --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi @@ -0,0 +1,71 @@ +# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +{ + 'targets': [ + { + 'target_name': 'audio_codec_speed_tests', + 'type': '<(gtest_target_type)', + 'dependencies': [ + 'audio_processing', + 'iSACFix', + 'webrtc_opus', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/test/test.gyp:test_support_main', + ], + 'sources': [ + 'audio_codec_speed_test.h', + 'audio_codec_speed_test.cc', + '<(webrtc_root)/modules/audio_coding/codecs/opus/opus_speed_test.cc', + '<(webrtc_root)/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc', + ], + 'conditions': [ + # TODO(henrike): remove build_with_chromium==1 when the bots are + # using Chromium's buildbots. + ['build_with_chromium==1 and OS=="android"', { + 'dependencies': [ + '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', + ], + }], + ], + }], + 'conditions': [ + # TODO(henrike): remove build_with_chromium==1 when the bots are using + # Chromium's buildbots. + ['build_with_chromium==1 and OS=="android"', { + 'targets': [ + { + 'target_name': 'audio_codec_speed_tests_apk_target', + 'type': 'none', + 'dependencies': [ + '<(apk_tests_path):audio_codec_speed_tests_apk', + ], + }, + ], + }], + ['test_isolation_mode != "noop"', { + 'targets': [ + { + 'target_name': 'audio_codec_speed_tests_run', + 'type': 'none', + 'dependencies': [ + 'audio_codec_speed_tests', + ], + 'includes': [ + '../../../../build/isolate.gypi', + 'audio_codec_speed_tests.isolate', + ], + 'sources': [ + 'audio_codec_speed_tests.isolate', + ], + }, + ], + }], + ], +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate new file mode 100644 index 00000000000..8c5a2bd0ec0 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate @@ -0,0 +1,40 @@ +# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. +{ + 'conditions': [ + ['OS=="android"', { + 'variables': { + 'isolate_dependency_untracked': [ + '<(DEPTH)/resources/', + '<(DEPTH)/data/', + ], + }, + }], + ['OS=="linux" or OS=="mac" or OS=="win"', { + 'variables': { + 'command': [ + '<(DEPTH)/testing/test_env.py', + '<(PRODUCT_DIR)/audio_codec_speed_tests<(EXECUTABLE_SUFFIX)', + ], + 'isolate_dependency_touched': [ + '<(DEPTH)/DEPS', + ], + 'isolate_dependency_tracked': [ + '<(DEPTH)/resources/audio_coding/music_stereo_48kHz.pcm', + '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm', + '<(DEPTH)/resources/audio_coding/speech_mono_32_48kHz.pcm', + '<(DEPTH)/testing/test_env.py', + '<(PRODUCT_DIR)/audio_codec_speed_tests<(EXECUTABLE_SUFFIX)', + ], + 'isolate_dependency_untracked': [ + '<(DEPTH)/tools/swarming_client/', + ], + }, + }], + ], +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/main/OWNERS index e1e6256ca48..83880d21dc8 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/OWNERS +++ b/chromium/third_party/webrtc/modules/audio_coding/main/OWNERS @@ -1,3 +1,4 @@ tina.legrand@webrtc.org turaj@webrtc.org jan.skoglund@webrtc.org +henrik.lundin@webrtc.org diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc index fd30a137ae0..e55b6c4660f 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc @@ -20,7 +20,7 @@ #include <assert.h> #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/system_wrappers/interface/trace.h" // Includes needed to create the codecs. diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h index 98869efeead..65be793e37b 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h @@ -18,7 +18,7 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.h index 7216a574af8..6197a9d93ad 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.h @@ -12,6 +12,7 @@ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_ #include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" typedef struct WebRtcG722EncInst G722EncInst; typedef struct WebRtcG722DecInst G722DecInst; @@ -40,7 +41,8 @@ class ACMG722 : public ACMGenericCodec { int32_t Add10MsDataSafe(const uint32_t timestamp, const int16_t* data, const uint16_t length_smpl, - const uint8_t audio_channel); + const uint8_t audio_channel) + EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); void DestructEncoderSafe(); diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc index aa8e8be0637..a4808c0e384 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc @@ -26,7 +26,7 @@ namespace acm2 { // Enum for CNG enum { kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER, - kNewCNGNumPLCParams = 8 + kNewCNGNumLPCParams = 8 }; // Interval for sending new CNG parameters (SID frames) is 100 msec. @@ -56,9 +56,10 @@ ACMGenericCodec::ACMGenericCodec() vad_mode_(VADNormal), dtx_enabled_(false), ptr_dtx_inst_(NULL), - num_lpc_params_(kNewCNGNumPLCParams), + num_lpc_params_(kNewCNGNumLPCParams), sent_cn_previous_(false), prev_frame_cng_(0), + has_internal_fec_(false), neteq_decode_lock_(NULL), codec_wrapper_lock_(*RWLockWrapper::CreateRWLock()), last_timestamp_(0xD87F3F9F), @@ -546,7 +547,7 @@ void ACMGenericCodec::DestructEncoder() { WebRtcCng_FreeEnc(ptr_dtx_inst_); ptr_dtx_inst_ = NULL; } - num_lpc_params_ = kNewCNGNumPLCParams; + num_lpc_params_ = kNewCNGNumLPCParams; DestructEncoderSafe(); } diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h index d41580fff54..fa21ca015aa 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h @@ -13,9 +13,10 @@ #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/system_wrappers/interface/trace.h" #define MAX_FRAME_SIZE_10MSEC 6 @@ -560,6 +561,46 @@ class ACMGenericCodec { // virtual AudioDecoder* Decoder(int /* codec_id */) { return NULL; } + /////////////////////////////////////////////////////////////////////////// + // bool HasInternalFEC() + // Used to check if the codec has internal FEC. + // + // Return value: + // true if the codec has an internal FEC, e.g. Opus. + // false otherwise. + // + bool HasInternalFEC() const { return has_internal_fec_; } + + /////////////////////////////////////////////////////////////////////////// + // int SetFEC(); + // Sets the codec internal FEC. No effects on codecs that do not provide + // internal FEC. + // + // Input: + // -enable_fec : if true FEC will be enabled otherwise the FEC is + // disabled. + // + // Return value: + // -1 if failed, or the codec does not support FEC + // 0 if succeeded. + // + virtual int SetFEC(bool /* enable_fec */) { return -1; } + + /////////////////////////////////////////////////////////////////////////// + // int SetPacketLossRate() + // Sets expected packet loss rate for encoding. Some encoders provide packet + // loss gnostic encoding to make stream less sensitive to packet losses, + // through e.g., FEC. No effects on codecs that do not provide such encoding. + // + // Input: + // -loss_rate : expected packet loss rate (0 -- 100 inclusive). + // + // Return value: + // -1 if failed, + // 0 if succeeded or packet loss rate is ignored. + // + virtual int SetPacketLossRate(int /* loss_rate */) { return 0; } + protected: /////////////////////////////////////////////////////////////////////////// // All the functions with FunctionNameSafe(...) contain the actual @@ -576,7 +617,8 @@ class ACMGenericCodec { virtual int32_t Add10MsDataSafe(const uint32_t timestamp, const int16_t* data, const uint16_t length, - const uint8_t audio_channel); + const uint8_t audio_channel) + EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // See EncoderParam() for the description of function, input(s)/output(s) @@ -588,14 +630,15 @@ class ACMGenericCodec { // See ResetEncoder() for the description of function, input(s)/output(s) // and return value. // - int16_t ResetEncoderSafe(); + int16_t ResetEncoderSafe() EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // See InitEncoder() for the description of function, input(s)/output(s) // and return value. // int16_t InitEncoderSafe(WebRtcACMCodecParams* codec_params, - bool force_initialization); + bool force_initialization) + EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // See InitDecoder() for the description of function, input(s)/output(s) @@ -641,7 +684,8 @@ class ACMGenericCodec { // See SetVAD() for the description of function, input(s)/output(s) and // return value. // - int16_t SetVADSafe(bool* enable_dtx, bool* enable_vad, ACMVADMode* mode); + int16_t SetVADSafe(bool* enable_dtx, bool* enable_vad, ACMVADMode* mode) + EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // See ReplaceInternalDTX() for the description of function, input and @@ -678,7 +722,8 @@ class ACMGenericCodec { // -1 if failed, // 0 if succeeded. // - int16_t EnableVAD(ACMVADMode mode); + int16_t EnableVAD(ACMVADMode mode) + EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // int16_t DisableVAD() @@ -688,7 +733,7 @@ class ACMGenericCodec { // -1 if failed, // 0 if succeeded. // - int16_t DisableVAD(); + int16_t DisableVAD() EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // int16_t EnableDTX() @@ -699,7 +744,7 @@ class ACMGenericCodec { // -1 if failed, // 0 if succeeded. // - virtual int16_t EnableDTX(); + virtual int16_t EnableDTX() EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // int16_t DisableDTX() @@ -710,7 +755,7 @@ class ACMGenericCodec { // -1 if failed, // 0 if succeeded. // - virtual int16_t DisableDTX(); + virtual int16_t DisableDTX() EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // int16_t InternalEncode() @@ -838,7 +883,8 @@ class ACMGenericCodec { // int16_t ProcessFrameVADDTX(uint8_t* bitstream, int16_t* bitstream_len_byte, - int16_t* samples_processed); + int16_t* samples_processed) + EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_); /////////////////////////////////////////////////////////////////////////// // CurrentRate() @@ -885,19 +931,23 @@ class ACMGenericCodec { // True if the encoder instance initialized bool encoder_initialized_; - bool registered_in_neteq_; + const bool registered_in_neteq_; // TODO(henrik.lundin) Remove? // VAD/DTX bool has_internal_dtx_; - WebRtcVadInst* ptr_vad_inst_; - bool vad_enabled_; - ACMVADMode vad_mode_; - int16_t vad_label_[MAX_FRAME_SIZE_10MSEC]; - bool dtx_enabled_; - WebRtcCngEncInst* ptr_dtx_inst_; - uint8_t num_lpc_params_; - bool sent_cn_previous_; - int16_t prev_frame_cng_; + WebRtcVadInst* ptr_vad_inst_ GUARDED_BY(codec_wrapper_lock_); + bool vad_enabled_ GUARDED_BY(codec_wrapper_lock_); + ACMVADMode vad_mode_ GUARDED_BY(codec_wrapper_lock_); + int16_t vad_label_[MAX_FRAME_SIZE_10MSEC] GUARDED_BY(codec_wrapper_lock_); + bool dtx_enabled_ GUARDED_BY(codec_wrapper_lock_); + WebRtcCngEncInst* ptr_dtx_inst_ GUARDED_BY(codec_wrapper_lock_); + uint8_t num_lpc_params_ // TODO(henrik.lundin) Delete and + GUARDED_BY(codec_wrapper_lock_); // replace with kNewCNGNumLPCParams. + bool sent_cn_previous_ GUARDED_BY(codec_wrapper_lock_); + int16_t prev_frame_cng_ GUARDED_BY(codec_wrapper_lock_); + + // FEC. + bool has_internal_fec_; WebRtcACMCodecParams encoder_params_; @@ -909,7 +959,7 @@ class ACMGenericCodec { // such as buffers and state variables. RWLockWrapper& codec_wrapper_lock_; - uint32_t last_timestamp_; + uint32_t last_timestamp_ GUARDED_BY(codec_wrapper_lock_); uint32_t unique_id_; }; diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc index e27284212fd..9fbcdd4cd8b 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc @@ -14,7 +14,8 @@ #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" #ifdef WEBRTC_CODEC_ISAC @@ -59,14 +60,15 @@ static const int32_t kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = { #if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)) ACMISAC::ACMISAC(int16_t /* codec_id */) - : codec_inst_ptr_(NULL), + : codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + codec_inst_ptr_(NULL), is_enc_initialized_(false), isac_coding_mode_(CHANNEL_INDEPENDENT), enforce_frame_size_(false), isac_currentBN_(32000), samples_in10MsAudio_(160), // Initiates to 16 kHz mode. - audio_decoder_(NULL), - decoder_initialized_(false) {} + decoder_initialized_(false) { +} ACMISAC::~ACMISAC() { return; @@ -261,81 +263,14 @@ static uint16_t ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) { #endif -// Decoder class to be injected into NetEq. -class AcmAudioDecoderIsac : public AudioDecoder { - public: - AcmAudioDecoderIsac(int codec_id, void* state) - : AudioDecoder(ACMCodecDB::neteq_decoders_[codec_id]) { - state_ = state; - } - - // ACMISAC is the owner of the object where |state_| is pointing to. - // Therefore, it should not be deleted in this destructor. - virtual ~AcmAudioDecoderIsac() {} - - virtual int Decode(const uint8_t* encoded, size_t encoded_len, - int16_t* decoded, SpeechType* speech_type) { - int16_t temp_type; - int ret = ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(state_), - reinterpret_cast<const uint16_t*>(encoded), - static_cast<int16_t>(encoded_len), decoded, - &temp_type); - *speech_type = ConvertSpeechType(temp_type); - return ret; - } - - virtual bool HasDecodePlc() const { return true; } - - virtual int DecodePlc(int num_frames, int16_t* decoded) { - return ACM_ISAC_DECODEPLC(static_cast<ACM_ISAC_STRUCT*>(state_), - decoded, static_cast<int16_t>(num_frames)); - } - - virtual int Init() { - return 0; // We expect that the initialized instance is injected in the - // constructor. - } - - virtual int IncomingPacket(const uint8_t* payload, - size_t payload_len, - uint16_t rtp_sequence_number, - uint32_t rtp_timestamp, - uint32_t arrival_timestamp) { - return ACM_ISAC_DECODE_BWE(static_cast<ACM_ISAC_STRUCT*>(state_), - reinterpret_cast<const uint16_t*>(payload), - static_cast<uint32_t>(payload_len), - rtp_sequence_number, - rtp_timestamp, - arrival_timestamp); - } - - virtual int DecodeRedundant(const uint8_t* encoded, - size_t encoded_len, int16_t* decoded, - SpeechType* speech_type) { - int16_t temp_type = 1; // Default is speech. - int16_t ret = ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(state_), - reinterpret_cast<const uint16_t*>(encoded), - static_cast<int16_t>(encoded_len), decoded, - &temp_type); - *speech_type = ConvertSpeechType(temp_type); - return ret; - } - - virtual int ErrorCode() { - return ACM_ISAC_GETERRORCODE(static_cast<ACM_ISAC_STRUCT*>(state_)); - } - - private: - DISALLOW_COPY_AND_ASSIGN(AcmAudioDecoderIsac); -}; - ACMISAC::ACMISAC(int16_t codec_id) - : is_enc_initialized_(false), + : AudioDecoder(ACMCodecDB::neteq_decoders_[codec_id]), + codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + is_enc_initialized_(false), isac_coding_mode_(CHANNEL_INDEPENDENT), enforce_frame_size_(false), isac_current_bn_(32000), samples_in_10ms_audio_(160), // Initiates to 16 kHz mode. - audio_decoder_(NULL), decoder_initialized_(false) { codec_id_ = codec_id; @@ -345,14 +280,10 @@ ACMISAC::ACMISAC(int16_t codec_id) return; } codec_inst_ptr_->inst = NULL; + state_ = codec_inst_ptr_; } ACMISAC::~ACMISAC() { - if (audio_decoder_ != NULL) { - delete audio_decoder_; - audio_decoder_ = NULL; - } - if (codec_inst_ptr_ != NULL) { if (codec_inst_ptr_->inst != NULL) { ACM_ISAC_FREE(codec_inst_ptr_->inst); @@ -364,6 +295,34 @@ ACMISAC::~ACMISAC() { return; } +int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { + // set decoder sampling frequency. + if (codec_params->codec_inst.plfreq == 32000 || + codec_params->codec_inst.plfreq == 48000) { + UpdateDecoderSampFreq(ACMCodecDB::kISACSWB); + } else { + UpdateDecoderSampFreq(ACMCodecDB::kISAC); + } + + // in a one-way communication we may never register send-codec. + // However we like that the BWE to work properly so it has to + // be initialized. The BWE is initialized when iSAC encoder is initialized. + // Therefore, we need this. + if (!encoder_initialized_) { + // Since we don't require a valid rate or a valid packet size when + // initializing the decoder, we set valid values before initializing encoder + codec_params->codec_inst.rate = kIsacWbDefaultRate; + codec_params->codec_inst.pacsize = kIsacPacSize960; + if (InternalInitEncoder(codec_params) < 0) { + return -1; + } + encoder_initialized_ = true; + } + + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst); +} + ACMGenericCodec* ACMISAC::CreateInstance(void) { return NULL; } int16_t ACMISAC::InternalEncode(uint8_t* bitstream, @@ -375,6 +334,7 @@ int16_t ACMISAC::InternalEncode(uint8_t* bitstream, // at the first 10ms pushed in to iSAC if the bit-rate is low, this is // sort of a bug in iSAC. to address this we treat iSAC as the // following. + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -428,6 +388,7 @@ int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { if (UpdateEncoderSampFreq((uint16_t)codec_params->codec_inst.plfreq) < 0) { return -1; } + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) { return -1; } @@ -450,38 +411,8 @@ int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { return 0; } -int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { - if (codec_inst_ptr_ == NULL) { - return -1; - } - - // set decoder sampling frequency. - if (codec_params->codec_inst.plfreq == 32000 || - codec_params->codec_inst.plfreq == 48000) { - UpdateDecoderSampFreq(ACMCodecDB::kISACSWB); - } else { - UpdateDecoderSampFreq(ACMCodecDB::kISAC); - } - - // in a one-way communication we may never register send-codec. - // However we like that the BWE to work properly so it has to - // be initialized. The BWE is initialized when iSAC encoder is initialized. - // Therefore, we need this. - if (!encoder_initialized_) { - // Since we don't require a valid rate or a valid packet size when - // initializing the decoder, we set valid values before initializing encoder - codec_params->codec_inst.rate = kIsacWbDefaultRate; - codec_params->codec_inst.pacsize = kIsacPacSize960; - if (InternalInitEncoder(codec_params) < 0) { - return -1; - } - encoder_initialized_ = true; - } - - return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst); -} - int16_t ACMISAC::InternalCreateEncoder() { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -493,19 +424,6 @@ int16_t ACMISAC::InternalCreateEncoder() { return status; } -void ACMISAC::DestructEncoderSafe() { - // codec with shared instance cannot delete. - encoder_initialized_ = false; - return; -} - -void ACMISAC::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - ACM_ISAC_FREE(static_cast<ACM_ISAC_STRUCT *>(ptr_inst)); - } - return; -} - int16_t ACMISAC::Transcode(uint8_t* bitstream, int16_t* bitstream_len_byte, int16_t q_bwe, @@ -513,6 +431,7 @@ int16_t ACMISAC::Transcode(uint8_t* bitstream, bool is_red) { int16_t jitter_info = 0; // transcode from a higher rate to lower rate sanity check + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -530,7 +449,27 @@ int16_t ACMISAC::Transcode(uint8_t* bitstream, } } +void ACMISAC::UpdateFrameLen() { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); + encoder_params_.codec_inst.pacsize = frame_len_smpl_; +} + +void ACMISAC::DestructEncoderSafe() { + // codec with shared instance cannot delete. + encoder_initialized_ = false; + return; +} + +void ACMISAC::InternalDestructEncoderInst(void* ptr_inst) { + if (ptr_inst != NULL) { + ACM_ISAC_FREE(static_cast<ACM_ISAC_STRUCT *>(ptr_inst)); + } + return; +} + int16_t ACMISAC::SetBitRateSafe(int32_t bit_rate) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -594,6 +533,7 @@ int32_t ACMISAC::GetEstimatedBandwidthSafe() { int samp_rate; // Get bandwidth information + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); ACM_ISAC_GETSENDBWE(codec_inst_ptr_->inst, &bandwidth_index, &delay_index); // Validy check of index @@ -615,6 +555,7 @@ int32_t ACMISAC::SetEstimatedBandwidthSafe(int32_t estimated_bandwidth) { int16_t bandwidth_index; // Check sample frequency and choose appropriate table + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); samp_rate = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst); if (samp_rate == 16000) { @@ -657,6 +598,7 @@ int32_t ACMISAC::GetRedPayloadSafe( return -1; #else uint8_t* red_payload, int16_t* payload_bytes) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); int16_t bytes = WebRtcIsac_GetRedPayload( codec_inst_ptr_->inst, reinterpret_cast<int16_t*>(red_payload)); @@ -672,6 +614,7 @@ int16_t ACMISAC::UpdateDecoderSampFreq( #ifdef WEBRTC_CODEC_ISAC int16_t codec_id) { // The decoder supports only wideband and super-wideband. + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (ACMCodecDB::kISAC == codec_id) { return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 16000); } else if (ACMCodecDB::kISACSWB == codec_id || @@ -700,6 +643,7 @@ int16_t ACMISAC::UpdateEncoderSampFreq( in_audio_ix_read_ = 0; in_audio_ix_write_ = 0; in_timestamp_ix_write_ = 0; + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (WebRtcIsac_SetEncSampRate(codec_inst_ptr_->inst, encoder_samp_freq_hz) < 0) { return -1; @@ -718,6 +662,7 @@ int16_t ACMISAC::UpdateEncoderSampFreq( } int16_t ACMISAC::EncoderSampFreq(uint16_t* samp_freq_hz) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); *samp_freq_hz = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst); return 0; } @@ -730,6 +675,7 @@ int32_t ACMISAC::ConfigISACBandwidthEstimator( { uint16_t samp_freq_hz; EncoderSampFreq(&samp_freq_hz); + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); // TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce // the frame-size otherwise we might get error. Revise if // control-bwe is changed. @@ -749,26 +695,25 @@ int32_t ACMISAC::ConfigISACBandwidthEstimator( return -1; } UpdateFrameLen(); + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_); return 0; } int32_t ACMISAC::SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); return ACM_ISAC_SETMAXPAYLOADSIZE(codec_inst_ptr_->inst, max_payload_len_bytes); } int32_t ACMISAC::SetISACMaxRate(const uint32_t max_rate_bit_per_sec) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); return ACM_ISAC_SETMAXRATE(codec_inst_ptr_->inst, max_rate_bit_per_sec); } -void ACMISAC::UpdateFrameLen() { - frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); - encoder_params_.codec_inst.pacsize = frame_len_smpl_; -} - void ACMISAC::CurrentRate(int32_t* rate_bit_per_sec) { if (isac_coding_mode_ == ADAPTIVE) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, rate_bit_per_sec); } } @@ -784,12 +729,71 @@ int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate, return status; } -AudioDecoder* ACMISAC::Decoder(int codec_id) { - if (audio_decoder_) - return audio_decoder_; +int ACMISAC::Decode(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) { + int16_t temp_type; + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + int ret = + ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), + decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int ACMISAC::DecodePlc(int num_frames, int16_t* decoded) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_DECODEPLC( + static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + decoded, + static_cast<int16_t>(num_frames)); +} + +int ACMISAC::IncomingPacket(const uint8_t* payload, + size_t payload_len, + uint16_t rtp_sequence_number, + uint32_t rtp_timestamp, + uint32_t arrival_timestamp) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_DECODE_BWE( + static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + reinterpret_cast<const uint16_t*>(payload), + static_cast<uint32_t>(payload_len), + rtp_sequence_number, + rtp_timestamp, + arrival_timestamp); +} + +int ACMISAC::DecodeRedundant(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) { + int16_t temp_type = 1; // Default is speech. + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + int16_t ret = + ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), + decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int ACMISAC::ErrorCode() { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_GETERRORCODE( + static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst)); +} +AudioDecoder* ACMISAC::Decoder(int codec_id) { // Create iSAC instance if it does not exist. if (!encoder_exist_) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); assert(codec_inst_ptr_->inst == NULL); encoder_initialized_ = false; decoder_initialized_ = false; @@ -822,8 +826,7 @@ AudioDecoder* ACMISAC::Decoder(int codec_id) { decoder_initialized_ = true; } - audio_decoder_ = new AcmAudioDecoderIsac(codec_id, codec_inst_ptr_->inst); - return audio_decoder_; + return this; } #endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.h index a3227d5d0b3..3249526f7d8 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.h @@ -12,86 +12,119 @@ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_ #include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { +class CriticalSectionWrapper; + namespace acm2 { struct ACMISACInst; -class AcmAudioDecoderIsac; enum IsacCodingMode { ADAPTIVE, CHANNEL_INDEPENDENT }; -class ACMISAC : public ACMGenericCodec { +class ACMISAC : public ACMGenericCodec, AudioDecoder { public: explicit ACMISAC(int16_t codec_id); ~ACMISAC(); - // for FEC - ACMGenericCodec* CreateInstance(void); + int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params); - int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); + // Methods below are inherited from ACMGenericCodec. + ACMGenericCodec* CreateInstance(void) OVERRIDE; - int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); + int16_t InternalEncode(uint8_t* bitstream, + int16_t* bitstream_len_byte) OVERRIDE; - int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params); + int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params) OVERRIDE; - int16_t UpdateDecoderSampFreq(int16_t codec_id); + int16_t UpdateDecoderSampFreq(int16_t codec_id) OVERRIDE; - int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz); + int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE; - int16_t EncoderSampFreq(uint16_t* samp_freq_hz); + int16_t EncoderSampFreq(uint16_t* samp_freq_hz) OVERRIDE; int32_t ConfigISACBandwidthEstimator(const uint8_t init_frame_size_msec, const uint16_t init_rate_bit_per_sec, - const bool enforce_frame_size); + const bool enforce_frame_size) OVERRIDE; - int32_t SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes); + int32_t SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes) OVERRIDE; - int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec); + int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec) OVERRIDE; int16_t REDPayloadISAC(const int32_t isac_rate, const int16_t isac_bw_estimate, uint8_t* payload, - int16_t* payload_len_bytes); + int16_t* payload_len_bytes) OVERRIDE; - protected: - void DestructEncoderSafe(); + // Methods below are inherited from AudioDecoder. + virtual int Decode(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) OVERRIDE; - int16_t SetBitRateSafe(const int32_t bit_rate); + virtual bool HasDecodePlc() const OVERRIDE { return true; } - int32_t GetEstimatedBandwidthSafe(); + virtual int DecodePlc(int num_frames, int16_t* decoded) OVERRIDE; - int32_t SetEstimatedBandwidthSafe(int32_t estimated_bandwidth); + virtual int Init() OVERRIDE { return 0; } - int32_t GetRedPayloadSafe(uint8_t* red_payload, int16_t* payload_bytes); + virtual int IncomingPacket(const uint8_t* payload, + size_t payload_len, + uint16_t rtp_sequence_number, + uint32_t rtp_timestamp, + uint32_t arrival_timestamp) OVERRIDE; - int16_t InternalCreateEncoder(); + virtual int DecodeRedundant(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) OVERRIDE; - void InternalDestructEncoderInst(void* ptr_inst); + virtual int ErrorCode() OVERRIDE; + protected: int16_t Transcode(uint8_t* bitstream, int16_t* bitstream_len_byte, int16_t q_bwe, int32_t rate, bool is_red); - void CurrentRate(int32_t* rate_bit_per_sec); - void UpdateFrameLen(); - virtual AudioDecoder* Decoder(int codec_id); + // Methods below are inherited from ACMGenericCodec. + void DestructEncoderSafe() OVERRIDE; + + int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE; + + int32_t GetEstimatedBandwidthSafe() OVERRIDE; + + int32_t SetEstimatedBandwidthSafe(int32_t estimated_bandwidth) OVERRIDE; + + int32_t GetRedPayloadSafe(uint8_t* red_payload, + int16_t* payload_bytes) OVERRIDE; + + int16_t InternalCreateEncoder() OVERRIDE; + + void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; + + void CurrentRate(int32_t* rate_bit_per_sec) OVERRIDE; + + virtual AudioDecoder* Decoder(int codec_id) OVERRIDE; - ACMISACInst* codec_inst_ptr_; + // |codec_inst_crit_sect_| protects |codec_inst_ptr_|. + const scoped_ptr<CriticalSectionWrapper> codec_inst_crit_sect_; + ACMISACInst* codec_inst_ptr_ GUARDED_BY(codec_inst_crit_sect_); bool is_enc_initialized_; IsacCodingMode isac_coding_mode_; bool enforce_frame_size_; int32_t isac_current_bn_; uint16_t samples_in_10ms_audio_; - AcmAudioDecoderIsac* audio_decoder_; bool decoder_initialized_; }; diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc index c00a9203a9d..544c932f39f 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc @@ -75,6 +75,8 @@ ACMOpus::ACMOpus(int16_t codec_id) // Opus has internal DTX, but we dont use it for now. has_internal_dtx_ = false; + has_internal_fec_ = true; + if (codec_id_ != ACMCodecDB::kOpus) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "Wrong codec id for Opus."); @@ -140,6 +142,20 @@ int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { // Store bitrate. bitrate_ = codec_params->codec_inst.rate; + // TODO(tlegrand): Remove this code when we have proper APIs to set the + // complexity at a higher level. +#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) + // If we are on Android, iOS and/or ARM, use a lower complexity setting as + // default, to save encoder complexity. + const int kOpusComplexity5 = 5; + WebRtcOpus_SetComplexity(encoder_inst_ptr_, kOpusComplexity5); + if (ret < 0) { + WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, + "Setting complexity failed for Opus"); + return ret; + } +#endif + return 0; } @@ -184,6 +200,31 @@ int16_t ACMOpus::SetBitRateSafe(const int32_t rate) { return -1; } +int ACMOpus::SetFEC(bool enable_fec) { + // Ask the encoder to enable FEC. + if (enable_fec) { + if (WebRtcOpus_EnableFec(encoder_inst_ptr_) == 0) { + fec_enabled_ = true; + return 0; + } + } else { + if (WebRtcOpus_DisableFec(encoder_inst_ptr_) == 0) { + fec_enabled_ = false; + return 0; + } + } + return -1; +} + +int ACMOpus::SetPacketLossRate(int loss_rate) { + // Ask the encoder to change the target packet loss rate. + if (WebRtcOpus_SetPacketLossRate(encoder_inst_ptr_, loss_rate) == 0) { + packet_loss_rate_ = loss_rate; + return 0; + } + return -1; +} + #endif // WEBRTC_CODEC_OPUS } // namespace acm2 diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.h index a346e3c8ff3..07ce0721686 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.h @@ -32,6 +32,10 @@ class ACMOpus : public ACMGenericCodec { int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); + virtual int SetFEC(bool enable_fec) OVERRIDE; + + virtual int SetPacketLossRate(int loss_rate) OVERRIDE; + protected: void DestructEncoderSafe(); @@ -43,8 +47,11 @@ class ACMOpus : public ACMGenericCodec { WebRtcOpusEncInst* encoder_inst_ptr_; uint16_t sample_freq_; - uint16_t bitrate_; + int32_t bitrate_; int channels_; + + bool fec_enabled_; + int packet_loss_rate_; }; } // namespace acm2 diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc index ac92198f92b..cb7c4184079 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc @@ -21,8 +21,9 @@ #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" #include "webrtc/modules/audio_coding/main/acm2/nack.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" @@ -35,7 +36,6 @@ namespace acm2 { namespace { -const int kNeteqInitSampleRateHz = 16000; const int kNackThresholdPackets = 2; // |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_| @@ -117,21 +117,23 @@ bool IsCng(int codec_id) { } // namespace -AcmReceiver::AcmReceiver() - : id_(0), - neteq_(NetEq::Create(kNeteqInitSampleRateHz)), +AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) + : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + id_(config.id), last_audio_decoder_(-1), // Invalid value. - decode_lock_(RWLockWrapper::CreateRWLock()), - neteq_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vad_enabled_(true), - previous_audio_activity_(AudioFrame::kVadUnknown), - current_sample_rate_hz_(kNeteqInitSampleRateHz), + previous_audio_activity_(AudioFrame::kVadPassive), + current_sample_rate_hz_(config.neteq_config.sample_rate_hz), nack_(), nack_enabled_(false), + neteq_(NetEq::Create(config.neteq_config)), + decode_lock_(RWLockWrapper::CreateRWLock()), + vad_enabled_(true), + clock_(config.clock), av_sync_(false), initial_delay_manager_(), missing_packets_sync_stream_(), late_packets_sync_stream_() { + assert(clock_); for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) { decoders_[n].registered = false; } @@ -148,7 +150,6 @@ AcmReceiver::AcmReceiver() AcmReceiver::~AcmReceiver() { delete neteq_; delete decode_lock_; - delete neteq_crit_sect_; } int AcmReceiver::SetMinimumDelay(int delay_ms) { @@ -162,7 +163,7 @@ int AcmReceiver::SetInitialDelay(int delay_ms) { if (delay_ms < 0 || delay_ms > 10000) { return -1; } - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (delay_ms == 0) { av_sync_ = false; @@ -206,7 +207,7 @@ int AcmReceiver::LeastRequiredDelayMs() const { } int AcmReceiver::current_sample_rate_hz() const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); return current_sample_rate_hz_; } @@ -269,7 +270,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, const RTPHeader* header = &rtp_header.header; // Just a shorthand. { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); int codec_id = RtpHeaderToCodecIndex(*header, incoming_payload); if (codec_id < 0) { @@ -328,7 +329,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz, missing_packets_sync_stream_.get()); } - } + } // |crit_sect_| is released. { WriteLockScoped lock_codecs(*decode_lock_); // Lock to prevent an encoding. @@ -359,7 +360,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { { // Accessing members, take the lock. - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (av_sync_) { assert(initial_delay_manager_.get()); @@ -404,7 +405,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { } // Accessing members, take the lock. - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); // Update NACK. int decoded_sequence_num = 0; @@ -426,9 +427,13 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { if (ptr_audio_buffer == audio_buffer_) { // Data is written to local buffer. if (need_resampling) { - samples_per_channel = resampler_.Resample10Msec( - audio_buffer_, current_sample_rate_hz_, desired_freq_hz, - num_channels, audio_frame->data_); + samples_per_channel = + resampler_.Resample10Msec(audio_buffer_, + current_sample_rate_hz_, + desired_freq_hz, + num_channels, + AudioFrame::kMaxDataSizeSamples, + audio_frame->data_); if (samples_per_channel < 0) { LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed."; return -1; @@ -442,9 +447,13 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { // Data is written into |audio_frame|. if (need_resampling) { // We might end up here ONLY if codec is changed. - samples_per_channel = resampler_.Resample10Msec( - audio_frame->data_, current_sample_rate_hz_, desired_freq_hz, - num_channels, audio_buffer_); + samples_per_channel = + resampler_.Resample10Msec(audio_frame->data_, + current_sample_rate_hz_, + desired_freq_hz, + num_channels, + AudioFrame::kMaxDataSizeSamples, + audio_buffer_); if (samples_per_channel < 0) { LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed."; return -1; @@ -463,6 +472,19 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame); previous_audio_activity_ = audio_frame->vad_activity_; call_stats_.DecodedByNetEq(audio_frame->speech_type_); + + // Computes the RTP timestamp of the first sample in |audio_frame| from + // |GetPlayoutTimestamp|, which is the timestamp of the last sample of + // |audio_frame|. + uint32_t playout_timestamp = 0; + if (GetPlayoutTimestamp(&playout_timestamp)) { + audio_frame->timestamp_ = + playout_timestamp - audio_frame->samples_per_channel_; + } else { + // Remain 0 until we have a valid |playout_timestamp|. + audio_frame->timestamp_ = 0; + } + return 0; } @@ -473,19 +495,25 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id, assert(acm_codec_id >= 0 && acm_codec_id < ACMCodecDB::kMaxNumCodecs); NetEqDecoder neteq_decoder = ACMCodecDB::neteq_decoders_[acm_codec_id]; - CriticalSectionScoped lock(neteq_crit_sect_); + // Make sure the right decoder is registered for Opus. + if (neteq_decoder == kDecoderOpus && channels == 2) { + neteq_decoder = kDecoderOpus_2ch; + } + + CriticalSectionScoped lock(crit_sect_.get()); // The corresponding NetEq decoder ID. // If this coder has been registered before. if (decoders_[acm_codec_id].registered) { - if (decoders_[acm_codec_id].payload_type == payload_type) { + if (decoders_[acm_codec_id].payload_type == payload_type && + decoders_[acm_codec_id].channels == channels) { // Re-registering the same codec with the same payload-type. Do nothing // and return. return 0; } - // Changing the payload-type of this codec. First unregister. Then register - // with new payload-type. + // Changing the payload-type or number of channels for this codec. + // First unregister. Then register with new payload-type/channels. if (neteq_->RemovePayloadType(decoders_[acm_codec_id].payload_type) != NetEq::kOK) { LOG_F(LS_ERROR) << "Cannot remover payload " @@ -499,8 +527,7 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id, ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type); } else { ret_val = neteq_->RegisterExternalDecoder( - audio_decoder, neteq_decoder, - ACMCodecDB::database_[acm_codec_id].plfreq, payload_type); + audio_decoder, neteq_decoder, payload_type); } if (ret_val != NetEq::kOK) { LOG_FERR3(LS_ERROR, "AcmReceiver::AddCodec", acm_codec_id, payload_type, @@ -519,13 +546,13 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id, void AcmReceiver::EnableVad() { neteq_->EnableVad(); - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); vad_enabled_ = true; } void AcmReceiver::DisableVad() { neteq_->DisableVad(); - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); vad_enabled_ = false; } @@ -537,7 +564,7 @@ void AcmReceiver::FlushBuffers() { // many as it can. int AcmReceiver::RemoveAllCodecs() { int ret_val = 0; - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) { if (decoders_[n].registered) { if (neteq_->RemovePayloadType(decoders_[n].payload_type) == 0) { @@ -557,15 +584,13 @@ int AcmReceiver::RemoveAllCodecs() { int AcmReceiver::RemoveCodec(uint8_t payload_type) { int codec_index = PayloadType2CodecIndex(payload_type); if (codec_index < 0) { // Such a payload-type is not registered. - LOG(LS_WARNING) << "payload_type " << payload_type << " is not registered," - " no action is taken."; return 0; } if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) { LOG_FERR1(LS_ERROR, "AcmReceiver::RemoveCodec", payload_type); return -1; } - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); decoders_[codec_index].registered = false; if (last_audio_decoder_ == codec_index) last_audio_decoder_ = -1; // Codec is removed, invalidate last decoder. @@ -573,26 +598,27 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) { } void AcmReceiver::set_id(int id) { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); id_ = id; } -uint32_t AcmReceiver::PlayoutTimestamp() { +bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) { if (av_sync_) { assert(initial_delay_manager_.get()); - if (initial_delay_manager_->buffering()) - return initial_delay_manager_->playout_timestamp(); + if (initial_delay_manager_->buffering()) { + return initial_delay_manager_->GetPlayoutTimestamp(timestamp); + } } - return neteq_->PlayoutTimestamp(); + return neteq_->GetPlayoutTimestamp(timestamp); } int AcmReceiver::last_audio_codec_id() const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); return last_audio_decoder_; } int AcmReceiver::last_audio_payload_type() const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (last_audio_decoder_ < 0) return -1; assert(decoders_[last_audio_decoder_].registered); @@ -600,7 +626,7 @@ int AcmReceiver::last_audio_payload_type() const { } int AcmReceiver::RedPayloadType() const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (ACMCodecDB::kRED < 0 || !decoders_[ACMCodecDB::kRED].registered) { LOG_F(LS_WARNING) << "RED is not registered."; @@ -610,9 +636,8 @@ int AcmReceiver::RedPayloadType() const { } int AcmReceiver::LastAudioCodec(CodecInst* codec) const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (last_audio_decoder_ < 0) { - LOG_F(LS_WARNING) << "No audio payload is received, yet."; return -1; } assert(decoders_[last_audio_decoder_].registered); @@ -636,6 +661,7 @@ void AcmReceiver::NetworkStatistics(ACMNetworkStatistics* acm_stat) { acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate; acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate; acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm; + acm_stat->addedSamples = neteq_stat.added_zero_samples; std::vector<int> waiting_times; neteq_->WaitingTimes(&waiting_times); @@ -665,7 +691,7 @@ void AcmReceiver::NetworkStatistics(ACMNetworkStatistics* acm_stat) { int AcmReceiver::DecoderByPayloadType(uint8_t payload_type, CodecInst* codec) const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); int codec_index = PayloadType2CodecIndex(payload_type); if (codec_index < 0) { LOG_FERR1(LS_ERROR, "AcmReceiver::DecoderByPayloadType", payload_type); @@ -691,7 +717,7 @@ int AcmReceiver::EnableNack(size_t max_nack_list_size) { if (max_nack_list_size == 0 || max_nack_list_size > Nack::kNackListSizeLimit) return -1; - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (!nack_enabled_) { nack_.reset(Nack::Create(kNackThresholdPackets)); nack_enabled_ = true; @@ -707,14 +733,14 @@ int AcmReceiver::EnableNack(size_t max_nack_list_size) { } void AcmReceiver::DisableNack() { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); nack_.reset(); // Memory is released. nack_enabled_ = false; } std::vector<uint16_t> AcmReceiver::GetNackList( int round_trip_time_ms) const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); if (round_trip_time_ms < 0) { WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, "GetNackList: round trip time cannot be negative." @@ -730,7 +756,7 @@ std::vector<uint16_t> AcmReceiver::GetNackList( void AcmReceiver::ResetInitialDelay() { { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); av_sync_ = false; initial_delay_manager_.reset(NULL); missing_packets_sync_stream_.reset(NULL); @@ -752,13 +778,9 @@ bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) { // exceeds a threshold. int num_packets; int max_num_packets; - int buffer_size_byte; - int max_buffer_size_byte; const float kBufferingThresholdScale = 0.9f; - neteq_->PacketBufferStatistics(&num_packets, &max_num_packets, - &buffer_size_byte, &max_buffer_size_byte); - if (num_packets > max_num_packets * kBufferingThresholdScale || - buffer_size_byte > max_buffer_size_byte * kBufferingThresholdScale) { + neteq_->PacketBufferStatistics(&num_packets, &max_num_packets); + if (num_packets > max_num_packets * kBufferingThresholdScale) { initial_delay_manager_->DisableBuffering(); return false; } @@ -771,7 +793,6 @@ bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) { current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq; frame->num_channels_ = decoders_[last_audio_decoder_].channels; } else { - current_sample_rate_hz_ = kNeteqInitSampleRateHz; frame->num_channels_ = 1; } @@ -785,7 +806,6 @@ bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) { frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms. frame->speech_type_ = AudioFrame::kCNG; frame->vad_activity_ = AudioFrame::kVadPassive; - frame->energy_ = 0; int samples = frame->samples_per_channel_ * frame->num_channels_; memset(frame->data_, 0, samples * sizeof(int16_t)); return true; @@ -815,7 +835,7 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const { // We masked 6 most significant bits of 32-bit so there is no overflow in // the conversion from milliseconds to timestamp. const uint32_t now_in_ms = static_cast<uint32_t>( - TickTime::MillisecondTimestamp() & 0x03ffffff); + clock_->TimeInMilliseconds() & 0x03ffffff); return static_cast<uint32_t>( (decoder_sampling_rate / 1000) * now_in_ms); } @@ -839,7 +859,7 @@ void AcmReceiver::InsertStreamOfSyncPackets( void AcmReceiver::GetDecodingCallStatistics( AudioDecodingCallStats* stats) const { - CriticalSectionScoped lock(neteq_crit_sect_); + CriticalSectionScoped lock(crit_sect_.get()); *stats = call_stats_.GetDecodingStatistics(); } diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.h index 81eb5206b8a..b6898f73f9a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.h @@ -20,9 +20,10 @@ #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -47,9 +48,7 @@ class AcmReceiver { }; // Constructor of the class - AcmReceiver(); - - explicit AcmReceiver(NetEq* neteq); + explicit AcmReceiver(const AudioCodingModule::Config& config); // Destructor of the class. ~AcmReceiver(); @@ -244,9 +243,10 @@ class AcmReceiver { void set_id(int id); // TODO(turajs): can be inline. // - // Returns the RTP timestamp of the last sample delivered by GetAudio(). + // Gets the RTP timestamp of the last sample delivered by GetAudio(). + // Returns true if the RTP timestamp is valid, otherwise false. // - uint32_t PlayoutTimestamp(); + bool GetPlayoutTimestamp(uint32_t* timestamp); // // Return the index of the codec associated with the last non-CNG/non-DTMF @@ -328,7 +328,8 @@ class AcmReceiver { private: int PayloadType2CodecIndex(uint8_t payload_type) const; - bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame); + bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); int GetNumSyncPacketToInsert(uint16_t received_squence_number); @@ -339,20 +340,23 @@ class AcmReceiver { void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream); - int id_; + scoped_ptr<CriticalSectionWrapper> crit_sect_; + int id_; // TODO(henrik.lundin) Make const. + int last_audio_decoder_ GUARDED_BY(crit_sect_); + AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_); + int current_sample_rate_hz_ GUARDED_BY(crit_sect_); + ACMResampler resampler_ GUARDED_BY(crit_sect_); + // Used in GetAudio, declared as member to avoid allocating every 10ms. + // TODO(henrik.lundin) Stack-allocate in GetAudio instead? + int16_t audio_buffer_[AudioFrame::kMaxDataSizeSamples] GUARDED_BY(crit_sect_); + scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_); + bool nack_enabled_ GUARDED_BY(crit_sect_); + CallStatistics call_stats_ GUARDED_BY(crit_sect_); NetEq* neteq_; Decoder decoders_[ACMCodecDB::kMaxNumCodecs]; - int last_audio_decoder_; RWLockWrapper* decode_lock_; - CriticalSectionWrapper* neteq_crit_sect_; bool vad_enabled_; - AudioFrame::VADActivity previous_audio_activity_; - int current_sample_rate_hz_; - ACMResampler resampler_; - // Used in GetAudio, declared as member to avoid allocating every 10ms. - int16_t audio_buffer_[AudioFrame::kMaxDataSizeSamples]; - scoped_ptr<Nack> nack_; - bool nack_enabled_; + Clock* clock_; // TODO(henrik.lundin) Make const if possible. // Indicates if a non-zero initial delay is set, and the receiver is in // AV-sync mode. @@ -366,8 +370,6 @@ class AcmReceiver { // initial delay is set. scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_; scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_; - - CallStatistics call_stats_; }; } // namespace acm2 diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc index 712eeb26877..4234f146474 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc @@ -16,7 +16,8 @@ #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" -#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/test_suite.h" #include "webrtc/test/testsupport/fileutils.h" @@ -42,12 +43,14 @@ class AcmReceiverTest : public AudioPacketizationCallback, public ::testing::Test { protected: AcmReceiverTest() - : receiver_(new AcmReceiver), - acm_(new AudioCodingModuleImpl(0)), - timestamp_(0), + : timestamp_(0), packet_sent_(false), last_packet_send_timestamp_(timestamp_), - last_frame_type_(kFrameEmpty) {} + last_frame_type_(kFrameEmpty) { + AudioCodingModule::Config config; + acm_.reset(new AudioCodingModuleImpl(config)); + receiver_.reset(new AcmReceiver(config)); + } ~AcmReceiverTest() {} @@ -302,55 +305,6 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PostdecodingVad)) { EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_); } -TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(FlushBuffer)) { - const int id = ACMCodecDB::kISAC; - EXPECT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels, - NULL)); - const int kNumPackets = 5; - const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100); - for (int n = 0; n < kNumPackets; ++n) - InsertOnePacketOfSilence(id); - ACMNetworkStatistics statistics; - receiver_->NetworkStatistics(&statistics); - ASSERT_EQ(num_10ms_frames * kNumPackets * 10, statistics.currentBufferSize); - - receiver_->FlushBuffers(); - receiver_->NetworkStatistics(&statistics); - ASSERT_EQ(0, statistics.currentBufferSize); -} - -TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PlayoutTimestamp)) { - const int id = ACMCodecDB::kPCM16Bwb; - EXPECT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels, - NULL)); - receiver_->SetPlayoutMode(fax); - const int kNumPackets = 5; - const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100); - uint32_t expected_timestamp; - AudioFrame frame; - int ts_offset = 0; - bool first_audio_frame = true; - for (int n = 0; n < kNumPackets; ++n) { - packet_sent_ = false; - InsertOnePacketOfSilence(id); - ASSERT_TRUE(packet_sent_); - expected_timestamp = last_packet_send_timestamp_; - for (int k = 0; k < num_10ms_frames; ++k) { - ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame)); - if (first_audio_frame) { - // There is an offset in playout timestamps. Perhaps, it is related to - // initial delay that NetEq applies - ts_offset = receiver_->PlayoutTimestamp() - expected_timestamp; - first_audio_frame = false; - } else { - EXPECT_EQ(expected_timestamp + ts_offset, - receiver_->PlayoutTimestamp()); - } - expected_timestamp += codecs_[id].plfreq / 100; // Increment by 10 ms. - } - } -} - TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { const int kCodecId[] = { ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB, diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc index 3abe4f1ec46..97d87b1b3a4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc @@ -10,61 +10,59 @@ #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" +#include <assert.h> #include <string.h> #include "webrtc/common_audio/resampler/include/resampler.h" -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { - namespace acm2 { -ACMResampler::ACMResampler() - : resampler_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) { +ACMResampler::ACMResampler() { } ACMResampler::~ACMResampler() { - delete resampler_crit_sect_; } int ACMResampler::Resample10Msec(const int16_t* in_audio, int in_freq_hz, int out_freq_hz, int num_audio_channels, + int out_capacity_samples, int16_t* out_audio) { - CriticalSectionScoped cs(resampler_crit_sect_); - + int in_length = in_freq_hz * num_audio_channels / 100; + int out_length = out_freq_hz * num_audio_channels / 100; if (in_freq_hz == out_freq_hz) { - size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100); - memcpy(out_audio, in_audio, length * sizeof(int16_t)); - return static_cast<int16_t>(in_freq_hz / 100); + if (out_capacity_samples < in_length) { + assert(false); + return -1; + } + memcpy(out_audio, in_audio, in_length * sizeof(int16_t)); + return in_length / num_audio_channels; } - // |maxLen| is maximum number of samples for 10ms at 48kHz. - int max_len = 480 * num_audio_channels; - int length_in = (in_freq_hz / 100) * num_audio_channels; - int out_len; - - ResamplerType type = (num_audio_channels == 1) ? kResamplerSynchronous : - kResamplerSynchronousStereo; - - if (resampler_.ResetIfNeeded(in_freq_hz, out_freq_hz, type) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0, - "Error in reset of resampler"); + if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz, + num_audio_channels) != 0) { + LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz, + num_audio_channels); return -1; } - if (resampler_.Push(in_audio, length_in, out_audio, max_len, out_len) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0, - "Error in resampler: resampler.Push"); + out_length = + resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); + if (out_length == -1) { + LOG_FERR4(LS_ERROR, + Resample, + in_audio, + in_length, + out_audio, + out_capacity_samples); return -1; } - return out_len / num_audio_channels; + return out_length / num_audio_channels; } } // namespace acm2 - } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.h index e992955f5f3..a8fc6b6f26a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.h @@ -11,13 +11,10 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ -#include "webrtc/common_audio/resampler/include/resampler.h" +#include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/typedefs.h" namespace webrtc { - -class CriticalSectionWrapper; - namespace acm2 { class ACMResampler { @@ -29,16 +26,14 @@ class ACMResampler { int in_freq_hz, int out_freq_hz, int num_audio_channels, + int out_capacity_samples, int16_t* out_audio); private: - // Use the Resampler class. - Resampler resampler_; - CriticalSectionWrapper* resampler_crit_sect_; + PushResampler<int16_t> resampler_; }; } // namespace acm2 - } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc index 60ed69cb29c..eca909cc49c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc @@ -13,22 +13,21 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" -#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { -const char kLegacyAcmVersion[] = "acm1"; -const char kExperimentalAcmVersion[] = "acm2"; - // Create module AudioCodingModule* AudioCodingModule::Create(int id) { - return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock()); + return Create(id, Clock::GetRealTimeClock()); } AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) { - return new acm1::AudioCodingModuleImpl(id, clock); + AudioCodingModule::Config config; + config.id = id; + config.clock = clock; + return new acm2::AudioCodingModuleImpl(config); } // Get number of supported codecs @@ -95,13 +94,4 @@ bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { } } -AudioCodingModule* AudioCodingModuleFactory::Create(int id) const { - return new acm1::AudioCodingModuleImpl(static_cast<int32_t>(id), - Clock::GetRealTimeClock()); -} - -AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const { - return new acm2::AudioCodingModuleImpl(id); -} - } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi index f51c3bf7d74..90dad6c55c0 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi @@ -7,16 +7,36 @@ # be found in the AUTHORS file in the root of the source tree. { + 'variables': { + 'audio_coding_dependencies': [ + 'CNG', + 'G711', + 'G722', + 'iLBC', + 'iSAC', + 'iSACFix', + 'PCM16B', + '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + ], + 'audio_coding_defines': [], + 'conditions': [ + ['include_opus==1', { + 'audio_coding_dependencies': ['webrtc_opus',], + 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',], + }], + ], + }, 'targets': [ { - 'target_name': 'acm2', + 'target_name': 'audio_coding_module', 'type': 'static_library', 'defines': [ '<@(audio_coding_defines)', ], 'dependencies': [ '<@(audio_coding_dependencies)', - 'NetEq4', + 'neteq', ], 'include_dirs': [ '../interface', @@ -93,4 +113,45 @@ ], }, ], + 'conditions': [ + ['include_tests==1', { + 'targets': [ + { + 'target_name': 'delay_test', + 'type': 'executable', + 'dependencies': [ + 'audio_coding_module', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/test/test.gyp:test_support', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + ], + 'sources': [ + '../test/delay_test.cc', + '../test/Channel.cc', + '../test/PCMFile.cc', + '../test/utility.cc', + ], + }, # delay_test + { + 'target_name': 'insert_packet_with_timing', + 'type': 'executable', + 'dependencies': [ + 'audio_coding_module', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/test/test.gyp:test_support', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + ], + 'sources': [ + '../test/insert_packet_with_timing.cc', + '../test/Channel.cc', + '../test/PCMFile.cc', + ], + }, # delay_test + ], + }], + ], } diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index 4c64e07dd5c..a07e8543347 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -39,11 +39,11 @@ enum { kMaxPacketSize = 2560 }; -// Maximum number of payloads that can be packed in one RED payload. For -// regular FEC, we only pack two payloads. In case of dual-streaming, in worst -// case we might pack 3 payloads in one RED payload. +// Maximum number of payloads that can be packed in one RED packet. For +// regular RED, we only pack two payloads. In case of dual-streaming, in worst +// case we might pack 3 payloads in one RED packet. enum { - kNumFecFragmentationVectors = 2, + kNumRedFragmentationVectors = 2, kMaxNumFragmentationVectors = 3 }; @@ -114,9 +114,10 @@ static int TimestampLessThan(uint32_t t1, uint32_t t2) { } // namespace -AudioCodingModuleImpl::AudioCodingModuleImpl(int id) - : packetization_callback_(NULL), - id_(id), +AudioCodingModuleImpl::AudioCodingModuleImpl( + const AudioCodingModule::Config& config) + : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + id_(config.id), expected_codec_ts_(0xD87F3F9F), expected_in_ts_(0xD87F3F9F), send_codec_inst_(), @@ -131,18 +132,20 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id) stereo_send_(false), current_send_codec_idx_(-1), send_codec_registered_(false), - acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vad_callback_(NULL), + receiver_(config), is_first_red_(true), - fec_enabled_(false), - last_fec_timestamp_(0), + red_enabled_(false), + last_red_timestamp_(0), + codec_fec_enabled_(false), previous_pltype_(255), aux_rtp_header_(NULL), receiver_initialized_(false), - callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), secondary_send_codec_inst_(), codec_timestamp_(expected_codec_ts_), - first_10ms_data_(false) { + first_10ms_data_(false), + callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + packetization_callback_(NULL), + vad_callback_(NULL) { // Nullify send codec memory, set payload type and set codec name to // invalid values. @@ -159,8 +162,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id) mirror_codec_idx_[i] = -1; } - receiver_.set_id(id_); - // Allocate memory for RED. red_buffer_ = new uint8_t[MAX_PAYLOAD_SIZE_BYTE]; @@ -201,7 +202,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id) WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, "Cannot initialize receiver"); } - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created"); + WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created"); } AudioCodingModuleImpl::~AudioCodingModuleImpl() { @@ -349,7 +350,7 @@ int AudioCodingModuleImpl::ProcessDualStream() { int16_t len_bytes = MAX_PAYLOAD_SIZE_BYTE; WebRtcACMEncodingType encoding_type; if (secondary_encoder_->Encode(red_buffer_, &len_bytes, - &last_fec_timestamp_, + &last_red_timestamp_, &encoding_type) < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, "ProcessDual(): Encoding of secondary encoder Failed"); @@ -372,7 +373,7 @@ int AudioCodingModuleImpl::ProcessDualStream() { index_primary = secondary_ready_to_encode ? TimestampLessThan(primary_timestamp, secondary_timestamp) : 0; index_primary += has_previous_payload ? - TimestampLessThan(primary_timestamp, last_fec_timestamp_) : 0; + TimestampLessThan(primary_timestamp, last_red_timestamp_) : 0; } if (secondary_ready_to_encode) { @@ -384,7 +385,7 @@ int AudioCodingModuleImpl::ProcessDualStream() { if (has_previous_payload) { index_previous_secondary = primary_ready_to_encode ? - (1 - TimestampLessThan(primary_timestamp, last_fec_timestamp_)) : 0; + (1 - TimestampLessThan(primary_timestamp, last_red_timestamp_)) : 0; // If secondary is ready it always have a timestamp larger than previous // secondary. So the index is either 0 or 1. index_previous_secondary += secondary_ready_to_encode ? 1 : 0; @@ -405,7 +406,7 @@ int AudioCodingModuleImpl::ProcessDualStream() { } else if (index_secondary == 0) { current_timestamp = secondary_timestamp; } else { - current_timestamp = last_fec_timestamp_; + current_timestamp = last_red_timestamp_; } fragmentation_.fragmentationVectorSize = 0; @@ -420,7 +421,7 @@ int AudioCodingModuleImpl::ProcessDualStream() { fragmentation_.fragmentationPlType[index_previous_secondary] = secondary_send_codec_inst_.pltype; fragmentation_.fragmentationTimeDiff[index_previous_secondary] = - static_cast<uint16_t>(current_timestamp - last_fec_timestamp_); + static_cast<uint16_t>(current_timestamp - last_red_timestamp_); fragmentation_.fragmentationVectorSize++; } @@ -462,7 +463,7 @@ int AudioCodingModuleImpl::ProcessDualStream() { { CriticalSectionScoped lock(callback_crit_sect_); if (packetization_callback_ != NULL) { - // Callback with payload data, including redundant data (FEC/RED). + // Callback with payload data, including redundant data (RED). if (packetization_callback_->SendData(kAudioFrameSpeech, my_red_payload_type, current_timestamp, stream, @@ -495,7 +496,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() { FrameType frame_type = kAudioFrameSpeech; uint8_t current_payload_type = 0; bool has_data_to_send = false; - bool fec_active = false; + bool red_active = false; RTPFragmentationHeader my_fragmentation; // Keep the scope of the ACM critical section limited. @@ -562,15 +563,15 @@ int AudioCodingModuleImpl::ProcessSingleStream() { // Redundancy encode is done here. The two bitstreams packetized into // one RTP packet and the fragmentation points are set. // Only apply RED on speech data. - if ((fec_enabled_) && + if ((red_enabled_) && ((encoding_type == kActiveNormalEncoded) || (encoding_type == kPassiveNormalEncoded))) { - // FEC is enabled within this scope. + // RED is enabled within this scope. // // Note that, a special solution exists for iSAC since it is the only // codec for which GetRedPayload has a non-empty implementation. // - // Summary of the FEC scheme below (use iSAC as example): + // Summary of the RED scheme below (use iSAC as example): // // 1st (is_first_red_ is true) encoded iSAC frame (primary #1) => // - call GetRedPayload() and store redundancy for packet #1 in @@ -581,7 +582,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() { // - store primary #2 in 1st fragment of RED buffer and send the // combined packet // - the transmitted packet contains primary #2 (new) and - // reduncancy for packet #1 (old) + // redundancy for packet #1 (old) // - call GetRed_Payload() and store redundancy for packet #2 in // second fragment of RED buffer // @@ -604,19 +605,19 @@ int AudioCodingModuleImpl::ProcessSingleStream() { // // Hence, even if every second packet is dropped, perfect // reconstruction is possible. - fec_active = true; + red_active = true; has_data_to_send = false; // Skip the following part for the first packet in a RED session. if (!is_first_red_) { - // Rearrange stream such that FEC packets are included. + // Rearrange stream such that RED packets are included. // Replace stream now that we have stored current stream. memcpy(stream + fragmentation_.fragmentationOffset[1], red_buffer_, fragmentation_.fragmentationLength[1]); // Update the fragmentation time difference vector, in number of // timestamps. uint16_t time_since_last = static_cast<uint16_t>( - rtp_timestamp - last_fec_timestamp_); + rtp_timestamp - last_red_timestamp_); // Update fragmentation vectors. fragmentation_.fragmentationPlType[1] = @@ -630,7 +631,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() { // Insert new packet payload type. fragmentation_.fragmentationPlType[0] = current_payload_type; - last_fec_timestamp_ = rtp_timestamp; + last_red_timestamp_ = rtp_timestamp; // Can be modified by the GetRedPayload() call if iSAC is utilized. red_length_bytes = length_bytes; @@ -650,7 +651,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() { if (codecs_[current_send_codec_idx_]->GetRedPayload( red_buffer_, &red_length_bytes) == -1) { // The codec was not iSAC => use current encoder output as redundant - // data instead (trivial FEC scheme). + // data instead (trivial RED scheme). memcpy(red_buffer_, stream, red_length_bytes); } @@ -658,7 +659,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() { // Update payload type with RED payload type. current_payload_type = red_pltype_; // We have packed 2 payloads. - fragmentation_.fragmentationVectorSize = kNumFecFragmentationVectors; + fragmentation_.fragmentationVectorSize = kNumRedFragmentationVectors; // Copy to local variable, as it will be used outside ACM lock. my_fragmentation.CopyFrom(fragmentation_); @@ -672,8 +673,8 @@ int AudioCodingModuleImpl::ProcessSingleStream() { CriticalSectionScoped lock(callback_crit_sect_); if (packetization_callback_ != NULL) { - if (fec_active) { - // Callback with payload data, including redundant data (FEC/RED). + if (red_active) { + // Callback with payload data, including redundant data (RED). packetization_callback_->SendData(frame_type, current_payload_type, rtp_timestamp, stream, length_bytes, &my_fragmentation); @@ -713,14 +714,14 @@ int AudioCodingModuleImpl::InitializeSender() { } } - // Initialize FEC/RED. + // Initialize RED. is_first_red_ = true; - if (fec_enabled_ || secondary_encoder_.get() != NULL) { + if (red_enabled_ || secondary_encoder_.get() != NULL) { if (red_buffer_ != NULL) { memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); } - if (fec_enabled_) { - ResetFragmentation(kNumFecFragmentationVectors); + if (red_enabled_) { + ResetFragmentation(kNumRedFragmentationVectors); } else { ResetFragmentation(0); } @@ -1031,10 +1032,20 @@ int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) { // Everything is fine so we can replace the previous codec with this one. if (send_codec_registered_) { - // If we change codec we start fresh with FEC. + // If we change codec we start fresh with RED. // This is not strictly required by the standard. is_first_red_ = true; codec_ptr->SetVAD(&dtx_enabled_, &vad_enabled_, &vad_mode_); + + if (!codec_ptr->HasInternalFEC()) { + codec_fec_enabled_ = false; + } else { + if (codec_ptr->SetFEC(codec_fec_enabled_) < 0) { + WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, + "Cannot set codec FEC"); + return -1; + } + } } current_send_codec_idx_ = codec_id; @@ -1120,8 +1131,18 @@ int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) { } send_codec_inst_.rate = send_codec.rate; } - previous_pltype_ = send_codec_inst_.pltype; + if (!codecs_[codec_id]->HasInternalFEC()) { + codec_fec_enabled_ = false; + } else { + if (codecs_[codec_id]->SetFEC(codec_fec_enabled_) < 0) { + WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, + "Cannot set codec FEC"); + return -1; + } + } + + previous_pltype_ = send_codec_inst_.pltype; return 0; } } @@ -1205,11 +1226,7 @@ int AudioCodingModuleImpl::Add10MsData( return -1; } - // Allow for 8, 16, 32 and 48kHz input audio. - if ((audio_frame.sample_rate_hz_ != 8000) - && (audio_frame.sample_rate_hz_ != 16000) - && (audio_frame.sample_rate_hz_ != 32000) - && (audio_frame.sample_rate_hz_ != 48000)) { + if (audio_frame.sample_rate_hz_ > 48000) { assert(false); WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, "Cannot Add 10 ms audio, input frequency not valid"); @@ -1365,13 +1382,17 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, // The result of the resampler is written to output frame. dest_ptr_audio = preprocess_frame_.data_; - preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec( - src_ptr_audio, in_frame.sample_rate_hz_, send_codec_inst_.plfreq, - preprocess_frame_.num_channels_, dest_ptr_audio); + preprocess_frame_.samples_per_channel_ = + resampler_.Resample10Msec(src_ptr_audio, + in_frame.sample_rate_hz_, + send_codec_inst_.plfreq, + preprocess_frame_.num_channels_, + AudioFrame::kMaxDataSizeSamples, + dest_ptr_audio); if (preprocess_frame_.samples_per_channel_ < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add 10 ms audio, resmapling failed"); + "Cannot add 10 ms audio, resampling failed"); return -1; } preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq; @@ -1384,42 +1405,87 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, } ///////////////////////////////////////// -// (FEC) Forward Error Correction +// (RED) Redundant Coding // -bool AudioCodingModuleImpl::FECStatus() const { +bool AudioCodingModuleImpl::REDStatus() const { CriticalSectionScoped lock(acm_crit_sect_); - return fec_enabled_; + + return red_enabled_; } -// Configure FEC status i.e on/off. -int AudioCodingModuleImpl::SetFECStatus( +// Configure RED status i.e on/off. +int AudioCodingModuleImpl::SetREDStatus( #ifdef WEBRTC_CODEC_RED - bool enable_fec) { + bool enable_red) { CriticalSectionScoped lock(acm_crit_sect_); - if (fec_enabled_ != enable_fec) { + if (enable_red == true && codec_fec_enabled_ == true) { + WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, + "Codec internal FEC and RED cannot be co-enabled."); + return -1; + } + + if (red_enabled_ != enable_red) { // Reset the RED buffer. memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); // Reset fragmentation buffers. - ResetFragmentation(kNumFecFragmentationVectors); - // Set fec_enabled_. - fec_enabled_ = enable_fec; + ResetFragmentation(kNumRedFragmentationVectors); + // Set red_enabled_. + red_enabled_ = enable_red; } - is_first_red_ = true; // Make sure we restart FEC. + is_first_red_ = true; // Make sure we restart RED. return 0; #else - bool /* enable_fec */) { - fec_enabled_ = false; + bool /* enable_red */) { + red_enabled_ = false; WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - " WEBRTC_CODEC_RED is undefined => fec_enabled_ = %d", - fec_enabled_); + " WEBRTC_CODEC_RED is undefined => red_enabled_ = %d", + red_enabled_); return -1; #endif } ///////////////////////////////////////// +// (FEC) Forward Error Correction (codec internal) +// + +bool AudioCodingModuleImpl::CodecFEC() const { + return codec_fec_enabled_; +} + +int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) { + CriticalSectionScoped lock(acm_crit_sect_); + + if (enable_codec_fec == true && red_enabled_ == true) { + WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, + "Codec internal FEC and RED cannot be co-enabled."); + return -1; + } + + // Set codec FEC. + if (HaveValidEncoder("SetCodecFEC") && + codecs_[current_send_codec_idx_]->SetFEC(enable_codec_fec) < 0) { + WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, + "Set codec internal FEC failed."); + return -1; + } + codec_fec_enabled_ = enable_codec_fec; + return 0; +} + +int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { + if (HaveValidEncoder("SetPacketLossRate") && + codecs_[current_send_codec_idx_]->SetPacketLossRate(loss_rate) < 0) { + WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, + "Set packet loss rate failed."); + return -1; + } + return 0; +} + +///////////////////////////////////////// // (VAD) Voice Activity Detection // int AudioCodingModuleImpl::SetVAD(bool enable_dtx, @@ -1710,8 +1776,6 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz, } audio_frame->id_ = id_; - audio_frame->energy_ = 0; - audio_frame->timestamp_ = 0; return 0; } @@ -1770,6 +1834,7 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload, aux_rtp_header_->type.Audio.channel = 1; } + aux_rtp_header_->header.timestamp = timestamp; IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_); // Get ready for the next payload. aux_rtp_header_->header.sequenceNumber++; @@ -1851,8 +1916,7 @@ int AudioCodingModuleImpl::ConfigISACBandwidthEstimator( } int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) { - *timestamp = receiver_.PlayoutTimestamp(); - return 0; + return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1; } bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const { @@ -1976,10 +2040,6 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const { return receiver_.LeastRequiredDelayMs(); } -const char* AudioCodingModuleImpl::Version() const { - return kExperimentalAcmVersion; -} - void AudioCodingModuleImpl::GetDecodingCallStatistics( AudioDecodingCallStats* call_stats) const { receiver_.GetDecodingCallStatistics(call_stats); diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h index bc4ea0f7a66..e54202bf6da 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h @@ -19,11 +19,11 @@ #include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { class CriticalSectionWrapper; -class RWLockWrapper; namespace acm2 { @@ -32,11 +32,9 @@ class ACMGenericCodec; class AudioCodingModuleImpl : public AudioCodingModule { public: - explicit AudioCodingModuleImpl(int id); + explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config); ~AudioCodingModuleImpl(); - virtual const char* Version() const; - // Change the unique identifier of this object. virtual int32_t ChangeUniqueId(const int32_t id); @@ -94,14 +92,27 @@ class AudioCodingModuleImpl : public AudioCodingModule { int Add10MsData(const AudioFrame& audio_frame); ///////////////////////////////////////// - // (FEC) Forward Error Correction + // (RED) Redundant Coding + // + + // Configure RED status i.e. on/off. + int SetREDStatus(bool enable_red); + + // Get RED status. + bool REDStatus() const; + + ///////////////////////////////////////// + // (FEC) Forward Error Correction (codec internal) // - // Configure FEC status i.e on/off. - int SetFECStatus(bool enable_fec); + // Configure FEC status i.e. on/off. + int SetCodecFEC(bool enabled_codec_fec); // Get FEC status. - bool FECStatus() const; + bool CodecFEC() const; + + // Set target packet loss rate + int SetPacketLossRate(int loss_rate); ///////////////////////////////////////// // (VAD) Voice Activity Detection @@ -235,13 +246,14 @@ class AudioCodingModuleImpl : public AudioCodingModule { ACMGenericCodec* CreateCodec(const CodecInst& codec); - int InitializeReceiverSafe(); + int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); bool HaveValidEncoder(const char* caller_name) const; // Set VAD/DTX status. This function does not acquire a lock, and it is // created to be called only from inside a critical section. - int SetVADSafe(bool enable_dtx, bool enable_vad, ACMVADMode mode); + int SetVADSafe(bool enable_dtx, bool enable_vad, ACMVADMode mode) + EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); // Process buffered audio when dual-streaming is not enabled (When RED is // enabled still this function is used.) @@ -263,18 +275,22 @@ class AudioCodingModuleImpl : public AudioCodingModule { // -1: if encountering an error. // 0: otherwise. int PreprocessToAddData(const AudioFrame& in_frame, - const AudioFrame** ptr_out); + const AudioFrame** ptr_out) + EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); // Change required states after starting to receive the codec corresponding // to |index|. int UpdateUponReceivingCodec(int index); - int EncodeFragmentation(int fragmentation_index, int payload_type, + int EncodeFragmentation(int fragmentation_index, + int payload_type, uint32_t current_timestamp, ACMGenericCodec* encoder, - uint8_t* stream); + uint8_t* stream) + EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); - void ResetFragmentation(int vector_size); + void ResetFragmentation(int vector_size) + EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); // Get a pointer to AudioDecoder of the given codec. For some codecs, e.g. // iSAC, encoding and decoding have to be performed on a shared @@ -289,50 +305,50 @@ class AudioCodingModuleImpl : public AudioCodingModule { int GetAudioDecoder(const CodecInst& codec, int codec_id, int mirror_id, AudioDecoder** decoder); - AudioPacketizationCallback* packetization_callback_; - - int id_; - uint32_t expected_codec_ts_; - uint32_t expected_in_ts_; - CodecInst send_codec_inst_; - - uint8_t cng_nb_pltype_; - uint8_t cng_wb_pltype_; - uint8_t cng_swb_pltype_; - uint8_t cng_fb_pltype_; - - uint8_t red_pltype_; - bool vad_enabled_; - bool dtx_enabled_; - ACMVADMode vad_mode_; + CriticalSectionWrapper* acm_crit_sect_; + int id_; // TODO(henrik.lundin) Make const. + uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_); + uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_); + CodecInst send_codec_inst_ GUARDED_BY(acm_crit_sect_); + + uint8_t cng_nb_pltype_ GUARDED_BY(acm_crit_sect_); + uint8_t cng_wb_pltype_ GUARDED_BY(acm_crit_sect_); + uint8_t cng_swb_pltype_ GUARDED_BY(acm_crit_sect_); + uint8_t cng_fb_pltype_ GUARDED_BY(acm_crit_sect_); + + uint8_t red_pltype_ GUARDED_BY(acm_crit_sect_); + bool vad_enabled_ GUARDED_BY(acm_crit_sect_); + bool dtx_enabled_ GUARDED_BY(acm_crit_sect_); + ACMVADMode vad_mode_ GUARDED_BY(acm_crit_sect_); ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs]; int mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs]; - bool stereo_send_; + bool stereo_send_ GUARDED_BY(acm_crit_sect_); int current_send_codec_idx_; bool send_codec_registered_; - ACMResampler resampler_; + ACMResampler resampler_ GUARDED_BY(acm_crit_sect_); AcmReceiver receiver_; - CriticalSectionWrapper* acm_crit_sect_; - ACMVADCallback* vad_callback_; - // RED/FEC. - bool is_first_red_; - bool fec_enabled_; + // RED. + bool is_first_red_ GUARDED_BY(acm_crit_sect_); + bool red_enabled_ GUARDED_BY(acm_crit_sect_); // TODO(turajs): |red_buffer_| is allocated in constructor, why having them // as pointers and not an array. If concerned about the memory, then make a // set-up function to allocate them only when they are going to be used, i.e. - // FEC or Dual-streaming is enabled. - uint8_t* red_buffer_; + // RED or Dual-streaming is enabled. + uint8_t* red_buffer_ GUARDED_BY(acm_crit_sect_); // TODO(turajs): we actually don't need |fragmentation_| as a member variable. // It is sufficient to keep the length & payload type of previous payload in // member variables. - RTPFragmentationHeader fragmentation_; - uint32_t last_fec_timestamp_; + RTPFragmentationHeader fragmentation_ GUARDED_BY(acm_crit_sect_); + uint32_t last_red_timestamp_ GUARDED_BY(acm_crit_sect_); + + // Codec internal FEC + bool codec_fec_enabled_; // This is to keep track of CN instances where we can send DTMFs. - uint8_t previous_pltype_; + uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_); // Used when payloads are pushed into ACM without any RTP info // One example is when pre-encoded bit-stream is pushed from @@ -342,15 +358,18 @@ class AudioCodingModuleImpl : public AudioCodingModule { // be used in other methods, locks need to be taken. WebRtcRTPHeader* aux_rtp_header_; - bool receiver_initialized_; + bool receiver_initialized_ GUARDED_BY(acm_crit_sect_); - CriticalSectionWrapper* callback_crit_sect_; + AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_); + CodecInst secondary_send_codec_inst_ GUARDED_BY(acm_crit_sect_); + scoped_ptr<ACMGenericCodec> secondary_encoder_ GUARDED_BY(acm_crit_sect_); + uint32_t codec_timestamp_ GUARDED_BY(acm_crit_sect_); + bool first_10ms_data_ GUARDED_BY(acm_crit_sect_); - AudioFrame preprocess_frame_; - CodecInst secondary_send_codec_inst_; - scoped_ptr<ACMGenericCodec> secondary_encoder_; - uint32_t codec_timestamp_; - bool first_10ms_data_; + CriticalSectionWrapper* callback_crit_sect_; + AudioPacketizationCallback* packetization_callback_ + GUARDED_BY(callback_crit_sect_); + ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_); }; } // namespace acm2 diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc new file mode 100644 index 00000000000..37cd70e5e84 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc @@ -0,0 +1,514 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <string.h> +#include <vector> + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" +#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/event_wrapper.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/sleep.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" +#include "webrtc/system_wrappers/interface/thread_wrapper.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/test/testsupport/gtest_disable.h" + +namespace webrtc { + +const int kSampleRateHz = 16000; +const int kNumSamples10ms = kSampleRateHz / 100; +const int kFrameSizeMs = 10; // Multiple of 10. +const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; +const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); +const uint8_t kPayloadType = 111; + +class RtpUtility { + public: + RtpUtility(int samples_per_packet, uint8_t payload_type) + : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} + + virtual ~RtpUtility() {} + + void Populate(WebRtcRTPHeader* rtp_header) { + rtp_header->header.sequenceNumber = 0xABCD; + rtp_header->header.timestamp = 0xABCDEF01; + rtp_header->header.payloadType = payload_type_; + rtp_header->header.markerBit = false; + rtp_header->header.ssrc = 0x1234; + rtp_header->header.numCSRCs = 0; + rtp_header->frameType = kAudioFrameSpeech; + + rtp_header->header.payload_type_frequency = kSampleRateHz; + rtp_header->type.Audio.channel = 1; + rtp_header->type.Audio.isCNG = false; + } + + void Forward(WebRtcRTPHeader* rtp_header) { + ++rtp_header->header.sequenceNumber; + rtp_header->header.timestamp += samples_per_packet_; + } + + private: + int samples_per_packet_; + uint8_t payload_type_; +}; + +class PacketizationCallbackStub : public AudioPacketizationCallback { + public: + PacketizationCallbackStub() + : num_calls_(0), + crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {} + + virtual int32_t SendData( + FrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + uint16_t payload_len_bytes, + const RTPFragmentationHeader* fragmentation) OVERRIDE { + CriticalSectionScoped lock(crit_sect_.get()); + ++num_calls_; + last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); + return 0; + } + + int num_calls() const { + CriticalSectionScoped lock(crit_sect_.get()); + return num_calls_; + } + + int last_payload_len_bytes() const { + CriticalSectionScoped lock(crit_sect_.get()); + return last_payload_vec_.size(); + } + + void SwapBuffers(std::vector<uint8_t>* payload) { + CriticalSectionScoped lock(crit_sect_.get()); + last_payload_vec_.swap(*payload); + } + + private: + int num_calls_ GUARDED_BY(crit_sect_); + std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_); + const scoped_ptr<CriticalSectionWrapper> crit_sect_; +}; + +class AudioCodingModuleTest : public ::testing::Test { + protected: + AudioCodingModuleTest() + : id_(1), + rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), + clock_(Clock::GetRealTimeClock()) {} + + ~AudioCodingModuleTest() {} + + void TearDown() {} + + void SetUp() { + acm_.reset(AudioCodingModule::Create(id_, clock_)); + + RegisterCodec(); + + rtp_utility_->Populate(&rtp_header_); + + input_frame_.sample_rate_hz_ = kSampleRateHz; + input_frame_.num_channels_ = 1; + input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000; // 10 ms. + COMPILE_ASSERT(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples, + audio_frame_too_small); + memset(input_frame_.data_, + 0, + input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0])); + + ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_)); + } + + virtual void RegisterCodec() { + AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1); + codec_.pltype = kPayloadType; + + // Register L16 codec in ACM. + ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); + ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); + } + + virtual void InsertPacketAndPullAudio() { + InsertPacket(); + PullAudio(); + } + + virtual void InsertPacket() { + const uint8_t kPayload[kPayloadSizeBytes] = {0}; + ASSERT_EQ(0, + acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); + rtp_utility_->Forward(&rtp_header_); + } + + virtual void PullAudio() { + AudioFrame audio_frame; + ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame)); + } + + virtual void InsertAudio() { + ASSERT_EQ(0, acm_->Add10MsData(input_frame_)); + input_frame_.timestamp_ += kNumSamples10ms; + } + + virtual void Encode() { + int32_t encoded_bytes = acm_->Process(); + // Expect to get one packet with two bytes per sample, or no packet at all, + // depending on how many 10 ms blocks go into |codec_.pacsize|. + EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0); + } + + const int id_; + scoped_ptr<RtpUtility> rtp_utility_; + scoped_ptr<AudioCodingModule> acm_; + PacketizationCallbackStub packet_cb_; + WebRtcRTPHeader rtp_header_; + AudioFrame input_frame_; + CodecInst codec_; + Clock* clock_; +}; + +// Check if the statistics are initialized correctly. Before any call to ACM +// all fields have to be zero. +TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) { + AudioDecodingCallStats stats; + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(0, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(0, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); +} + +// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms() +// should result in generating silence, check the associated field. +TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { + AudioDecodingCallStats stats; + const int kInitialDelay = 100; + + acm_->SetInitialPlayoutDelay(kInitialDelay); + + int num_calls = 0; + for (int time_ms = 0; time_ms < kInitialDelay; + time_ms += kFrameSizeMs, ++num_calls) { + InsertPacketAndPullAudio(); + } + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(0, stats.calls_to_neteq); + EXPECT_EQ(num_calls, stats.calls_to_silence_generator); + EXPECT_EQ(0, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); +} + +// Insert some packets and pull audio. Check statistics are valid. Then, +// simulate packet loss and check if PLC and PLC-to-CNG statistics are +// correctly updated. +TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) { + AudioDecodingCallStats stats; + const int kNumNormalCalls = 10; + + for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) { + InsertPacketAndPullAudio(); + } + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); + + const int kNumPlc = 3; + const int kNumPlcCng = 5; + + // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG. + for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) { + PullAudio(); + } + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(kNumPlc, stats.decoded_plc); + EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng); +} + +TEST_F(AudioCodingModuleTest, VerifyOutputFrame) { + AudioFrame audio_frame; + const int kSampleRateHz = 32000; + EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame)); + EXPECT_EQ(id_, audio_frame.id_); + EXPECT_EQ(0u, audio_frame.timestamp_); + EXPECT_GT(audio_frame.num_channels_, 0); + EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_); + EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); +} + +TEST_F(AudioCodingModuleTest, FailOnZeroDesiredFrequency) { + AudioFrame audio_frame; + EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame)); +} + +// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz +// codec, while the derive class AcmIsacMtTest is using iSAC. +class AudioCodingModuleMtTest : public AudioCodingModuleTest { + protected: + static const int kNumPackets = 500; + static const int kNumPullCalls = 500; + + AudioCodingModuleMtTest() + : AudioCodingModuleTest(), + send_thread_(ThreadWrapper::CreateThread(CbSendThread, + this, + kRealtimePriority, + "send")), + insert_packet_thread_(ThreadWrapper::CreateThread(CbInsertPacketThread, + this, + kRealtimePriority, + "insert_packet")), + pull_audio_thread_(ThreadWrapper::CreateThread(CbPullAudioThread, + this, + kRealtimePriority, + "pull_audio")), + test_complete_(EventWrapper::Create()), + send_count_(0), + insert_packet_count_(0), + pull_audio_count_(0), + crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + next_insert_packet_time_ms_(0), + fake_clock_(new SimulatedClock(0)) { + clock_ = fake_clock_.get(); + } + + void SetUp() { + AudioCodingModuleTest::SetUp(); + StartThreads(); + } + + void StartThreads() { + unsigned int thread_id = 0; + ASSERT_TRUE(send_thread_->Start(thread_id)); + ASSERT_TRUE(insert_packet_thread_->Start(thread_id)); + ASSERT_TRUE(pull_audio_thread_->Start(thread_id)); + } + + void TearDown() { + AudioCodingModuleTest::TearDown(); + pull_audio_thread_->Stop(); + send_thread_->Stop(); + insert_packet_thread_->Stop(); + } + + EventTypeWrapper RunTest() { + return test_complete_->Wait(10 * 60 * 1000); // 10 minutes' timeout. + } + + virtual bool TestDone() { + if (packet_cb_.num_calls() > kNumPackets) { + CriticalSectionScoped lock(crit_sect_.get()); + if (pull_audio_count_ > kNumPullCalls) { + // Both conditions for completion are met. End the test. + return true; + } + } + return false; + } + + static bool CbSendThread(void* context) { + return reinterpret_cast<AudioCodingModuleMtTest*>(context)->CbSendImpl(); + } + + // The send thread doesn't have to care about the current simulated time, + // since only the AcmReceiver is using the clock. + bool CbSendImpl() { + SleepMs(1); + if (HasFatalFailure()) { + // End the test early if a fatal failure (ASSERT_*) has occurred. + test_complete_->Set(); + } + ++send_count_; + InsertAudio(); + Encode(); + if (TestDone()) { + test_complete_->Set(); + } + return true; + } + + static bool CbInsertPacketThread(void* context) { + return reinterpret_cast<AudioCodingModuleMtTest*>(context) + ->CbInsertPacketImpl(); + } + + bool CbInsertPacketImpl() { + SleepMs(1); + { + CriticalSectionScoped lock(crit_sect_.get()); + if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { + return true; + } + next_insert_packet_time_ms_ += 10; + } + // Now we're not holding the crit sect when calling ACM. + ++insert_packet_count_; + InsertPacket(); + return true; + } + + static bool CbPullAudioThread(void* context) { + return reinterpret_cast<AudioCodingModuleMtTest*>(context) + ->CbPullAudioImpl(); + } + + bool CbPullAudioImpl() { + SleepMs(1); + { + CriticalSectionScoped lock(crit_sect_.get()); + // Don't let the insert thread fall behind. + if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) { + return true; + } + ++pull_audio_count_; + } + // Now we're not holding the crit sect when calling ACM. + PullAudio(); + fake_clock_->AdvanceTimeMilliseconds(10); + return true; + } + + scoped_ptr<ThreadWrapper> send_thread_; + scoped_ptr<ThreadWrapper> insert_packet_thread_; + scoped_ptr<ThreadWrapper> pull_audio_thread_; + const scoped_ptr<EventWrapper> test_complete_; + int send_count_; + int insert_packet_count_; + int pull_audio_count_ GUARDED_BY(crit_sect_); + const scoped_ptr<CriticalSectionWrapper> crit_sect_; + int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_); + scoped_ptr<SimulatedClock> fake_clock_; +}; + +TEST_F(AudioCodingModuleMtTest, DoTest) { + EXPECT_EQ(kEventSignaled, RunTest()); +} + +// This is a multi-threaded ACM test using iSAC. The test encodes audio +// from a PCM file. The most recent encoded frame is used as input to the +// receiving part. Depending on timing, it may happen that the same RTP packet +// is inserted into the receiver multiple times, but this is a valid use-case, +// and simplifies the test code a lot. +class AcmIsacMtTest : public AudioCodingModuleMtTest { + protected: + static const int kNumPackets = 500; + static const int kNumPullCalls = 500; + + AcmIsacMtTest() + : AudioCodingModuleMtTest(), + last_packet_number_(0) {} + + ~AcmIsacMtTest() {} + + void SetUp() { + AudioCodingModuleTest::SetUp(); + + // Set up input audio source to read from specified file, loop after 5 + // seconds, and deliver blocks of 10 ms. + const std::string input_file_name = + webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm"); + audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms); + + // Generate one packet to have something to insert. + int loop_counter = 0; + while (packet_cb_.last_payload_len_bytes() == 0) { + InsertAudio(); + Encode(); + ASSERT_LT(loop_counter++, 10); + } + // Set |last_packet_number_| to one less that |num_calls| so that the packet + // will be fetched in the next InsertPacket() call. + last_packet_number_ = packet_cb_.num_calls() - 1; + + StartThreads(); + } + + virtual void RegisterCodec() { + COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz); + AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1); + codec_.pltype = kPayloadType; + + // Register iSAC codec in ACM, effectively unregistering the PCM16B codec + // registered in AudioCodingModuleTest::SetUp(); + ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); + ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); + } + + void InsertPacket() { + int num_calls = packet_cb_.num_calls(); // Store locally for thread safety. + if (num_calls > last_packet_number_) { + // Get the new payload out from the callback handler. + // Note that since we swap buffers here instead of directly inserting + // a pointer to the data in |packet_cb_|, we avoid locking the callback + // for the duration of the IncomingPacket() call. + packet_cb_.SwapBuffers(&last_payload_vec_); + ASSERT_GT(last_payload_vec_.size(), 0u); + rtp_utility_->Forward(&rtp_header_); + last_packet_number_ = num_calls; + } + ASSERT_GT(last_payload_vec_.size(), 0u); + ASSERT_EQ( + 0, + acm_->IncomingPacket( + &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); + } + + void InsertAudio() { + memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms); + AudioCodingModuleTest::InsertAudio(); + } + + void Encode() { ASSERT_GE(acm_->Process(), 0); } + + // This method is the same as AudioCodingModuleMtTest::TestDone(), but here + // it is using the constants defined in this class (i.e., shorter test run). + virtual bool TestDone() { + if (packet_cb_.num_calls() > kNumPackets) { + CriticalSectionScoped lock(crit_sect_.get()); + if (pull_audio_count_ > kNumPullCalls) { + // Both conditions for completion are met. End the test. + return true; + } + } + return false; + } + + int last_packet_number_; + std::vector<uint8_t> last_payload_vec_; + test::AudioLoop audio_loop_; +}; + +TEST_F(AcmIsacMtTest, DoTest) { + EXPECT_EQ(kEventSignaled, RunTest()); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc index 9153325afaf..4c3e9fc3939 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc @@ -10,7 +10,7 @@ #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" -#include <cassert> +#include <assert.h> namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc index c2b218cb6cf..786fb2e5275 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc @@ -219,6 +219,14 @@ void InitialDelayManager::LatePackets( return; } +bool InitialDelayManager::GetPlayoutTimestamp(uint32_t* playout_timestamp) { + if (!buffering_) { + return false; + } + *playout_timestamp = playout_timestamp_; + return true; +} + void InitialDelayManager::DisableBuffering() { buffering_ = false; } diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h index 3c5ba3c0139..6edc1150843 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h @@ -65,8 +65,9 @@ class InitialDelayManager { // sequence of late (or perhaps missing) packets is computed. void LatePackets(uint32_t timestamp_now, SyncStream* sync_stream); - // Playout timestamp, valid when buffering. - uint32_t playout_timestamp() { return playout_timestamp_; } + // Get playout timestamp. + // Returns true if the timestamp is valid (when buffering), otherwise false. + bool GetPlayoutTimestamp(uint32_t* playout_timestamp); // True if buffered audio is less than the given initial delay (specified at // the constructor). Buffering might be disabled by the client of this class. diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc index 1e129f37e90..38b7cfc2714 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include <cstring> +#include <string.h> #include "gtest/gtest.h" #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" @@ -359,7 +359,9 @@ TEST_F(InitialDelayManagerTest, BufferingAudio) { EXPECT_TRUE(manager_->buffering()); const uint32_t expected_playout_timestamp = rtp_info_.header.timestamp - kInitDelayMs * kSamplingRateHz / 1000; - EXPECT_EQ(expected_playout_timestamp, manager_->playout_timestamp()); + uint32_t actual_playout_timestamp = 0; + EXPECT_TRUE(manager_->GetPlayoutTimestamp(&actual_playout_timestamp)); + EXPECT_EQ(expected_playout_timestamp, actual_playout_timestamp); NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_); } diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc index 8011d8856c0..5837c31a899 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc @@ -398,7 +398,7 @@ TEST(NackTest, ChangeOfListSizeAppliedAndOldElementsRemoved) { // Packet lost more than NACK-list size limit. uint16_t num_lost_packets = kNackThreshold + kNackListSize + 5; - scoped_array<uint16_t> seq_num_lost(new uint16_t[num_lost_packets]); + scoped_ptr<uint16_t[]> seq_num_lost(new uint16_t[num_lost_packets]); for (int n = 0; n < num_lost_packets; ++n) { seq_num_lost[n] = ++seq_num; } diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/chromium/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h index db45addde22..cb0953aa400 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h +++ b/chromium/third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h @@ -15,7 +15,9 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/modules/interface/module.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -25,7 +27,6 @@ struct CodecInst; struct WebRtcRTPHeader; class AudioFrame; class RTPFragmentationHeader; -class Clock; #define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz @@ -73,15 +74,22 @@ class ACMVQMonCallback { const uint16_t delayMS) = 0; // average delay in ms }; -// Version string for testing, to distinguish instances of ACM1 from ACM2. -extern const char kLegacyAcmVersion[]; -extern const char kExperimentalAcmVersion[]; - class AudioCodingModule: public Module { protected: AudioCodingModule() {} public: + struct Config { + Config() + : id(0), + neteq_config(), + clock(Clock::GetRealTimeClock()) {} + + int id; + NetEq::Config neteq_config; + Clock* clock; + }; + /////////////////////////////////////////////////////////////////////////// // Creation and destruction of a ACM. // @@ -178,11 +186,6 @@ class AudioCodingModule: public Module { // static bool IsCodecValid(const CodecInst& codec); - // Returns the version of ACM. This facilitates distinguishing instances of - // ACM1 from ACM2 while testing. This API will be removed when ACM1 is - // completely removed. - virtual const char* Version() const = 0; - /////////////////////////////////////////////////////////////////////////// // Sender // @@ -370,12 +373,12 @@ class AudioCodingModule: public Module { virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0; /////////////////////////////////////////////////////////////////////////// - // (FEC) Forward Error Correction + // (RED) Redundant Coding // /////////////////////////////////////////////////////////////////////////// - // int32_t SetFECStatus(const bool enable) - // configure FEC status i.e. on/off. + // int32_t SetREDStatus() + // configure RED status i.e. on/off. // // RFC 2198 describes a solution which has a single payload type which // signifies a packet with redundancy. That packet then becomes a container, @@ -385,27 +388,69 @@ class AudioCodingModule: public Module { // since each encapsulated payload must be preceded by a header indicating // the type of data enclosed. // - // This means that FEC is actually a RED scheme. + // Input: + // -enable_red : if true RED is enabled, otherwise RED is + // disabled. + // + // Return value: + // -1 if failed to set RED status, + // 0 if succeeded. + // + virtual int32_t SetREDStatus(bool enable_red) = 0; + + /////////////////////////////////////////////////////////////////////////// + // bool REDStatus() + // Get RED status + // + // Return value: + // true if RED is enabled, + // false if RED is disabled. + // + virtual bool REDStatus() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // (FEC) Forward Error Correction (codec internal) + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t SetCodecFEC() + // Configures codec internal FEC status i.e. on/off. No effects on codecs that + // do not provide internal FEC. // // Input: - // -enable_fec : if true FEC is enabled, otherwise FEC is + // -enable_fec : if true FEC will be enabled otherwise the FEC is // disabled. // // Return value: - // -1 if failed to set FEC status, + // -1 if failed, or the codec does not support FEC // 0 if succeeded. // - virtual int32_t SetFECStatus(const bool enable_fec) = 0; + virtual int SetCodecFEC(bool enable_codec_fec) = 0; /////////////////////////////////////////////////////////////////////////// - // bool FECStatus() - // Get FEC status + // bool CodecFEC() + // Gets status of codec internal FEC. // - // Return value + // Return value: // true if FEC is enabled, // false if FEC is disabled. // - virtual bool FECStatus() const = 0; + virtual bool CodecFEC() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int SetPacketLossRate() + // Sets expected packet loss rate for encoding. Some encoders provide packet + // loss gnostic encoding to make stream less sensitive to packet losses, + // through e.g., FEC. No effects on codecs that do not provide such encoding. + // + // Input: + // -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive). + // + // Return value + // -1 if failed to set packet loss rate, + // 0 if succeeded. + // + virtual int SetPacketLossRate(int packet_loss_rate) = 0; /////////////////////////////////////////////////////////////////////////// // (VAD) Voice Activity Detection @@ -936,20 +981,6 @@ class AudioCodingModule: public Module { AudioDecodingCallStats* call_stats) const = 0; }; -struct AudioCodingModuleFactory { - AudioCodingModuleFactory() {} - virtual ~AudioCodingModuleFactory() {} - - virtual AudioCodingModule* Create(int id) const; -}; - -struct NewAudioCodingModuleFactory : AudioCodingModuleFactory { - NewAudioCodingModuleFactory() {} - virtual ~NewAudioCodingModuleFactory() {} - - virtual AudioCodingModule* Create(int id) const; -}; - } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/Android.mk b/chromium/third_party/webrtc/modules/audio_coding/main/source/Android.mk deleted file mode 100644 index 90214a9c408..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/Android.mk +++ /dev/null @@ -1,67 +0,0 @@ -# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -LOCAL_PATH := $(call my-dir) - -include $(CLEAR_VARS) - -include $(LOCAL_PATH)/../../../../../android-webrtc.mk - -LOCAL_ARM_MODE := arm -LOCAL_MODULE_CLASS := STATIC_LIBRARIES -LOCAL_MODULE := libwebrtc_audio_coding -LOCAL_MODULE_TAGS := optional -LOCAL_CPP_EXTENSION := .cc -LOCAL_SRC_FILES := \ - acm_cng.cc \ - acm_codec_database.cc \ - acm_dtmf_detection.cc \ - acm_dtmf_playout.cc \ - acm_g722.cc \ - acm_generic_codec.cc \ - acm_ilbc.cc \ - acm_isac.cc \ - acm_neteq.cc \ - acm_pcm16b.cc \ - acm_pcma.cc \ - acm_pcmu.cc \ - acm_red.cc \ - acm_resampler.cc \ - audio_coding_module.cc \ - audio_coding_module_impl.cc - -# Flags passed to both C and C++ files. -LOCAL_CFLAGS := \ - $(MY_WEBRTC_COMMON_DEFS) - -LOCAL_C_INCLUDES := \ - $(LOCAL_PATH)/../interface \ - $(LOCAL_PATH)/../../codecs/cng/include \ - $(LOCAL_PATH)/../../codecs/g711/include \ - $(LOCAL_PATH)/../../codecs/g722/include \ - $(LOCAL_PATH)/../../codecs/ilbc/interface \ - $(LOCAL_PATH)/../../codecs/iSAC/main/interface \ - $(LOCAL_PATH)/../../codecs/iSAC/fix/interface \ - $(LOCAL_PATH)/../../codecs/pcm16b/include \ - $(LOCAL_PATH)/../../neteq/interface \ - $(LOCAL_PATH)/../../../.. \ - $(LOCAL_PATH)/../../../interface \ - $(LOCAL_PATH)/../../../../common_audio/resampler/include \ - $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \ - $(LOCAL_PATH)/../../../../common_audio/vad/include \ - $(LOCAL_PATH)/../../../../system_wrappers/interface - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libdl \ - libstlport - -ifndef NDK_ROOT -include external/stlport/libstlport.mk -endif -include $(BUILD_STATIC_LIBRARY) diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc deleted file mode 100644 index d398607789b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc +++ /dev/null @@ -1,430 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_amr.h" - -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_AMR -// NOTE! GSM AMR is not included in the open-source package. The following -// interface file is needed: -// -// /modules/audio_coding/codecs/amr/main/interface/amr_interface.h -// -// The API in the header file should match the one below. -// -// int16_t WebRtcAmr_CreateEnc(AMR_encinst_t_** enc_inst); -// int16_t WebRtcAmr_CreateDec(AMR_decinst_t_** dec_inst); -// int16_t WebRtcAmr_FreeEnc(AMR_encinst_t_* enc_inst); -// int16_t WebRtcAmr_FreeDec(AMR_decinst_t_* dec_inst); -// int16_t WebRtcAmr_Encode(AMR_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t*output, -// int16_t mode); -// int16_t WebRtcAmr_EncoderInit(AMR_encinst_t_* enc_inst, -// int16_t dtx_mode); -// int16_t WebRtcAmr_EncodeBitmode(AMR_encinst_t_* enc_inst, -// int format); -// int16_t WebRtcAmr_Decode(AMR_decinst_t_* dec_inst); -// int16_t WebRtcAmr_DecodePlc(AMR_decinst_t_* dec_inst); -// int16_t WebRtcAmr_DecoderInit(AMR_decinst_t_* dec_inst); -// int16_t WebRtcAmr_DecodeBitmode(AMR_decinst_t_* dec_inst, -// int format); -#include "amr_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_AMR -ACMAMR::ACMAMR(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - encoding_mode_(-1), // Invalid value. - encoding_rate_(0), // Invalid value. - encoder_packing_format_(AMRBandwidthEfficient), - decoder_packing_format_(AMRBandwidthEfficient) { - return; -} - -ACMAMR::~ACMAMR() { - return; -} - -int16_t ACMAMR::InternalEncode(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMAMR::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMAMR::EnableDTX() { - return -1; -} - -int16_t ACMAMR::DisableDTX() { - return -1; -} - -int16_t ACMAMR::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMAMR::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMAMR::CreateInstance(void) { - return NULL; -} - -int16_t ACMAMR::InternalCreateEncoder() { - return -1; -} - -void ACMAMR::DestructEncoderSafe() { - return; -} - -int16_t ACMAMR::InternalCreateDecoder() { - return -1; -} - -void ACMAMR::DestructDecoderSafe() { - return; -} - -int16_t ACMAMR::SetBitRateSafe(const int32_t /* rate */) { - return -1; -} - -void ACMAMR::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -int16_t ACMAMR::SetAMREncoderPackingFormat( - ACMAMRPackingFormat /* packing_format */) { - return -1; -} - -ACMAMRPackingFormat ACMAMR::AMREncoderPackingFormat() const { - return AMRUndefined; -} - -int16_t ACMAMR::SetAMRDecoderPackingFormat( - ACMAMRPackingFormat /* packing_format */) { - return -1; -} - -ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const { - return AMRUndefined; -} - -#else //===================== Actual Implementation ======================= - -#define WEBRTC_AMR_MR475 0 -#define WEBRTC_AMR_MR515 1 -#define WEBRTC_AMR_MR59 2 -#define WEBRTC_AMR_MR67 3 -#define WEBRTC_AMR_MR74 4 -#define WEBRTC_AMR_MR795 5 -#define WEBRTC_AMR_MR102 6 -#define WEBRTC_AMR_MR122 7 - -ACMAMR::ACMAMR(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - encoding_mode_(-1), // invalid value - encoding_rate_(0) { // invalid value - codec_id_ = codec_id; - has_internal_dtx_ = true; - encoder_packing_format_ = AMRBandwidthEfficient; - decoder_packing_format_ = AMRBandwidthEfficient; - return; -} - -ACMAMR::~ACMAMR() { - if (encoder_inst_ptr_ != NULL) { - WebRtcAmr_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcAmr_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMAMR::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - int16_t vad_decision = 1; - // sanity check, if the rate is set correctly. we might skip this - // sanity check. if rate is not set correctly, initialization flag - // should be false and should not be here. - if ((encoding_mode_ < WEBRTC_AMR_MR475) || - (encoding_mode_ > WEBRTC_AMR_MR122)) { - *bitstream_len_byte = 0; - return -1; - } - *bitstream_len_byte = WebRtcAmr_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - frame_len_smpl_, - (int16_t*)bitstream, - encoding_mode_); - - // Update VAD, if internal DTX is used - if (has_internal_dtx_ && dtx_enabled_) { - if (*bitstream_len_byte <= (7 * frame_len_smpl_ / 160)) { - vad_decision = 0; - } - for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) { - vad_label_[n] = vad_decision; - } - } - // increment the read index - in_audio_ix_read_ += frame_len_smpl_; - return *bitstream_len_byte; -} - -int16_t ACMAMR::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMAMR::EnableDTX() { - if (dtx_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // enable DTX - if (WebRtcAmr_EncoderInit(encoder_inst_ptr_, 1) < 0) { - return -1; - } - dtx_enabled_ = true; - return 0; - } else { - return -1; - } -} - -int16_t ACMAMR::DisableDTX() { - if (!dtx_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // disable DTX - if (WebRtcAmr_EncoderInit(encoder_inst_ptr_, 0) < 0) { - return -1; - } - dtx_enabled_ = false; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } -} - -int16_t ACMAMR::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - int16_t status = SetBitRateSafe((codec_params->codec_inst).rate); - status += (WebRtcAmr_EncoderInit( - encoder_inst_ptr_, ((codec_params->enable_dtx) ? 1 : 0)) < 0) ? -1 : 0; - status += (WebRtcAmr_EncodeBitmode( - encoder_inst_ptr_, encoder_packing_format_) < 0) ? -1 : 0; - return (status < 0) ? -1 : 0; -} - -int16_t ACMAMR::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - int16_t status = - ((WebRtcAmr_DecoderInit(decoder_inst_ptr_) < 0) ? -1 : 0); - status += WebRtcAmr_DecodeBitmode(decoder_inst_ptr_, decoder_packing_format_); - return (status < 0) ? -1 : 0; -} - -int32_t ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - // Todo: - // log error - return -1; - } - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_AMR_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderAMR, codec_inst.pltype, decoder_inst_ptr_, - 8000); - SET_AMR_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMAMR::CreateInstance(void) { - return NULL; -} - -int16_t ACMAMR::InternalCreateEncoder() { - return WebRtcAmr_CreateEnc(&encoder_inst_ptr_); -} - -void ACMAMR::DestructEncoderSafe() { - if (encoder_inst_ptr_ != NULL) { - WebRtcAmr_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - // there is no encoder set the following - encoder_exist_ = false; - encoder_initialized_ = false; - encoding_mode_ = -1; // invalid value - encoding_rate_ = 0; // invalid value -} - -int16_t ACMAMR::InternalCreateDecoder() { - return WebRtcAmr_CreateDec(&decoder_inst_ptr_); -} - -void ACMAMR::DestructDecoderSafe() { - if (decoder_inst_ptr_ != NULL) { - WebRtcAmr_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - // there is no encoder instance set the followings - decoder_exist_ = false; - decoder_initialized_ = false; -} - -int16_t ACMAMR::SetBitRateSafe(const int32_t rate) { - switch (rate) { - case 4750: { - encoding_mode_ = WEBRTC_AMR_MR475; - encoding_rate_ = 4750; - break; - } - case 5150: { - encoding_mode_ = WEBRTC_AMR_MR515; - encoding_rate_ = 5150; - break; - } - case 5900: { - encoding_mode_ = WEBRTC_AMR_MR59; - encoding_rate_ = 5900; - break; - } - case 6700: { - encoding_mode_ = WEBRTC_AMR_MR67; - encoding_rate_ = 6700; - break; - } - case 7400: { - encoding_mode_ = WEBRTC_AMR_MR74; - encoding_rate_ = 7400; - break; - } - case 7950: { - encoding_mode_ = WEBRTC_AMR_MR795; - encoding_rate_ = 7950; - break; - } - case 10200: { - encoding_mode_ = WEBRTC_AMR_MR102; - encoding_rate_ = 10200; - break; - } - case 12200: { - encoding_mode_ = WEBRTC_AMR_MR122; - encoding_rate_ = 12200; - break; - } - default: { - return -1; - } - } - return 0; -} - -void ACMAMR::InternalDestructEncoderInst(void* ptr_inst) { - // Free the memory where ptr_inst is pointing to - if (ptr_inst != NULL) { - WebRtcAmr_FreeEnc(reinterpret_cast<AMR_encinst_t_*>(ptr_inst)); - } - return; -} - -int16_t ACMAMR::SetAMREncoderPackingFormat( - ACMAMRPackingFormat packing_format) { - if ((packing_format != AMRBandwidthEfficient) && - (packing_format != AMROctetAlligned) && - (packing_format != AMRFileStorage)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Invalid AMR Encoder packing-format."); - return -1; - } else { - if (WebRtcAmr_EncodeBitmode(encoder_inst_ptr_, packing_format) < 0) { - return -1; - } else { - encoder_packing_format_ = packing_format; - return 0; - } - } -} - -ACMAMRPackingFormat ACMAMR::AMREncoderPackingFormat() const { - return encoder_packing_format_; -} - -int16_t ACMAMR::SetAMRDecoderPackingFormat( - ACMAMRPackingFormat packing_format) { - if ((packing_format != AMRBandwidthEfficient) && - (packing_format != AMROctetAlligned) && - (packing_format != AMRFileStorage)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Invalid AMR decoder packing-format."); - return -1; - } else { - if (WebRtcAmr_DecodeBitmode(decoder_inst_ptr_, packing_format) < 0) { - return -1; - } else { - decoder_packing_format_ = packing_format; - return 0; - } - } -} - -ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const { - return decoder_packing_format_; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc - diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amr.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amr.h deleted file mode 100644 index 19c657246a2..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amr.h +++ /dev/null @@ -1,87 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct AMR_encinst_t_; -struct AMR_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMAMR : public ACMGenericCodec { - public: - explicit ACMAMR(int16_t codec_id); - virtual ~ACMAMR(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - int16_t SetAMREncoderPackingFormat(const ACMAMRPackingFormat packing_format); - - ACMAMRPackingFormat AMREncoderPackingFormat() const; - - int16_t SetAMRDecoderPackingFormat(const ACMAMRPackingFormat packing_format); - - ACMAMRPackingFormat AMRDecoderPackingFormat() const; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual int16_t SetBitRateSafe(const int32_t rate) OVERRIDE; - - virtual int16_t EnableDTX() OVERRIDE; - - virtual int16_t DisableDTX() OVERRIDE; - - AMR_encinst_t_* encoder_inst_ptr_; - AMR_decinst_t_* decoder_inst_ptr_; - int16_t encoding_mode_; - int16_t encoding_rate_; - ACMAMRPackingFormat encoder_packing_format_; - ACMAMRPackingFormat decoder_packing_format_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc deleted file mode 100644 index 8b1b58d03cf..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc +++ /dev/null @@ -1,436 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_amrwb.h" - -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_AMRWB -// NOTE! GSM AMR-wb is not included in the open-source package. The -// following interface file is needed: -// -// /modules/audio_coding/codecs/amrwb/main/interface/amrwb_interface.h -// -// The API in the header file should match the one below. -// -// int16_t WebRtcAmrWb_CreateEnc(AMRWB_encinst_t_** enc_inst); -// int16_t WebRtcAmrWb_CreateDec(AMRWB_decinst_t_** dec_inst); -// int16_t WebRtcAmrWb_FreeEnc(AMRWB_encinst_t_* enc_inst); -// int16_t WebRtcAmrWb_FreeDec(AMRWB_decinst_t_* dec_inst); -// int16_t WebRtcAmrWb_Encode(AMRWB_encinst_t_* enc_inst, int16_t* input, -// int16_t len, int16_t* output, int16_t mode); -// int16_t WebRtcAmrWb_EncoderInit(AMRWB_encinst_t_* enc_inst, -// int16_t dtx_mode); -// int16_t WebRtcAmrWb_EncodeBitmode(AMRWB_encinst_t_* enc_inst, -// int format); -// int16_t WebRtcAmrWb_Decode(AMRWB_decinst_t_* dec_inst); -// int16_t WebRtcAmrWb_DecodePlc(AMRWB_decinst_t_* dec_inst); -// int16_t WebRtcAmrWb_DecoderInit(AMRWB_decinst_t_* dec_inst); -// int16_t WebRtcAmrWb_DecodeBitmode(AMRWB_decinst_t_* dec_inst, -// int format); -#include "amrwb_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_AMRWB -ACMAMRwb::ACMAMRwb(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - encoding_mode_(-1), // invalid value - encoding_rate_(0), // invalid value - encoder_packing_format_(AMRBandwidthEfficient), - decoder_packing_format_(AMRBandwidthEfficient) { -} - -ACMAMRwb::~ACMAMRwb() { -} - -int16_t ACMAMRwb::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMAMRwb::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMAMRwb::EnableDTX() { - return -1; -} - -int16_t ACMAMRwb::DisableDTX() { - return -1; -} - -int16_t ACMAMRwb::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMAMRwb::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* -ACMAMRwb::CreateInstance(void) { - return NULL; -} - -int16_t ACMAMRwb::InternalCreateEncoder() { - return -1; -} - -void ACMAMRwb::DestructEncoderSafe() { - return; -} - -int16_t ACMAMRwb::InternalCreateDecoder() { - return -1; -} - -void ACMAMRwb::DestructDecoderSafe() { - return; -} - -int16_t ACMAMRwb::SetBitRateSafe(const int32_t /* rate */) { - return -1; -} - -void ACMAMRwb::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -int16_t ACMAMRwb::SetAMRwbEncoderPackingFormat( - ACMAMRPackingFormat /* packing_format */) { - return -1; -} - -ACMAMRPackingFormat ACMAMRwb::AMRwbEncoderPackingFormat() const { - return AMRUndefined; -} - -int16_t ACMAMRwb::SetAMRwbDecoderPackingFormat( - ACMAMRPackingFormat /* packing_format */) { - return -1; -} - -ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const { - return AMRUndefined; -} - -#else //===================== Actual Implementation ======================= - -#define AMRWB_MODE_7k 0 -#define AMRWB_MODE_9k 1 -#define AMRWB_MODE_12k 2 -#define AMRWB_MODE_14k 3 -#define AMRWB_MODE_16k 4 -#define AMRWB_MODE_18k 5 -#define AMRWB_MODE_20k 6 -#define AMRWB_MODE_23k 7 -#define AMRWB_MODE_24k 8 - -ACMAMRwb::ACMAMRwb(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - encoding_mode_(-1), // invalid value - encoding_rate_(0) { // invalid value - codec_id_ = codec_id; - has_internal_dtx_ = true; - encoder_packing_format_ = AMRBandwidthEfficient; - decoder_packing_format_ = AMRBandwidthEfficient; - return; -} - -ACMAMRwb::~ACMAMRwb() { - if (encoder_inst_ptr_ != NULL) { - WebRtcAmrWb_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcAmrWb_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMAMRwb::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - int16_t vad_decision = 1; - // sanity check, if the rate is set correctly. we might skip this - // sanity check. if rate is not set correctly, initialization flag - // should be false and should not be here. - if ((encoding_mode_ < AMRWB_MODE_7k) || (encoding_mode_ > AMRWB_MODE_24k)) { - *bitstream_len_byte = 0; - return -1; - } - *bitstream_len_byte = WebRtcAmrWb_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - frame_len_smpl_, - (int16_t*)bitstream, - encoding_mode_); - - // Update VAD, if internal DTX is used - if (has_internal_dtx_ && dtx_enabled_) { - if (*bitstream_len_byte <= (7 * frame_len_smpl_ / 160)) { - vad_decision = 0; - } - for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) { - vad_label_[n] = vad_decision; - } - } - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += frame_len_smpl_; - return *bitstream_len_byte; -} - -int16_t ACMAMRwb::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMAMRwb::EnableDTX() { - if (dtx_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // enable DTX - if (WebRtcAmrWb_EncoderInit(encoder_inst_ptr_, 1) < 0) { - return -1; - } - dtx_enabled_ = true; - return 0; - } else { - return -1; - } -} - -int16_t ACMAMRwb::DisableDTX() { - if (!dtx_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // disable DTX - if (WebRtcAmrWb_EncoderInit(encoder_inst_ptr_, 0) < 0) { - return -1; - } - dtx_enabled_ = false; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } -} - -int16_t ACMAMRwb::InternalInitEncoder( - WebRtcACMCodecParams* codec_params) { - // sanity check - if (encoder_inst_ptr_ == NULL) { - return -1; - } - - int16_t status = SetBitRateSafe((codec_params->codec_inst).rate); - status += (WebRtcAmrWb_EncoderInit( - encoder_inst_ptr_, ((codec_params->enable_dtx) ? 1 : 0)) < 0) ? -1 : 0; - status += (WebRtcAmrWb_EncodeBitmode( - encoder_inst_ptr_, encoder_packing_format_) < 0) ? -1 : 0; - return (status < 0) ? -1 : 0; -} - -int16_t ACMAMRwb::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - int16_t status = WebRtcAmrWb_DecodeBitmode(decoder_inst_ptr_, - decoder_packing_format_); - status += ((WebRtcAmrWb_DecoderInit(decoder_inst_ptr_) < 0) ? -1 : 0); - return (status < 0) ? -1 : 0; -} - -int32_t ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - return -1; - } - - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_AMRWB_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderAMRWB, codec_inst.pltype, - decoder_inst_ptr_, 16000); - SET_AMRWB_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMAMRwb::CreateInstance(void) { - return NULL; -} - -int16_t ACMAMRwb::InternalCreateEncoder() { - return WebRtcAmrWb_CreateEnc(&encoder_inst_ptr_); -} - -void ACMAMRwb::DestructEncoderSafe() { - if (encoder_inst_ptr_ != NULL) { - WebRtcAmrWb_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - // there is no encoder set the following - encoder_exist_ = false; - encoder_initialized_ = false; - encoding_mode_ = -1; // invalid value - encoding_rate_ = 0; -} - -int16_t ACMAMRwb::InternalCreateDecoder() { - return WebRtcAmrWb_CreateDec(&decoder_inst_ptr_); -} - -void ACMAMRwb::DestructDecoderSafe() { - if (decoder_inst_ptr_ != NULL) { - WebRtcAmrWb_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - // there is no encoder instance set the followings - decoder_exist_ = false; - decoder_initialized_ = false; -} - -int16_t ACMAMRwb::SetBitRateSafe(const int32_t rate) { - switch (rate) { - case 7000: { - encoding_mode_ = AMRWB_MODE_7k; - encoding_rate_ = 7000; - break; - } - case 9000: { - encoding_mode_ = AMRWB_MODE_9k; - encoding_rate_ = 9000; - break; - } - case 12000: { - encoding_mode_ = AMRWB_MODE_12k; - encoding_rate_ = 12000; - break; - } - case 14000: { - encoding_mode_ = AMRWB_MODE_14k; - encoding_rate_ = 14000; - break; - } - case 16000: { - encoding_mode_ = AMRWB_MODE_16k; - encoding_rate_ = 16000; - break; - } - case 18000: { - encoding_mode_ = AMRWB_MODE_18k; - encoding_rate_ = 18000; - break; - } - case 20000: { - encoding_mode_ = AMRWB_MODE_20k; - encoding_rate_ = 20000; - break; - } - case 23000: { - encoding_mode_ = AMRWB_MODE_23k; - encoding_rate_ = 23000; - break; - } - case 24000: { - encoding_mode_ = AMRWB_MODE_24k; - encoding_rate_ = 24000; - break; - } - default: { - return -1; - } - } - return 0; -} - -void ACMAMRwb::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcAmrWb_FreeEnc(static_cast<AMRWB_encinst_t_*>(ptr_inst)); - } - return; -} - -int16_t ACMAMRwb::SetAMRwbEncoderPackingFormat( - ACMAMRPackingFormat packing_format) { - if ((packing_format != AMRBandwidthEfficient) && - (packing_format != AMROctetAlligned) && - (packing_format != AMRFileStorage)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Invalid AMRwb encoder packing-format."); - return -1; - } else { - if (WebRtcAmrWb_EncodeBitmode(encoder_inst_ptr_, packing_format) < 0) { - return -1; - } else { - encoder_packing_format_ = packing_format; - return 0; - } - } -} - -ACMAMRPackingFormat ACMAMRwb::AMRwbEncoderPackingFormat() const { - return encoder_packing_format_; -} - -int16_t ACMAMRwb::SetAMRwbDecoderPackingFormat( - ACMAMRPackingFormat packing_format) { - if ((packing_format != AMRBandwidthEfficient) && - (packing_format != AMROctetAlligned) && - (packing_format != AMRFileStorage)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Invalid AMRwb decoder packing-format."); - return -1; - } else { - if (WebRtcAmrWb_DecodeBitmode(decoder_inst_ptr_, packing_format) < 0) { - return -1; - } else { - decoder_packing_format_ = packing_format; - return 0; - } - } -} - -ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const { - return decoder_packing_format_; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.h deleted file mode 100644 index 25934187e55..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.h +++ /dev/null @@ -1,90 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct AMRWB_encinst_t_; -struct AMRWB_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMAMRwb : public ACMGenericCodec { - public: - explicit ACMAMRwb(int16_t codec_id); - virtual ~ACMAMRwb(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t SetAMRwbEncoderPackingFormat( - const ACMAMRPackingFormat packing_format); - - virtual ACMAMRPackingFormat AMRwbEncoderPackingFormat() const; - - virtual int16_t SetAMRwbDecoderPackingFormat( - const ACMAMRPackingFormat packing_format); - - virtual ACMAMRPackingFormat AMRwbDecoderPackingFormat() const; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual int16_t SetBitRateSafe(const int32_t rate) OVERRIDE; - - virtual int16_t EnableDTX() OVERRIDE; - - virtual int16_t DisableDTX() OVERRIDE; - - AMRWB_encinst_t_* encoder_inst_ptr_; - AMRWB_decinst_t_* decoder_inst_ptr_; - - int16_t encoding_mode_; - int16_t encoding_rate_; - ACMAMRPackingFormat encoder_packing_format_; - ACMAMRPackingFormat decoder_packing_format_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc deleted file mode 100644 index 3b838143d4a..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc +++ /dev/null @@ -1,339 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_celt.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_CELT -// NOTE! Celt is not included in the open-source package. Modify this file or -// your codec API to match the function call and name of used Celt API file. -#include "celt_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_CELT - -ACMCELT::ACMCELT(int16_t /* codec_id */) - : enc_inst_ptr_(NULL), - dec_inst_ptr_(NULL), - sampling_freq_(0), - bitrate_(0), - channels_(1), - dec_channels_(1) { - return; -} - -ACMCELT::~ACMCELT() { - return; -} - -int16_t ACMCELT::InternalEncode(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMCELT::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMCELT::CreateInstance(void) { - return NULL; -} - -int16_t ACMCELT::InternalCreateEncoder() { - return -1; -} - -void ACMCELT::DestructEncoderSafe() { - return; -} - -int16_t ACMCELT::InternalCreateDecoder() { - return -1; -} - -void ACMCELT::DestructDecoderSafe() { - return; -} - -void ACMCELT::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -bool ACMCELT::IsTrueStereoCodec() { - return true; -} - -int16_t ACMCELT::SetBitRateSafe(const int32_t /*rate*/) { - return -1; -} - -void ACMCELT::SplitStereoPacket(uint8_t* /*payload*/, - int32_t* /*payload_length*/) {} - -#else //===================== Actual Implementation ======================= - -ACMCELT::ACMCELT(int16_t codec_id) - : enc_inst_ptr_(NULL), - dec_inst_ptr_(NULL), - sampling_freq_(32000), // Default sampling frequency. - bitrate_(64000), // Default rate. - channels_(1), // Default send mono. - dec_channels_(1) { // Default receive mono. - // TODO(tlegrand): remove later when ACMGenericCodec has a new constructor. - codec_id_ = codec_id; - - return; -} - -ACMCELT::~ACMCELT() { - if (enc_inst_ptr_ != NULL) { - WebRtcCelt_FreeEnc(enc_inst_ptr_); - enc_inst_ptr_ = NULL; - } - if (dec_inst_ptr_ != NULL) { - WebRtcCelt_FreeDec(dec_inst_ptr_); - dec_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMCELT::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - *bitstream_len_byte = 0; - - // Call Encoder. - *bitstream_len_byte = WebRtcCelt_Encode(enc_inst_ptr_, - &in_audio_[in_audio_ix_read_], - bitstream); - - // Increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer. - in_audio_ix_read_ += frame_len_smpl_ * channels_; - - if (*bitstream_len_byte < 0) { - // Error reported from the encoder. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalEncode: Encode error for Celt"); - *bitstream_len_byte = 0; - return -1; - } - - return *bitstream_len_byte; -} - -int16_t ACMCELT::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - // Set bitrate and check that it is within the valid range. - int16_t status = SetBitRateSafe((codec_params->codec_inst).rate); - if (status < 0) { - return -1; - } - - // If number of channels changed we need to re-create memory. - if (codec_params->codec_inst.channels != channels_) { - WebRtcCelt_FreeEnc(enc_inst_ptr_); - enc_inst_ptr_ = NULL; - // Store new number of channels. - channels_ = codec_params->codec_inst.channels; - if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, channels_) < 0) { - return -1; - } - } - - // Initiate encoder. - if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) { - return 0; - } else { - return -1; - } -} - -int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { - // If number of channels changed we need to re-create memory. - if (codec_params->codec_inst.channels != dec_channels_) { - WebRtcCelt_FreeDec(dec_inst_ptr_); - dec_inst_ptr_ = NULL; - // Store new number of channels. - dec_channels_ = codec_params->codec_inst.channels; - if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) { - return -1; - } - } - - // Initiate decoder, both master and slave parts. - if (WebRtcCelt_DecoderInit(dec_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: init decoder failed for Celt."); - return -1; - } - if (WebRtcCelt_DecoderInitSlave(dec_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: init decoder failed for Celt."); - return -1; - } - return 0; -} - -int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodecDef: Decoder uninitialized for Celt"); - return -1; - } - - // Fill up the structure by calling - // "SET_CODEC_PAR" and "SET_CELT_FUNCTIONS" or "SET_CELTSLAVE_FUNCTIONS". - // Then call NetEQ to add the codec to it's - // database. - if (codec_inst.channels == 1) { - SET_CODEC_PAR(codec_def, kDecoderCELT_32, codec_inst.pltype, dec_inst_ptr_, - 32000); - } else { - SET_CODEC_PAR(codec_def, kDecoderCELT_32_2ch, codec_inst.pltype, - dec_inst_ptr_, 32000); - } - - // If this is the master of NetEQ, regular decoder will be added, otherwise - // the slave decoder will be used. - if (is_master_) { - SET_CELT_FUNCTIONS(codec_def); - } else { - SET_CELTSLAVE_FUNCTIONS(codec_def); - } - return 0; -} - -ACMGenericCodec* ACMCELT::CreateInstance(void) { - return NULL; -} - -int16_t ACMCELT::InternalCreateEncoder() { - if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, num_channels_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: create encoder failed for Celt"); - return -1; - } - channels_ = num_channels_; - return 0; -} - -void ACMCELT::DestructEncoderSafe() { - encoder_exist_ = false; - encoder_initialized_ = false; - if (enc_inst_ptr_ != NULL) { - WebRtcCelt_FreeEnc(enc_inst_ptr_); - enc_inst_ptr_ = NULL; - } -} - -int16_t ACMCELT::InternalCreateDecoder() { - if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateDecoder: create decoder failed for Celt"); - return -1; - } - - return 0; -} - -void ACMCELT::DestructDecoderSafe() { - decoder_exist_ = false; - decoder_initialized_ = false; - if (dec_inst_ptr_ != NULL) { - WebRtcCelt_FreeDec(dec_inst_ptr_); - dec_inst_ptr_ = NULL; - } -} - -void ACMCELT::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcCelt_FreeEnc(static_cast<CELT_encinst_t*>(ptr_inst)); - } - return; -} - -bool ACMCELT::IsTrueStereoCodec() { - return true; -} - -int16_t ACMCELT::SetBitRateSafe(const int32_t rate) { - // Check that rate is in the valid range. - if ((rate >= 48000) && (rate <= 128000)) { - // Store new rate. - bitrate_ = rate; - - // Initiate encoder with new rate. - if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) { - return 0; - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetBitRateSafe: Failed to initiate Celt with rate %d", - rate); - return -1; - } - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetBitRateSafe: Invalid rate Celt, %d", rate); - return -1; - } -} - -// Copy the stereo packet so that NetEq will insert into both master and slave. -void ACMCELT::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { - // Check for valid inputs. - assert(payload != NULL); - assert(*payload_length > 0); - - // Duplicate the payload. - memcpy(&payload[*payload_length], &payload[0], - sizeof(uint8_t) * (*payload_length)); - // Double the size of the packet. - *payload_length *= 2; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_celt.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_celt.h deleted file mode 100644 index 4a4610e0d47..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_celt.h +++ /dev/null @@ -1,79 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct CELT_encinst_t_; -struct CELT_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMCELT : public ACMGenericCodec { - public: - explicit ACMCELT(int16_t codec_id); - virtual ~ACMCELT(); - - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual bool IsTrueStereoCodec() OVERRIDE; - - virtual int16_t SetBitRateSafe(const int32_t rate) OVERRIDE; - - virtual void SplitStereoPacket(uint8_t* payload, - int32_t* payload_length) OVERRIDE; - - CELT_encinst_t_* enc_inst_ptr_; - CELT_decinst_t_* dec_inst_ptr_; - uint16_t sampling_freq_; - int32_t bitrate_; - uint16_t channels_; - uint16_t dec_channels_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc deleted file mode 100644 index 6f3a5057e0b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc +++ /dev/null @@ -1,150 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_cng.h" - -#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -namespace acm1 { - -ACMCNG::ACMCNG(int16_t codec_id) { - encoder_inst_ptr_ = NULL; - decoder_inst_ptr_ = NULL; - codec_id_ = codec_id; - samp_freq_hz_ = ACMCodecDB::CodecFreq(codec_id_); - return; -} - -ACMCNG::~ACMCNG() { - if (encoder_inst_ptr_ != NULL) { - WebRtcCng_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcCng_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -// CNG is not like a regular encoder, this function -// should not be called normally -// instead the following function is called from inside -// ACMGenericCodec::ProcessFrameVADDTX -int16_t ACMCNG::InternalEncode(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMCNG::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -// CNG is not like a regular encoder, -// this function should not be called normally -// instead the following function is called from inside -// ACMGenericCodec::ProcessFrameVADDTX -int16_t ACMCNG::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMCNG::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return WebRtcCng_InitDec(decoder_inst_ptr_); -} - -int32_t ACMCNG::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - // TODO(tlegrand): log error - return -1; - } - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_CNG_FUNCTION." - // Then return the structure back to NetEQ to add the codec to it's - // database. - - if (samp_freq_hz_ == 8000 || samp_freq_hz_ == 16000 || - samp_freq_hz_ == 32000 || samp_freq_hz_ == 48000) { - SET_CODEC_PAR((codec_def), kDecoderCNG, codec_inst.pltype, - decoder_inst_ptr_, samp_freq_hz_); - SET_CNG_FUNCTIONS((codec_def)); - return 0; - } else { - return -1; - } -} - -ACMGenericCodec* ACMCNG::CreateInstance(void) { - return NULL; -} - -int16_t ACMCNG::InternalCreateEncoder() { - if (WebRtcCng_CreateEnc(&encoder_inst_ptr_) < 0) { - encoder_inst_ptr_ = NULL; - return -1; - } else { - return 0; - } -} - -void ACMCNG::DestructEncoderSafe() { - if (encoder_inst_ptr_ != NULL) { - WebRtcCng_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - encoder_exist_ = false; - encoder_initialized_ = false; -} - -int16_t ACMCNG::InternalCreateDecoder() { - if (WebRtcCng_CreateDec(&decoder_inst_ptr_) < 0) { - decoder_inst_ptr_ = NULL; - return -1; - } else { - return 0; - } -} - -void ACMCNG::DestructDecoderSafe() { - if (decoder_inst_ptr_ != NULL) { - WebRtcCng_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - decoder_exist_ = false; - decoder_initialized_ = false; -} - -void ACMCNG::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcCng_FreeEnc(static_cast<CNG_enc_inst*>(ptr_inst)); - } - return; -} - -int16_t ACMCNG::EnableDTX() { return -1; } -int16_t ACMCNG::DisableDTX() { return -1; } - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_cng.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_cng.h deleted file mode 100644 index 728312d55fb..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_cng.h +++ /dev/null @@ -1,73 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct WebRtcCngEncInst; -struct WebRtcCngDecInst; - -namespace webrtc { - -namespace acm1 { - -class ACMCNG: public ACMGenericCodec { - public: - explicit ACMCNG(int16_t codec_id); - virtual ~ACMCNG(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual int16_t EnableDTX() OVERRIDE; - virtual int16_t DisableDTX() OVERRIDE; - - WebRtcCngEncInst* encoder_inst_ptr_; - WebRtcCngDecInst* decoder_inst_ptr_; - uint16_t samp_freq_hz_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc deleted file mode 100644 index 138effd6a9f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc +++ /dev/null @@ -1,956 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file generates databases with information about all supported audio - * codecs. - */ - -// TODO(tlegrand): Change constant input pointers in all functions to constant -// references, where appropriate. -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/system_wrappers/interface/trace.h" - -// Includes needed to create the codecs. -// G.711, PCM mu-law and A-law. -#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_pcma.h" -#include "webrtc/modules/audio_coding/main/source/acm_pcmu.h" -// CNG. -#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" -#include "webrtc/modules/audio_coding/main/source/acm_cng.h" -// NetEQ. -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#ifdef WEBRTC_CODEC_ISAC -#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" -#endif -#ifdef WEBRTC_CODEC_ISACFX -#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" -#endif -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) -#include "webrtc/modules/audio_coding/main/source/acm_isac.h" -#include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h" -#endif -#ifdef WEBRTC_CODEC_PCM16 -#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" -#include "webrtc/modules/audio_coding/main/source/acm_pcm16b.h" -#endif -#ifdef WEBRTC_CODEC_ILBC -#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" -#include "webrtc/modules/audio_coding/main/source/acm_ilbc.h" -#endif -#ifdef WEBRTC_CODEC_AMR -#include "webrtc/modules/audio_coding/codecs/amr/include/amr_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_amr.h" -#endif -#ifdef WEBRTC_CODEC_AMRWB -#include "webrtc/modules/audio_coding/codecs/amrwb/include/amrwb_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_amrwb.h" -#endif -#ifdef WEBRTC_CODEC_CELT -#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_celt.h" -#endif -#ifdef WEBRTC_CODEC_G722 -#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_g722.h" -#endif -#ifdef WEBRTC_CODEC_G722_1 -#include "webrtc/modules/audio_coding/codecs/g7221/include/g7221_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_g7221.h" -#endif -#ifdef WEBRTC_CODEC_G722_1C -#include "webrtc/modules/audio_coding/codecs/g7221c/include/g7221c_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_g7221c.h" -#endif -#ifdef WEBRTC_CODEC_G729 -#include "webrtc/modules/audio_coding/codecs/g729/include/g729_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_g729.h" -#endif -#ifdef WEBRTC_CODEC_G729_1 -#include "webrtc/modules/audio_coding/codecs/g7291/include/g7291_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_g7291.h" -#endif -#ifdef WEBRTC_CODEC_GSMFR -#include "webrtc/modules/audio_coding/codecs/gsmfr/include/gsmfr_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h" -#endif -#ifdef WEBRTC_CODEC_OPUS -#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_opus.h" -#endif -#ifdef WEBRTC_CODEC_SPEEX -#include "webrtc/modules/audio_coding/codecs/speex/include/speex_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_speex.h" -#endif -#ifdef WEBRTC_CODEC_AVT -#include "webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h" -#endif -#ifdef WEBRTC_CODEC_RED -#include "webrtc/modules/audio_coding/main/source/acm_red.h" -#endif - -namespace webrtc { - -namespace acm1 { - -// Not yet used payload-types. -// 83, 82, 81, 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68, -// 67, 66, 65 - -const CodecInst ACMCodecDB::database_[] = { -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - {103, "ISAC", 16000, kIsacPacSize480, 1, kIsacWbDefaultRate}, -# if (defined(WEBRTC_CODEC_ISAC)) - {104, "ISAC", 32000, kIsacPacSize960, 1, kIsacSwbDefaultRate}, - {105, "ISAC", 48000, kIsacPacSize1440, 1, kIsacSwbDefaultRate}, -# endif -#endif -#ifdef WEBRTC_CODEC_PCM16 - // Mono - {107, "L16", 8000, 80, 1, 128000}, - {108, "L16", 16000, 160, 1, 256000}, - {109, "L16", 32000, 320, 1, 512000}, - // Stereo - {111, "L16", 8000, 80, 2, 128000}, - {112, "L16", 16000, 160, 2, 256000}, - {113, "L16", 32000, 320, 2, 512000}, -#endif - // G.711, PCM mu-law and A-law. - // Mono - {0, "PCMU", 8000, 160, 1, 64000}, - {8, "PCMA", 8000, 160, 1, 64000}, - // Stereo - {110, "PCMU", 8000, 160, 2, 64000}, - {118, "PCMA", 8000, 160, 2, 64000}, -#ifdef WEBRTC_CODEC_ILBC - {102, "ILBC", 8000, 240, 1, 13300}, -#endif -#ifdef WEBRTC_CODEC_AMR - {114, "AMR", 8000, 160, 1, 12200}, -#endif -#ifdef WEBRTC_CODEC_AMRWB - {115, "AMR-WB", 16000, 320, 1, 20000}, -#endif -#ifdef WEBRTC_CODEC_CELT - // Mono - {116, "CELT", 32000, 640, 1, 64000}, - // Stereo - {117, "CELT", 32000, 640, 2, 64000}, -#endif -#ifdef WEBRTC_CODEC_G722 - // Mono - {9, "G722", 16000, 320, 1, 64000}, - // Stereo - {119, "G722", 16000, 320, 2, 64000}, -#endif -#ifdef WEBRTC_CODEC_G722_1 - {92, "G7221", 16000, 320, 1, 32000}, - {91, "G7221", 16000, 320, 1, 24000}, - {90, "G7221", 16000, 320, 1, 16000}, -#endif -#ifdef WEBRTC_CODEC_G722_1C - {89, "G7221", 32000, 640, 1, 48000}, - {88, "G7221", 32000, 640, 1, 32000}, - {87, "G7221", 32000, 640, 1, 24000}, -#endif -#ifdef WEBRTC_CODEC_G729 - {18, "G729", 8000, 240, 1, 8000}, -#endif -#ifdef WEBRTC_CODEC_G729_1 - {86, "G7291", 16000, 320, 1, 32000}, -#endif -#ifdef WEBRTC_CODEC_GSMFR - {3, "GSM", 8000, 160, 1, 13200}, -#endif -#ifdef WEBRTC_CODEC_OPUS - // Opus internally supports 48, 24, 16, 12, 8 kHz. - // Mono and stereo. - {120, "opus", 48000, 960, 2, 64000}, -#endif -#ifdef WEBRTC_CODEC_SPEEX - {85, "speex", 8000, 160, 1, 11000}, - {84, "speex", 16000, 320, 1, 22000}, -#endif - // Comfort noise for four different sampling frequencies. - {13, "CN", 8000, 240, 1, 0}, - {98, "CN", 16000, 480, 1, 0}, - {99, "CN", 32000, 960, 1, 0}, -#ifdef ENABLE_48000_HZ - {100, "CN", 48000, 1440, 1, 0}, -#endif -#ifdef WEBRTC_CODEC_AVT - {106, "telephone-event", 8000, 240, 1, 0}, -#endif -#ifdef WEBRTC_CODEC_RED - {127, "red", 8000, 0, 1, 0}, -#endif - // To prevent compile errors due to trailing commas. - {-1, "Null", -1, -1, -1, -1} -}; - -// Create database with all codec settings at compile time. -// Each entry needs the following parameters in the given order: -// Number of allowed packet sizes, a vector with the allowed packet sizes, -// Basic block samples, max number of channels that are supported. -const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = { -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - {2, {kIsacPacSize480, kIsacPacSize960}, 0, 1}, -# if (defined(WEBRTC_CODEC_ISAC)) - {1, {kIsacPacSize960}, 0, 1}, - {1, {kIsacPacSize1440}, 0, 1}, -# endif -#endif -#ifdef WEBRTC_CODEC_PCM16 - // Mono - {4, {80, 160, 240, 320}, 0, 2}, - {4, {160, 320, 480, 640}, 0, 2}, - {2, {320, 640}, 0, 2}, - // Stereo - {4, {80, 160, 240, 320}, 0, 2}, - {4, {160, 320, 480, 640}, 0, 2}, - {2, {320, 640}, 0, 2}, -#endif - // G.711, PCM mu-law and A-law. - // Mono - {6, {80, 160, 240, 320, 400, 480}, 0, 2}, - {6, {80, 160, 240, 320, 400, 480}, 0, 2}, - // Stereo - {6, {80, 160, 240, 320, 400, 480}, 0, 2}, - {6, {80, 160, 240, 320, 400, 480}, 0, 2}, -#ifdef WEBRTC_CODEC_ILBC - {4, {160, 240, 320, 480}, 0, 1}, -#endif -#ifdef WEBRTC_CODEC_AMR - {3, {160, 320, 480}, 0, 1}, -#endif -#ifdef WEBRTC_CODEC_AMRWB - {3, {320, 640, 960}, 0, 1}, -#endif -#ifdef WEBRTC_CODEC_CELT - // Mono - {1, {640}, 0, 2}, - // Stereo - {1, {640}, 0, 2}, -#endif -#ifdef WEBRTC_CODEC_G722 - // Mono - {6, {160, 320, 480, 640, 800, 960}, 0, 2}, - // Stereo - {6, {160, 320, 480, 640, 800, 960}, 0, 2}, -#endif -#ifdef WEBRTC_CODEC_G722_1 - {1, {320}, 320, 1}, - {1, {320}, 320, 1}, - {1, {320}, 320, 1}, -#endif -#ifdef WEBRTC_CODEC_G722_1C - {1, {640}, 640, 1}, - {1, {640}, 640, 1}, - {1, {640}, 640, 1}, -#endif -#ifdef WEBRTC_CODEC_G729 - {6, {80, 160, 240, 320, 400, 480}, 0, 1}, -#endif -#ifdef WEBRTC_CODEC_G729_1 - {3, {320, 640, 960}, 0, 1}, -#endif -#ifdef WEBRTC_CODEC_GSMFR - {3, {160, 320, 480}, 160, 1}, -#endif -#ifdef WEBRTC_CODEC_OPUS - // Opus supports frames shorter than 10ms, - // but it doesn't help us to use them. - // Mono and stereo. - {4, {480, 960, 1920, 2880}, 0, 2}, -#endif -#ifdef WEBRTC_CODEC_SPEEX - {3, {160, 320, 480}, 0, 1}, - {3, {320, 640, 960}, 0, 1}, -#endif - // Comfort noise for three different sampling frequencies. - {1, {240}, 240, 1}, - {1, {480}, 480, 1}, - {1, {960}, 960, 1}, -#ifdef ENABLE_48000_HZ - {1, {1440}, 1440, 1}, -#endif -#ifdef WEBRTC_CODEC_AVT - {1, {240}, 240, 1}, -#endif -#ifdef WEBRTC_CODEC_RED - {1, {0}, 0, 1}, -#endif - // To prevent compile errors due to trailing commas. - {-1, {-1}, -1, -1} -}; - -// Create a database of all NetEQ decoders at compile time. -const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = { -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - kDecoderISAC, -# if (defined(WEBRTC_CODEC_ISAC)) - kDecoderISACswb, - kDecoderISACfb, -# endif -#endif -#ifdef WEBRTC_CODEC_PCM16 - // Mono - kDecoderPCM16B, - kDecoderPCM16Bwb, - kDecoderPCM16Bswb32kHz, - // Stereo - kDecoderPCM16B_2ch, - kDecoderPCM16Bwb_2ch, - kDecoderPCM16Bswb32kHz_2ch, -#endif - // G.711, PCM mu-las and A-law. - // Mono - kDecoderPCMu, - kDecoderPCMa, - // Stereo - kDecoderPCMu_2ch, - kDecoderPCMa_2ch, -#ifdef WEBRTC_CODEC_ILBC - kDecoderILBC, -#endif -#ifdef WEBRTC_CODEC_AMR - kDecoderAMR, -#endif -#ifdef WEBRTC_CODEC_AMRWB - kDecoderAMRWB, -#endif -#ifdef WEBRTC_CODEC_CELT - // Mono - kDecoderCELT_32, - // Stereo - kDecoderCELT_32_2ch, -#endif -#ifdef WEBRTC_CODEC_G722 - // Mono - kDecoderG722, - // Stereo - kDecoderG722_2ch, -#endif -#ifdef WEBRTC_CODEC_G722_1 - kDecoderG722_1_32, - kDecoderG722_1_24, - kDecoderG722_1_16, -#endif -#ifdef WEBRTC_CODEC_G722_1C - kDecoderG722_1C_48, - kDecoderG722_1C_32, - kDecoderG722_1C_24, -#endif -#ifdef WEBRTC_CODEC_G729 - kDecoderG729, -#endif -#ifdef WEBRTC_CODEC_G729_1 - kDecoderG729_1, -#endif -#ifdef WEBRTC_CODEC_GSMFR - kDecoderGSMFR, -#endif -#ifdef WEBRTC_CODEC_OPUS - // Mono and stereo. - kDecoderOpus, -#endif -#ifdef WEBRTC_CODEC_SPEEX - kDecoderSPEEX_8, - kDecoderSPEEX_16, -#endif - // Comfort noise for three different sampling frequencies. - kDecoderCNG, - kDecoderCNG, - kDecoderCNG, -#ifdef ENABLE_48000_HZ - kDecoderCNG, -#endif -#ifdef WEBRTC_CODEC_AVT - kDecoderAVT, -#endif -#ifdef WEBRTC_CODEC_RED - kDecoderRED, -#endif - kDecoderReservedEnd -}; - -// Get codec information from database. -// TODO(tlegrand): replace memcpy with a pointer to the data base memory. -int ACMCodecDB::Codec(int codec_id, CodecInst* codec_inst) { - // Error check to see that codec_id is not out of bounds. - if ((codec_id < 0) || (codec_id >= kNumCodecs)) { - return -1; - } - - // Copy database information for the codec to the output. - memcpy(codec_inst, &database_[codec_id], sizeof(CodecInst)); - - return 0; -} - -// Enumerator for error codes when asking for codec database id. -enum { - kInvalidCodec = -10, - kInvalidPayloadtype = -30, - kInvalidPacketSize = -40, - kInvalidRate = -50 -}; - -// Gets the codec id number from the database. If there is some mismatch in -// the codec settings, the function will return an error code. -// NOTE! The first mismatch found will generate the return value. -int ACMCodecDB::CodecNumber(const CodecInst* codec_inst, int* mirror_id) { - // Look for a matching codec in the database. - int codec_id = CodecId(codec_inst); - - // Checks if we found a matching codec. - if (codec_id == -1) { - return kInvalidCodec; - } - - // Checks the validity of payload type - if (!ValidPayloadType(codec_inst->pltype)) { - return kInvalidPayloadtype; - } - - // Comfort Noise is special case, packet-size & rate is not checked. - if (STR_CASE_CMP(database_[codec_id].plname, "CN") == 0) { - *mirror_id = codec_id; - return codec_id; - } - - // RED is special case, packet-size & rate is not checked. - if (STR_CASE_CMP(database_[codec_id].plname, "red") == 0) { - *mirror_id = codec_id; - return codec_id; - } - - // Checks the validity of packet size. - if (codec_settings_[codec_id].num_packet_sizes > 0) { - bool packet_size_ok = false; - int i; - int packet_size_samples; - for (i = 0; i < codec_settings_[codec_id].num_packet_sizes; i++) { - packet_size_samples = - codec_settings_[codec_id].packet_sizes_samples[i]; - if (codec_inst->pacsize == packet_size_samples) { - packet_size_ok = true; - break; - } - } - - if (!packet_size_ok) { - return kInvalidPacketSize; - } - } - - if (codec_inst->pacsize < 1) { - return kInvalidPacketSize; - } - - // Check the validity of rate. Codecs with multiple rates have their own - // function for this. - *mirror_id = codec_id; - if (STR_CASE_CMP("isac", codec_inst->plname) == 0) { - if (IsISACRateValid(codec_inst->rate)) { - // Set mirrorID to iSAC WB which is only created once to be used both for - // iSAC WB and SWB, because they need to share struct. - *mirror_id = kISAC; - return codec_id; - } else { - return kInvalidRate; - } - } else if (STR_CASE_CMP("ilbc", codec_inst->plname) == 0) { - return IsILBCRateValid(codec_inst->rate, codec_inst->pacsize) - ? codec_id : kInvalidRate; - } else if (STR_CASE_CMP("amr", codec_inst->plname) == 0) { - return IsAMRRateValid(codec_inst->rate) - ? codec_id : kInvalidRate; - } else if (STR_CASE_CMP("amr-wb", codec_inst->plname) == 0) { - return IsAMRwbRateValid(codec_inst->rate) - ? codec_id : kInvalidRate; - } else if (STR_CASE_CMP("g7291", codec_inst->plname) == 0) { - return IsG7291RateValid(codec_inst->rate) - ? codec_id : kInvalidRate; - } else if (STR_CASE_CMP("opus", codec_inst->plname) == 0) { - return IsOpusRateValid(codec_inst->rate) - ? codec_id : kInvalidRate; - } else if (STR_CASE_CMP("speex", codec_inst->plname) == 0) { - return IsSpeexRateValid(codec_inst->rate) - ? codec_id : kInvalidRate; - } else if (STR_CASE_CMP("celt", codec_inst->plname) == 0) { - return IsCeltRateValid(codec_inst->rate) - ? codec_id : kInvalidRate; - } - - return IsRateValid(codec_id, codec_inst->rate) ? - codec_id : kInvalidRate; -} - -// Looks for a matching payload name, frequency, and channels in the -// codec list. Need to check all three since some codecs have several codec -// entries with different frequencies and/or channels. -// Does not check other codec settings, such as payload type and packet size. -// Returns the id of the codec, or -1 if no match is found. -int ACMCodecDB::CodecId(const CodecInst* codec_inst) { - return (CodecId(codec_inst->plname, codec_inst->plfreq, - codec_inst->channels)); -} - -int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) { - for (int id = 0; id < kNumCodecs; id++) { - bool name_match = false; - bool frequency_match = false; - bool channels_match = false; - - // Payload name, sampling frequency and number of channels need to match. - // NOTE! If |frequency| is -1, the frequency is not applicable, and is - // always treated as true, like for RED. - name_match = (STR_CASE_CMP(database_[id].plname, payload_name) == 0); - frequency_match = (frequency == database_[id].plfreq) || (frequency == -1); - // The number of channels must match for all codecs but Opus. - if (STR_CASE_CMP(payload_name, "opus") != 0) { - channels_match = (channels == database_[id].channels); - } else { - // For opus we just check that number of channels is valid. - channels_match = (channels == 1 || channels == 2); - } - - if (name_match && frequency_match && channels_match) { - // We have found a matching codec in the list. - return id; - } - } - - // We didn't find a matching codec. - return -1; -} -// Gets codec id number, and mirror id, from database for the receiver. -int ACMCodecDB::ReceiverCodecNumber(const CodecInst* codec_inst, - int* mirror_id) { - // Look for a matching codec in the database. - int codec_id = CodecId(codec_inst); - - // Set |mirror_id| to |codec_id|, except for iSAC. In case of iSAC we always - // set |mirror_id| to iSAC WB (kISAC) which is only created once to be used - // both for iSAC WB and SWB, because they need to share struct. - if (STR_CASE_CMP(codec_inst->plname, "ISAC") != 0) { - *mirror_id = codec_id; - } else { - *mirror_id = kISAC; - } - - return codec_id; -} - -// Returns the codec sampling frequency for codec with id = "codec_id" in -// database. -int ACMCodecDB::CodecFreq(int codec_id) { - // Error check to see that codec_id is not out of bounds. - if (codec_id < 0 || codec_id >= kNumCodecs) { - return -1; - } - - return database_[codec_id].plfreq; -} - -// Returns the codec's basic coding block size in samples. -int ACMCodecDB::BasicCodingBlock(int codec_id) { - // Error check to see that codec_id is not out of bounds. - if (codec_id < 0 || codec_id >= kNumCodecs) { - return -1; - } - - return codec_settings_[codec_id].basic_block_samples; -} - -// Returns the NetEQ decoder database. -const WebRtcNetEQDecoder* ACMCodecDB::NetEQDecoders() { - return neteq_decoders_; -} - -// Gets mirror id. The Id is used for codecs sharing struct for settings that -// need different payload types. -int ACMCodecDB::MirrorID(int codec_id) { - if (STR_CASE_CMP(database_[codec_id].plname, "isac") == 0) { - return kISAC; - } else { - return codec_id; - } -} - -// Creates memory/instance for storing codec state. -ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) { - // All we have support for right now. - if (!STR_CASE_CMP(codec_inst->plname, "ISAC")) { -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - return new ACMISAC(kISAC); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "PCMU")) { - if (codec_inst->channels == 1) { - return new ACMPCMU(kPCMU); - } else { - return new ACMPCMU(kPCMU_2ch); - } - } else if (!STR_CASE_CMP(codec_inst->plname, "PCMA")) { - if (codec_inst->channels == 1) { - return new ACMPCMA(kPCMA); - } else { - return new ACMPCMA(kPCMA_2ch); - } - } else if (!STR_CASE_CMP(codec_inst->plname, "ILBC")) { -#ifdef WEBRTC_CODEC_ILBC - return new ACMILBC(kILBC); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "AMR")) { -#ifdef WEBRTC_CODEC_AMR - return new ACMAMR(kGSMAMR); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "AMR-WB")) { -#ifdef WEBRTC_CODEC_AMRWB - return new ACMAMRwb(kGSMAMRWB); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "CELT")) { -#ifdef WEBRTC_CODEC_CELT - if (codec_inst->channels == 1) { - return new ACMCELT(kCELT32); - } else { - return new ACMCELT(kCELT32_2ch); - } -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "G722")) { -#ifdef WEBRTC_CODEC_G722 - if (codec_inst->channels == 1) { - return new ACMG722(kG722); - } else { - return new ACMG722(kG722_2ch); - } -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "G7221")) { - switch (codec_inst->plfreq) { - case 16000: { -#ifdef WEBRTC_CODEC_G722_1 - int codec_id; - switch (codec_inst->rate) { - case 16000 : { - codec_id = kG722_1_16; - break; - } - case 24000 : { - codec_id = kG722_1_24; - break; - } - case 32000 : { - codec_id = kG722_1_32; - break; - } - default: { - return NULL; - } - } - return new ACMG722_1(codec_id); -#endif - } - case 32000: { -#ifdef WEBRTC_CODEC_G722_1C - int codec_id; - switch (codec_inst->rate) { - case 24000 : { - codec_id = kG722_1C_24; - break; - } - case 32000 : { - codec_id = kG722_1C_32; - break; - } - case 48000 : { - codec_id = kG722_1C_48; - break; - } - default: { - return NULL; - } - } - return new ACMG722_1C(codec_id); -#endif - } - } - } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) { - // For CN we need to check sampling frequency to know what codec to create. - int codec_id; - switch (codec_inst->plfreq) { - case 8000: { - codec_id = kCNNB; - break; - } - case 16000: { - codec_id = kCNWB; - break; - } - case 32000: { - codec_id = kCNSWB; - break; - } -#ifdef ENABLE_48000_HZ - case 48000: { - codec_id = kCNFB; - break; - } -#endif - default: { - return NULL; - } - } - return new ACMCNG(codec_id); - } else if (!STR_CASE_CMP(codec_inst->plname, "G729")) { -#ifdef WEBRTC_CODEC_G729 - return new ACMG729(kG729); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "G7291")) { -#ifdef WEBRTC_CODEC_G729_1 - return new ACMG729_1(kG729_1); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "opus")) { -#ifdef WEBRTC_CODEC_OPUS - return new ACMOpus(kOpus); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "speex")) { -#ifdef WEBRTC_CODEC_SPEEX - int codec_id; - switch (codec_inst->plfreq) { - case 8000: { - codec_id = kSPEEX8; - break; - } - case 16000: { - codec_id = kSPEEX16; - break; - } - default: { - return NULL; - } - } - return new ACMSPEEX(codec_id); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) { - // For CN we need to check sampling frequency to know what codec to create. - int codec_id; - switch (codec_inst->plfreq) { - case 8000: { - codec_id = kCNNB; - break; - } - case 16000: { - codec_id = kCNWB; - break; - } - case 32000: { - codec_id = kCNSWB; - break; - } -#ifdef ENABLE_48000_HZ - case 48000: { - codec_id = kCNFB; - break; - } -#endif - default: { - return NULL; - } - } - return new ACMCNG(codec_id); - } else if (!STR_CASE_CMP(codec_inst->plname, "L16")) { -#ifdef WEBRTC_CODEC_PCM16 - // For L16 we need to check sampling frequency to know what codec to create. - int codec_id; - if (codec_inst->channels == 1) { - switch (codec_inst->plfreq) { - case 8000: { - codec_id = kPCM16B; - break; - } - case 16000: { - codec_id = kPCM16Bwb; - break; - } - case 32000: { - codec_id = kPCM16Bswb32kHz; - break; - } - default: { - return NULL; - } - } - } else { - switch (codec_inst->plfreq) { - case 8000: { - codec_id = kPCM16B_2ch; - break; - } - case 16000: { - codec_id = kPCM16Bwb_2ch; - break; - } - case 32000: { - codec_id = kPCM16Bswb32kHz_2ch; - break; - } - default: { - return NULL; - } - } - } - return new ACMPCM16B(codec_id); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "telephone-event")) { -#ifdef WEBRTC_CODEC_AVT - return new ACMDTMFPlayout(kAVT); -#endif - } else if (!STR_CASE_CMP(codec_inst->plname, "red")) { -#ifdef WEBRTC_CODEC_RED - return new ACMRED(kRED); -#endif - } - return NULL; -} - -// Checks if the bitrate is valid for the codec. -bool ACMCodecDB::IsRateValid(int codec_id, int rate) { - if (database_[codec_id].rate == rate) { - return true; - } else { - return false; - } -} - -// Checks if the bitrate is valid for iSAC. -bool ACMCodecDB::IsISACRateValid(int rate) { - if ((rate == -1) || ((rate <= 56000) && (rate >= 10000))) { - return true; - } else { - return false; - } -} - -// Checks if the bitrate is valid for iLBC. -bool ACMCodecDB::IsILBCRateValid(int rate, int frame_size_samples) { - if (((frame_size_samples == 240) || (frame_size_samples == 480)) && - (rate == 13300)) { - return true; - } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) && - (rate == 15200)) { - return true; - } else { - return false; - } -} - -// Check if the bitrate is valid for the GSM-AMR. -bool ACMCodecDB::IsAMRRateValid(int rate) { - switch (rate) { - case 4750: - case 5150: - case 5900: - case 6700: - case 7400: - case 7950: - case 10200: - case 12200: { - return true; - } - default: { - return false; - } - } -} - -// Check if the bitrate is valid for GSM-AMR-WB. -bool ACMCodecDB::IsAMRwbRateValid(int rate) { - switch (rate) { - case 7000: - case 9000: - case 12000: - case 14000: - case 16000: - case 18000: - case 20000: - case 23000: - case 24000: { - return true; - } - default: { - return false; - } - } -} - -// Check if the bitrate is valid for G.729.1. -bool ACMCodecDB::IsG7291RateValid(int rate) { - switch (rate) { - case 8000: - case 12000: - case 14000: - case 16000: - case 18000: - case 20000: - case 22000: - case 24000: - case 26000: - case 28000: - case 30000: - case 32000: { - return true; - } - default: { - return false; - } - } -} - -// Checks if the bitrate is valid for Speex. -bool ACMCodecDB::IsSpeexRateValid(int rate) { - if (rate > 2000) { - return true; - } else { - return false; - } -} - -// Checks if the bitrate is valid for Opus. -bool ACMCodecDB::IsOpusRateValid(int rate) { - if ((rate < 6000) || (rate > 510000)) { - return false; - } - return true; -} - -// Checks if the bitrate is valid for Celt. -bool ACMCodecDB::IsCeltRateValid(int rate) { - if ((rate >= 48000) && (rate <= 128000)) { - return true; - } else { - return false; - } -} - -// Checks if the payload type is in the valid range. -bool ACMCodecDB::ValidPayloadType(int payload_type) { - if ((payload_type < 0) || (payload_type > 127)) { - return false; - } - return true; -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.h deleted file mode 100644 index 7a7054dd1aa..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.h +++ /dev/null @@ -1,336 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file generates databases with information about all supported audio - * codecs. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ - -#include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" - -namespace webrtc { - -namespace acm1 { - -// TODO(tlegrand): replace class ACMCodecDB with a namespace. -class ACMCodecDB { - public: - // Enum with array indexes for the supported codecs. NOTE! The order MUST - // be the same as when creating the database in acm_codec_database.cc. - enum { - kNone = -1 -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - , kISAC -# if (defined(WEBRTC_CODEC_ISAC)) - , kISACSWB - , kISACFB -# endif -#endif -#ifdef WEBRTC_CODEC_PCM16 - // Mono - , kPCM16B - , kPCM16Bwb - , kPCM16Bswb32kHz - // Stereo - , kPCM16B_2ch - , kPCM16Bwb_2ch - , kPCM16Bswb32kHz_2ch -#endif - // Mono - , kPCMU - , kPCMA - // Stereo - , kPCMU_2ch - , kPCMA_2ch -#ifdef WEBRTC_CODEC_ILBC - , kILBC -#endif -#ifdef WEBRTC_CODEC_AMR - , kGSMAMR -#endif -#ifdef WEBRTC_CODEC_AMRWB - , kGSMAMRWB -#endif -#ifdef WEBRTC_CODEC_CELT - // Mono - , kCELT32 - // Stereo - , kCELT32_2ch -#endif -#ifdef WEBRTC_CODEC_G722 - // Mono - , kG722 - // Stereo - , kG722_2ch -#endif -#ifdef WEBRTC_CODEC_G722_1 - , kG722_1_32 - , kG722_1_24 - , kG722_1_16 -#endif -#ifdef WEBRTC_CODEC_G722_1C - , kG722_1C_48 - , kG722_1C_32 - , kG722_1C_24 -#endif -#ifdef WEBRTC_CODEC_G729 - , kG729 -#endif -#ifdef WEBRTC_CODEC_G729_1 - , kG729_1 -#endif -#ifdef WEBRTC_CODEC_GSMFR - , kGSMFR -#endif -#ifdef WEBRTC_CODEC_OPUS - // Mono and stereo - , kOpus -#endif -#ifdef WEBRTC_CODEC_SPEEX - , kSPEEX8 - , kSPEEX16 -#endif - , kCNNB - , kCNWB - , kCNSWB -#ifdef ENABLE_48000_HZ - , kCNFB -#endif -#ifdef WEBRTC_CODEC_AVT - , kAVT -#endif -#ifdef WEBRTC_CODEC_RED - , kRED -#endif - , kNumCodecs - }; - - // Set unsupported codecs to -1 -#ifndef WEBRTC_CODEC_ISAC - enum {kISACSWB = -1}; - enum {kISACFB = -1}; -# ifndef WEBRTC_CODEC_ISACFX - enum {kISAC = -1}; -# endif -#endif -#ifndef WEBRTC_CODEC_PCM16 - // Mono - enum {kPCM16B = -1}; - enum {kPCM16Bwb = -1}; - enum {kPCM16Bswb32kHz = -1}; - // Stereo - enum {kPCM16B_2ch = -1}; - enum {kPCM16Bwb_2ch = -1}; - enum {kPCM16Bswb32kHz_2ch = -1}; -#endif - // 48 kHz not supported, always set to -1. - enum {kPCM16Bswb48kHz = -1}; -#ifndef WEBRTC_CODEC_ILBC - enum {kILBC = -1}; -#endif -#ifndef WEBRTC_CODEC_AMR - enum {kGSMAMR = -1}; -#endif -#ifndef WEBRTC_CODEC_AMRWB - enum {kGSMAMRWB = -1}; -#endif -#ifndef WEBRTC_CODEC_CELT - // Mono - enum {kCELT32 = -1}; - // Stereo - enum {kCELT32_2ch = -1}; -#endif -#ifndef WEBRTC_CODEC_G722 - // Mono - enum {kG722 = -1}; - // Stereo - enum {kG722_2ch = -1}; -#endif -#ifndef WEBRTC_CODEC_G722_1 - enum {kG722_1_32 = -1}; - enum {kG722_1_24 = -1}; - enum {kG722_1_16 = -1}; -#endif -#ifndef WEBRTC_CODEC_G722_1C - enum {kG722_1C_48 = -1}; - enum {kG722_1C_32 = -1}; - enum {kG722_1C_24 = -1}; -#endif -#ifndef WEBRTC_CODEC_G729 - enum {kG729 = -1}; -#endif -#ifndef WEBRTC_CODEC_G729_1 - enum {kG729_1 = -1}; -#endif -#ifndef WEBRTC_CODEC_GSMFR - enum {kGSMFR = -1}; -#endif -#ifndef WEBRTC_CODEC_SPEEX - enum {kSPEEX8 = -1}; - enum {kSPEEX16 = -1}; -#endif -#ifndef WEBRTC_CODEC_OPUS - // Mono and stereo - enum {kOpus = -1}; -#endif -#ifndef WEBRTC_CODEC_AVT - enum {kAVT = -1}; -#endif -#ifndef WEBRTC_CODEC_RED - enum {kRED = -1}; -#endif - - // kMaxNumCodecs - Maximum number of codecs that can be activated in one - // build. - // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec. - // These might need to be increased if adding a new codec to the database - static const int kMaxNumCodecs = 50; - static const int kMaxNumPacketSize = 6; - - // Codec specific settings - // - // num_packet_sizes - number of allowed packet sizes. - // packet_sizes_samples - list of the allowed packet sizes. - // basic_block_samples - assigned a value different from 0 if the codec - // requires to be fed with a specific number of samples - // that can be different from packet size. - // channel_support - number of channels supported to encode; - // 1 = mono, 2 = stereo, etc. - struct CodecSettings { - int num_packet_sizes; - int packet_sizes_samples[kMaxNumPacketSize]; - int basic_block_samples; - int channel_support; - }; - - // Gets codec information from database at the position in database given by - // [codec_id]. - // Input: - // [codec_id] - number that specifies at what position in the database to - // get the information. - // Output: - // [codec_inst] - filled with information about the codec. - // Return: - // 0 if successful, otherwise -1. - static int Codec(int codec_id, CodecInst* codec_inst); - - // Returns codec id and mirror id from database, given the information - // received in the input [codec_inst]. Mirror id is a number that tells - // where to find the codec's memory (instance). The number is either the - // same as codec id (most common), or a number pointing at a different - // entry in the database, if the codec has several entries with different - // payload types. This is used for codecs that must share one struct even if - // the payload type differs. - // One example is the codec iSAC which has the same struct for both 16 and - // 32 khz, but they have different entries in the database. Let's say the - // function is called with iSAC 32kHz. The function will return 1 as that is - // the entry in the data base, and [mirror_id] = 0, as that is the entry for - // iSAC 16 kHz, which holds the shared memory. - // Input: - // [codec_inst] - Information about the codec for which we require the - // database id. - // Output: - // [mirror_id] - mirror id, which most often is the same as the return - // value, see above. - // Return: - // codec id if successful, otherwise < 0. - static int CodecNumber(const CodecInst* codec_inst, int* mirror_id); - static int CodecId(const CodecInst* codec_inst); - static int CodecId(const char* payload_name, int frequency, int channels); - static int ReceiverCodecNumber(const CodecInst* codec_inst, int* mirror_id); - - // Returns the codec sampling frequency for codec with id = "codec_id" in - // database. - // TODO(tlegrand): Check if function is needed, or if we can change - // to access database directly. - // Input: - // [codec_id] - number that specifies at what position in the database to - // get the information. - // Return: - // codec sampling frequency if successful, otherwise -1. - static int CodecFreq(int codec_id); - - // Return the codec's basic coding block size in samples. - // TODO(tlegrand): Check if function is needed, or if we can change - // to access database directly. - // Input: - // [codec_id] - number that specifies at what position in the database to - // get the information. - // Return: - // codec basic block size if successful, otherwise -1. - static int BasicCodingBlock(int codec_id); - - // Returns the NetEQ decoder database. - static const WebRtcNetEQDecoder* NetEQDecoders(); - - // Returns mirror id, which is a number that tells where to find the codec's - // memory (instance). It is either the same as codec id (most common), or a - // number pointing at a different entry in the database, if the codec have - // several entries with different payload types. This is used for codecs that - // must share struct even if the payload type differs. - // TODO(tlegrand): Check if function is needed, or if we can change - // to access database directly. - // Input: - // [codec_id] - number that specifies codec's position in the database. - // Return: - // Mirror id on success, otherwise -1. - static int MirrorID(int codec_id); - - // Create memory/instance for storing codec state. - // Input: - // [codec_inst] - information about codec. Only name of codec, "plname", is - // used in this function. - static ACMGenericCodec* CreateCodecInstance(const CodecInst* codec_inst); - - // Checks if the bitrate is valid for the codec. - // Input: - // [codec_id] - number that specifies codec's position in the database. - // [rate] - bitrate to check. - // [frame_size_samples] - (used for iLBC) specifies which frame size to go - // with the rate. - static bool IsRateValid(int codec_id, int rate); - static bool IsISACRateValid(int rate); - static bool IsILBCRateValid(int rate, int frame_size_samples); - static bool IsAMRRateValid(int rate); - static bool IsAMRwbRateValid(int rate); - static bool IsG7291RateValid(int rate); - static bool IsSpeexRateValid(int rate); - static bool IsOpusRateValid(int rate); - static bool IsCeltRateValid(int rate); - - // Check if the payload type is valid, meaning that it is in the valid range - // of 0 to 127. - // Input: - // [payload_type] - payload type. - static bool ValidPayloadType(int payload_type); - - // Databases with information about the supported codecs - // database_ - stored information about all codecs: payload type, name, - // sampling frequency, packet size in samples, default channel - // support, and default rate. - // codec_settings_ - stored codec settings: number of allowed packet sizes, - // a vector with the allowed packet sizes, basic block - // samples, and max number of channels that are supported. - // neteq_decoders_ - list of supported decoders in NetEQ. - static const CodecInst database_[kMaxNumCodecs]; - static const CodecSettings codec_settings_[kMaxNumCodecs]; - static const WebRtcNetEQDecoder neteq_decoders_[kMaxNumCodecs]; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc deleted file mode 100644 index edb62987689..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc +++ /dev/null @@ -1,42 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h" - -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" - -namespace webrtc { - -namespace acm1 { - -ACMDTMFDetection::ACMDTMFDetection() {} - -ACMDTMFDetection::~ACMDTMFDetection() {} - -int16_t ACMDTMFDetection::Enable(ACMCountries /* cpt */) { - return -1; -} - -int16_t ACMDTMFDetection::Disable() { - return -1; -} - -int16_t ACMDTMFDetection::Detect( - const int16_t* /* in_audio_buff */, - const uint16_t /* in_buff_len_word16 */, - const int32_t /* in_freq_hz */, - bool& /* tone_detected */, - int16_t& /* tone */) { - return -1; -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h deleted file mode 100644 index 74553107a36..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h +++ /dev/null @@ -1,42 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_ - -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/source/acm_resampler.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -namespace acm1 { - -class ACMDTMFDetection { - public: - ACMDTMFDetection(); - ~ACMDTMFDetection(); - int16_t Enable(ACMCountries cpt = ACMDisableCountryDetection); - int16_t Disable(); - int16_t Detect(const int16_t* in_audio_buff, - const uint16_t in_buff_len_word16, - const int32_t in_freq_hz, - bool& tone_detected, - int16_t& tone); - - private: - ACMResampler resampler_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc deleted file mode 100644 index 32195e6fe82..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc +++ /dev/null @@ -1,171 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_AVT - -ACMDTMFPlayout::ACMDTMFPlayout( - int16_t /* codec_id */) { - return; -} - -ACMDTMFPlayout::~ACMDTMFPlayout() { - return; -} - -int16_t ACMDTMFPlayout::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMDTMFPlayout::DecodeSafe( - uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMDTMFPlayout::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMDTMFPlayout::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMDTMFPlayout::CreateInstance(void) { - return NULL; -} - -int16_t ACMDTMFPlayout::InternalCreateEncoder() { - return -1; -} - -int16_t ACMDTMFPlayout::InternalCreateDecoder() { - return -1; -} - -void ACMDTMFPlayout::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -void ACMDTMFPlayout::DestructEncoderSafe() { - return; -} - -void ACMDTMFPlayout::DestructDecoderSafe() { - return; -} - -#else //===================== Actual Implementation ======================= - -ACMDTMFPlayout::ACMDTMFPlayout(int16_t codec_id) { - codec_id_ = codec_id; -} - -ACMDTMFPlayout::~ACMDTMFPlayout() { - return; -} - -int16_t ACMDTMFPlayout::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return 0; -} - -int16_t ACMDTMFPlayout::DecodeSafe( - uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMDTMFPlayout::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, - // DTMFPlayout has no instance - return 0; -} - -int16_t ACMDTMFPlayout::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, - // DTMFPlayout has no instance - return 0; -} - -int32_t ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_AVT_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderAVT, codec_inst.pltype, NULL, 8000); - SET_AVT_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMDTMFPlayout::CreateInstance(void) { - return NULL; -} - -int16_t ACMDTMFPlayout::InternalCreateEncoder() { - // DTMFPlayout has no instance - return 0; -} - -int16_t ACMDTMFPlayout::InternalCreateDecoder() { - // DTMFPlayout has no instance - return 0; -} - -void ACMDTMFPlayout::InternalDestructEncoderInst(void* /* ptr_inst */) { - // DTMFPlayout has no instance - return; -} - -void ACMDTMFPlayout::DestructEncoderSafe() { - // DTMFPlayout has no instance - return; -} - -void ACMDTMFPlayout::DestructDecoderSafe() { - // DTMFPlayout has no instance - return; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h deleted file mode 100644 index 46175f59e66..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h +++ /dev/null @@ -1,62 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -namespace webrtc { - -namespace acm1 { - -class ACMDTMFPlayout: public ACMGenericCodec { - public: - explicit ACMDTMFPlayout(int16_t codec_id); - virtual ~ACMDTMFPlayout(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc deleted file mode 100644 index 1c19109b6a3..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc +++ /dev/null @@ -1,358 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_g722.h" - -#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_G722 - -ACMG722::ACMG722(int16_t /* codec_id */) - : ptr_enc_str_(NULL), - ptr_dec_str_(NULL), - encoder_inst_ptr_(NULL), - encoder_inst_ptr_right_(NULL), - decoder_inst_ptr_(NULL) {} - -ACMG722::~ACMG722() {} - -int32_t ACMG722::Add10MsDataSafe( - const uint32_t /* timestamp */, - const int16_t* /* data */, - const uint16_t /* length_smpl */, - const uint8_t /* audio_channel */) { - return -1; -} - -int16_t ACMG722::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMG722::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMG722::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMG722::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMG722::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMG722::CreateInstance(void) { - return NULL; -} - -int16_t ACMG722::InternalCreateEncoder() { - return -1; -} - -void ACMG722::DestructEncoderSafe() { - return; -} - -int16_t ACMG722::InternalCreateDecoder() { - return -1; -} - -void ACMG722::DestructDecoderSafe() { - return; -} - -void ACMG722::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -void ACMG722::SplitStereoPacket(uint8_t* /*payload*/, - int32_t* /*payload_length*/) {} - -#else //===================== Actual Implementation ======================= - -// Encoder and decoder memory -struct ACMG722EncStr { - G722EncInst* inst; // instance for left channel in case of stereo - G722EncInst* inst_right; // instance for right channel in case of stereo -}; -struct ACMG722DecStr { - G722DecInst* inst; // instance for left channel in case of stereo - G722DecInst* inst_right; // instance for right channel in case of stereo -}; - -ACMG722::ACMG722(int16_t codec_id) - : encoder_inst_ptr_(NULL), - encoder_inst_ptr_right_(NULL), - decoder_inst_ptr_(NULL) { - // Encoder - ptr_enc_str_ = new ACMG722EncStr; - if (ptr_enc_str_ != NULL) { - ptr_enc_str_->inst = NULL; - ptr_enc_str_->inst_right = NULL; - } - // Decoder - ptr_dec_str_ = new ACMG722DecStr; - if (ptr_dec_str_ != NULL) { - ptr_dec_str_->inst = NULL; - ptr_dec_str_->inst_right = NULL; // Not used - } - codec_id_ = codec_id; - return; -} - -ACMG722::~ACMG722() { - // Encoder - if (ptr_enc_str_ != NULL) { - if (ptr_enc_str_->inst != NULL) { - WebRtcG722_FreeEncoder(ptr_enc_str_->inst); - ptr_enc_str_->inst = NULL; - } - if (ptr_enc_str_->inst_right != NULL) { - WebRtcG722_FreeEncoder(ptr_enc_str_->inst_right); - ptr_enc_str_->inst_right = NULL; - } - delete ptr_enc_str_; - ptr_enc_str_ = NULL; - } - // Decoder - if (ptr_dec_str_ != NULL) { - if (ptr_dec_str_->inst != NULL) { - WebRtcG722_FreeDecoder(ptr_dec_str_->inst); - ptr_dec_str_->inst = NULL; - } - if (ptr_dec_str_->inst_right != NULL) { - WebRtcG722_FreeDecoder(ptr_dec_str_->inst_right); - ptr_dec_str_->inst_right = NULL; - } - delete ptr_dec_str_; - ptr_dec_str_ = NULL; - } - return; -} - -int32_t ACMG722::Add10MsDataSafe(const uint32_t timestamp, - const int16_t* data, - const uint16_t length_smpl, - const uint8_t audio_channel) { - return ACMGenericCodec::Add10MsDataSafe((timestamp >> 1), data, length_smpl, - audio_channel); -} - -int16_t ACMG722::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - // If stereo, split input signal in left and right channel before encoding - if (num_channels_ == 2) { - int16_t left_channel[960]; - int16_t right_channel[960]; - uint8_t out_left[480]; - uint8_t out_right[480]; - int16_t len_in_bytes; - for (int i = 0, j = 0; i < frame_len_smpl_ * 2; i += 2, j++) { - left_channel[j] = in_audio_[in_audio_ix_read_ + i]; - right_channel[j] = in_audio_[in_audio_ix_read_ + i + 1]; - } - len_in_bytes = WebRtcG722_Encode(encoder_inst_ptr_, left_channel, - frame_len_smpl_, - (int16_t*)out_left); - len_in_bytes += WebRtcG722_Encode(encoder_inst_ptr_right_, right_channel, - frame_len_smpl_, - (int16_t*)out_right); - *bitstream_len_byte = len_in_bytes; - - // Interleave the 4 bits per sample from left and right channel - for (int i = 0, j = 0; i < len_in_bytes; i += 2, j++) { - bitstream[i] = (out_left[j] & 0xF0) + (out_right[j] >> 4); - bitstream[i + 1] = ((out_left[j] & 0x0F) << 4) + (out_right[j] & 0x0F); - } - } else { - *bitstream_len_byte = WebRtcG722_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - frame_len_smpl_, - (int16_t*)bitstream); - } - - // increment the read index this tell the caller how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += frame_len_smpl_ * num_channels_; - return *bitstream_len_byte; -} - -int16_t ACMG722::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMG722::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - if (codec_params->codec_inst.channels == 2) { - // Create codec struct for right channel - if (ptr_enc_str_->inst_right == NULL) { - WebRtcG722_CreateEncoder(&ptr_enc_str_->inst_right); - if (ptr_enc_str_->inst_right == NULL) { - return -1; - } - } - encoder_inst_ptr_right_ = ptr_enc_str_->inst_right; - if (WebRtcG722_EncoderInit(encoder_inst_ptr_right_) < 0) { - return -1; - } - } - - return WebRtcG722_EncoderInit(encoder_inst_ptr_); -} - -int16_t ACMG722::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return WebRtcG722_DecoderInit(decoder_inst_ptr_); -} - -int32_t ACMG722::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - // TODO(turajs): log error - return -1; - } - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_G722_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - if (codec_inst.channels == 1) { - SET_CODEC_PAR(codec_def, kDecoderG722, codec_inst.pltype, decoder_inst_ptr_, - 16000); - } else { - SET_CODEC_PAR(codec_def, kDecoderG722_2ch, codec_inst.pltype, - decoder_inst_ptr_, 16000); - } - SET_G722_FUNCTIONS(codec_def); - return 0; -} - -ACMGenericCodec* ACMG722::CreateInstance(void) { - return NULL; -} - -int16_t ACMG722::InternalCreateEncoder() { - if (ptr_enc_str_ == NULL) { - // this structure must be created at the costructor - // if it is still NULL then there is a probelm and - // we dont continue - return -1; - } - WebRtcG722_CreateEncoder(&ptr_enc_str_->inst); - if (ptr_enc_str_->inst == NULL) { - return -1; - } - encoder_inst_ptr_ = ptr_enc_str_->inst; - return 0; -} - -void ACMG722::DestructEncoderSafe() { - if (ptr_enc_str_ != NULL) { - if (ptr_enc_str_->inst != NULL) { - WebRtcG722_FreeEncoder(ptr_enc_str_->inst); - ptr_enc_str_->inst = NULL; - } - } - encoder_exist_ = false; - encoder_initialized_ = false; -} - -int16_t ACMG722::InternalCreateDecoder() { - if (ptr_dec_str_ == NULL) { - // this structure must be created at the costructor - // if it is still NULL then there is a probelm and - // we dont continue - return -1; - } - - WebRtcG722_CreateDecoder(&ptr_dec_str_->inst); - if (ptr_dec_str_->inst == NULL) { - return -1; - } - decoder_inst_ptr_ = ptr_dec_str_->inst; - return 0; -} - -void ACMG722::DestructDecoderSafe() { - decoder_exist_ = false; - decoder_initialized_ = false; - if (ptr_dec_str_ != NULL) { - if (ptr_dec_str_->inst != NULL) { - WebRtcG722_FreeDecoder(ptr_dec_str_->inst); - ptr_dec_str_->inst = NULL; - } - } -} - -void ACMG722::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcG722_FreeEncoder(static_cast<G722EncInst*>(ptr_inst)); - } - return; -} - -// Split the stereo packet and place left and right channel after each other -// in the payload vector. -void ACMG722::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { - uint8_t right_byte; - - // Check for valid inputs. - assert(payload != NULL); - assert(*payload_length > 0); - - // Regroup the 4 bits/sample so to |l1 l2| |r1 r2| |l3 l4| |r3 r4| ..., - // where "lx" is 4 bits representing left sample number x, and "rx" right - // sample. Two samples fits in one byte, represented with |...|. - for (int i = 0; i < *payload_length; i += 2) { - right_byte = ((payload[i] & 0x0F) << 4) + (payload[i + 1] & 0x0F); - payload[i] = (payload[i] & 0xF0) + (payload[i + 1] >> 4); - payload[i + 1] = right_byte; - } - - // Move one byte representing right channel each loop, and place it at the - // end of the bytestream vector. After looping the data is reordered to: - // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|, - // where N is the total number of samples. - for (int i = 0; i < *payload_length / 2; i++) { - right_byte = payload[i + 1]; - memmove(&payload[i + 1], &payload[i + 2], *payload_length - i - 2); - payload[*payload_length - 1] = right_byte; - } -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g722.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g722.h deleted file mode 100644 index cf7ebe1e223..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g722.h +++ /dev/null @@ -1,84 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -typedef struct WebRtcG722EncInst G722EncInst; -typedef struct WebRtcG722DecInst G722DecInst; - -namespace webrtc { - -namespace acm1 { - -// forward declaration -struct ACMG722EncStr; -struct ACMG722DecStr; - -class ACMG722 : public ACMGenericCodec { - public: - explicit ACMG722(int16_t codec_id); - virtual ~ACMG722(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual int32_t Add10MsDataSafe(const uint32_t timestamp, - const int16_t* data, - const uint16_t length_smpl, - const uint8_t audio_channel) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual void SplitStereoPacket(uint8_t* payload, - int32_t* payload_length) OVERRIDE; - - ACMG722EncStr* ptr_enc_str_; - ACMG722DecStr* ptr_dec_str_; - - G722EncInst* encoder_inst_ptr_; - G722EncInst* encoder_inst_ptr_right_; // Prepared for stereo - G722DecInst* decoder_inst_ptr_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc deleted file mode 100644 index ed172fd3e1b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc +++ /dev/null @@ -1,500 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_g7221.h" - -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_G722_1 -// NOTE! G.722.1 is not included in the open-source package. The following -// interface file is needed: -// -// /modules/audio_coding/codecs/g7221/main/interface/g7221_interface.h -// -// The API in the header file should match the one below. -// -// int16_t WebRtcG7221_CreateEnc16(G722_1_16_encinst_t_** enc_inst); -// int16_t WebRtcG7221_CreateEnc24(G722_1_24_encinst_t_** enc_inst); -// int16_t WebRtcG7221_CreateEnc32(G722_1_32_encinst_t_** enc_inst); -// int16_t WebRtcG7221_CreateDec16(G722_1_16_decinst_t_** dec_inst); -// int16_t WebRtcG7221_CreateDec24(G722_1_24_decinst_t_** dec_inst); -// int16_t WebRtcG7221_CreateDec32(G722_1_32_decinst_t_** dec_inst); -// -// int16_t WebRtcG7221_FreeEnc16(G722_1_16_encinst_t_** enc_inst); -// int16_t WebRtcG7221_FreeEnc24(G722_1_24_encinst_t_** enc_inst); -// int16_t WebRtcG7221_FreeEnc32(G722_1_32_encinst_t_** enc_inst); -// int16_t WebRtcG7221_FreeDec16(G722_1_16_decinst_t_** dec_inst); -// int16_t WebRtcG7221_FreeDec24(G722_1_24_decinst_t_** dec_inst); -// int16_t WebRtcG7221_FreeDec32(G722_1_32_decinst_t_** dec_inst); -// -// int16_t WebRtcG7221_EncoderInit16(G722_1_16_encinst_t_* enc_inst); -// int16_t WebRtcG7221_EncoderInit24(G722_1_24_encinst_t_* enc_inst); -// int16_t WebRtcG7221_EncoderInit32(G722_1_32_encinst_t_* enc_inst); -// int16_t WebRtcG7221_DecoderInit16(G722_1_16_decinst_t_* dec_inst); -// int16_t WebRtcG7221_DecoderInit24(G722_1_24_decinst_t_* dec_inst); -// int16_t WebRtcG7221_DecoderInit32(G722_1_32_decinst_t_* dec_inst); -// -// int16_t WebRtcG7221_Encode16(G722_1_16_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221_Encode24(G722_1_24_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221_Encode32(G722_1_32_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t* output); -// -// int16_t WebRtcG7221_Decode16(G722_1_16_decinst_t_* dec_inst, -// int16_t* bitstream, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221_Decode24(G722_1_24_decinst_t_* dec_inst, -// int16_t* bitstream, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221_Decode32(G722_1_32_decinst_t_* dec_inst, -// int16_t* bitstream, -// int16_t len, -// int16_t* output); -// -// int16_t WebRtcG7221_DecodePlc16(G722_1_16_decinst_t_* dec_inst, -// int16_t* output, -// int16_t nr_lost_frames); -// int16_t WebRtcG7221_DecodePlc24(G722_1_24_decinst_t_* dec_inst, -// int16_t* output, -// int16_t nr_lost_frames); -// int16_t WebRtcG7221_DecodePlc32(G722_1_32_decinst_t_* dec_inst, -// int16_t* output, -// int16_t nr_lost_frames); -#include "g7221_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_G722_1 - -ACMG722_1::ACMG722_1(int16_t /* codec_id */) - : operational_rate_(-1), - encoder_inst_ptr_(NULL), - encoder_inst_ptr_right_(NULL), - decoder_inst_ptr_(NULL), - encoder_inst16_ptr_(NULL), - encoder_inst16_ptr_right_(NULL), - encoder_inst24_ptr_(NULL), - encoder_inst24_ptr_right_(NULL), - encoder_inst32_ptr_(NULL), - encoder_inst32_ptr_right_(NULL), - decoder_inst16_ptr_(NULL), - decoder_inst24_ptr_(NULL), - decoder_inst32_ptr_(NULL) { - return; -} - -ACMG722_1::~ACMG722_1() { - return; -} - -int16_t ACMG722_1::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMG722_1::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMG722_1::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMG722_1::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMG722_1::CreateInstance(void) { - return NULL; -} - -int16_t ACMG722_1::InternalCreateEncoder() { - return -1; -} - -void ACMG722_1::DestructEncoderSafe() { - return; -} - -int16_t ACMG722_1::InternalCreateDecoder() { - return -1; -} - -void ACMG722_1::DestructDecoderSafe() { - return; -} - -void ACMG722_1::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -#else //===================== Actual Implementation ======================= -ACMG722_1::ACMG722_1(int16_t codec_id) - : encoder_inst_ptr_(NULL), - encoder_inst_ptr_right_(NULL), - decoder_inst_ptr_(NULL), - encoder_inst16_ptr_(NULL), - encoder_inst16_ptr_right_(NULL), - encoder_inst24_ptr_(NULL), - encoder_inst24_ptr_right_(NULL), - encoder_inst32_ptr_(NULL), - encoder_inst32_ptr_right_(NULL), - decoder_inst16_ptr_(NULL), - decoder_inst24_ptr_(NULL), - decoder_inst32_ptr_(NULL) { - codec_id_ = codec_id; - if (codec_id_ == ACMCodecDB::kG722_1_16) { - operational_rate_ = 16000; - } else if (codec_id_ == ACMCodecDB::kG722_1_24) { - operational_rate_ = 24000; - } else if (codec_id_ == ACMCodecDB::kG722_1_32) { - operational_rate_ = 32000; - } else { - operational_rate_ = -1; - } - return; -} - -ACMG722_1::~ACMG722_1() { - if (encoder_inst_ptr_ != NULL) { - delete encoder_inst_ptr_; - encoder_inst_ptr_ = NULL; - } - if (encoder_inst_ptr_right_ != NULL) { - delete encoder_inst_ptr_right_; - encoder_inst_ptr_right_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - delete decoder_inst_ptr_; - decoder_inst_ptr_ = NULL; - } - - switch (operational_rate_) { - case 16000: { - encoder_inst16_ptr_ = NULL; - encoder_inst16_ptr_right_ = NULL; - decoder_inst16_ptr_ = NULL; - break; - } - case 24000: { - encoder_inst24_ptr_ = NULL; - encoder_inst24_ptr_right_ = NULL; - decoder_inst24_ptr_ = NULL; - break; - } - case 32000: { - encoder_inst32_ptr_ = NULL; - encoder_inst32_ptr_right_ = NULL; - decoder_inst32_ptr_ = NULL; - break; - } - default: { - break; - } - } - return; -} - -int16_t ACMG722_1::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - int16_t left_channel[320]; - int16_t right_channel[320]; - int16_t len_in_bytes; - int16_t out_bits[160]; - - // If stereo, split input signal in left and right channel before encoding - if (num_channels_ == 2) { - for (int i = 0, j = 0; i < frame_len_smpl_ * 2; i += 2, j++) { - left_channel[j] = in_audio_[in_audio_ix_read_ + i]; - right_channel[j] = in_audio_[in_audio_ix_read_ + i + 1]; - } - } else { - memcpy(left_channel, &in_audio_[in_audio_ix_read_], 320); - } - - switch (operational_rate_) { - case 16000: { - len_in_bytes = WebRtcG7221_Encode16(encoder_inst16_ptr_, left_channel, - 320, &out_bits[0]); - if (num_channels_ == 2) { - len_in_bytes += WebRtcG7221_Encode16(encoder_inst16_ptr_right_, - right_channel, 320, - &out_bits[len_in_bytes / 2]); - } - break; - } - case 24000: { - len_in_bytes = WebRtcG7221_Encode24(encoder_inst24_ptr_, left_channel, - 320, &out_bits[0]); - if (num_channels_ == 2) { - len_in_bytes += WebRtcG7221_Encode24(encoder_inst24_ptr_right_, - right_channel, 320, - &out_bits[len_in_bytes / 2]); - } - break; - } - case 32000: { - len_in_bytes = WebRtcG7221_Encode32(encoder_inst32_ptr_, left_channel, - 320, &out_bits[0]); - if (num_channels_ == 2) { - len_in_bytes += WebRtcG7221_Encode32(encoder_inst32_ptr_right_, - right_channel, 320, - &out_bits[len_in_bytes / 2]); - } - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitEncode: Wrong rate for G722_1."); - return -1; - } - } - memcpy(bitstream, out_bits, len_in_bytes); - *bitstream_len_byte = len_in_bytes; - - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += 320 * num_channels_; - return *bitstream_len_byte; -} - -int16_t ACMG722_1::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMG722_1::InternalInitEncoder( - WebRtcACMCodecParams* codec_params) { - int16_t ret; - - switch (operational_rate_) { - case 16000: { - ret = WebRtcG7221_EncoderInit16(encoder_inst16_ptr_right_); - if (ret < 0) { - return ret; - } - return WebRtcG7221_EncoderInit16(encoder_inst16_ptr_); - } - case 24000: { - ret = WebRtcG7221_EncoderInit24(encoder_inst24_ptr_right_); - if (ret < 0) { - return ret; - } - return WebRtcG7221_EncoderInit24(encoder_inst24_ptr_); - } - case 32000: { - ret = WebRtcG7221_EncoderInit32(encoder_inst32_ptr_right_); - if (ret < 0) { - return ret; - } - return WebRtcG7221_EncoderInit32(encoder_inst32_ptr_); - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, - unique_id_, "InternalInitEncoder: Wrong rate for G722_1."); - return -1; - } - } -} - -int16_t ACMG722_1::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - switch (operational_rate_) { - case 16000: { - return WebRtcG7221_DecoderInit16(decoder_inst16_ptr_); - } - case 24000: { - return WebRtcG7221_DecoderInit24(decoder_inst24_ptr_); - } - case 32000: { - return WebRtcG7221_DecoderInit32(decoder_inst32_ptr_); - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: Wrong rate for G722_1."); - return -1; - } - } -} - -int32_t ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - // Todo: - // log error - return -1; - } - // NetEq has an array of pointers to WebRtcNetEQ_CodecDef. - // Get an entry of that array (neteq wrapper will allocate memory) - // by calling "netEq->CodecDef", where "NETEQ_CODEC_G722_1_XX" would - // be the index of the entry. - // Fill up the given structure by calling - // "SET_CODEC_PAR" & "SET_G722_1_XX_FUNCTION." - // Then return the structure back to NetEQ to add the codec to it's - // database. - switch (operational_rate_) { - case 16000: { - SET_CODEC_PAR((codec_def), kDecoderG722_1_16, codec_inst.pltype, - decoder_inst16_ptr_, 16000); - SET_G722_1_16_FUNCTIONS((codec_def)); - break; - } - case 24000: { - SET_CODEC_PAR((codec_def), kDecoderG722_1_24, codec_inst.pltype, - decoder_inst24_ptr_, 16000); - SET_G722_1_24_FUNCTIONS((codec_def)); - break; - } - case 32000: { - SET_CODEC_PAR((codec_def), kDecoderG722_1_32, codec_inst.pltype, - decoder_inst32_ptr_, 16000); - SET_G722_1_32_FUNCTIONS((codec_def)); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodecDef: Wrong rate for G722_1."); - return -1; - } - } - return 0; -} - -ACMGenericCodec* ACMG722_1::CreateInstance(void) { - return NULL; -} - -int16_t ACMG722_1::InternalCreateEncoder() { - if ((encoder_inst_ptr_ == NULL) || (encoder_inst_ptr_right_ == NULL)) { - return -1; - } - switch (operational_rate_) { - case 16000: { - WebRtcG7221_CreateEnc16(&encoder_inst16_ptr_); - WebRtcG7221_CreateEnc16(&encoder_inst16_ptr_right_); - break; - } - case 24000: { - WebRtcG7221_CreateEnc24(&encoder_inst24_ptr_); - WebRtcG7221_CreateEnc24(&encoder_inst24_ptr_right_); - break; - } - case 32000: { - WebRtcG7221_CreateEnc32(&encoder_inst32_ptr_); - WebRtcG7221_CreateEnc32(&encoder_inst32_ptr_right_); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: Wrong rate for G722_1."); - return -1; - } - } - return 0; -} - -void ACMG722_1::DestructEncoderSafe() { - encoder_exist_ = false; - encoder_initialized_ = false; - if (encoder_inst_ptr_ != NULL) { - delete encoder_inst_ptr_; - encoder_inst_ptr_ = NULL; - } - if (encoder_inst_ptr_right_ != NULL) { - delete encoder_inst_ptr_right_; - encoder_inst_ptr_right_ = NULL; - } - encoder_inst16_ptr_ = NULL; - encoder_inst24_ptr_ = NULL; - encoder_inst32_ptr_ = NULL; -} - -int16_t ACMG722_1::InternalCreateDecoder() { - if (decoder_inst_ptr_ == NULL) { - return -1; - } - switch (operational_rate_) { - case 16000: { - WebRtcG7221_CreateDec16(&decoder_inst16_ptr_); - break; - } - case 24000: { - WebRtcG7221_CreateDec24(&decoder_inst24_ptr_); - break; - } - case 32000: { - WebRtcG7221_CreateDec32(&decoder_inst32_ptr_); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateDecoder: Wrong rate for G722_1."); - return -1; - } - } - return 0; -} - -void ACMG722_1::DestructDecoderSafe() { - decoder_exist_ = false; - decoder_initialized_ = false; - if (decoder_inst_ptr_ != NULL) { - delete decoder_inst_ptr_; - decoder_inst_ptr_ = NULL; - } - decoder_inst16_ptr_ = NULL; - decoder_inst24_ptr_ = NULL; - decoder_inst32_ptr_ = NULL; -} - -void ACMG722_1::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - delete ptr_inst; - } - return; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221.h deleted file mode 100644 index 8ea66742c97..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221.h +++ /dev/null @@ -1,86 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct G722_1_16_encinst_t_; -struct G722_1_16_decinst_t_; -struct G722_1_24_encinst_t_; -struct G722_1_24_decinst_t_; -struct G722_1_32_encinst_t_; -struct G722_1_32_decinst_t_; -struct G722_1_Inst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMG722_1: public ACMGenericCodec { - public: - explicit ACMG722_1(int16_t codec_id); - ~ACMG722_1(); - - // for FEC - ACMGenericCodec* CreateInstance(void); - - int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); - - int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); - - int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params); - - protected: - int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, int16_t* audio_samples, - int8_t* speech_type); - - int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst); - - void DestructEncoderSafe(); - - void DestructDecoderSafe(); - - int16_t InternalCreateEncoder(); - - int16_t InternalCreateDecoder(); - - void InternalDestructEncoderInst(void* ptr_inst); - - int32_t operational_rate_; - - G722_1_Inst_t_* encoder_inst_ptr_; - G722_1_Inst_t_* encoder_inst_ptr_right_; // Used in stereo mode - G722_1_Inst_t_* decoder_inst_ptr_; - - // Only one set of these pointer is valid at any instance - G722_1_16_encinst_t_* encoder_inst16_ptr_; - G722_1_16_encinst_t_* encoder_inst16_ptr_right_; - G722_1_24_encinst_t_* encoder_inst24_ptr_; - G722_1_24_encinst_t_* encoder_inst24_ptr_right_; - G722_1_32_encinst_t_* encoder_inst32_ptr_; - G722_1_32_encinst_t_* encoder_inst32_ptr_right_; - - // Only one of these pointer is valid at any instance - G722_1_16_decinst_t_* decoder_inst16_ptr_; - G722_1_24_decinst_t_* decoder_inst24_ptr_; - G722_1_32_decinst_t_* decoder_inst32_ptr_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc deleted file mode 100644 index 96caba0a08f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc +++ /dev/null @@ -1,510 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_g7221c.h" - -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_G722_1C -// NOTE! G.722.1C is not included in the open-source package. The following -// interface file is needed: -// -// /modules/audio_coding/codecs/g7221c/main/interface/g7221c_interface.h -// -// The API in the header file should match the one below. -// - -// int16_t WebRtcG7221C_CreateEnc24(G722_1C_24_encinst_t_** enc_inst); -// int16_t WebRtcG7221C_CreateEnc32(G722_1C_32_encinst_t_** enc_inst); -// int16_t WebRtcG7221C_CreateEnc48(G722_1C_48_encinst_t_** enc_inst); -// int16_t WebRtcG7221C_CreateDec24(G722_1C_24_decinst_t_** dec_inst); -// int16_t WebRtcG7221C_CreateDec32(G722_1C_32_decinst_t_** dec_inst); -// int16_t WebRtcG7221C_CreateDec48(G722_1C_48_decinst_t_** dec_inst); -// -// int16_t WebRtcG7221C_FreeEnc24(G722_1C_24_encinst_t_** enc_inst); -// int16_t WebRtcG7221C_FreeEnc32(G722_1C_32_encinst_t_** enc_inst); -// int16_t WebRtcG7221C_FreeEnc48(G722_1C_48_encinst_t_** enc_inst); -// int16_t WebRtcG7221C_FreeDec24(G722_1C_24_decinst_t_** dec_inst); -// int16_t WebRtcG7221C_FreeDec32(G722_1C_32_decinst_t_** dec_inst); -// int16_t WebRtcG7221C_FreeDec48(G722_1C_48_decinst_t_** dec_inst); -// -// int16_t WebRtcG7221C_EncoderInit24(G722_1C_24_encinst_t_* enc_inst); -// int16_t WebRtcG7221C_EncoderInit32(G722_1C_32_encinst_t_* enc_inst); -// int16_t WebRtcG7221C_EncoderInit48(G722_1C_48_encinst_t_* enc_inst); -// int16_t WebRtcG7221C_DecoderInit24(G722_1C_24_decinst_t_* dec_inst); -// int16_t WebRtcG7221C_DecoderInit32(G722_1C_32_decinst_t_* dec_inst); -// int16_t WebRtcG7221C_DecoderInit48(G722_1C_48_decinst_t_* dec_inst); -// -// int16_t WebRtcG7221C_Encode24(G722_1C_24_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221C_Encode32(G722_1C_32_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221C_Encode48(G722_1C_48_encinst_t_* enc_inst, -// int16_t* input, -// int16_t len, -// int16_t* output); -// -// int16_t WebRtcG7221C_Decode24(G722_1C_24_decinst_t_* dec_inst, -// int16_t* bitstream, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221C_Decode32(G722_1C_32_decinst_t_* dec_inst, -// int16_t* bitstream, -// int16_t len, -// int16_t* output); -// int16_t WebRtcG7221C_Decode48(G722_1C_48_decinst_t_* dec_inst, -// int16_t* bitstream, -// int16_t len, -// int16_t* output); -// -// int16_t WebRtcG7221C_DecodePlc24(G722_1C_24_decinst_t_* dec_inst, -// int16_t* output, -// int16_t nr_lost_frames); -// int16_t WebRtcG7221C_DecodePlc32(G722_1C_32_decinst_t_* dec_inst, -// int16_t* output, -// int16_t nr_lost_frames); -// int16_t WebRtcG7221C_DecodePlc48(G722_1C_48_decinst_t_* dec_inst, -// int16_t* output, -// int16_t nr_lost_frames); -#include "g7221c_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_G722_1C - -ACMG722_1C::ACMG722_1C(int16_t /* codec_id */) - : operational_rate_(-1), - encoder_inst_ptr_(NULL), - encoder_inst_ptr_right_(NULL), - decoder_inst_ptr_(NULL), - encoder_inst24_ptr_(NULL), - encoder_inst24_ptr_right_(NULL), - encoder_inst32_ptr_(NULL), - encoder_inst32_ptr_right_(NULL), - encoder_inst48_ptr_(NULL), - encoder_inst48_ptr_right_(NULL), - decoder_inst24_ptr_(NULL), - decoder_inst32_ptr_(NULL), - decoder_inst48_ptr_(NULL) { - return; -} - -ACMG722_1C::~ACMG722_1C() { - return; -} - -int16_t ACMG722_1C::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMG722_1C::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMG722_1C::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMG722_1C::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMG722_1C::CreateInstance(void) { - return NULL; -} - -int16_t ACMG722_1C::InternalCreateEncoder() { - return -1; -} - -void ACMG722_1C::DestructEncoderSafe() { - return; -} - -int16_t ACMG722_1C::InternalCreateDecoder() { - return -1; -} - -void ACMG722_1C::DestructDecoderSafe() { - return; -} - -void ACMG722_1C::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -#else //===================== Actual Implementation ======================= -ACMG722_1C::ACMG722_1C(int16_t codec_id) - : encoder_inst_ptr_(NULL), - encoder_inst_ptr_right_(NULL), - decoder_inst_ptr_(NULL), - encoder_inst24_ptr_(NULL), - encoder_inst24_ptr_right_(NULL), - encoder_inst32_ptr_(NULL), - encoder_inst32_ptr_right_(NULL), - encoder_inst48_ptr_(NULL), - encoder_inst48_ptr_right_(NULL), - decoder_inst24_ptr_(NULL), - decoder_inst32_ptr_(NULL), - decoder_inst48_ptr_(NULL) { - codec_id_ = codec_id; - if (codec_id_ == ACMCodecDB::kG722_1C_24) { - operational_rate_ = 24000; - } else if (codec_id_ == ACMCodecDB::kG722_1C_32) { - operational_rate_ = 32000; - } else if (codec_id_ == ACMCodecDB::kG722_1C_48) { - operational_rate_ = 48000; - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Wrong codec id for G722_1c."); - operational_rate_ = -1; - } - return; -} - -ACMG722_1C::~ACMG722_1C() { - if (encoder_inst_ptr_ != NULL) { - delete encoder_inst_ptr_; - encoder_inst_ptr_ = NULL; - } - if (encoder_inst_ptr_right_ != NULL) { - delete encoder_inst_ptr_right_; - encoder_inst_ptr_right_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - delete decoder_inst_ptr_; - decoder_inst_ptr_ = NULL; - } - - switch (operational_rate_) { - case 24000: { - encoder_inst24_ptr_ = NULL; - encoder_inst24_ptr_right_ = NULL; - decoder_inst24_ptr_ = NULL; - break; - } - case 32000: { - encoder_inst32_ptr_ = NULL; - encoder_inst32_ptr_right_ = NULL; - decoder_inst32_ptr_ = NULL; - break; - } - case 48000: { - encoder_inst48_ptr_ = NULL; - encoder_inst48_ptr_right_ = NULL; - decoder_inst48_ptr_ = NULL; - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Wrong rate for G722_1c."); - break; - } - } - return; -} - -int16_t ACMG722_1C::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - int16_t left_channel[640]; - int16_t right_channel[640]; - int16_t len_in_bytes; - int16_t out_bits[240]; - - // If stereo, split input signal in left and right channel before encoding - if (num_channels_ == 2) { - for (int i = 0, j = 0; i < frame_len_smpl_ * 2; i += 2, j++) { - left_channel[j] = in_audio_[in_audio_ix_read_ + i]; - right_channel[j] = in_audio_[in_audio_ix_read_ + i + 1]; - } - } else { - memcpy(left_channel, &in_audio_[in_audio_ix_read_], 640); - } - - switch (operational_rate_) { - case 24000: { - len_in_bytes = WebRtcG7221C_Encode24(encoder_inst24_ptr_, left_channel, - 640, &out_bits[0]); - if (num_channels_ == 2) { - len_in_bytes += WebRtcG7221C_Encode24(encoder_inst24_ptr_right_, - right_channel, 640, - &out_bits[len_in_bytes / 2]); - } - break; - } - case 32000: { - len_in_bytes = WebRtcG7221C_Encode32(encoder_inst32_ptr_, left_channel, - 640, &out_bits[0]); - if (num_channels_ == 2) { - len_in_bytes += WebRtcG7221C_Encode32(encoder_inst32_ptr_right_, - right_channel, 640, - &out_bits[len_in_bytes / 2]); - } - break; - } - case 48000: { - len_in_bytes = WebRtcG7221C_Encode48(encoder_inst48_ptr_, left_channel, - 640, &out_bits[0]); - if (num_channels_ == 2) { - len_in_bytes += WebRtcG7221C_Encode48(encoder_inst48_ptr_right_, - right_channel, 640, - &out_bits[len_in_bytes / 2]); - } - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalEncode: Wrong rate for G722_1c."); - return -1; - } - } - - memcpy(bitstream, out_bits, len_in_bytes); - *bitstream_len_byte = len_in_bytes; - - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += 640 * num_channels_; - - return *bitstream_len_byte; -} - -int16_t ACMG722_1C::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMG722_1C::InternalInitEncoder( - WebRtcACMCodecParams* codec_params) { - int16_t ret; - - switch (operational_rate_) { - case 24000: { - ret = WebRtcG7221C_EncoderInit24(encoder_inst24_ptr_right_); - if (ret < 0) { - return ret; - } - return WebRtcG7221C_EncoderInit24(encoder_inst24_ptr_); - } - case 32000: { - ret = WebRtcG7221C_EncoderInit32(encoder_inst32_ptr_right_); - if (ret < 0) { - return ret; - } - return WebRtcG7221C_EncoderInit32(encoder_inst32_ptr_); - } - case 48000: { - ret = WebRtcG7221C_EncoderInit48(encoder_inst48_ptr_right_); - if (ret < 0) { - return ret; - } - return WebRtcG7221C_EncoderInit48(encoder_inst48_ptr_); - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitEncode: Wrong rate for G722_1c."); - return -1; - } - } -} - -int16_t ACMG722_1C::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - switch (operational_rate_) { - case 24000: { - return WebRtcG7221C_DecoderInit24(decoder_inst24_ptr_); - } - case 32000: { - return WebRtcG7221C_DecoderInit32(decoder_inst32_ptr_); - } - case 48000: { - return WebRtcG7221C_DecoderInit48(decoder_inst48_ptr_); - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: Wrong rate for G722_1c."); - return -1; - } - } -} - -int32_t ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodeDef: decoder not initialized for G722_1c"); - return -1; - } - // NetEq has an array of pointers to WebRtcNetEQ_CodecDef. - // get an entry of that array (neteq wrapper will allocate memory) - // by calling "netEq->CodecDef", where "NETEQ_CODEC_G722_1_XX" would - // be the index of the entry. - // Fill up the given structure by calling - // "SET_CODEC_PAR" & "SET_G722_1_XX_FUNCTION." - // Then return the structure back to NetEQ to add the codec to it's - // database. - switch (operational_rate_) { - case 24000: { - SET_CODEC_PAR((codec_def), kDecoderG722_1C_24, codec_inst.pltype, - decoder_inst24_ptr_, 32000); - SET_G722_1C_24_FUNCTIONS((codec_def)); - break; - } - case 32000: { - SET_CODEC_PAR((codec_def), kDecoderG722_1C_32, codec_inst.pltype, - decoder_inst32_ptr_, 32000); - SET_G722_1C_32_FUNCTIONS((codec_def)); - break; - } - case 48000: { - SET_CODEC_PAR((codec_def), kDecoderG722_1C_32, codec_inst.pltype, - decoder_inst48_ptr_, 32000); - SET_G722_1C_48_FUNCTIONS((codec_def)); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodeDef: Wrong rate for G722_1c."); - return -1; - } - } - return 0; -} - -ACMGenericCodec* -ACMG722_1C::CreateInstance(void) { - return NULL; -} - -int16_t ACMG722_1C::InternalCreateEncoder() { - if ((encoder_inst_ptr_ == NULL) || (encoder_inst_ptr_right_ == NULL)) { - return -1; - } - switch (operational_rate_) { - case 24000: { - WebRtcG7221C_CreateEnc24(&encoder_inst24_ptr_); - WebRtcG7221C_CreateEnc24(&encoder_inst24_ptr_right_); - break; - } - case 32000: { - WebRtcG7221C_CreateEnc32(&encoder_inst32_ptr_); - WebRtcG7221C_CreateEnc32(&encoder_inst32_ptr_right_); - break; - } - case 48000: { - WebRtcG7221C_CreateEnc48(&encoder_inst48_ptr_); - WebRtcG7221C_CreateEnc48(&encoder_inst48_ptr_right_); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: Wrong rate for G722_1c."); - return -1; - } - } - return 0; -} - -void ACMG722_1C::DestructEncoderSafe() { - encoder_exist_ = false; - encoder_initialized_ = false; - if (encoder_inst_ptr_ != NULL) { - delete encoder_inst_ptr_; - encoder_inst_ptr_ = NULL; - } - if (encoder_inst_ptr_right_ != NULL) { - delete encoder_inst_ptr_right_; - encoder_inst_ptr_right_ = NULL; - } - encoder_inst24_ptr_ = NULL; - encoder_inst32_ptr_ = NULL; - encoder_inst48_ptr_ = NULL; -} - -int16_t ACMG722_1C::InternalCreateDecoder() { - if (decoder_inst_ptr_ == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: cannot create decoder"); - return -1; - } - switch (operational_rate_) { - case 24000: { - WebRtcG7221C_CreateDec24(&decoder_inst24_ptr_); - break; - } - case 32000: { - WebRtcG7221C_CreateDec32(&decoder_inst32_ptr_); - break; - } - case 48000: { - WebRtcG7221C_CreateDec48(&decoder_inst48_ptr_); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: Wrong rate for G722_1c."); - return -1; - } - } - return 0; -} - -void ACMG722_1C::DestructDecoderSafe() { - decoder_exist_ = false; - decoder_initialized_ = false; - if (decoder_inst_ptr_ != NULL) { - delete decoder_inst_ptr_; - decoder_inst_ptr_ = NULL; - } - decoder_inst24_ptr_ = NULL; - decoder_inst32_ptr_ = NULL; - decoder_inst48_ptr_ = NULL; -} - -void ACMG722_1C::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - delete ptr_inst; - } - return; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.h deleted file mode 100644 index d8875aa2fbf..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.h +++ /dev/null @@ -1,94 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct G722_1C_24_encinst_t_; -struct G722_1C_24_decinst_t_; -struct G722_1C_32_encinst_t_; -struct G722_1C_32_decinst_t_; -struct G722_1C_48_encinst_t_; -struct G722_1C_48_decinst_t_; -struct G722_1_Inst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMG722_1C : public ACMGenericCodec { - public: - explicit ACMG722_1C(int16_t codec_id); - ~ACMG722_1C(); - - // for FEC - ACMGenericCodec* CreateInstance(void); - - int16_t InternalEncode( - uint8_t* bitstream, - int16_t* bitstream_len_byte); - - int16_t InternalInitEncoder( - WebRtcACMCodecParams *codec_params); - - int16_t InternalInitDecoder( - WebRtcACMCodecParams *codec_params); - - protected: - int16_t DecodeSafe( - uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type); - - int32_t CodecDef( - WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst); - - void DestructEncoderSafe(); - - void DestructDecoderSafe(); - - int16_t InternalCreateEncoder(); - - int16_t InternalCreateDecoder(); - - void InternalDestructEncoderInst( - void* ptr_inst); - - int32_t operational_rate_; - - G722_1_Inst_t_* encoder_inst_ptr_; - G722_1_Inst_t_* encoder_inst_ptr_right_; // Used in stereo mode - G722_1_Inst_t_* decoder_inst_ptr_; - - // Only one set of these pointer is valid at any instance - G722_1C_24_encinst_t_* encoder_inst24_ptr_; - G722_1C_24_encinst_t_* encoder_inst24_ptr_right_; - G722_1C_32_encinst_t_* encoder_inst32_ptr_; - G722_1C_32_encinst_t_* encoder_inst32_ptr_right_; - G722_1C_48_encinst_t_* encoder_inst48_ptr_; - G722_1C_48_encinst_t_* encoder_inst48_ptr_right_; - - // Only one of these pointer is valid at any instance - G722_1C_24_decinst_t_* decoder_inst24_ptr_; - G722_1C_32_decinst_t_* decoder_inst32_ptr_; - G722_1C_48_decinst_t_* decoder_inst48_ptr_; -}; - -} // namespace acm1 - -} // namespace webrtc; - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7221C_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc deleted file mode 100644 index 406bb61e48d..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc +++ /dev/null @@ -1,366 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_g729.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_G729 -// NOTE! G.729 is not included in the open-source package. Modify this file -// or your codec API to match the function calls and names of used G.729 API -// file. -#include "g729_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_G729 - -ACMG729::ACMG729(int16_t /* codec_id */) -: encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - return; -} - -ACMG729::~ACMG729() { - return; -} - -int16_t ACMG729::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMG729::EnableDTX() { - return -1; -} - -int16_t ACMG729::DisableDTX() { - return -1; -} - -int32_t ACMG729::ReplaceInternalDTXSafe( - const bool /*replace_internal_dtx */) { - return -1; -} - -int32_t ACMG729::IsInternalDTXReplacedSafe( - bool* /* internal_dtx_replaced */) { - return -1; -} - -int16_t ACMG729::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMG729::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMG729::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMG729::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMG729::CreateInstance(void) { - return NULL; -} - -int16_t ACMG729::InternalCreateEncoder() { - return -1; -} - -void ACMG729::DestructEncoderSafe() { - return; -} - -int16_t ACMG729::InternalCreateDecoder() { - return -1; -} - -void ACMG729::DestructDecoderSafe() { - return; -} - -void ACMG729::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -#else //===================== Actual Implementation ======================= -ACMG729::ACMG729(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - codec_id_ = codec_id; - has_internal_dtx_ = true; - return; -} - -ACMG729::~ACMG729() { - if (encoder_inst_ptr_ != NULL) { - // Delete encoder memory - WebRtcG729_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - // Delete decoder memory - WebRtcG729_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMG729::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - // Initialize before entering the loop - int16_t num_encoded_samples = 0; - int16_t tmp_len_byte = 0; - int16_t vad_decision = 0; - *bitstream_len_byte = 0; - while (num_encoded_samples < frame_len_smpl_) { - // Call G.729 encoder with pointer to encoder memory, input - // audio, number of samples and bitsream - tmp_len_byte = WebRtcG729_Encode( - encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], 80, - (int16_t*)(&(bitstream[*bitstream_len_byte]))); - - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += 80; - - // sanity check - if (tmp_len_byte < 0) { - // error has happened - *bitstream_len_byte = 0; - return -1; - } - - // increment number of written bytes - *bitstream_len_byte += tmp_len_byte; - switch (tmp_len_byte) { - case 0: { - if (0 == num_encoded_samples) { - // this is the first 10 ms in this packet and there is - // no data generated, perhaps DTX is enabled and the - // codec is not generating any bit-stream for this 10 ms. - // we do not continue encoding this frame. - return 0; - } - break; - } - case 2: { - // check if G.729 internal DTX is enabled - if (has_internal_dtx_ && dtx_enabled_) { - vad_decision = 0; - for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) { - vad_label_[n] = vad_decision; - } - } - // we got a SID and have to send out this packet no matter - // how much audio we have encoded - return *bitstream_len_byte; - } - case 10: { - vad_decision = 1; - // this is a valid length just continue encoding - break; - } - default: { - return -1; - } - } - - // update number of encoded samples - num_encoded_samples += 80; - } - - // update VAD decision vector - if (has_internal_dtx_ && !vad_decision && dtx_enabled_) { - for (int16_t n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) { - vad_label_[n] = vad_decision; - } - } - - // done encoding, return number of encoded bytes - return *bitstream_len_byte; -} - -int16_t ACMG729::EnableDTX() { - if (dtx_enabled_) { - // DTX already enabled, do nothing - return 0; - } else if (encoder_exist_) { - // Re-init the G.729 encoder to turn on DTX - if (WebRtcG729_EncoderInit(encoder_inst_ptr_, 1) < 0) { - return -1; - } - dtx_enabled_ = true; - return 0; - } else { - return -1; - } -} - -int16_t ACMG729::DisableDTX() { - if (!dtx_enabled_) { - // DTX already dissabled, do nothing - return 0; - } else if (encoder_exist_) { - // Re-init the G.729 decoder to turn off DTX - if (WebRtcG729_EncoderInit(encoder_inst_ptr_, 0) < 0) { - return -1; - } - dtx_enabled_ = false; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } -} - -int32_t ACMG729::ReplaceInternalDTXSafe(const bool replace_internal_dtx) { - // This function is used to disable the G.729 built in DTX and use an - // external instead. - - if (replace_internal_dtx == has_internal_dtx_) { - // Make sure we keep the DTX/VAD setting if possible - bool old_enable_dtx = dtx_enabled_; - bool old_enable_vad = vad_enabled_; - ACMVADMode old_mode = vad_mode_; - if (replace_internal_dtx) { - // Disable internal DTX before enabling external DTX - DisableDTX(); - } else { - // Disable external DTX before enabling internal - ACMGenericCodec::DisableDTX(); - } - has_internal_dtx_ = !replace_internal_dtx; - int16_t status = SetVADSafe(old_enable_dtx, old_enable_vad, old_mode); - // Check if VAD status has changed from inactive to active, or if error was - // reported - if (status == 1) { - vad_enabled_ = true; - return status; - } else if (status < 0) { - has_internal_dtx_ = replace_internal_dtx; - return -1; - } - } - return 0; -} - -int32_t ACMG729::IsInternalDTXReplacedSafe(bool* internal_dtx_replaced) { - // Get status of wether DTX is replaced or not - *internal_dtx_replaced = !has_internal_dtx_; - return 0; -} - -int16_t ACMG729::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - // This function is not used. G.729 decoder is called from inside NetEQ - return 0; -} - -int16_t ACMG729::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - // Init G.729 encoder - return WebRtcG729_EncoderInit(encoder_inst_ptr_, - ((codec_params->enable_dtx) ? 1 : 0)); -} - -int16_t ACMG729::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - // Init G.729 decoder - return WebRtcG729_DecoderInit(decoder_inst_ptr_); -} - -int32_t ACMG729::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - // Todo: - // log error - return -1; - } - - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_G729_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderG729, codec_inst.pltype, decoder_inst_ptr_, - 8000); - SET_G729_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMG729::CreateInstance(void) { - // Function not used - return NULL; -} - -int16_t ACMG729::InternalCreateEncoder() { - // Create encoder memory - return WebRtcG729_CreateEnc(&encoder_inst_ptr_); -} - -void ACMG729::DestructEncoderSafe() { - // Free encoder memory - encoder_exist_ = false; - encoder_initialized_ = false; - if (encoder_inst_ptr_ != NULL) { - WebRtcG729_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } -} - -int16_t ACMG729::InternalCreateDecoder() { - // Create decoder memory - return WebRtcG729_CreateDec(&decoder_inst_ptr_); -} - -void ACMG729::DestructDecoderSafe() { - // Free decoder memory - decoder_exist_ = false; - decoder_initialized_ = false; - if (decoder_inst_ptr_ != NULL) { - WebRtcG729_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } -} - -void ACMG729::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcG729_FreeEnc((G729_encinst_t_*) ptr_inst); - } - return; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g729.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g729.h deleted file mode 100644 index 5cfff63b69a..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g729.h +++ /dev/null @@ -1,76 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct G729_encinst_t_; -struct G729_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMG729 : public ACMGenericCodec { - public: - explicit ACMG729(int16_t codec_id); - ~ACMG729(); - - // for FEC - ACMGenericCodec* CreateInstance(void); - - int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte); - - int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); - - int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params); - - protected: - int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type); - - int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst); - - void DestructEncoderSafe(); - - void DestructDecoderSafe(); - - int16_t InternalCreateEncoder(); - - int16_t InternalCreateDecoder(); - - void InternalDestructEncoderInst(void* ptr_inst); - - int16_t EnableDTX(); - - int16_t DisableDTX(); - - int32_t ReplaceInternalDTXSafe(const bool replace_internal_dtx); - - int32_t IsInternalDTXReplacedSafe(bool* internal_dtx_replaced); - - G729_encinst_t_* encoder_inst_ptr_; - G729_decinst_t_* decoder_inst_ptr_; - -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc deleted file mode 100644 index 0da6c99d21e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc +++ /dev/null @@ -1,349 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_g7291.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" -#ifdef WEBRTC_CODEC_G729_1 -// NOTE! G.729.1 is not included in the open-source package. Modify this file -// or your codec API to match the function calls and names of used G.729.1 API -// file. -#include "g7291_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_G729_1 - -ACMG729_1::ACMG729_1(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - my_rate_(32000), - flag_8khz_(0), - flag_g729_mode_(0) { - return; -} - -ACMG729_1::~ACMG729_1() { - return; -} - -int16_t ACMG729_1::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMG729_1::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMG729_1::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMG729_1::CreateInstance(void) { - return NULL; -} - -int16_t ACMG729_1::InternalCreateEncoder() { - return -1; -} - -void ACMG729_1::DestructEncoderSafe() { - return; -} - -int16_t ACMG729_1::InternalCreateDecoder() { - return -1; -} - -void ACMG729_1::DestructDecoderSafe() { - return; -} - -void ACMG729_1::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) { - return -1; -} - -#else //===================== Actual Implementation ======================= - -struct G729_1_inst_t_; - -ACMG729_1::ACMG729_1(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - my_rate_(32000), // Default rate. - flag_8khz_(0), - flag_g729_mode_(0) { - // TODO(tlegrand): We should add codec_id as a input variable to the - // constructor of ACMGenericCodec. - codec_id_ = codec_id; - return; -} - -ACMG729_1::~ACMG729_1() { - if (encoder_inst_ptr_ != NULL) { - WebRtcG7291_Free(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcG7291_Free(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMG729_1::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - - // Initialize before entering the loop - int16_t num_encoded_samples = 0; - *bitstream_len_byte = 0; - - int16_t byte_length_frame = 0; - - // Derive number of 20ms frames per encoded packet. - // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples - int16_t num_20ms_frames = (frame_len_smpl_ / 320); - // Byte length for the frame. +1 is for rate information. - byte_length_frame = my_rate_ / (8 * 50) * num_20ms_frames + (1 - - flag_g729_mode_); - - // The following might be revised if we have G729.1 Annex C (support for DTX); - do { - *bitstream_len_byte = WebRtcG7291_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - (int16_t*) bitstream, - my_rate_, num_20ms_frames); - - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += 160; - - // sanity check - if (*bitstream_len_byte < 0) { - // error has happened - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalEncode: Encode error for G729_1"); - *bitstream_len_byte = 0; - return -1; - } - - num_encoded_samples += 160; - } while (*bitstream_len_byte == 0); - - // This criteria will change if we have Annex C. - if (*bitstream_len_byte != byte_length_frame) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalEncode: Encode error for G729_1"); - *bitstream_len_byte = 0; - return -1; - } - - if (num_encoded_samples != frame_len_smpl_) { - *bitstream_len_byte = 0; - return -1; - } - - return *bitstream_len_byte; -} - -int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMG729_1::InternalInitEncoder( - WebRtcACMCodecParams* codec_params) { - //set the bit rate and initialize - my_rate_ = codec_params->codec_inst.rate; - return SetBitRateSafe((uint32_t) my_rate_); -} - -int16_t ACMG729_1::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - if (WebRtcG7291_DecoderInit(decoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: init decoder failed for G729_1"); - return -1; - } - return 0; -} - -int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodeDef: Decoder uninitialized for G729_1"); - return -1; - } - - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_G729_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderG729_1, codec_inst.pltype, - decoder_inst_ptr_, 16000); - SET_G729_1_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMG729_1::CreateInstance(void) { - return NULL; -} - -int16_t ACMG729_1::InternalCreateEncoder() { - if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: create encoder failed for G729_1"); - return -1; - } - return 0; -} - -void ACMG729_1::DestructEncoderSafe() { - encoder_exist_ = false; - encoder_initialized_ = false; - if (encoder_inst_ptr_ != NULL) { - WebRtcG7291_Free(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } -} - -int16_t ACMG729_1::InternalCreateDecoder() { - if (WebRtcG7291_Create(&decoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateDecoder: create decoder failed for G729_1"); - return -1; - } - return 0; -} - -void ACMG729_1::DestructDecoderSafe() { - decoder_exist_ = false; - decoder_initialized_ = false; - if (decoder_inst_ptr_ != NULL) { - WebRtcG7291_Free(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } -} - -void ACMG729_1::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - // WebRtcG7291_Free((G729_1_inst_t*)ptrInst); - } - return; -} - -int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) { - // allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000, - // 22000, 24000, 26000, 28000, 30000, 32000}; - // TODO(tlegrand): This check exists in one other place two. Should be - // possible to reuse code. - switch (rate) { - case 8000: { - my_rate_ = 8000; - break; - } - case 12000: { - my_rate_ = 12000; - break; - } - case 14000: { - my_rate_ = 14000; - break; - } - case 16000: { - my_rate_ = 16000; - break; - } - case 18000: { - my_rate_ = 18000; - break; - } - case 20000: { - my_rate_ = 20000; - break; - } - case 22000: { - my_rate_ = 22000; - break; - } - case 24000: { - my_rate_ = 24000; - break; - } - case 26000: { - my_rate_ = 26000; - break; - } - case 28000: { - my_rate_ = 28000; - break; - } - case 30000: { - my_rate_ = 30000; - break; - } - case 32000: { - my_rate_ = 32000; - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetBitRateSafe: Invalid rate G729_1"); - return -1; - } - } - - // Re-init with new rate - if (WebRtcG7291_EncoderInit(encoder_inst_ptr_, my_rate_, flag_8khz_, - flag_g729_mode_) >= 0) { - encoder_params_.codec_inst.rate = my_rate_; - return 0; - } else { - return -1; - } -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7291.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7291.h deleted file mode 100644 index bac7faf8368..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_g7291.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct G729_1_inst_t_; -struct G729_1_inst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMG729_1 : public ACMGenericCodec { - public: - explicit ACMG729_1(int16_t codec_id); - ~ACMG729_1(); - - // for FEC - ACMGenericCodec* CreateInstance(void); - - int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); - - int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); - - int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params); - - protected: - int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type); - - int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst); - - void DestructEncoderSafe(); - - void DestructDecoderSafe(); - - int16_t InternalCreateEncoder(); - - int16_t InternalCreateDecoder(); - - void InternalDestructEncoderInst(void* ptr_inst); - - int16_t SetBitRateSafe(const int32_t rate); - - G729_1_inst_t_* encoder_inst_ptr_; - G729_1_inst_t_* decoder_inst_ptr_; - - uint16_t my_rate_; - int16_t flag_8khz_; - int16_t flag_g729_mode_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G7291_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc deleted file mode 100644 index 4e53b873a1f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc +++ /dev/null @@ -1,1263 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -#include <assert.h> -#include <string.h> - -#include "webrtc/common_audio/vad/include/webrtc_vad.h" -#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -namespace acm1 { - -// Enum for CNG -enum { - kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER, - kNewCNGNumPLCParams = 8 -}; - -// Interval for sending new CNG parameters (SID frames) is 100 msec. -enum { - kCngSidIntervalMsec = 100 -}; - -// We set some of the variables to invalid values as a check point -// if a proper initialization has happened. Another approach is -// to initialize to a default codec that we are sure is always included. -ACMGenericCodec::ACMGenericCodec() - : in_audio_ix_write_(0), - in_audio_ix_read_(0), - in_timestamp_ix_write_(0), - in_audio_(NULL), - in_timestamp_(NULL), - frame_len_smpl_(-1), // invalid value - num_channels_(1), - codec_id_(-1), // invalid value - num_missed_samples_(0), - encoder_exist_(false), - decoder_exist_(false), - encoder_initialized_(false), - decoder_initialized_(false), - registered_in_neteq_(false), - has_internal_dtx_(false), - ptr_vad_inst_(NULL), - vad_enabled_(false), - vad_mode_(VADNormal), - dtx_enabled_(false), - ptr_dtx_inst_(NULL), - num_lpc_params_(kNewCNGNumPLCParams), - sent_cn_previous_(false), - is_master_(true), - prev_frame_cng_(0), - neteq_decode_lock_(NULL), - codec_wrapper_lock_(*RWLockWrapper::CreateRWLock()), - last_encoded_timestamp_(0), - last_timestamp_(0xD87F3F9F), - is_audio_buff_fresh_(true), - unique_id_(0) { - // Initialize VAD vector. - for (int i = 0; i < MAX_FRAME_SIZE_10MSEC; i++) { - vad_label_[i] = 0; - } - // Nullify memory for encoder and decoder, and set payload type to an - // invalid value. - memset(&encoder_params_, 0, sizeof(WebRtcACMCodecParams)); - encoder_params_.codec_inst.pltype = -1; - memset(&decoder_params_, 0, sizeof(WebRtcACMCodecParams)); - decoder_params_.codec_inst.pltype = -1; -} - -ACMGenericCodec::~ACMGenericCodec() { - // Check all the members which are pointers, and if they are not NULL - // delete/free them. - if (ptr_vad_inst_ != NULL) { - WebRtcVad_Free(ptr_vad_inst_); - ptr_vad_inst_ = NULL; - } - if (in_audio_ != NULL) { - delete[] in_audio_; - in_audio_ = NULL; - } - if (in_timestamp_ != NULL) { - delete[] in_timestamp_; - in_timestamp_ = NULL; - } - if (ptr_dtx_inst_ != NULL) { - WebRtcCng_FreeEnc(ptr_dtx_inst_); - ptr_dtx_inst_ = NULL; - } - delete &codec_wrapper_lock_; -} - -int32_t ACMGenericCodec::Add10MsData(const uint32_t timestamp, - const int16_t* data, - const uint16_t length_smpl, - const uint8_t audio_channel) { - WriteLockScoped wl(codec_wrapper_lock_); - return Add10MsDataSafe(timestamp, data, length_smpl, audio_channel); -} - -int32_t ACMGenericCodec::Add10MsDataSafe(const uint32_t timestamp, - const int16_t* data, - const uint16_t length_smpl, - const uint8_t audio_channel) { - // The codec expects to get data in correct sampling rate. Get the sampling - // frequency of the codec. - uint16_t plfreq_hz; - if (EncoderSampFreq(plfreq_hz) < 0) { - return -1; - } - - // Sanity check to make sure the length of the input corresponds to 10 ms. - if ((plfreq_hz / 100) != length_smpl) { - // This is not 10 ms of audio, given the sampling frequency of the codec. - return -1; - } - - if (last_timestamp_ == timestamp) { - // Same timestamp as the last time, overwrite. - if ((in_audio_ix_write_ >= length_smpl * audio_channel) && - (in_timestamp_ix_write_ > 0)) { - in_audio_ix_write_ -= length_smpl * audio_channel; - in_timestamp_ix_write_--; - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, unique_id_, - "Adding 10ms with previous timestamp, overwriting the " - "previous 10ms"); - } else { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, unique_id_, - "Adding 10ms with previous timestamp, this will sound bad"); - } - } - - last_timestamp_ = timestamp; - - // If the data exceeds the buffer size, we throw away the oldest data and - // add the newly received 10 msec at the end. - if ((in_audio_ix_write_ + length_smpl * audio_channel) > - AUDIO_BUFFER_SIZE_W16) { - // Get the number of samples to be overwritten. - int16_t missed_samples = in_audio_ix_write_ + length_smpl * audio_channel - - AUDIO_BUFFER_SIZE_W16; - - // Move the data (overwrite the old data). - memmove(in_audio_, in_audio_ + missed_samples, - (AUDIO_BUFFER_SIZE_W16 - length_smpl * audio_channel) * - sizeof(int16_t)); - - // Copy the new data. - memcpy(in_audio_ + (AUDIO_BUFFER_SIZE_W16 - length_smpl * audio_channel), - data, length_smpl * audio_channel * sizeof(int16_t)); - - // Get the number of 10 ms blocks which are overwritten. - int16_t missed_10ms_blocks =static_cast<int16_t>( - (missed_samples / audio_channel * 100) / plfreq_hz); - - // Move the timestamps. - memmove(in_timestamp_, in_timestamp_ + missed_10ms_blocks, - (in_timestamp_ix_write_ - missed_10ms_blocks) * sizeof(uint32_t)); - in_timestamp_ix_write_ -= missed_10ms_blocks; - assert(in_timestamp_ix_write_ >= 0); - in_timestamp_[in_timestamp_ix_write_] = timestamp; - in_timestamp_ix_write_++; - - // Buffer is full. - in_audio_ix_write_ = AUDIO_BUFFER_SIZE_W16; - IncreaseNoMissedSamples(missed_samples); - is_audio_buff_fresh_ = false; - return -missed_samples; - } - - // Store the input data in our data buffer. - memcpy(in_audio_ + in_audio_ix_write_, data, - length_smpl * audio_channel * sizeof(int16_t)); - in_audio_ix_write_ += length_smpl * audio_channel; - - assert(in_timestamp_ix_write_ < TIMESTAMP_BUFFER_SIZE_W32); - assert(in_timestamp_ix_write_ >= 0); - - in_timestamp_[in_timestamp_ix_write_] = timestamp; - in_timestamp_ix_write_++; - is_audio_buff_fresh_ = false; - return 0; -} - -bool ACMGenericCodec::HasFrameToEncode() const { - ReadLockScoped lockCodec(codec_wrapper_lock_); - if (in_audio_ix_write_ < frame_len_smpl_ * num_channels_) - return false; - return true; -} - -int16_t ACMGenericCodec::Encode(uint8_t* bitstream, - int16_t* bitstream_len_byte, - uint32_t* timestamp, - WebRtcACMEncodingType* encoding_type) { - if (!HasFrameToEncode()) { - // There is not enough audio - *timestamp = 0; - *bitstream_len_byte = 0; - // Doesn't really matter what this parameter set to - *encoding_type = kNoEncoding; - return 0; - } - WriteLockScoped lockCodec(codec_wrapper_lock_); - ReadLockScoped lockNetEq(*neteq_decode_lock_); - - // Not all codecs accept the whole frame to be pushed into encoder at once. - // Some codecs needs to be feed with a specific number of samples different - // from the frame size. If this is the case, |myBasicCodingBlockSmpl| will - // report a number different from 0, and we will loop over calls to encoder - // further down, until we have encode a complete frame. - const int16_t my_basic_coding_block_smpl = - ACMCodecDB::BasicCodingBlock(codec_id_); - if (my_basic_coding_block_smpl < 0 || !encoder_initialized_ || - !encoder_exist_) { - // This should not happen, but in case it does, report no encoding done. - *timestamp = 0; - *bitstream_len_byte = 0; - *encoding_type = kNoEncoding; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EncodeSafe: error, basic coding sample block is negative"); - return -1; - } - // This makes the internal encoder read from the beginning of the buffer. - in_audio_ix_read_ = 0; - *timestamp = in_timestamp_[0]; - - // Process the audio through VAD. The function will set |_vad_labels|. - // If VAD is disabled all entries in |_vad_labels| are set to ONE (active). - int16_t status = 0; - int16_t dtx_processed_samples = 0; - status = ProcessFrameVADDTX(bitstream, bitstream_len_byte, - &dtx_processed_samples); - if (status < 0) { - *timestamp = 0; - *bitstream_len_byte = 0; - *encoding_type = kNoEncoding; - } else { - if (dtx_processed_samples > 0) { - // Dtx have processed some samples, and even if a bit-stream is generated - // we should not do any encoding (normally there won't be enough data). - - // Setting the following makes sure that the move of audio data and - // timestamps done correctly. - in_audio_ix_read_ = dtx_processed_samples; - // This will let the owner of ACMGenericCodec to know that the - // generated bit-stream is DTX to use correct payload type. - uint16_t samp_freq_hz; - EncoderSampFreq(samp_freq_hz); - if (samp_freq_hz == 8000) { - *encoding_type = kPassiveDTXNB; - } else if (samp_freq_hz == 16000) { - *encoding_type = kPassiveDTXWB; - } else if (samp_freq_hz == 32000) { - *encoding_type = kPassiveDTXSWB; - } else if (samp_freq_hz == 48000) { - *encoding_type = kPassiveDTXFB; - } else { - status = -1; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EncodeSafe: Wrong sampling frequency for DTX."); - } - - // Transport empty frame if we have an empty bitstream. - if ((*bitstream_len_byte == 0) && - (sent_cn_previous_ || - ((in_audio_ix_write_ - in_audio_ix_read_) <= 0))) { - // Makes sure we transmit an empty frame. - *bitstream_len_byte = 1; - *encoding_type = kNoEncoding; - } - sent_cn_previous_ = true; - } else { - // We should encode the audio frame. Either VAD and/or DTX is off, or the - // audio was considered "active". - - sent_cn_previous_ = false; - if (my_basic_coding_block_smpl == 0) { - // This codec can handle all allowed frame sizes as basic coding block. - status = InternalEncode(bitstream, bitstream_len_byte); - if (status < 0) { - // TODO(tlegrand): Maybe reseting the encoder to be fresh for the next - // frame. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, - unique_id_, "EncodeSafe: error in internal_encode"); - *bitstream_len_byte = 0; - *encoding_type = kNoEncoding; - } - } else { - // A basic-coding-block for this codec is defined so we loop over the - // audio with the steps of the basic-coding-block. - int16_t tmp_bitstream_len_byte; - - // Reset the variables which will be incremented in the loop. - *bitstream_len_byte = 0; - do { - status = InternalEncode(&bitstream[*bitstream_len_byte], - &tmp_bitstream_len_byte); - *bitstream_len_byte += tmp_bitstream_len_byte; - - // Guard Against errors and too large payloads. - if ((status < 0) || (*bitstream_len_byte > MAX_PAYLOAD_SIZE_BYTE)) { - // Error has happened, and even if we are in the middle of a full - // frame we have to exit. Before exiting, whatever bits are in the - // buffer are probably corrupted, so we ignore them. - *bitstream_len_byte = 0; - *encoding_type = kNoEncoding; - // We might have come here because of the second condition. - status = -1; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, - unique_id_, "EncodeSafe: error in InternalEncode"); - // break from the loop - break; - } - } while (in_audio_ix_read_ < frame_len_smpl_ * num_channels_); - } - if (status >= 0) { - *encoding_type = (vad_label_[0] == 1) ? kActiveNormalEncoded : - kPassiveNormalEncoded; - // Transport empty frame if we have an empty bitstream. - if ((*bitstream_len_byte == 0) && - ((in_audio_ix_write_ - in_audio_ix_read_) <= 0)) { - // Makes sure we transmit an empty frame. - *bitstream_len_byte = 1; - *encoding_type = kNoEncoding; - } - } - } - } - - // Move the timestamp buffer according to the number of 10 ms blocks - // which are read. - uint16_t samp_freq_hz; - EncoderSampFreq(samp_freq_hz); - int16_t num_10ms_blocks = static_cast<int16_t>( - (in_audio_ix_read_ / num_channels_ * 100) / samp_freq_hz); - if (in_timestamp_ix_write_ > num_10ms_blocks) { - memmove(in_timestamp_, in_timestamp_ + num_10ms_blocks, - (in_timestamp_ix_write_ - num_10ms_blocks) * sizeof(int32_t)); - } - in_timestamp_ix_write_ -= num_10ms_blocks; - assert(in_timestamp_ix_write_ >= 0); - // Remove encoded audio and move next audio to be encoded to the beginning - // of the buffer. Accordingly, adjust the read and write indices. - if (in_audio_ix_read_ < in_audio_ix_write_) { - memmove(in_audio_, &in_audio_[in_audio_ix_read_], - (in_audio_ix_write_ - in_audio_ix_read_) * sizeof(int16_t)); - } - in_audio_ix_write_ -= in_audio_ix_read_; - assert(in_timestamp_ix_write_ >= 0); - in_audio_ix_read_ = 0; - last_encoded_timestamp_ = *timestamp; - return (status < 0) ? (-1) : (*bitstream_len_byte); -} - -int16_t ACMGenericCodec::Decode(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) { - WriteLockScoped wl(codec_wrapper_lock_); - return DecodeSafe(bitstream, bitstream_len_byte, audio, audio_samples, - speech_type); -} - -bool ACMGenericCodec::EncoderInitialized() { - ReadLockScoped rl(codec_wrapper_lock_); - return encoder_initialized_; -} - -bool ACMGenericCodec::DecoderInitialized() { - ReadLockScoped rl(codec_wrapper_lock_); - return decoder_initialized_; -} - -int32_t ACMGenericCodec::RegisterInNetEq(ACMNetEQ* neteq, - const CodecInst& codec_inst) { - WebRtcNetEQ_CodecDef codec_def; - WriteLockScoped wl(codec_wrapper_lock_); - - if (CodecDef(codec_def, codec_inst) < 0) { - // Failed to register the decoder. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "RegisterInNetEq: error, failed to register"); - registered_in_neteq_ = false; - return -1; - } else { - if (neteq->AddCodec(&codec_def, is_master_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "RegisterInNetEq: error, failed to add codec"); - registered_in_neteq_ = false; - return -1; - } - // Succeeded registering the decoder. - registered_in_neteq_ = true; - return 0; - } -} - -int16_t ACMGenericCodec::EncoderParams(WebRtcACMCodecParams* enc_params) { - ReadLockScoped rl(codec_wrapper_lock_); - return EncoderParamsSafe(enc_params); -} - -int16_t ACMGenericCodec::EncoderParamsSafe(WebRtcACMCodecParams* enc_params) { - // Codec parameters are valid only if the encoder is initialized. - if (encoder_initialized_) { - int32_t current_rate; - memcpy(enc_params, &encoder_params_, sizeof(WebRtcACMCodecParams)); - current_rate = enc_params->codec_inst.rate; - CurrentRate(current_rate); - enc_params->codec_inst.rate = current_rate; - return 0; - } else { - enc_params->codec_inst.plname[0] = '\0'; - enc_params->codec_inst.pltype = -1; - enc_params->codec_inst.pacsize = 0; - enc_params->codec_inst.rate = 0; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EncoderParamsSafe: error, encoder not initialized"); - return -1; - } -} - -bool ACMGenericCodec::DecoderParams(WebRtcACMCodecParams* dec_params, - const uint8_t payload_type) { - ReadLockScoped rl(codec_wrapper_lock_); - return DecoderParamsSafe(dec_params, payload_type); -} - -bool ACMGenericCodec::DecoderParamsSafe(WebRtcACMCodecParams* dec_params, - const uint8_t payload_type) { - // Decoder parameters are valid only if decoder is initialized. - if (decoder_initialized_) { - if (payload_type == decoder_params_.codec_inst.pltype) { - memcpy(dec_params, &decoder_params_, sizeof(WebRtcACMCodecParams)); - return true; - } - } - - dec_params->codec_inst.plname[0] = '\0'; - dec_params->codec_inst.pltype = -1; - dec_params->codec_inst.pacsize = 0; - dec_params->codec_inst.rate = 0; - return false; -} - -int16_t ACMGenericCodec::ResetEncoder() { - WriteLockScoped lockCodec(codec_wrapper_lock_); - ReadLockScoped lockNetEq(*neteq_decode_lock_); - return ResetEncoderSafe(); -} - -int16_t ACMGenericCodec::ResetEncoderSafe() { - if (!encoder_exist_ || !encoder_initialized_) { - // We don't reset if encoder doesn't exists or isn't initialized yet. - return 0; - } - - in_audio_ix_write_ = 0; - in_audio_ix_read_ = 0; - in_timestamp_ix_write_ = 0; - num_missed_samples_ = 0; - is_audio_buff_fresh_ = true; - memset(in_audio_, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t)); - memset(in_timestamp_, 0, TIMESTAMP_BUFFER_SIZE_W32 * sizeof(int32_t)); - - // Store DTX/VAD parameters. - bool enable_vad = vad_enabled_; - bool enable_dtx = dtx_enabled_; - ACMVADMode mode = vad_mode_; - - // Reset the encoder. - if (InternalResetEncoder() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "ResetEncoderSafe: error in reset encoder"); - return -1; - } - - // Disable DTX & VAD to delete the states and have a fresh start. - DisableDTX(); - DisableVAD(); - - // Set DTX/VAD. - int status = SetVADSafe(&enable_dtx, &enable_vad, &mode); - dtx_enabled_ = enable_dtx; - vad_enabled_ = enable_vad; - vad_mode_ = mode; - return status; -} - -int16_t ACMGenericCodec::InternalResetEncoder() { - // Call the codecs internal encoder initialization/reset function. - return InternalInitEncoder(&encoder_params_); -} - -int16_t ACMGenericCodec::InitEncoder(WebRtcACMCodecParams* codec_params, - bool force_initialization) { - WriteLockScoped lockCodec(codec_wrapper_lock_); - ReadLockScoped lockNetEq(*neteq_decode_lock_); - return InitEncoderSafe(codec_params, force_initialization); -} - -int16_t ACMGenericCodec::InitEncoderSafe(WebRtcACMCodecParams* codec_params, - bool force_initialization) { - // Check if we got a valid set of parameters. - int mirrorID; - int codec_number = ACMCodecDB::CodecNumber(&(codec_params->codec_inst), - &mirrorID); - if (codec_number < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitEncoderSafe: error, codec number negative"); - return -1; - } - // Check if the parameters are for this codec. - if ((codec_id_ >= 0) && (codec_id_ != codec_number) && - (codec_id_ != mirrorID)) { - // The current codec is not the same as the one given by codec_params. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitEncoderSafe: current codec is not the same as the one " - "given by codec_params"); - return -1; - } - - if (!CanChangeEncodingParam(codec_params->codec_inst)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitEncoderSafe: cannot change encoding parameters"); - return -1; - } - - if (encoder_initialized_ && !force_initialization) { - // The encoder is already initialized, and we don't want to force - // initialization. - return 0; - } - int16_t status; - if (!encoder_exist_) { - // New encoder, start with creating. - encoder_initialized_ = false; - status = CreateEncoder(); - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitEncoderSafe: cannot create encoder"); - return -1; - } else { - encoder_exist_ = true; - } - } - frame_len_smpl_ = (codec_params->codec_inst).pacsize; - num_channels_ = codec_params->codec_inst.channels; - status = InternalInitEncoder(codec_params); - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitEncoderSafe: error in init encoder"); - encoder_initialized_ = false; - return -1; - } else { - // Store encoder parameters. - memcpy(&encoder_params_, codec_params, sizeof(WebRtcACMCodecParams)); - encoder_initialized_ = true; - if (in_audio_ == NULL) { - in_audio_ = new int16_t[AUDIO_BUFFER_SIZE_W16]; - if (in_audio_ == NULL) { - return -1; - } - } - if (in_timestamp_ == NULL) { - in_timestamp_ = new uint32_t[TIMESTAMP_BUFFER_SIZE_W32]; - if (in_timestamp_ == NULL) { - return -1; - } - } - // Fresh start for audio buffer. - is_audio_buff_fresh_ = true; - memset(in_audio_, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t)); - memset(in_timestamp_, 0, sizeof(uint32_t) * TIMESTAMP_BUFFER_SIZE_W32); - in_audio_ix_write_ = 0; - in_audio_ix_read_ = 0; - in_timestamp_ix_write_ = 0; - } - status = SetVADSafe(&codec_params->enable_dtx, &codec_params->enable_vad, - &codec_params->vad_mode); - return status; -} - -// TODO(tlegrand): Remove the function CanChangeEncodingParam. Returns true -// for all codecs. -bool ACMGenericCodec::CanChangeEncodingParam(CodecInst& /*codec_inst*/) { - return true; -} - -void ACMGenericCodec::CurrentRate(int32_t& /* rate_bps */) { - return; -} - -int16_t ACMGenericCodec::InitDecoder(WebRtcACMCodecParams* codec_params, - bool force_initialization) { - WriteLockScoped lockCodc(codec_wrapper_lock_); - WriteLockScoped lockNetEq(*neteq_decode_lock_); - return InitDecoderSafe(codec_params, force_initialization); -} - -int16_t ACMGenericCodec::InitDecoderSafe(WebRtcACMCodecParams* codec_params, - bool force_initialization) { - int mirror_id; - // Check if we got a valid set of parameters. - int codec_number = ACMCodecDB::ReceiverCodecNumber(&codec_params->codec_inst, - &mirror_id); - if (codec_number < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitDecoderSafe: error, invalid codec number"); - return -1; - } - // Check if the parameters are for this codec. - if ((codec_id_ >= 0) && (codec_id_ != codec_number) && - (codec_id_ != mirror_id)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitDecoderSafe: current codec is not the same as the one " - "given by codec_params"); - // The current codec is not the same as the one given by codec_params. - return -1; - } - - if (decoder_initialized_ && !force_initialization) { - // The decoder is already initialized, and we don't want to force - // initialization. - return 0; - } - - int16_t status; - if (!decoder_exist_) { - // New decoder, start with creating. - decoder_initialized_ = false; - status = CreateDecoder(); - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitDecoderSafe: cannot create decoder"); - return -1; - } else { - decoder_exist_ = true; - } - } - - status = InternalInitDecoder(codec_params); - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InitDecoderSafe: cannot init decoder"); - decoder_initialized_ = false; - return -1; - } else { - // Store decoder parameters. - SaveDecoderParamSafe(codec_params); - decoder_initialized_ = true; - } - return 0; -} - -int16_t ACMGenericCodec::ResetDecoder(int16_t payload_type) { - WriteLockScoped lockCodec(codec_wrapper_lock_); - WriteLockScoped lockNetEq(*neteq_decode_lock_); - return ResetDecoderSafe(payload_type); -} - -int16_t ACMGenericCodec::ResetDecoderSafe(int16_t payload_type) { - WebRtcACMCodecParams decoder_params; - if (!decoder_exist_ || !decoder_initialized_) { - return 0; - } - // Initialization of the decoder should work for all the codec. For codecs - // that needs to keep some states an overloading implementation of - // |DecoderParamsSafe| exists. - DecoderParamsSafe(&decoder_params, static_cast<uint8_t>(payload_type)); - return InternalInitDecoder(&decoder_params); -} - -void ACMGenericCodec::ResetNoMissedSamples() { - WriteLockScoped cs(codec_wrapper_lock_); - num_missed_samples_ = 0; -} - -void ACMGenericCodec::IncreaseNoMissedSamples(const int16_t num_samples) { - num_missed_samples_ += num_samples; -} - -// Get the number of missed samples, this can be public. -uint32_t ACMGenericCodec::NoMissedSamples() const { - ReadLockScoped cs(codec_wrapper_lock_); - return num_missed_samples_; -} - -void ACMGenericCodec::DestructEncoder() { - WriteLockScoped wl(codec_wrapper_lock_); - - // Disable VAD and delete the instance. - if (ptr_vad_inst_ != NULL) { - WebRtcVad_Free(ptr_vad_inst_); - ptr_vad_inst_ = NULL; - } - vad_enabled_ = false; - vad_mode_ = VADNormal; - - // Disable DTX and delete the instance. - dtx_enabled_ = false; - if (ptr_dtx_inst_ != NULL) { - WebRtcCng_FreeEnc(ptr_dtx_inst_); - ptr_dtx_inst_ = NULL; - } - num_lpc_params_ = kNewCNGNumPLCParams; - - DestructEncoderSafe(); -} - -void ACMGenericCodec::DestructDecoder() { - WriteLockScoped wl(codec_wrapper_lock_); - decoder_params_.codec_inst.pltype = -1; - DestructDecoderSafe(); -} - -int16_t ACMGenericCodec::SetBitRate(const int32_t bitrate_bps) { - WriteLockScoped wl(codec_wrapper_lock_); - return SetBitRateSafe(bitrate_bps); -} - -int16_t ACMGenericCodec::SetBitRateSafe(const int32_t bitrate_bps) { - // If the codec can change the bit-rate this function is overloaded. - // Otherwise the only acceptable value is the one that is in the database. - CodecInst codec_params; - if (ACMCodecDB::Codec(codec_id_, &codec_params) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetBitRateSafe: error in ACMCodecDB::Codec"); - return -1; - } - if (codec_params.rate != bitrate_bps) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetBitRateSafe: rate value is not acceptable"); - return -1; - } else { - return 0; - } -} - -// iSAC specific functions: -int32_t ACMGenericCodec::GetEstimatedBandwidth() { - WriteLockScoped wl(codec_wrapper_lock_); - return GetEstimatedBandwidthSafe(); -} - -int32_t ACMGenericCodec::GetEstimatedBandwidthSafe() { - // All codecs but iSAC will return -1. - return -1; -} - -int32_t ACMGenericCodec::SetEstimatedBandwidth(int32_t estimated_bandwidth) { - WriteLockScoped wl(codec_wrapper_lock_); - return SetEstimatedBandwidthSafe(estimated_bandwidth); -} - -int32_t ACMGenericCodec::SetEstimatedBandwidthSafe( - int32_t /*estimated_bandwidth*/) { - // All codecs but iSAC will return -1. - return -1; -} -// End of iSAC specific functions. - -int32_t ACMGenericCodec::GetRedPayload(uint8_t* red_payload, - int16_t* payload_bytes) { - WriteLockScoped wl(codec_wrapper_lock_); - return GetRedPayloadSafe(red_payload, payload_bytes); -} - -int32_t ACMGenericCodec::GetRedPayloadSafe(uint8_t* /* red_payload */, - int16_t* /* payload_bytes */) { - return -1; // Do nothing by default. -} - -int16_t ACMGenericCodec::CreateEncoder() { - int16_t status = 0; - if (!encoder_exist_) { - status = InternalCreateEncoder(); - // We just created the codec and obviously it is not initialized. - encoder_initialized_ = false; - } - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CreateEncoder: error in internal create encoder"); - encoder_exist_ = false; - } else { - encoder_exist_ = true; - } - return status; -} - -int16_t ACMGenericCodec::CreateDecoder() { - int16_t status = 0; - if (!decoder_exist_) { - status = InternalCreateDecoder(); - // Decoder just created and obviously it is not initialized. - decoder_initialized_ = false; - } - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CreateDecoder: error in internal create decoder"); - decoder_exist_ = false; - } else { - decoder_exist_ = true; - } - return status; -} - -void ACMGenericCodec::DestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WriteLockScoped lockCodec(codec_wrapper_lock_); - ReadLockScoped lockNetEq(*neteq_decode_lock_); - InternalDestructEncoderInst(ptr_inst); - } -} - -// Get the current audio buffer including read and write states, and timestamps. -int16_t ACMGenericCodec::AudioBuffer(WebRtcACMAudioBuff& audio_buff) { - ReadLockScoped cs(codec_wrapper_lock_); - memcpy(audio_buff.in_audio, in_audio_, - AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t)); - audio_buff.in_audio_ix_read = in_audio_ix_read_; - audio_buff.in_audio_ix_write = in_audio_ix_write_; - memcpy(audio_buff.in_timestamp, in_timestamp_, - TIMESTAMP_BUFFER_SIZE_W32 * sizeof(uint32_t)); - audio_buff.in_timestamp_ix_write = in_timestamp_ix_write_; - audio_buff.last_timestamp = last_timestamp_; - return 0; -} - -// Set the audio buffer. -int16_t ACMGenericCodec::SetAudioBuffer(WebRtcACMAudioBuff& audio_buff) { - WriteLockScoped cs(codec_wrapper_lock_); - memcpy(in_audio_, audio_buff.in_audio, - AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t)); - in_audio_ix_read_ = audio_buff.in_audio_ix_read; - in_audio_ix_write_ = audio_buff.in_audio_ix_write; - memcpy(in_timestamp_, audio_buff.in_timestamp, - TIMESTAMP_BUFFER_SIZE_W32 * sizeof(uint32_t)); - in_timestamp_ix_write_ = audio_buff.in_timestamp_ix_write; - last_timestamp_ = audio_buff.last_timestamp; - is_audio_buff_fresh_ = false; - return 0; -} - -uint32_t ACMGenericCodec::LastEncodedTimestamp() const { - ReadLockScoped cs(codec_wrapper_lock_); - return last_encoded_timestamp_; -} - -uint32_t ACMGenericCodec::EarliestTimestamp() const { - ReadLockScoped cs(codec_wrapper_lock_); - return in_timestamp_[0]; -} - -int16_t ACMGenericCodec::SetVAD(bool* enable_dtx, bool* enable_vad, - ACMVADMode* mode) { - WriteLockScoped cs(codec_wrapper_lock_); - return SetVADSafe(enable_dtx, enable_vad, mode); -} - -int16_t ACMGenericCodec::SetVADSafe(bool* enable_dtx, bool* enable_vad, - ACMVADMode* mode) { - if (!STR_CASE_CMP(encoder_params_.codec_inst.plname, "OPUS") || - encoder_params_.codec_inst.channels == 2 ) { - // VAD/DTX is not supported for Opus (even if sending mono), or other - // stereo codecs. - DisableDTX(); - DisableVAD(); - *enable_dtx = false; - *enable_vad = false; - return 0; - } - - if (*enable_dtx) { - // Make G729 AnnexB a special case. - if (!STR_CASE_CMP(encoder_params_.codec_inst.plname, "G729") - && !has_internal_dtx_) { - if (ACMGenericCodec::EnableDTX() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetVADSafe: error in enable DTX"); - *enable_dtx = false; - *enable_vad = vad_enabled_; - return -1; - } - } else { - if (EnableDTX() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetVADSafe: error in enable DTX"); - *enable_dtx = false; - *enable_vad = vad_enabled_; - return -1; - } - } - - // If codec does not have internal DTX (normal case) enabling DTX requires - // an active VAD. '*enable_dtx == true' overwrites VAD status. - // If codec has internal DTX, practically we don't need WebRtc VAD, however, - // we let the user to turn it on if they need call-backs on silence. - if (!has_internal_dtx_) { - // DTX is enabled, and VAD will be activated. - *enable_vad = true; - } - } else { - // Make G729 AnnexB a special case. - if (!STR_CASE_CMP(encoder_params_.codec_inst.plname, "G729") - && !has_internal_dtx_) { - ACMGenericCodec::DisableDTX(); - *enable_dtx = false; - } else { - DisableDTX(); - *enable_dtx = false; - } - } - - int16_t status = (*enable_vad) ? EnableVAD(*mode) : DisableVAD(); - if (status < 0) { - // Failed to set VAD, disable DTX. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetVADSafe: error in enable VAD"); - DisableDTX(); - *enable_dtx = false; - *enable_vad = false; - } - return status; -} - -int16_t ACMGenericCodec::EnableDTX() { - if (has_internal_dtx_) { - // We should not be here if we have internal DTX this function should be - // overloaded by the derived class in this case. - return -1; - } - if (!dtx_enabled_) { - if (WebRtcCng_CreateEnc(&ptr_dtx_inst_) < 0) { - ptr_dtx_inst_ = NULL; - return -1; - } - uint16_t freq_hz; - EncoderSampFreq(freq_hz); - if (WebRtcCng_InitEnc(ptr_dtx_inst_, freq_hz, kCngSidIntervalMsec, - num_lpc_params_) < 0) { - // Couldn't initialize, has to return -1, and free the memory. - WebRtcCng_FreeEnc(ptr_dtx_inst_); - ptr_dtx_inst_ = NULL; - return -1; - } - dtx_enabled_ = true; - } - return 0; -} - -int16_t ACMGenericCodec::DisableDTX() { - if (has_internal_dtx_) { - // We should not be here if we have internal DTX this function should be - // overloaded by the derived class in this case. - return -1; - } - if (ptr_dtx_inst_ != NULL) { - WebRtcCng_FreeEnc(ptr_dtx_inst_); - ptr_dtx_inst_ = NULL; - } - dtx_enabled_ = false; - return 0; -} - -int16_t ACMGenericCodec::EnableVAD(ACMVADMode mode) { - if ((mode < VADNormal) || (mode > VADVeryAggr)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EnableVAD: error in VAD mode range"); - return -1; - } - - if (!vad_enabled_) { - if (WebRtcVad_Create(&ptr_vad_inst_) < 0) { - ptr_vad_inst_ = NULL; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EnableVAD: error in create VAD"); - return -1; - } - if (WebRtcVad_Init(ptr_vad_inst_) < 0) { - WebRtcVad_Free(ptr_vad_inst_); - ptr_vad_inst_ = NULL; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EnableVAD: error in init VAD"); - return -1; - } - } - - // Set the VAD mode to the given value. - if (WebRtcVad_set_mode(ptr_vad_inst_, mode) < 0) { - // We failed to set the mode and we have to return -1. If we already have a - // working VAD (vad_enabled_ == true) then we leave it to work. Otherwise, - // the following will be executed. - if (!vad_enabled_) { - // We just created the instance but cannot set the mode we have to free - // the memory. - WebRtcVad_Free(ptr_vad_inst_); - ptr_vad_inst_ = NULL; - } - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, unique_id_, - "EnableVAD: failed to set the VAD mode"); - return -1; - } - vad_mode_ = mode; - vad_enabled_ = true; - return 0; -} - -int16_t ACMGenericCodec::DisableVAD() { - if (ptr_vad_inst_ != NULL) { - WebRtcVad_Free(ptr_vad_inst_); - ptr_vad_inst_ = NULL; - } - vad_enabled_ = false; - return 0; -} - -int32_t ACMGenericCodec::ReplaceInternalDTX(const bool replace_internal_dtx) { - WriteLockScoped cs(codec_wrapper_lock_); - return ReplaceInternalDTXSafe(replace_internal_dtx); -} - -int32_t ACMGenericCodec::ReplaceInternalDTXSafe( - const bool /* replace_internal_dtx */) { - return -1; -} - -int32_t ACMGenericCodec::IsInternalDTXReplaced(bool* internal_dtx_replaced) { - WriteLockScoped cs(codec_wrapper_lock_); - return IsInternalDTXReplacedSafe(internal_dtx_replaced); -} - -int32_t ACMGenericCodec::IsInternalDTXReplacedSafe( - bool* internal_dtx_replaced) { - *internal_dtx_replaced = false; - return 0; -} - -int16_t ACMGenericCodec::ProcessFrameVADDTX(uint8_t* bitstream, - int16_t* bitstream_len_byte, - int16_t* samples_processed) { - if (!vad_enabled_) { - // VAD not enabled, set all |vad_lable_[]| to 1 (speech detected). - for (int n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) { - vad_label_[n] = 1; - } - *samples_processed = 0; - return 0; - } - - uint16_t freq_hz; - EncoderSampFreq(freq_hz); - - // Calculate number of samples in 10 ms blocks, and number ms in one frame. - int16_t samples_in_10ms = static_cast<int16_t>(freq_hz / 100); - int32_t frame_len_ms = static_cast<int32_t>(frame_len_smpl_) * 1000 / freq_hz; - int16_t status; - - // Vector for storing maximum 30 ms of mono audio at 48 kHz. - int16_t audio[1440]; - - // Calculate number of VAD-blocks to process, and number of samples in each - // block. - int num_samples_to_process[2]; - if (frame_len_ms == 40) { - // 20 ms in each VAD block. - num_samples_to_process[0] = num_samples_to_process[1] = 2 * samples_in_10ms; - } else { - // For 10-30 ms framesizes, second VAD block will be size zero ms, - // for 50 and 60 ms first VAD block will be 30 ms. - num_samples_to_process[0] = - (frame_len_ms > 30) ? 3 * samples_in_10ms : frame_len_smpl_; - num_samples_to_process[1] = frame_len_smpl_ - num_samples_to_process[0]; - } - - int offset = 0; - int loops = (num_samples_to_process[1] > 0) ? 2 : 1; - for (int i = 0; i < loops; i++) { - // TODO(turajs): Do we need to care about VAD together with stereo? - // If stereo, calculate mean of the two channels. - if (num_channels_ == 2) { - for (int j = 0; j < num_samples_to_process[i]; j++) { - audio[j] = (in_audio_[(offset + j) * 2] + - in_audio_[(offset + j) * 2 + 1]) / 2; - } - offset = num_samples_to_process[0]; - } else { - // Mono, copy data from in_audio_ to continue work on. - memcpy(audio, in_audio_, sizeof(int16_t) * num_samples_to_process[i]); - } - - // Call VAD. - status = static_cast<int16_t>(WebRtcVad_Process(ptr_vad_inst_, - static_cast<int>(freq_hz), - audio, - num_samples_to_process[i])); - vad_label_[i] = status; - - if (status < 0) { - // This will force that the data be removed from the buffer. - *samples_processed += num_samples_to_process[i]; - return -1; - } - - // If VAD decision non-active, update DTX. NOTE! We only do this if the - // first part of a frame gets the VAD decision "inactive". Otherwise DTX - // might say it is time to transmit SID frame, but we will encode the whole - // frame, because the first part is active. - *samples_processed = 0; - if ((status == 0) && (i == 0) && dtx_enabled_ && !has_internal_dtx_) { - int16_t bitstream_len; - int num_10ms_frames = num_samples_to_process[i] / samples_in_10ms; - *bitstream_len_byte = 0; - for (int n = 0; n < num_10ms_frames; n++) { - // This block is (passive) && (vad enabled). If first CNG after - // speech, force SID by setting last parameter to "1". - status = WebRtcCng_Encode(ptr_dtx_inst_, &audio[n * samples_in_10ms], - samples_in_10ms, bitstream, &bitstream_len, - !prev_frame_cng_); - if (status < 0) { - return -1; - } - - // Update previous frame was CNG. - prev_frame_cng_ = 1; - - *samples_processed += samples_in_10ms * num_channels_; - - // |bitstream_len_byte| will only be > 0 once per 100 ms. - *bitstream_len_byte += bitstream_len; - } - - // Check if all samples got processed by the DTX. - if (*samples_processed != num_samples_to_process[i] * num_channels_) { - // Set to zero since something went wrong. Shouldn't happen. - *samples_processed = 0; - } - } else { - // Update previous frame was not CNG. - prev_frame_cng_ = 0; - } - - if (*samples_processed > 0) { - // The block contains inactive speech, and is processed by DTX. - // Discontinue running VAD. - break; - } - } - - return status; -} - -int16_t ACMGenericCodec::SamplesLeftToEncode() { - ReadLockScoped rl(codec_wrapper_lock_); - return (frame_len_smpl_ <= in_audio_ix_write_) ? 0 : - (frame_len_smpl_ - in_audio_ix_write_); -} - -void ACMGenericCodec::SetUniqueID(const uint32_t id) { - unique_id_ = id; -} - -bool ACMGenericCodec::IsAudioBufferFresh() const { - ReadLockScoped rl(codec_wrapper_lock_); - return is_audio_buff_fresh_; -} - -int16_t ACMGenericCodec::UpdateDecoderSampFreq(int16_t /* codec_id */) { - return 0; -} - -// This function is replaced by codec specific functions for some codecs. -int16_t ACMGenericCodec::EncoderSampFreq(uint16_t& samp_freq_hz) { - int32_t f; - f = ACMCodecDB::CodecFreq(codec_id_); - if (f < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EncoderSampFreq: codec frequency is negative"); - return -1; - } else { - samp_freq_hz = static_cast<uint16_t>(f); - return 0; - } -} - -int32_t ACMGenericCodec::ConfigISACBandwidthEstimator( - const uint8_t /* init_frame_size_msec */, - const uint16_t /* init_rate_bit_per_sec */, - const bool /* enforce_frame_size */) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, unique_id_, - "The send-codec is not iSAC, failed to config iSAC bandwidth " - "estimator."); - return -1; -} - -int32_t ACMGenericCodec::SetISACMaxRate( - const uint32_t /* max_rate_bit_per_sec */) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, unique_id_, - "The send-codec is not iSAC, failed to set iSAC max rate."); - return -1; -} - -int32_t ACMGenericCodec::SetISACMaxPayloadSize( - const uint16_t /* max_payload_len_bytes */) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, unique_id_, - "The send-codec is not iSAC, failed to set iSAC max " - "payload-size."); - return -1; -} - -void ACMGenericCodec::SaveDecoderParam( - const WebRtcACMCodecParams* codec_params) { - WriteLockScoped wl(codec_wrapper_lock_); - SaveDecoderParamSafe(codec_params); -} - -void ACMGenericCodec::SaveDecoderParamSafe( - const WebRtcACMCodecParams* codec_params) { - memcpy(&decoder_params_, codec_params, sizeof(WebRtcACMCodecParams)); -} - -int16_t ACMGenericCodec::UpdateEncoderSampFreq( - uint16_t /* samp_freq_hz */) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "It is asked for a change in smapling frequency while the " - "current send-codec supports only one sampling rate."); - return -1; -} - -void ACMGenericCodec::SetIsMaster(bool is_master) { - WriteLockScoped wl(codec_wrapper_lock_); - is_master_ = is_master; -} - -int16_t ACMGenericCodec::REDPayloadISAC(const int32_t /* isac_rate */, - const int16_t /* isac_bw_estimate */, - uint8_t* /* payload */, - int16_t* /* payload_len_bytes */) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Error: REDPayloadISAC is an iSAC specific function"); - return -1; -} - -bool ACMGenericCodec::IsTrueStereoCodec() { return false; } - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.h deleted file mode 100644 index c1f9cdc554a..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.h +++ /dev/null @@ -1,1224 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ - -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#define MAX_FRAME_SIZE_10MSEC 6 - -// forward declaration -struct WebRtcVadInst; -struct WebRtcCngEncInst; - -namespace webrtc { - -// forward declaration -struct CodecInst; -struct WebRtcACMCodecParams; - -namespace acm1 { - -class ACMNetEQ; - -class ACMGenericCodec { - public: - /////////////////////////////////////////////////////////////////////////// - // Constructor of the class - // - ACMGenericCodec(); - - /////////////////////////////////////////////////////////////////////////// - // Destructor of the class. - // - virtual ~ACMGenericCodec(); - - /////////////////////////////////////////////////////////////////////////// - // ACMGenericCodec* CreateInstance(); - // The function will be used for FEC. It is not implemented yet. - // - virtual ACMGenericCodec* CreateInstance() = 0; - - /////////////////////////////////////////////////////////////////////////// - // int16_t Encode() - // The function is called to perform an encoding of the audio stored in - // audio buffer. An encoding is performed only if enough audio, i.e. equal - // to the frame-size of the codec, exist. The audio frame will be processed - // by VAD and CN/DTX if required. There are few different cases. - // - // A) Neither VAD nor DTX is active; the frame is encoded by the encoder. - // - // B) VAD is enabled but not DTX; in this case the audio is processed by VAD - // and encoded by the encoder. The "*encoding_type" will be either - // "kActiveNormalEncode" or "kPassiveNormalEncode" if frame is active or - // passive, respectively. - // - // C) DTX is enabled; if the codec has internal VAD/DTX we just encode the - // frame by the encoder. Otherwise, the frame is passed through VAD and - // if identified as passive, then it will be processed by CN/DTX. If the - // frame is active it will be encoded by the encoder. - // - // This function acquires the appropriate locks and calls EncodeSafe() for - // the actual processing. - // - // Outputs: - // -bitstream : a buffer where bit-stream will be written to. - // -bitstream_len_byte : contains the length of the bit-stream in - // bytes. - // -timestamp : contains the RTP timestamp, this is the - // sampling time of the first sample encoded - // (measured in number of samples). - // -encoding_type : contains the type of encoding applied on the - // audio samples. The alternatives are - // (c.f. acm_common_types.h) - // -kNoEncoding: - // there was not enough data to encode. or - // some error has happened that we could - // not do encoding. - // -kActiveNormalEncoded: - // the audio frame is active and encoded by - // the given codec. - // -kPassiveNormalEncoded: - // the audio frame is passive but coded with - // the given codec (NO DTX). - // -kPassiveDTXWB: - // The audio frame is passive and used - // wide-band CN to encode. - // -kPassiveDTXNB: - // The audio frame is passive and used - // narrow-band CN to encode. - // - // Return value: - // -1 if error is occurred, otherwise the length of the bit-stream in - // bytes. - // - int16_t Encode(uint8_t* bitstream, - int16_t* bitstream_len_byte, - uint32_t* timestamp, - WebRtcACMEncodingType* encoding_type); - - /////////////////////////////////////////////////////////////////////////// - // int16_t Decode() - // This function is used to decode a given bit-stream, without engaging - // NetEQ. - // - // This function acquires the appropriate locks and calls DecodeSafe() for - // the actual processing. Please note that this is not functional yet. - // - // Inputs: - // -bitstream : a buffer where bit-stream will be read. - // -bitstream_len_byte : the length of the bit-stream in bytes. - // - // Outputs: - // -audio : pointer to a buffer where the audio will written. - // -audio_samples : number of audio samples out of decoding the given - // bit-stream. - // -speech_type : speech type (for future use). - // - // Return value: - // -1 if failed to decode, - // 0 if succeeded. - // - int16_t Decode(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type); - - /////////////////////////////////////////////////////////////////////////// - // void SplitStereoPacket() - // This function is used to split stereo payloads in left and right channel. - // Codecs which has stereo support has there own implementation of the - // function. - // - // Input/Output: - // -payload : a vector with the received payload data. - // The function will reorder the data so that - // first half holds the left channel data, and the - // second half the right channel data. - // -payload_length : length of payload in bytes. Will be changed to - // twice the input in case of true stereo, where - // we simply copy the data and return it both for - // left channel and right channel decoding. - // - virtual void SplitStereoPacket(uint8_t* /* payload */, - int32_t* /* payload_length */) {} - - /////////////////////////////////////////////////////////////////////////// - // bool EncoderInitialized(); - // - // Return value: - // True if the encoder is successfully initialized, - // false otherwise. - // - bool EncoderInitialized(); - - /////////////////////////////////////////////////////////////////////////// - // bool DecoderInitialized(); - // - // Return value: - // True if the decoder is successfully initialized, - // false otherwise. - // - bool DecoderInitialized(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t EncoderParams() - // It is called to get encoder parameters. It will call - // EncoderParamsSafe() in turn. - // - // Output: - // -enc_params : a buffer where the encoder parameters is - // written to. If the encoder is not - // initialized this buffer is filled with - // invalid values - // Return value: - // -1 if the encoder is not initialized, - // 0 otherwise. - // - int16_t EncoderParams(WebRtcACMCodecParams *enc_params); - - /////////////////////////////////////////////////////////////////////////// - // int16_t DecoderParams(...) - // It is called to get decoder parameters. It will call DecoderParamsSafe() - // in turn. - // - // Output: - // -dec_params : a buffer where the decoder parameters is - // written to. If the decoder is not initialized - // this buffer is filled with invalid values - // - // Return value: - // -1 if the decoder is not initialized, - // 0 otherwise. - // - // - bool DecoderParams(WebRtcACMCodecParams *dec_params, - const uint8_t payload_type); - - /////////////////////////////////////////////////////////////////////////// - // int16_t InitEncoder(...) - // This function is called to initialize the encoder with the given - // parameters. - // - // Input: - // -codec_params : parameters of encoder. - // -force_initialization: if false the initialization is invoked only if - // the encoder is not initialized. If true the - // encoder is forced to (re)initialize. - // - // Return value: - // 0 if could initialize successfully, - // -1 if failed to initialize. - // - // - int16_t InitEncoder(WebRtcACMCodecParams* codec_params, - bool force_initialization); - - /////////////////////////////////////////////////////////////////////////// - // int16_t InitDecoder() - // This function is called to initialize the decoder with the given - // parameters. (c.f. acm_common_defs.h & common_types.h for the - // definition of the structure) - // - // Input: - // -codec_params : parameters of decoder. - // -force_initialization: if false the initialization is invoked only - // if the decoder is not initialized. If true - // the encoder is forced to(re)initialize. - // - // Return value: - // 0 if could initialize successfully, - // -1 if failed to initialize. - // - // - int16_t InitDecoder(WebRtcACMCodecParams* codec_params, - bool force_initialization); - - /////////////////////////////////////////////////////////////////////////// - // int32_t RegisterInNetEq(...) - // This function is called to register the decoder in NetEq, with the given - // payload type. - // - // Inputs: - // -neteq : pointer to NetEq Instance - // -codec_inst : instance with of the codec settings of the codec - // - // Return values - // -1 if failed to register, - // 0 if successfully initialized. - // - int32_t RegisterInNetEq(ACMNetEQ* neteq, const CodecInst& codec_inst); - - /////////////////////////////////////////////////////////////////////////// - // int32_t Add10MsData(...) - // This function is called to add 10 ms of audio to the audio buffer of - // the codec. - // - // Inputs: - // -timestamp : the timestamp of the 10 ms audio. the timestamp - // is the sampling time of the - // first sample measured in number of samples. - // -data : a buffer that contains the audio. The codec - // expects to get the audio in correct sampling - // frequency - // -length : the length of the audio buffer - // -audio_channel : 0 for mono, 1 for stereo (not supported yet) - // - // Return values: - // -1 if failed - // 0 otherwise. - // - int32_t Add10MsData(const uint32_t timestamp, - const int16_t* data, - const uint16_t length, - const uint8_t audio_channel); - - /////////////////////////////////////////////////////////////////////////// - // uint32_t NoMissedSamples() - // This function returns the number of samples which are overwritten in - // the audio buffer. The audio samples are overwritten if the input audio - // buffer is full, but Add10MsData() is called. (We might remove this - // function if it is not used) - // - // Return Value: - // Number of samples which are overwritten. - // - uint32_t NoMissedSamples() const; - - /////////////////////////////////////////////////////////////////////////// - // void ResetNoMissedSamples() - // This function resets the number of overwritten samples to zero. - // (We might remove this function if we remove NoMissedSamples()) - // - void ResetNoMissedSamples(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t SetBitRate() - // The function is called to set the encoding rate. - // - // Input: - // -bitrate_bps : encoding rate in bits per second - // - // Return value: - // -1 if failed to set the rate, due to invalid input or given - // codec is not rate-adjustable. - // 0 if the rate is adjusted successfully - // - int16_t SetBitRate(const int32_t bitrate_bps); - - /////////////////////////////////////////////////////////////////////////// - // DestructEncoderInst() - // This API is used in conferencing. It will free the memory that is pointed - // by |ptr_inst|. |ptr_inst| is a pointer to encoder instance, created and - // filled up by calling EncoderInst(...). - // - // Inputs: - // -ptr_inst : pointer to an encoder instance to be deleted. - // - // - void DestructEncoderInst(void* ptr_inst); - - /////////////////////////////////////////////////////////////////////////// - // int16_t AudioBuffer() - // This is used when synchronization of codecs is required. There are cases - // that the audio buffers of two codecs have to be synched. By calling this - // function on can get the audio buffer and other related parameters, such - // as timestamps... - // - // Output: - // -audio_buff : a pointer to WebRtcACMAudioBuff where the audio - // buffer of this codec will be written to. - // - // Return value: - // -1 if fails to copy the audio buffer, - // 0 if succeeded. - // - int16_t AudioBuffer(WebRtcACMAudioBuff& audio_buff); - - /////////////////////////////////////////////////////////////////////////// - // uint32_t EarliestTimestamp() - // Returns the timestamp of the first 10 ms in audio buffer. This is used - // to identify if a synchronization of two encoders is required. - // - // Return value: - // timestamp of the first 10 ms audio in the audio buffer. - // - uint32_t EarliestTimestamp() const; - - /////////////////////////////////////////////////////////////////////////// - // int16_t SetAudioBuffer() - // This function is called to set the audio buffer and the associated - // parameters to a given value. - // - // Return value: - // -1 if fails to copy the audio buffer, - // 0 if succeeded. - // - int16_t SetAudioBuffer(WebRtcACMAudioBuff& audio_buff); - - /////////////////////////////////////////////////////////////////////////// - // int16_t SetVAD() - // This is called to set VAD & DTX. If the codec has internal DTX, it will - // be used. If DTX is enabled and the codec does not have internal DTX, - // WebRtc-VAD will be used to decide if the frame is active. If DTX is - // disabled but VAD is enabled the audio is passed through VAD to label it - // as active or passive, but the frame is encoded normally. However the - // bit-stream is labeled properly so that ACM::Process() can use this - // information. In case of failure, the previous states of the VAD & DTX - // are kept. - // - // Input/Output: - // -enable_dtx : if true DTX will be enabled otherwise the DTX is - // disabled. If codec has internal DTX that will be - // used, otherwise WebRtc-CNG is used. In the latter - // case VAD is automatically activated. - // -enable_vad : if true WebRtc-VAD is enabled, otherwise VAD is - // disabled, except for the case that DTX is enabled - // but codec doesn't have internal DTX. In this case - // VAD is enabled regardless of the value of - // |enable_vad|. - // -mode : this specifies the aggressiveness of VAD. - // - // Return value - // -1 if failed to set DTX & VAD as specified, - // 0 if succeeded. - // - int16_t SetVAD(bool* enable_dtx, - bool* enable_vad, - ACMVADMode* mode); - - /////////////////////////////////////////////////////////////////////////// - // int32_t ReplaceInternalDTX() - // This is called to replace the codec internal DTX with WebRtc DTX. - // This is only valid for G729 where the user has possibility to replace - // AnnexB with WebRtc DTX. For other codecs this function has no effect. - // - // Input: - // -replace_internal_dtx : if true the internal DTX is replaced with WebRtc. - // - // Return value - // -1 if failed to replace internal DTX, - // 0 if succeeded. - // - int32_t ReplaceInternalDTX(const bool replace_internal_dtx); - - /////////////////////////////////////////////////////////////////////////// - // int32_t IsInternalDTXReplaced() - // This is called to check if the codec internal DTX is replaced by WebRtc - // DTX. This is only valid for G729 where the user has possibility to replace - // AnnexB with WebRtc DTX. For other codecs this function has no effect. - // - // Output: - // -internal_dtx_replaced: if true the internal DTX is replaced with WebRtc. - // - // Return value - // -1 if failed to check - // 0 if succeeded. - // - int32_t IsInternalDTXReplaced(bool* internal_dtx_replaced); - - /////////////////////////////////////////////////////////////////////////// - // void SetNetEqDecodeLock() - // Passes the NetEq lock to the codec. - // - // Input: - // -neteq_decode_lock : pointer to the lock associated with NetEQ of ACM. - // - void SetNetEqDecodeLock(RWLockWrapper* neteq_decode_lock) { - neteq_decode_lock_ = neteq_decode_lock; - } - - /////////////////////////////////////////////////////////////////////////// - // bool HasInternalDTX() - // Used to check if the codec has internal DTX. - // - // Return value: - // true if the codec has an internal DTX, e.g. G729, - // false otherwise. - // - bool HasInternalDTX() const { - return has_internal_dtx_; - } - - /////////////////////////////////////////////////////////////////////////// - // int32_t GetEstimatedBandwidth() - // Used to get decoder estimated bandwidth. Only iSAC will provide a value. - // - // - // Return value: - // -1 if fails to get decoder estimated bandwidth, - // >0 estimated bandwidth in bits/sec. - // - int32_t GetEstimatedBandwidth(); - - /////////////////////////////////////////////////////////////////////////// - // int32_t SetEstimatedBandwidth() - // Used to set estiamted bandwidth sent out of band from other side. Only - // iSAC will have use for the value. - // - // Input: - // -estimated_bandwidth: estimated bandwidth in bits/sec - // - // Return value: - // -1 if fails to set estimated bandwidth, - // 0 on success. - // - int32_t SetEstimatedBandwidth(int32_t estimated_bandwidth); - - /////////////////////////////////////////////////////////////////////////// - // int32_t GetRedPayload() - // Used to get codec specific RED payload (if such is implemented). - // Currently only done in iSAC. - // - // Outputs: - // -red_payload : a pointer to the data for RED payload. - // -payload_bytes : number of bytes in RED payload. - // - // Return value: - // -1 if fails to get codec specific RED, - // 0 if succeeded. - // - int32_t GetRedPayload(uint8_t* red_payload, - int16_t* payload_bytes); - - /////////////////////////////////////////////////////////////////////////// - // int16_t ResetEncoder() - // By calling this function you would re-initialize the encoder with the - // current parameters. All the settings, e.g. VAD/DTX, frame-size... should - // remain unchanged. (In case of iSAC we don't want to lose BWE history.) - // - // Return value - // -1 if failed, - // 0 if succeeded. - // - int16_t ResetEncoder(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t ResetEncoder() - // By calling this function you would re-initialize the decoder with the - // current parameters. - // - // Return value - // -1 if failed, - // 0 if succeeded. - // - int16_t ResetDecoder(int16_t payload_type); - - /////////////////////////////////////////////////////////////////////////// - // void DestructEncoder() - // This function is called to delete the encoder instance, if possible, to - // have a fresh start. For codecs where encoder and decoder share the same - // instance we cannot delete the encoder and instead we will initialize the - // encoder. We also delete VAD and DTX if they have been created. - // - void DestructEncoder(); - - /////////////////////////////////////////////////////////////////////////// - // void DestructDecoder() - // This function is called to delete the decoder instance, if possible, to - // have a fresh start. For codecs where encoder and decoder share the same - // instance we cannot delete the encoder and instead we will initialize the - // decoder. Before deleting decoder instance it has to be removed from the - // NetEq list. - // - void DestructDecoder(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t SamplesLeftToEncode() - // Returns the number of samples required to be able to do encoding. - // - // Return value: - // Number of samples. - // - int16_t SamplesLeftToEncode(); - - /////////////////////////////////////////////////////////////////////////// - // uint32_t LastEncodedTimestamp() - // Returns the timestamp of the last frame it encoded. - // - // Return value: - // Timestamp. - // - uint32_t LastEncodedTimestamp() const; - - /////////////////////////////////////////////////////////////////////////// - // SetUniqueID() - // Set a unique ID for the codec to be used for tracing and debugging - // - // Input - // -id : A number to identify the codec. - // - void SetUniqueID(const uint32_t id); - - /////////////////////////////////////////////////////////////////////////// - // IsAudioBufferFresh() - // Specifies if ever audio is injected to this codec. - // - // Return value - // -true; no audio is feed into this codec - // -false; audio has already been fed to the codec. - // - bool IsAudioBufferFresh() const; - - /////////////////////////////////////////////////////////////////////////// - // UpdateDecoderSampFreq() - // For most of the codecs this function does nothing. It must be - // implemented for those codecs that one codec instance serves as the - // decoder for different flavors of the codec. One example is iSAC. there, - // iSAC 16 kHz and iSAC 32 kHz are treated as two different codecs with - // different payload types, however, there is only one iSAC instance to - // decode. The reason for that is we would like to decode and encode with - // the same codec instance for bandwidth estimator to work. - // - // Each time that we receive a new payload type, we call this function to - // prepare the decoder associated with the new payload. Normally, decoders - // doesn't have to do anything. For iSAC the decoder has to change it's - // sampling rate. The input parameter specifies the current flavor of the - // codec in codec database. For instance, if we just got a SWB payload then - // the input parameter is ACMCodecDB::isacswb. - // - // Input: - // -codec_id : the ID of the codec associated with the - // payload type that we just received. - // - // Return value: - // 0 if succeeded in updating the decoder. - // -1 if failed to update. - // - virtual int16_t UpdateDecoderSampFreq(int16_t /* codec_id */); - - /////////////////////////////////////////////////////////////////////////// - // UpdateEncoderSampFreq() - // Call this function to update the encoder sampling frequency. This - // is for codecs where one payload-name supports several encoder sampling - // frequencies. Otherwise, to change the sampling frequency we need to - // register new codec. ACM will consider that as registration of a new - // codec, not a change in parameter. For iSAC, switching from WB to SWB - // is treated as a change in parameter. Therefore, we need this function. - // - // Input: - // -samp_freq_hz : encoder sampling frequency. - // - // Return value: - // -1 if failed, or if this is meaningless for the given codec. - // 0 if succeeded. - // - virtual int16_t UpdateEncoderSampFreq( - uint16_t samp_freq_hz); - - /////////////////////////////////////////////////////////////////////////// - // EncoderSampFreq() - // Get the sampling frequency that the encoder (WebRtc wrapper) expects. - // - // Output: - // -samp_freq_hz : sampling frequency, in Hertz, which the encoder - // should be fed with. - // - // Return value: - // -1 if failed to output sampling rate. - // 0 if the sample rate is returned successfully. - // - virtual int16_t EncoderSampFreq(uint16_t& samp_freq_hz); - - /////////////////////////////////////////////////////////////////////////// - // int32_t ConfigISACBandwidthEstimator() - // Call this function to configure the bandwidth estimator of ISAC. - // During the adaptation of bit-rate, iSAC automatically adjusts the - // frame-size (either 30 or 60 ms) to save on RTP header. The initial - // frame-size can be specified by the first argument. The configuration also - // regards the initial estimate of bandwidths. The estimator starts from - // this point and converges to the actual bottleneck. This is given by the - // second parameter. Furthermore, it is also possible to control the - // adaptation of frame-size. This is specified by the last parameter. - // - // Input: - // -init_frame_fize_ms : initial frame-size in milliseconds. For iSAC-wb - // 30 ms and 60 ms (default) are acceptable values, - // and for iSAC-swb 30 ms is the only acceptable - // value. Zero indicates default value. - // -init_rate_bps : initial estimate of the bandwidth. Values - // between 10000 and 58000 are acceptable. - // -enforce_frame_size : if true, the frame-size will not be adapted. - // - // Return value: - // -1 if failed to configure the bandwidth estimator, - // 0 if the configuration was successfully applied. - // - virtual int32_t ConfigISACBandwidthEstimator( - const uint8_t init_frame_size_msec, - const uint16_t init_rate_bps, - const bool enforce_frame_size); - - /////////////////////////////////////////////////////////////////////////// - // SetISACMaxPayloadSize() - // Set the maximum payload size of iSAC packets. No iSAC payload, - // regardless of its frame-size, may exceed the given limit. For - // an iSAC payload of size B bits and frame-size T sec we have; - // (B < max_payload_len_bytes * 8) and (B/T < max_rate_bit_per_sec), c.f. - // SetISACMaxRate(). - // - // Input: - // -max_payload_len_bytes : maximum payload size in bytes. - // - // Return value: - // -1 if failed to set the maximum payload-size. - // 0 if the given length is set successfully. - // - virtual int32_t SetISACMaxPayloadSize( - const uint16_t max_payload_len_bytes); - - /////////////////////////////////////////////////////////////////////////// - // SetISACMaxRate() - // Set the maximum instantaneous rate of iSAC. For a payload of B bits - // with a frame-size of T sec the instantaneous rate is B/T bits per - // second. Therefore, (B/T < max_rate_bit_per_sec) and - // (B < max_payload_len_bytes * 8) are always satisfied for iSAC payloads, - // c.f SetISACMaxPayloadSize(). - // - // Input: - // -max_rate_bps : maximum instantaneous bit-rate given in bits/sec. - // - // Return value: - // -1 if failed to set the maximum rate. - // 0 if the maximum rate is set successfully. - // - virtual int32_t SetISACMaxRate(const uint32_t max_rate_bps); - - /////////////////////////////////////////////////////////////////////////// - // SaveDecoderParamS() - // Save the parameters of decoder. - // - // Input: - // -codec_params : pointer to a structure where the parameters of - // decoder is stored in. - // - void SaveDecoderParam(const WebRtcACMCodecParams* codec_params); - - int32_t FrameSize() { - return frame_len_smpl_; - } - - void SetIsMaster(bool is_master); - - /////////////////////////////////////////////////////////////////////////// - // REDPayloadISAC() - // This is an iSAC-specific function. The function is called to get RED - // payload from a default-encoder. - // - // Inputs: - // -isac_rate : the target rate of the main payload. A RED - // payload is generated according to the rate of - // main payload. Note that we are not specifying the - // rate of RED payload, but the main payload. - // -isac_bw_estimate : bandwidth information should be inserted in - // RED payload. - // - // Output: - // -payload : pointer to a buffer where the RED payload will - // written to. - // -payload_len_bytes : a place-holder to write the length of the RED - // payload in Bytes. - // - // Return value: - // -1 if an error occurs, otherwise the length of the payload (in Bytes) - // is returned. - // - virtual int16_t REDPayloadISAC(const int32_t isac_rate, - const int16_t isac_bw_estimate, - uint8_t* payload, - int16_t* payload_len_bytes); - - /////////////////////////////////////////////////////////////////////////// - // IsTrueStereoCodec() - // Call to see if current encoder is a true stereo codec. This function - // should be overwritten for codecs which are true stereo codecs - // Return value: - // -true if stereo codec - // -false if not stereo codec. - // - virtual bool IsTrueStereoCodec(); - - /////////////////////////////////////////////////////////////////////////// - // HasFrameToEncode() - // Returns true if there is enough audio buffered for encoding, such that - // calling Encode() will return a payload. - // - bool HasFrameToEncode() const; - - protected: - /////////////////////////////////////////////////////////////////////////// - // All the functions with FunctionNameSafe(...) contain the actual - // implementation of FunctionName(...). FunctionName() acquires an - // appropriate lock and calls FunctionNameSafe() to do the actual work. - // Therefore, for the description of functionality, input/output arguments - // and return value we refer to FunctionName() - // - - /////////////////////////////////////////////////////////////////////////// - // See Decode() for the description of function, input(s)/output(s) and - // return value. - // - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) = 0; - - /////////////////////////////////////////////////////////////////////////// - // See Add10MsSafe() for the description of function, input(s)/output(s) - // and return value. - // - virtual int32_t Add10MsDataSafe(const uint32_t timestamp, - const int16_t* data, - const uint16_t length, - const uint8_t audio_channel); - - /////////////////////////////////////////////////////////////////////////// - // See RegisterInNetEq() for the description of function, - // input(s)/output(s) and return value. - // - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) = 0; - - /////////////////////////////////////////////////////////////////////////// - // See EncoderParam() for the description of function, input(s)/output(s) - // and return value. - // - int16_t EncoderParamsSafe(WebRtcACMCodecParams *enc_params); - - /////////////////////////////////////////////////////////////////////////// - // See DecoderParam for the description of function, input(s)/output(s) - // and return value. - // - // Note: - // Any Class where a single instance handle several flavors of the - // same codec, therefore, several payload types are associated with - // the same instance have to implement this function. - // - // Currently only iSAC is implementing it. A single iSAC instance is - // used for decoding both WB & SWB stream. At one moment both WB & SWB - // can be registered as receive codec. Hence two payloads are associated - // with a single codec instance. - // - virtual bool DecoderParamsSafe(WebRtcACMCodecParams *dec_params, - const uint8_t payload_type); - - /////////////////////////////////////////////////////////////////////////// - // See ResetEncoder() for the description of function, input(s)/output(s) - // and return value. - // - int16_t ResetEncoderSafe(); - - /////////////////////////////////////////////////////////////////////////// - // See InitEncoder() for the description of function, input(s)/output(s) - // and return value. - // - int16_t InitEncoderSafe(WebRtcACMCodecParams *codec_params, - bool force_initialization); - - /////////////////////////////////////////////////////////////////////////// - // See InitDecoder() for the description of function, input(s)/output(s) - // and return value. - // - int16_t InitDecoderSafe(WebRtcACMCodecParams *codec_params, - bool force_initialization); - - /////////////////////////////////////////////////////////////////////////// - // See ResetDecoder() for the description of function, input(s)/output(s) - // and return value. - // - int16_t ResetDecoderSafe(int16_t payload_type); - - /////////////////////////////////////////////////////////////////////////// - // See DestructEncoder() for the description of function, - // input(s)/output(s) and return value. - // - virtual void DestructEncoderSafe() = 0; - - /////////////////////////////////////////////////////////////////////////// - // See DestructDecoder() for the description of function, - // input(s)/output(s) and return value. - // - virtual void DestructDecoderSafe() = 0; - - /////////////////////////////////////////////////////////////////////////// - // See SetBitRate() for the description of function, input(s)/output(s) - // and return value. - // - // Any codec that can change the bit-rate has to implement this. - // - virtual int16_t SetBitRateSafe(const int32_t bitrate_bps); - - /////////////////////////////////////////////////////////////////////////// - // See GetEstimatedBandwidth() for the description of function, - // input(s)/output(s) and return value. - // - virtual int32_t GetEstimatedBandwidthSafe(); - - /////////////////////////////////////////////////////////////////////////// - // See SetEstimatedBandwidth() for the description of function, - // input(s)/output(s) and return value. - // - virtual int32_t SetEstimatedBandwidthSafe( - int32_t estimated_bandwidth); - - /////////////////////////////////////////////////////////////////////////// - // See GetRedPayload() for the description of function, input(s)/output(s) - // and return value. - // - virtual int32_t GetRedPayloadSafe(uint8_t* red_payload, - int16_t* payload_bytes); - - /////////////////////////////////////////////////////////////////////////// - // See SetVAD() for the description of function, input(s)/output(s) and - // return value. - // - int16_t SetVADSafe(bool* enable_dtx, - bool* enable_vad, - ACMVADMode* mode); - - /////////////////////////////////////////////////////////////////////////// - // See ReplaceInternalDTX() for the description of function, input and - // return value. - // - virtual int32_t ReplaceInternalDTXSafe(const bool replace_internal_dtx); - - /////////////////////////////////////////////////////////////////////////// - // See IsInternalDTXReplaced() for the description of function, input and - // return value. - // - virtual int32_t IsInternalDTXReplacedSafe(bool* internal_dtx_replaced); - - /////////////////////////////////////////////////////////////////////////// - // int16_t CreateEncoder() - // Creates the encoder instance. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - int16_t CreateEncoder(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t CreateDecoder() - // Creates the decoder instance. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - int16_t CreateDecoder(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t EnableVAD(); - // Enables VAD with the given mode. The VAD instance will be created if - // it does not exists. - // - // Input: - // -mode : VAD mode c.f. audio_coding_module_typedefs.h for - // the options. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - int16_t EnableVAD(ACMVADMode mode); - - /////////////////////////////////////////////////////////////////////////// - // int16_t DisableVAD() - // Disables VAD. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - int16_t DisableVAD(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t EnableDTX() - // Enables DTX. This method should be overwritten for codecs which have - // internal DTX. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t EnableDTX(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t DisableDTX() - // Disables usage of DTX. This method should be overwritten for codecs which - // have internal DTX. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t DisableDTX(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t InternalEncode() - // This is a codec-specific function called in EncodeSafe() to actually - // encode a frame of audio. - // - // Outputs: - // -bitstream : pointer to a buffer where the bit-stream is - // written to. - // -bitstream_len_byte : the length of the bit-stream in bytes, - // a negative value indicates error. - // - // Return value: - // -1 if failed, - // otherwise the length of the bit-stream is returned. - // - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) = 0; - - /////////////////////////////////////////////////////////////////////////// - // int16_t InternalInitEncoder() - // This is a codec-specific function called in InitEncoderSafe(), it has to - // do all codec-specific operation to initialize the encoder given the - // encoder parameters. - // - // Input: - // -codec_params : pointer to a structure that contains parameters to - // initialize encoder. - // Set codec_params->codec_inst.rate to -1 for - // iSAC to operate in adaptive mode. - // (to do: if frame-length is -1 frame-length will be - // automatically adjusted, otherwise, given - // frame-length is forced) - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams *codec_params) = 0; - - /////////////////////////////////////////////////////////////////////////// - // int16_t InternalInitDecoder() - // This is a codec-specific function called in InitDecoderSafe(), it has to - // do all codec-specific operation to initialize the decoder given the - // decoder parameters. - // - // Input: - // -codec_params : pointer to a structure that contains parameters to - // initialize encoder. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams *codec_params) = 0; - - /////////////////////////////////////////////////////////////////////////// - // void IncreaseNoMissedSamples() - // This method is called to increase the number of samples that are - // overwritten in the audio buffer. - // - // Input: - // -num_samples : the number of overwritten samples is incremented - // by this value. - // - void IncreaseNoMissedSamples(const int16_t num_samples); - - /////////////////////////////////////////////////////////////////////////// - // int16_t InternalCreateEncoder() - // This is a codec-specific method called in CreateEncoderSafe() it is - // supposed to perform all codec-specific operations to create encoder - // instance. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t InternalCreateEncoder() = 0; - - /////////////////////////////////////////////////////////////////////////// - // int16_t InternalCreateDecoder() - // This is a codec-specific method called in CreateDecoderSafe() it is - // supposed to perform all codec-specific operations to create decoder - // instance. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t InternalCreateDecoder() = 0; - - /////////////////////////////////////////////////////////////////////////// - // void InternalDestructEncoderInst() - // This is a codec-specific method, used in conferencing, called from - // DestructEncoderInst(). The input argument is pointer to encoder instance - // (codec instance for codecs that encoder and decoder share the same - // instance). This method is called to free the memory that |ptr_inst| is - // pointing to. - // - // Input: - // -ptr_inst : pointer to encoder instance. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual void InternalDestructEncoderInst(void* ptr_inst) = 0; - - /////////////////////////////////////////////////////////////////////////// - // int16_t InternalResetEncoder() - // This method is called to reset the states of encoder. However, the - // current parameters, e.g. frame-length, should remain as they are. For - // most of the codecs a re-initialization of the encoder is what needs to - // be down. But for iSAC we like to keep the BWE history so we cannot - // re-initialize. As soon as such an API is implemented in iSAC this method - // has to be overwritten in ACMISAC class. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - virtual int16_t InternalResetEncoder(); - - /////////////////////////////////////////////////////////////////////////// - // int16_t ProcessFrameVADDTX() - // This function is called when a full frame of audio is available. It will - // break the audio frame into blocks such that each block could be processed - // by VAD & CN/DTX. If a frame is divided into two blocks then there are two - // cases. First, the first block is active, the second block will not be - // processed by CN/DTX but only by VAD and return to caller with - // '*samples_processed' set to zero. There, the audio frame will be encoded - // by the encoder. Second, the first block is inactive and is processed by - // CN/DTX, then we stop processing the next block and return to the caller - // which is EncodeSafe(), with "*samples_processed" equal to the number of - // samples in first block. - // - // Output: - // -bitstream : pointer to a buffer where DTX frame, if - // generated, will be written to. - // -bitstream_len_byte : contains the length of bit-stream in bytes, if - // generated. Zero if no bit-stream is generated. - // -samples_processed : contains no of samples that actually CN has - // processed. Those samples processed by CN will not - // be encoded by the encoder, obviously. If - // contains zero, it means that the frame has been - // identified as active by VAD. Note that - // "*samples_processed" might be non-zero but - // "*bitstream_len_byte" be zero. - // - // Return value: - // -1 if failed, - // 0 if succeeded. - // - int16_t ProcessFrameVADDTX(uint8_t* bitstream, - int16_t* bitstream_len_byte, - int16_t* samples_processed); - - /////////////////////////////////////////////////////////////////////////// - // CanChangeEncodingParam() - // Check if the codec parameters can be changed. In conferencing normally - // codec parameters cannot be changed. The exception is bit-rate of isac. - // - // return value: - // -true if codec parameters are allowed to change. - // -false otherwise. - // - virtual bool CanChangeEncodingParam(CodecInst& codec_inst); - - /////////////////////////////////////////////////////////////////////////// - // CurrentRate() - // Call to get the current encoding rate of the encoder. This function - // should be overwritten for codecs which automatically change their - // target rate. One example is iSAC. The output of the function is the - // current target rate. - // - // Output: - // -rate_bps : the current target rate of the codec. - // - virtual void CurrentRate(int32_t& /* rate_bps */); - - virtual void SaveDecoderParamSafe(const WebRtcACMCodecParams* codec_params); - - // &in_audio_[in_audio_ix_write_] always point to where new audio can be - // written to - int16_t in_audio_ix_write_; - - // &in_audio_[in_audio_ix_read_] points to where audio has to be read from - int16_t in_audio_ix_read_; - - int16_t in_timestamp_ix_write_; - - // Where the audio is stored before encoding, - // To save memory the following buffer can be allocated - // dynamically for 80 ms depending on the sampling frequency - // of the codec. - int16_t* in_audio_; - uint32_t* in_timestamp_; - - int16_t frame_len_smpl_; - uint16_t num_channels_; - - // This will point to a static database of the supported codecs - int16_t codec_id_; - - // This will account for the number of samples were not encoded - // the case is rare, either samples are missed due to overwrite - // at input buffer or due to encoding error - uint32_t num_missed_samples_; - - // True if the encoder instance created - bool encoder_exist_; - bool decoder_exist_; - // True if the encoder instance initialized - bool encoder_initialized_; - bool decoder_initialized_; - - bool registered_in_neteq_; - - // VAD/DTX - bool has_internal_dtx_; - WebRtcVadInst* ptr_vad_inst_; - bool vad_enabled_; - ACMVADMode vad_mode_; - int16_t vad_label_[MAX_FRAME_SIZE_10MSEC]; - bool dtx_enabled_; - WebRtcCngEncInst* ptr_dtx_inst_; - uint8_t num_lpc_params_; - bool sent_cn_previous_; - bool is_master_; - int16_t prev_frame_cng_; - - WebRtcACMCodecParams encoder_params_; - WebRtcACMCodecParams decoder_params_; - - // Used as a global lock for all available decoders - // so that no decoder is used when NetEQ decodes. - RWLockWrapper* neteq_decode_lock_; - // Used to lock wrapper internal data - // such as buffers and state variables. - RWLockWrapper& codec_wrapper_lock_; - - uint32_t last_encoded_timestamp_; - uint32_t last_timestamp_; - bool is_audio_buff_fresh_; - uint32_t unique_id_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc deleted file mode 100644 index 5ea0c56d9f2..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc +++ /dev/null @@ -1,267 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" -#ifdef WEBRTC_CODEC_GSMFR -// NOTE! GSM-FR is not included in the open-source package. Modify this file -// or your codec API to match the function calls and names of used GSM-FR API -// file. -#include "gsmfr_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_GSMFR - -ACMGSMFR::ACMGSMFR(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - return; -} - -ACMGSMFR::~ACMGSMFR() { - return; -} - -int16_t ACMGSMFR::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMGSMFR::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMGSMFR::EnableDTX() { - return -1; -} - -int16_t ACMGSMFR::DisableDTX() { - return -1; -} - -int16_t ACMGSMFR::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMGSMFR::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMGSMFR::CreateInstance(void) { - return NULL; -} - -int16_t ACMGSMFR::InternalCreateEncoder() { - return -1; -} - -void ACMGSMFR::DestructEncoderSafe() { - return; -} - -int16_t ACMGSMFR::InternalCreateDecoder() { - return -1; -} - -void ACMGSMFR::DestructDecoderSafe() { - return; -} - -void ACMGSMFR::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -#else //===================== Actual Implementation ======================= - -ACMGSMFR::ACMGSMFR(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - codec_id_ = codec_id; - has_internal_dtx_ = true; - return; -} - -ACMGSMFR::~ACMGSMFR() { - if (encoder_inst_ptr_ != NULL) { - WebRtcGSMFR_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcGSMFR_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMGSMFR::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - *bitstream_len_byte = WebRtcGSMFR_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - frame_len_smpl_, - (int16_t*)bitstream); - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += frame_len_smpl_; - return *bitstream_len_byte; -} - -int16_t ACMGSMFR::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMGSMFR::EnableDTX() { - if (dtx_enabled_) { - return 0; - } else if (encoder_exist_) { - if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, 1) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "EnableDTX: cannot init encoder for GSMFR"); - return -1; - } - dtx_enabled_ = true; - return 0; - } else { - return -1; - } -} - -int16_t ACMGSMFR::DisableDTX() { - if (!dtx_enabled_) { - return 0; - } else if (encoder_exist_) { - if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, 0) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "DisableDTX: cannot init encoder for GSMFR"); - return -1; - } - dtx_enabled_ = false; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } -} - -int16_t ACMGSMFR::InternalInitEncoder( - WebRtcACMCodecParams* codec_params) { - if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, - ((codec_params->enable_dtx) ? 1 : 0)) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitEncoder: cannot init encoder for GSMFR"); - } - return 0; -} - -int16_t ACMGSMFR::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - if (WebRtcGSMFR_DecoderInit(decoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: cannot init decoder for GSMFR"); - return -1; - } - return 0; -} - -int32_t ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodecDef: decoder is not initialized for GSMFR"); - return -1; - } - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_GSMFR_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderGSMFR, codec_inst.pltype, - decoder_inst_ptr_, 8000); - SET_GSMFR_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMGSMFR::CreateInstance(void) { - return NULL; -} - -int16_t ACMGSMFR::InternalCreateEncoder() { - if (WebRtcGSMFR_CreateEnc(&encoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: cannot create instance for GSMFR " - "encoder"); - return -1; - } - return 0; -} - -void ACMGSMFR::DestructEncoderSafe() { - if (encoder_inst_ptr_ != NULL) { - WebRtcGSMFR_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - encoder_exist_ = false; - encoder_initialized_ = false; -} - -int16_t ACMGSMFR::InternalCreateDecoder() { - if (WebRtcGSMFR_CreateDec(&decoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateDecoder: cannot create instance for GSMFR " - "decoder"); - return -1; - } - return 0; -} - -void ACMGSMFR::DestructDecoderSafe() { - if (decoder_inst_ptr_ != NULL) { - WebRtcGSMFR_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - decoder_exist_ = false; - decoder_initialized_ = false; -} - -void ACMGSMFR::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcGSMFR_FreeEnc((GSMFR_encinst_t_*) ptr_inst); - } - return; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.h deleted file mode 100644 index aa499734af9..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.h +++ /dev/null @@ -1,71 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct GSMFR_encinst_t_; -struct GSMFR_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMGSMFR : public ACMGenericCodec { - public: - explicit ACMGSMFR(int16_t codec_id); - ~ACMGSMFR(); - - // for FEC - ACMGenericCodec* CreateInstance(void); - - int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte); - - int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); - - int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params); - - protected: - int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type); - - int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst); - - void DestructEncoderSafe(); - - void DestructDecoderSafe(); - - int16_t InternalCreateEncoder(); - - int16_t InternalCreateDecoder(); - - void InternalDestructEncoderInst(void* ptr_inst); - - int16_t EnableDTX(); - - int16_t DisableDTX(); - - GSMFR_encinst_t_* encoder_inst_ptr_; - GSMFR_decinst_t_* decoder_inst_ptr_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc deleted file mode 100644 index 0f8049e8047..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc +++ /dev/null @@ -1,259 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#include "webrtc/modules/audio_coding/main/source/acm_ilbc.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_ILBC -#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_ILBC - -ACMILBC::ACMILBC(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - return; -} - -ACMILBC::~ACMILBC() { - return; -} - -int16_t ACMILBC::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMILBC::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMILBC::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMILBC::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMILBC::CreateInstance(void) { - return NULL; -} - -int16_t ACMILBC::InternalCreateEncoder() { - return -1; -} - -void ACMILBC::DestructEncoderSafe() { - return; -} - -int16_t ACMILBC::InternalCreateDecoder() { - return -1; -} - -void ACMILBC::DestructDecoderSafe() { - return; -} - -void ACMILBC::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -int16_t ACMILBC::SetBitRateSafe(const int32_t /* rate */) { - return -1; -} - -#else //===================== Actual Implementation ======================= - -ACMILBC::ACMILBC(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - codec_id_ = codec_id; - return; -} - -ACMILBC::~ACMILBC() { - if (encoder_inst_ptr_ != NULL) { - WebRtcIlbcfix_EncoderFree(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcIlbcfix_DecoderFree(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMILBC::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - *bitstream_len_byte = WebRtcIlbcfix_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - frame_len_smpl_, - (int16_t*)bitstream); - if (*bitstream_len_byte < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalEncode: error in encode for ILBC"); - return -1; - } - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += frame_len_smpl_; - return *bitstream_len_byte; -} - -int16_t ACMILBC::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMILBC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - // initialize with a correct processing block length - if ((160 == (codec_params->codec_inst).pacsize) || - (320 == (codec_params->codec_inst).pacsize)) { - // processing block of 20ms - return WebRtcIlbcfix_EncoderInit(encoder_inst_ptr_, 20); - } else if ((240 == (codec_params->codec_inst).pacsize) || - (480 == (codec_params->codec_inst).pacsize)) { - // processing block of 30ms - return WebRtcIlbcfix_EncoderInit(encoder_inst_ptr_, 30); - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitEncoder: invalid processing block"); - return -1; - } -} - -int16_t ACMILBC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { - // initialize with a correct processing block length - if ((160 == (codec_params->codec_inst).pacsize) || - (320 == (codec_params->codec_inst).pacsize)) { - // processing block of 20ms - return WebRtcIlbcfix_DecoderInit(decoder_inst_ptr_, 20); - } else if ((240 == (codec_params->codec_inst).pacsize) || - (480 == (codec_params->codec_inst).pacsize)) { - // processing block of 30ms - return WebRtcIlbcfix_DecoderInit(decoder_inst_ptr_, 30); - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalInitDecoder: invalid processing block"); - return -1; - } -} - -int32_t ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodeDef: decoder not initialized for ILBC"); - return -1; - } - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_ILBC_FUNCTION." - // Then return the structure back to NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderILBC, codec_inst.pltype, decoder_inst_ptr_, - 8000); - SET_ILBC_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMILBC::CreateInstance(void) { - return NULL; -} - -int16_t ACMILBC::InternalCreateEncoder() { - if (WebRtcIlbcfix_EncoderCreate(&encoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateEncoder: cannot create instance for ILBC " - "encoder"); - return -1; - } - return 0; -} - -void ACMILBC::DestructEncoderSafe() { - encoder_initialized_ = false; - encoder_exist_ = false; - if (encoder_inst_ptr_ != NULL) { - WebRtcIlbcfix_EncoderFree(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } -} - -int16_t ACMILBC::InternalCreateDecoder() { - if (WebRtcIlbcfix_DecoderCreate(&decoder_inst_ptr_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalCreateDecoder: cannot create instance for ILBC " - "decoder"); - return -1; - } - return 0; -} - -void ACMILBC::DestructDecoderSafe() { - decoder_initialized_ = false; - decoder_exist_ = false; - if (decoder_inst_ptr_ != NULL) { - WebRtcIlbcfix_DecoderFree(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } -} - -void ACMILBC::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcIlbcfix_EncoderFree((iLBC_encinst_t_*) ptr_inst); - } - return; -} - -int16_t ACMILBC::SetBitRateSafe(const int32_t rate) { - // Check that rate is valid. No need to store the value - if (rate == 13300) { - WebRtcIlbcfix_EncoderInit(encoder_inst_ptr_, 30); - } else if (rate == 15200) { - WebRtcIlbcfix_EncoderInit(encoder_inst_ptr_, 20); - } else { - return -1; - } - encoder_params_.codec_inst.rate = rate; - - return 0; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.h deleted file mode 100644 index bd2495fe316..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.h +++ /dev/null @@ -1,71 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct iLBC_encinst_t_; -struct iLBC_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMILBC : public ACMGenericCodec { - public: - explicit ACMILBC(int16_t codec_id); - virtual ~ACMILBC(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual int16_t SetBitRateSafe(const int32_t rate) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - iLBC_encinst_t_* encoder_inst_ptr_; - iLBC_decinst_t_* decoder_inst_ptr_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc deleted file mode 100644 index 61fa32f6d6d..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc +++ /dev/null @@ -1,903 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#include "webrtc/modules/audio_coding/main/source/acm_isac.h" - -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_ISAC -#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" -#include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h" -#endif - -#ifdef WEBRTC_CODEC_ISACFX -#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" -#include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h" -#endif - -namespace webrtc { - -namespace acm1 { - -// we need this otherwise we cannot use forward declaration -// in the header file -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) -struct ACMISACInst { - ACM_ISAC_STRUCT *inst; -}; -#endif - -#define ISAC_MIN_RATE 10000 -#define ISAC_MAX_RATE 56000 - -// Tables for bandwidth estimates -#define NR_ISAC_BANDWIDTHS 24 -static const int32_t kIsacRatesWb[NR_ISAC_BANDWIDTHS] = { - 10000, 11100, 12300, 13700, 15200, 16900, - 18800, 20900, 23300, 25900, 28700, 31900, - 10100, 11200, 12400, 13800, 15300, 17000, - 18900, 21000, 23400, 26000, 28800, 32000 -}; - -static const int32_t kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = { - 10000, 11000, 12400, 13800, 15300, 17000, - 18900, 21000, 23200, 25400, 27600, 29800, - 32000, 34100, 36300, 38500, 40700, 42900, - 45100, 47300, 49500, 51700, 53900, 56000, -}; - -#if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)) - -ACMISAC::ACMISAC(int16_t /* codec_id */) - : codec_inst_ptr_(NULL), - is_enc_initialized_(false), - isac_coding_mode_(CHANNEL_INDEPENDENT), - enforce_frame_size_(false), - isac_currentBN_(32000), - samples_in10MsAudio_(160) { // Initiates to 16 kHz mode. - // Initiate decoder parameters for the 32 kHz mode. - memset(&decoder_params32kHz_, 0, sizeof(WebRtcACMCodecParams)); - decoder_params32kHz_.codec_inst.pltype = -1; - - return; -} - -ACMISAC::~ACMISAC() { - return; -} - -ACMGenericCodec* ACMISAC::CreateInstance(void) { - return NULL; -} - -int16_t ACMISAC::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMISAC::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMISAC::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMISAC::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMISAC::InternalCreateDecoder() { - return -1; -} - -void ACMISAC::DestructDecoderSafe() { - return; -} - -int16_t ACMISAC::InternalCreateEncoder() { - return -1; -} - -void ACMISAC::DestructEncoderSafe() { - return; -} - -int32_t ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -void ACMISAC::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -int16_t ACMISAC::DeliverCachedIsacData( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */, - uint32_t* /* timestamp */, - WebRtcACMEncodingType* /* encoding_type */, - const uint16_t /* isac_rate */, - const uint8_t /* isac_bw_estimate */) { - return -1; -} - -int16_t ACMISAC::Transcode(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */, - int16_t /* q_bwe */, - int32_t /* scale */, - bool /* is_red */) { - return -1; -} - -int16_t ACMISAC::SetBitRateSafe(int32_t /* bit_rate */) { - return -1; -} - -int32_t ACMISAC::GetEstimatedBandwidthSafe() { - return -1; -} - -int32_t ACMISAC::SetEstimatedBandwidthSafe( - int32_t /* estimated_bandwidth */) { - return -1; -} - -int32_t ACMISAC::GetRedPayloadSafe(uint8_t* /* red_payload */, - int16_t* /* payload_bytes */) { - return -1; -} - -int16_t ACMISAC::UpdateDecoderSampFreq(int16_t /* codec_id */) { - return -1; -} - -int16_t ACMISAC::UpdateEncoderSampFreq( - uint16_t /* encoder_samp_freq_hz */) { - return -1; -} - -int16_t ACMISAC::EncoderSampFreq(uint16_t& /* samp_freq_hz */) { - return -1; -} - -int32_t ACMISAC::ConfigISACBandwidthEstimator( - const uint8_t /* init_frame_size_msec */, - const uint16_t /* init_rate_bit_per_sec */, - const bool /* enforce_frame_size */) { - return -1; -} - -int32_t ACMISAC::SetISACMaxPayloadSize( - const uint16_t /* max_payload_len_bytes */) { - return -1; -} - -int32_t ACMISAC::SetISACMaxRate( - const uint32_t /* max_rate_bit_per_sec */) { - return -1; -} - -void ACMISAC::UpdateFrameLen() { - return; -} - -void ACMISAC::CurrentRate(int32_t& /*rate_bit_per_sec */) { - return; -} - -bool -ACMISAC::DecoderParamsSafe( - WebRtcACMCodecParams* /* dec_params */, - const uint8_t /* payload_type */) { - return false; -} - -void -ACMISAC::SaveDecoderParamSafe( - const WebRtcACMCodecParams* /* codec_params */) { - return; -} - -int16_t ACMISAC::REDPayloadISAC( - const int32_t /* isac_rate */, - const int16_t /* isac_bw_estimate */, - uint8_t* /* payload */, - int16_t* /* payload_len_bytes */) { - return -1; -} - -#else //===================== Actual Implementation ======================= - -#ifdef WEBRTC_CODEC_ISACFX - -// How the scaling is computed. iSAC computes a gain based on the -// bottleneck. It follows the following expression for that -// -// G(BN_kbps) = pow(10, (a + b * BN_kbps + c * BN_kbps * BN_kbps) / 20.0) -// / 3.4641; -// -// Where for 30 ms framelength we have, -// -// a = -23; b = 0.48; c = 0; -// -// As the default encoder is operating at 32kbps we have the scale as -// -// S(BN_kbps) = G(BN_kbps) / G(32); - -#define ISAC_NUM_SUPPORTED_RATES 9 - -static const uint16_t kIsacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = { - 32000, 30000, 26000, 23000, 21000, - 19000, 17000, 15000, 12000 -}; - -static const float kIsacScale[ISAC_NUM_SUPPORTED_RATES] = { - 1.0f, 0.8954f, 0.7178f, 0.6081f, 0.5445f, - 0.4875f, 0.4365f, 0.3908f, 0.3311f -}; - -enum IsacSamplingRate { - kIsacWideband = 16, - kIsacSuperWideband = 32 -}; - -static float ACMISACFixTranscodingScale(uint16_t rate) { - // find the scale for transcoding, the scale is rounded - // downward - float scale = -1; - for (int16_t n = 0; n < ISAC_NUM_SUPPORTED_RATES; n++) { - if (rate >= kIsacSuportedRates[n]) { - scale = kIsacScale[n]; - break; - } - } - return scale; -} - -static void ACMISACFixGetSendBitrate(ACM_ISAC_STRUCT* inst, - int32_t* bottleneck) { - *bottleneck = WebRtcIsacfix_GetUplinkBw(inst); -} - -static int16_t ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst, - int16_t bwe_index, - int16_t /* jitter_index */, - int32_t rate, - int16_t* bitstream, - bool is_red) { - if (is_red) { - // RED not supported with iSACFIX - return -1; - } - float scale = ACMISACFixTranscodingScale((uint16_t) rate); - return WebRtcIsacfix_GetNewBitStream(inst, bwe_index, scale, bitstream); -} - -static int16_t ACMISACFixGetSendBWE(ACM_ISAC_STRUCT* inst, - int16_t* rate_index, - int16_t* /* dummy */) { - int16_t local_rate_index; - int16_t status = WebRtcIsacfix_GetDownLinkBwIndex(inst, - &local_rate_index); - if (status < 0) { - return -1; - } else { - *rate_index = local_rate_index; - return 0; - } -} - -static int16_t ACMISACFixControlBWE(ACM_ISAC_STRUCT* inst, - int32_t rate_bps, - int16_t frame_size_ms, - int16_t enforce_frame_size) { - return WebRtcIsacfix_ControlBwe(inst, (int16_t) rate_bps, frame_size_ms, - enforce_frame_size); -} - -static int16_t ACMISACFixControl(ACM_ISAC_STRUCT* inst, - int32_t rate_bps, - int16_t frame_size_ms) { - return WebRtcIsacfix_Control(inst, (int16_t) rate_bps, frame_size_ms); -} - -// The following two function should have the same signature as their counter -// part in iSAC floating-point, i.e. WebRtcIsac_EncSampRate & -// WebRtcIsac_DecSampRate. -static uint16_t ACMISACFixGetEncSampRate(ACM_ISAC_STRUCT* /* inst */) { - return 16000; -} - -static uint16_t ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) { - return 16000; -} - -#endif - -ACMISAC::ACMISAC(int16_t codec_id) - : is_enc_initialized_(false), - isac_coding_mode_(CHANNEL_INDEPENDENT), - enforce_frame_size_(false), - isac_current_bn_(32000), - samples_in_10ms_audio_(160) { // Initiates to 16 kHz mode. - codec_id_ = codec_id; - - // Create codec instance. - codec_inst_ptr_ = new ACMISACInst; - if (codec_inst_ptr_ == NULL) { - return; - } - codec_inst_ptr_->inst = NULL; - - // Initiate decoder parameters for the 32 kHz mode. - memset(&decoder_params_32khz_, 0, sizeof(WebRtcACMCodecParams)); - decoder_params_32khz_.codec_inst.pltype = -1; - - // TODO(tlegrand): Check if the following is really needed, now that - // ACMGenericCodec has been updated to initialize this value. - // Initialize values that can be used uninitialized otherwise - decoder_params_.codec_inst.pltype = -1; -} - -ACMISAC::~ACMISAC() { - if (codec_inst_ptr_ != NULL) { - if (codec_inst_ptr_->inst != NULL) { - ACM_ISAC_FREE(codec_inst_ptr_->inst); - codec_inst_ptr_->inst = NULL; - } - delete codec_inst_ptr_; - codec_inst_ptr_ = NULL; - } - return; -} - -ACMGenericCodec* ACMISAC::CreateInstance(void) { - return NULL; -} - -int16_t ACMISAC::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - // ISAC takes 10ms audio everytime we call encoder, therefor, - // it should be treated like codecs with 'basic coding block' - // non-zero, and the following 'while-loop' should not be necessary. - // However, due to a mistake in the codec the frame-size might change - // at the first 10ms pushed in to iSAC if the bit-rate is low, this is - // sort of a bug in iSAC. to address this we treat iSAC as the - // following. - if (codec_inst_ptr_ == NULL) { - return -1; - } - *bitstream_len_byte = 0; - while ((*bitstream_len_byte == 0) && (in_audio_ix_read_ < frame_len_smpl_)) { - if (in_audio_ix_read_ > in_audio_ix_write_) { - // something is wrong. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "The actual fram-size of iSAC appears to be larger that " - "expected. All audio pushed in but no bit-stream is " - "generated."); - return -1; - } - *bitstream_len_byte = ACM_ISAC_ENCODE(codec_inst_ptr_->inst, - &in_audio_[in_audio_ix_read_], - (int16_t*)bitstream); - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += samples_in_10ms_audio_; - } - if (*bitstream_len_byte == 0) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, unique_id_, - "ISAC Has encoded the whole frame but no bit-stream is " - "generated."); - } - - // a packet is generated iSAC, is set in adaptive mode may change - // the frame length and we like to update the bottleneck value as - // well, although updating bottleneck is not crucial - if ((*bitstream_len_byte > 0) && (isac_coding_mode_ == ADAPTIVE)) { - ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_); - } - UpdateFrameLen(); - return *bitstream_len_byte; -} - -int16_t ACMISAC::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_sample */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - // if rate is set to -1 then iSAC has to be in adaptive mode - if (codec_params->codec_inst.rate == -1) { - isac_coding_mode_ = ADAPTIVE; - } else if ((codec_params->codec_inst.rate >= ISAC_MIN_RATE) && - (codec_params->codec_inst.rate <= ISAC_MAX_RATE)) { - // sanity check that rate is in acceptable range - isac_coding_mode_ = CHANNEL_INDEPENDENT; - isac_current_bn_ = codec_params->codec_inst.rate; - } else { - return -1; - } - - // we need to set the encoder sampling frequency. - if (UpdateEncoderSampFreq((uint16_t) codec_params->codec_inst.plfreq) - < 0) { - return -1; - } - if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) { - return -1; - } - - // apply the frame-size and rate if operating in - // channel-independent mode - if (isac_coding_mode_ == CHANNEL_INDEPENDENT) { - if (ACM_ISAC_CONTROL(codec_inst_ptr_->inst, - codec_params->codec_inst.rate, - codec_params->codec_inst.pacsize / - (codec_params->codec_inst.plfreq / 1000)) < 0) { - return -1; - } - } else { - // We need this for adaptive case and has to be called - // after initialization - ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_); - } - frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); - return 0; -} - -int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { - if (codec_inst_ptr_ == NULL) { - return -1; - } - - // set decoder sampling frequency. - if (codec_params->codec_inst.plfreq == 32000 || - codec_params->codec_inst.plfreq == 48000) { - UpdateDecoderSampFreq(ACMCodecDB::kISACSWB); - } else { - UpdateDecoderSampFreq(ACMCodecDB::kISAC); - } - - // in a one-way communication we may never register send-codec. - // However we like that the BWE to work properly so it has to - // be initialized. The BWE is initialized when iSAC encoder is initialized. - // Therefore, we need this. - if (!encoder_initialized_) { - // Since we don't require a valid rate or a valid packet size when - // initializing the decoder, we set valid values before initializing encoder - codec_params->codec_inst.rate = kIsacWbDefaultRate; - codec_params->codec_inst.pacsize = kIsacPacSize960; - if (InternalInitEncoder(codec_params) < 0) { - return -1; - } - encoder_initialized_ = true; - } - - return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst); -} - -int16_t ACMISAC::InternalCreateDecoder() { - if (codec_inst_ptr_ == NULL) { - return -1; - } - int16_t status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst)); - - // specific to codecs with one instance for encoding and decoding - encoder_initialized_ = false; - if (status < 0) { - encoder_exist_ = false; - } else { - encoder_exist_ = true; - } - return status; -} - -void ACMISAC::DestructDecoderSafe() { - // codec with shared instance cannot delete. - decoder_initialized_ = false; - return; -} - -int16_t ACMISAC::InternalCreateEncoder() { - if (codec_inst_ptr_ == NULL) { - return -1; - } - int16_t status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst)); - - // specific to codecs with one instance for encoding and decoding - decoder_initialized_ = false; - if (status < 0) { - decoder_exist_ = false; - } else { - decoder_exist_ = true; - } - return status; -} - -void ACMISAC::DestructEncoderSafe() { - // codec with shared instance cannot delete. - encoder_initialized_ = false; - return; -} - -int32_t ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - // Sanity checks - if (codec_inst_ptr_ == NULL) { - return -1; - } - if (!decoder_initialized_ || !decoder_exist_) { - return -1; - } - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_ISAC_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - if (codec_inst.plfreq == 16000) { - SET_CODEC_PAR((codec_def), kDecoderISAC, codec_inst.pltype, - codec_inst_ptr_->inst, 16000); -#ifdef WEBRTC_CODEC_ISAC - SET_ISAC_FUNCTIONS((codec_def)); -#else - SET_ISACfix_FUNCTIONS((codec_def)); -#endif - } else { -#ifdef WEBRTC_CODEC_ISAC - // Decoder is either @ 16 kHz or 32 kHz. Even if encoder is set @ 48 kHz - // decoding is @ 32 kHz. - if (codec_inst.plfreq == 32000) { - SET_CODEC_PAR((codec_def), kDecoderISACswb, codec_inst.pltype, - codec_inst_ptr_->inst, 32000); - SET_ISACSWB_FUNCTIONS((codec_def)); - } else { - SET_CODEC_PAR((codec_def), kDecoderISACfb, codec_inst.pltype, - codec_inst_ptr_->inst, 32000); - SET_ISACFB_FUNCTIONS((codec_def)); - } -#else - return -1; -#endif - } - return 0; -} - -void ACMISAC::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - ACM_ISAC_FREE((ACM_ISAC_STRUCT *) ptr_inst); - } - return; -} - -int16_t ACMISAC::Transcode(uint8_t* bitstream, - int16_t* bitstream_len_byte, - int16_t q_bwe, - int32_t rate, - bool is_red) { - int16_t jitter_info = 0; - // transcode from a higher rate to lower rate sanity check - if (codec_inst_ptr_ == NULL) { - return -1; - } - - *bitstream_len_byte = ACM_ISAC_GETNEWBITSTREAM(codec_inst_ptr_->inst, q_bwe, - jitter_info, rate, - (int16_t*)bitstream, - (is_red) ? 1 : 0); - - if (*bitstream_len_byte < 0) { - // error happened - *bitstream_len_byte = 0; - return -1; - } else { - return *bitstream_len_byte; - } -} - -int16_t ACMISAC::SetBitRateSafe(int32_t bit_rate) { - if (codec_inst_ptr_ == NULL) { - return -1; - } - uint16_t encoder_samp_freq; - EncoderSampFreq(encoder_samp_freq); - bool reinit = false; - // change the BN of iSAC - if (bit_rate == -1) { - // ADAPTIVE MODE - // Check if it was already in adaptive mode - if (isac_coding_mode_ != ADAPTIVE) { - // was not in adaptive, then set the mode to adaptive - // and flag for re-initialization - isac_coding_mode_ = ADAPTIVE; - reinit = true; - } - } else if ((bit_rate >= ISAC_MIN_RATE) && (bit_rate <= ISAC_MAX_RATE)) { - // Sanity check if the rate valid - // check if it was in channel-independent mode before - if (isac_coding_mode_ != CHANNEL_INDEPENDENT) { - // was not in channel independent, set the mode to - // channel-independent and flag for re-initialization - isac_coding_mode_ = CHANNEL_INDEPENDENT; - reinit = true; - } - // store the bottleneck - isac_current_bn_ = (uint16_t) bit_rate; - } else { - // invlaid rate - return -1; - } - - int16_t status = 0; - if (reinit) { - // initialize and check if it is successful - if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) { - // failed initialization - return -1; - } - } - if (isac_coding_mode_ == CHANNEL_INDEPENDENT) { - status = ACM_ISAC_CONTROL( - codec_inst_ptr_->inst, isac_current_bn_, - (encoder_samp_freq == 32000 || encoder_samp_freq == 48000) ? 30 : - (frame_len_smpl_ / 16)); - if (status < 0) { - status = -1; - } - } - - // Update encoder parameters - encoder_params_.codec_inst.rate = bit_rate; - - UpdateFrameLen(); - return status; -} - -int32_t ACMISAC::GetEstimatedBandwidthSafe() { - int16_t bandwidth_index = 0; - int16_t delay_index = 0; - int samp_rate; - - // Get bandwidth information - ACM_ISAC_GETSENDBWE(codec_inst_ptr_->inst, &bandwidth_index, &delay_index); - - // Validy check of index - if ((bandwidth_index < 0) || (bandwidth_index >= NR_ISAC_BANDWIDTHS)) { - return -1; - } - - // Check sample frequency - samp_rate = ACM_ISAC_GETDECSAMPRATE(codec_inst_ptr_->inst); - if (samp_rate == 16000) { - return kIsacRatesWb[bandwidth_index]; - } else { - return kIsacRatesSwb[bandwidth_index]; - } -} - -int32_t ACMISAC::SetEstimatedBandwidthSafe( - int32_t estimated_bandwidth) { - int samp_rate; - int16_t bandwidth_index; - - // Check sample frequency and choose appropriate table - samp_rate = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst); - - if (samp_rate == 16000) { - // Search through the WB rate table to find the index - bandwidth_index = NR_ISAC_BANDWIDTHS / 2 - 1; - for (int i = 0; i < (NR_ISAC_BANDWIDTHS / 2); i++) { - if (estimated_bandwidth == kIsacRatesWb[i]) { - bandwidth_index = i; - break; - } else if (estimated_bandwidth - == kIsacRatesWb[i + NR_ISAC_BANDWIDTHS / 2]) { - bandwidth_index = i + NR_ISAC_BANDWIDTHS / 2; - break; - } else if (estimated_bandwidth < kIsacRatesWb[i]) { - bandwidth_index = i; - break; - } - } - } else { - // Search through the SWB rate table to find the index - bandwidth_index = NR_ISAC_BANDWIDTHS - 1; - for (int i = 0; i < NR_ISAC_BANDWIDTHS; i++) { - if (estimated_bandwidth <= kIsacRatesSwb[i]) { - bandwidth_index = i; - break; - } - } - } - - // Set iSAC Bandwidth Estimate - ACM_ISAC_SETBWE(codec_inst_ptr_->inst, bandwidth_index); - - return 0; -} - -int32_t ACMISAC::GetRedPayloadSafe( -#if (!defined(WEBRTC_CODEC_ISAC)) - uint8_t* /* red_payload */, int16_t* /* payload_bytes */) { - return -1; -#else - uint8_t* red_payload, int16_t* payload_bytes) { - int16_t bytes = WebRtcIsac_GetRedPayload(codec_inst_ptr_->inst, - (int16_t*)red_payload); - if (bytes < 0) { - return -1; - } - *payload_bytes = bytes; - return 0; -#endif -} - -int16_t ACMISAC::UpdateDecoderSampFreq( -#ifdef WEBRTC_CODEC_ISAC - int16_t codec_id) { - // The decoder supports only wideband and super-wideband. - if (ACMCodecDB::kISAC == codec_id) { - return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 16000); - } else if (ACMCodecDB::kISACSWB == codec_id || - ACMCodecDB::kISACFB == codec_id) { - return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 32000); - } else { - return -1; - } -#else - int16_t /* codec_id */) { - return 0; -#endif -} - -int16_t ACMISAC::UpdateEncoderSampFreq( -#ifdef WEBRTC_CODEC_ISAC - uint16_t encoder_samp_freq_hz) { - uint16_t current_samp_rate_hz; - EncoderSampFreq(current_samp_rate_hz); - - if (current_samp_rate_hz != encoder_samp_freq_hz) { - if ((encoder_samp_freq_hz != 16000) && - (encoder_samp_freq_hz != 32000) && - (encoder_samp_freq_hz != 48000)) { - return -1; - } else { - in_audio_ix_read_ = 0; - in_audio_ix_write_ = 0; - in_timestamp_ix_write_ = 0; - if (WebRtcIsac_SetEncSampRate(codec_inst_ptr_->inst, - encoder_samp_freq_hz) < 0) { - return -1; - } - samples_in_10ms_audio_ = encoder_samp_freq_hz / 100; - frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); - encoder_params_.codec_inst.pacsize = frame_len_smpl_; - encoder_params_.codec_inst.plfreq = encoder_samp_freq_hz; - return 0; - } - } -#else - uint16_t /* codec_id */) { -#endif - return 0; -} - -int16_t ACMISAC::EncoderSampFreq(uint16_t& samp_freq_hz) { - samp_freq_hz = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst); - return 0; -} - -int32_t ACMISAC::ConfigISACBandwidthEstimator( - const uint8_t init_frame_size_msec, - const uint16_t init_rate_bit_per_sec, - const bool enforce_frame_size) { - int16_t status; - { - uint16_t samp_freq_hz; - EncoderSampFreq(samp_freq_hz); - // TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce - // the frame-size otherwise we might get error. Revise if - // control-bwe is changed. - if (samp_freq_hz == 32000 || samp_freq_hz == 48000) { - status = ACM_ISAC_CONTROL_BWE(codec_inst_ptr_->inst, - init_rate_bit_per_sec, 30, 1); - } else { - status = ACM_ISAC_CONTROL_BWE(codec_inst_ptr_->inst, - init_rate_bit_per_sec, - init_frame_size_msec, - enforce_frame_size ? 1 : 0); - } - } - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Couldn't config iSAC BWE."); - return -1; - } - UpdateFrameLen(); - ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_); - return 0; -} - -int32_t ACMISAC::SetISACMaxPayloadSize( - const uint16_t max_payload_len_bytes) { - return ACM_ISAC_SETMAXPAYLOADSIZE(codec_inst_ptr_->inst, - max_payload_len_bytes); -} - -int32_t ACMISAC::SetISACMaxRate( - const uint32_t max_rate_bit_per_sec) { - return ACM_ISAC_SETMAXRATE(codec_inst_ptr_->inst, max_rate_bit_per_sec); -} - -void ACMISAC::UpdateFrameLen() { - frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); - encoder_params_.codec_inst.pacsize = frame_len_smpl_; -} - -void ACMISAC::CurrentRate(int32_t& rate_bit_per_sec) { - if (isac_coding_mode_ == ADAPTIVE) { - ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &rate_bit_per_sec); - } -} - -bool ACMISAC::DecoderParamsSafe(WebRtcACMCodecParams* dec_params, - const uint8_t payload_type) { - if (decoder_initialized_) { - if (payload_type == decoder_params_.codec_inst.pltype) { - memcpy(dec_params, &decoder_params_, sizeof(WebRtcACMCodecParams)); - return true; - } - if (payload_type == decoder_params_32khz_.codec_inst.pltype) { - memcpy(dec_params, &decoder_params_32khz_, sizeof(WebRtcACMCodecParams)); - return true; - } - } - return false; -} - -void ACMISAC::SaveDecoderParamSafe(const WebRtcACMCodecParams* codec_params) { - // set decoder sampling frequency. - if (codec_params->codec_inst.plfreq == 32000 || - codec_params->codec_inst.plfreq == 48000) { - memcpy(&decoder_params_32khz_, codec_params, sizeof(WebRtcACMCodecParams)); - } else { - memcpy(&decoder_params_, codec_params, sizeof(WebRtcACMCodecParams)); - } -} - -int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate, - const int16_t isac_bw_estimate, - uint8_t* payload, - int16_t* payload_len_bytes) { - int16_t status; - ReadLockScoped rl(codec_wrapper_lock_); - status = Transcode(payload, payload_len_bytes, isac_bw_estimate, isac_rate, - true); - return status; -} - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac.h deleted file mode 100644 index 20b6c5391be..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac.h +++ /dev/null @@ -1,138 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -namespace webrtc { - -namespace acm1 { - -struct ACMISACInst; - -enum IsacCodingMode { - ADAPTIVE, - CHANNEL_INDEPENDENT -}; - -class ACMISAC : public ACMGenericCodec { - public: - explicit ACMISAC(int16_t codec_id); - virtual ~ACMISAC(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - int16_t DeliverCachedIsacData(uint8_t* bitstream, - int16_t* bitstream_len_byte, - uint32_t* timestamp, - WebRtcACMEncodingType* encoding_type, - const uint16_t isac_rate, - const uint8_t isac_bwestimate); - - int16_t DeliverCachedData(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */, - uint32_t* /* timestamp */, - WebRtcACMEncodingType* /* encoding_type */) { - return -1; - } - - virtual int16_t UpdateDecoderSampFreq(int16_t codec_id) OVERRIDE; - - virtual int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE; - - virtual int16_t EncoderSampFreq(uint16_t& samp_freq_hz) OVERRIDE; - - virtual int32_t ConfigISACBandwidthEstimator( - const uint8_t init_frame_size_msec, - const uint16_t init_rate_bit_per_sec, - const bool enforce_frame_size) OVERRIDE; - - virtual int32_t SetISACMaxPayloadSize( - const uint16_t max_payload_len_bytes) OVERRIDE; - - virtual int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec) OVERRIDE; - - virtual int16_t REDPayloadISAC(const int32_t isac_rate, - const int16_t isac_bw_estimate, - uint8_t* payload, - int16_t* payload_len_bytes) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE; - - virtual int32_t GetEstimatedBandwidthSafe() OVERRIDE; - - virtual int32_t SetEstimatedBandwidthSafe( - int32_t estimated_bandwidth) OVERRIDE; - - virtual int32_t GetRedPayloadSafe(uint8_t* red_payload, - int16_t* payload_bytes) OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - int16_t Transcode(uint8_t* bitstream, - int16_t* bitstream_len_byte, - int16_t q_bwe, - int32_t rate, - bool is_red); - - virtual void CurrentRate(int32_t& rate_bit_per_sec) OVERRIDE; - - void UpdateFrameLen(); - - virtual bool DecoderParamsSafe(WebRtcACMCodecParams* dec_params, - const uint8_t payload_type) OVERRIDE; - - virtual void SaveDecoderParamSafe( - const WebRtcACMCodecParams* codec_params) OVERRIDE; - - ACMISACInst* codec_inst_ptr_; - bool is_enc_initialized_; - IsacCodingMode isac_coding_mode_; - bool enforce_frame_size_; - int32_t isac_current_bn_; - uint16_t samples_in_10ms_audio_; - WebRtcACMCodecParams decoder_params_32khz_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac_macros.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac_macros.h deleted file mode 100644 index 01e1e44b3e6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_isac_macros.h +++ /dev/null @@ -1,77 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ - -#include "webrtc/engine_configurations.h" - -namespace webrtc { - -namespace acm1 { - -#ifdef WEBRTC_CODEC_ISAC -#define ACM_ISAC_CREATE WebRtcIsac_Create -#define ACM_ISAC_FREE WebRtcIsac_Free -#define ACM_ISAC_ENCODERINIT WebRtcIsac_EncoderInit -#define ACM_ISAC_ENCODE WebRtcIsac_Encode -#define ACM_ISAC_DECODERINIT WebRtcIsac_DecoderInit -#define ACM_ISAC_DECODE_BWE WebRtcIsac_UpdateBwEstimate -#define ACM_ISAC_DECODE_B WebRtcIsac_Decode -#define ACM_ISAC_DECODEPLC WebRtcIsac_DecodePlc -#define ACM_ISAC_CONTROL WebRtcIsac_Control -#define ACM_ISAC_CONTROL_BWE WebRtcIsac_ControlBwe -#define ACM_ISAC_GETFRAMELEN WebRtcIsac_ReadFrameLen -#define ACM_ISAC_GETERRORCODE WebRtcIsac_GetErrorCode -#define ACM_ISAC_GETSENDBITRATE WebRtcIsac_GetUplinkBw -#define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsac_SetMaxPayloadSize -#define ACM_ISAC_SETMAXRATE WebRtcIsac_SetMaxRate -#define ACM_ISAC_GETNEWBITSTREAM WebRtcIsac_GetNewBitStream -#define ACM_ISAC_GETSENDBWE WebRtcIsac_GetDownLinkBwIndex -#define ACM_ISAC_SETBWE WebRtcIsac_UpdateUplinkBw -#define ACM_ISAC_GETBWE WebRtcIsac_ReadBwIndex -#define ACM_ISAC_GETNEWFRAMELEN WebRtcIsac_GetNewFrameLen -#define ACM_ISAC_STRUCT ISACStruct -#define ACM_ISAC_GETENCSAMPRATE WebRtcIsac_EncSampRate -#define ACM_ISAC_GETDECSAMPRATE WebRtcIsac_DecSampRate -#endif - -#ifdef WEBRTC_CODEC_ISACFX -#define ACM_ISAC_CREATE WebRtcIsacfix_Create -#define ACM_ISAC_FREE WebRtcIsacfix_Free -#define ACM_ISAC_ENCODERINIT WebRtcIsacfix_EncoderInit -#define ACM_ISAC_ENCODE WebRtcIsacfix_Encode -#define ACM_ISAC_DECODERINIT WebRtcIsacfix_DecoderInit -#define ACM_ISAC_DECODE_BWE WebRtcIsacfix_UpdateBwEstimate -#define ACM_ISAC_DECODE_B WebRtcIsacfix_Decode -#define ACM_ISAC_DECODEPLC WebRtcIsacfix_DecodePlc -#define ACM_ISAC_CONTROL ACMISACFixControl // local Impl -#define ACM_ISAC_CONTROL_BWE ACMISACFixControlBWE // local Impl -#define ACM_ISAC_GETFRAMELEN WebRtcIsacfix_ReadFrameLen -#define ACM_ISAC_GETERRORCODE WebRtcIsacfix_GetErrorCode -#define ACM_ISAC_GETSENDBITRATE ACMISACFixGetSendBitrate // local Impl -#define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsacfix_SetMaxPayloadSize -#define ACM_ISAC_SETMAXRATE WebRtcIsacfix_SetMaxRate -#define ACM_ISAC_GETNEWBITSTREAM ACMISACFixGetNewBitstream // local Impl -#define ACM_ISAC_GETSENDBWE ACMISACFixGetSendBWE // local Impl -#define ACM_ISAC_SETBWE WebRtcIsacfix_UpdateUplinkBw -#define ACM_ISAC_GETBWE WebRtcIsacfix_ReadBwIndex -#define ACM_ISAC_GETNEWFRAMELEN WebRtcIsacfix_GetNewFrameLen -#define ACM_ISAC_STRUCT ISACFIX_MainStruct -#define ACM_ISAC_GETENCSAMPRATE ACMISACFixGetEncSampRate // local Impl -#define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl -#endif - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_ - diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc deleted file mode 100644 index 154cc54d004..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc +++ /dev/null @@ -1,1151 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" - -#include <stdlib.h> // malloc - -#include <algorithm> // sort -#include <vector> - -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/system_wrappers/interface/trace_event.h" - -namespace webrtc { - -namespace acm1 { - -#define RTP_HEADER_SIZE 12 -#define NETEQ_INIT_FREQ 8000 -#define NETEQ_INIT_FREQ_KHZ (NETEQ_INIT_FREQ/1000) -#define NETEQ_ERR_MSG_LEN_BYTE (WEBRTC_NETEQ_MAX_ERROR_NAME + 1) - -ACMNetEQ::ACMNetEQ() - : id_(0), - current_samp_freq_khz_(NETEQ_INIT_FREQ_KHZ), - avt_playout_(false), - playout_mode_(voice), - neteq_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vad_status_(false), - vad_mode_(VADNormal), - decode_lock_(RWLockWrapper::CreateRWLock()), - num_slaves_(0), - received_stereo_(false), - master_slave_info_(NULL), - previous_audio_activity_(AudioFrame::kVadUnknown), - callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - min_of_max_num_packets_(0), - min_of_buffer_size_bytes_(0), - per_packet_overhead_bytes_(0), - av_sync_(false), - minimum_delay_ms_(0), - maximum_delay_ms_(0) { - for (int n = 0; n < MAX_NUM_SLAVE_NETEQ + 1; n++) { - is_initialized_[n] = false; - ptr_vadinst_[n] = NULL; - inst_[n] = NULL; - inst_mem_[n] = NULL; - neteq_packet_buffer_[n] = NULL; - } -} - -ACMNetEQ::~ACMNetEQ() { - { - CriticalSectionScoped lock(neteq_crit_sect_); - RemoveNetEQSafe(0); // Master. - RemoveSlavesSafe(); - } - if (neteq_crit_sect_ != NULL) { - delete neteq_crit_sect_; - } - - if (decode_lock_ != NULL) { - delete decode_lock_; - } - - if (callback_crit_sect_ != NULL) { - delete callback_crit_sect_; - } -} - -int32_t ACMNetEQ::Init() { - CriticalSectionScoped lock(neteq_crit_sect_); - - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (InitByIdxSafe(idx) < 0) { - return -1; - } - // delete VAD instance and start fresh if required. - if (ptr_vadinst_[idx] != NULL) { - WebRtcVad_Free(ptr_vadinst_[idx]); - ptr_vadinst_[idx] = NULL; - } - if (vad_status_) { - // Has to enable VAD - if (EnableVADByIdxSafe(idx) < 0) { - // Failed to enable VAD. - // Delete VAD instance, if it is created - if (ptr_vadinst_[idx] != NULL) { - WebRtcVad_Free(ptr_vadinst_[idx]); - ptr_vadinst_[idx] = NULL; - } - // We are at initialization of NetEq, if failed to - // enable VAD, we delete the NetEq instance. - if (inst_mem_[idx] != NULL) { - free(inst_mem_[idx]); - inst_mem_[idx] = NULL; - inst_[idx] = NULL; - } - is_initialized_[idx] = false; - return -1; - } - } - is_initialized_[idx] = true; - } - if (EnableVAD() == -1) { - return -1; - } - return 0; -} - -int16_t ACMNetEQ::InitByIdxSafe(const int16_t idx) { - int memory_size_bytes; - if (WebRtcNetEQ_AssignSize(&memory_size_bytes) != 0) { - LogError("AssignSize", idx); - return -1; - } - - if (inst_mem_[idx] != NULL) { - free(inst_mem_[idx]); - inst_mem_[idx] = NULL; - inst_[idx] = NULL; - } - inst_mem_[idx] = malloc(memory_size_bytes); - if (inst_mem_[idx] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitByIdxSafe: NetEq Initialization error: could not " - "allocate memory for NetEq"); - is_initialized_[idx] = false; - return -1; - } - if (WebRtcNetEQ_Assign(&inst_[idx], inst_mem_[idx]) != 0) { - if (inst_mem_[idx] != NULL) { - free(inst_mem_[idx]); - inst_mem_[idx] = NULL; - inst_[idx] = NULL; - } - LogError("Assign", idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitByIdxSafe: NetEq Initialization error: could not Assign"); - is_initialized_[idx] = false; - return -1; - } - if (WebRtcNetEQ_Init(inst_[idx], NETEQ_INIT_FREQ) != 0) { - if (inst_mem_[idx] != NULL) { - free(inst_mem_[idx]); - inst_mem_[idx] = NULL; - inst_[idx] = NULL; - } - LogError("Init", idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitByIdxSafe: NetEq Initialization error: could not " - "initialize NetEq"); - is_initialized_[idx] = false; - return -1; - } - is_initialized_[idx] = true; - return 0; -} - -int16_t ACMNetEQ::EnableVADByIdxSafe(const int16_t idx) { - if (ptr_vadinst_[idx] == NULL) { - if (WebRtcVad_Create(&ptr_vadinst_[idx]) < 0) { - ptr_vadinst_[idx] = NULL; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "EnableVADByIdxSafe: NetEq Initialization error: could not " - "create VAD"); - return -1; - } - } - - if (WebRtcNetEQ_SetVADInstance( - inst_[idx], ptr_vadinst_[idx], - (WebRtcNetEQ_VADInitFunction) WebRtcVad_Init, - (WebRtcNetEQ_VADSetmodeFunction) WebRtcVad_set_mode, - (WebRtcNetEQ_VADFunction) WebRtcVad_Process) < 0) { - LogError("setVADinstance", idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "EnableVADByIdxSafe: NetEq Initialization error: could not " - "set VAD instance"); - return -1; - } - - if (WebRtcNetEQ_SetVADMode(inst_[idx], vad_mode_) < 0) { - LogError("setVADmode", idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "EnableVADByIdxSafe: NetEq Initialization error: could not " - "set VAD mode"); - return -1; - } - return 0; -} - -int32_t ACMNetEQ::AllocatePacketBuffer( - const WebRtcNetEQDecoder* used_codecs, - int16_t num_codecs) { - // Due to WebRtcNetEQ_GetRecommendedBufferSize - // the following has to be int otherwise we will have compiler error - // if not casted - - CriticalSectionScoped lock(neteq_crit_sect_); - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (AllocatePacketBufferByIdxSafe(used_codecs, num_codecs, idx) < 0) { - return -1; - } - } - return 0; -} - -int16_t ACMNetEQ::AllocatePacketBufferByIdxSafe( - const WebRtcNetEQDecoder* used_codecs, - int16_t num_codecs, - const int16_t idx) { - int max_num_packets; - int buffer_size_in_bytes; - int per_packet_overhead_bytes; - - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AllocatePacketBufferByIdxSafe: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_GetRecommendedBufferSize(inst_[idx], used_codecs, - num_codecs, - kTCPXLargeJitter, - &max_num_packets, - &buffer_size_in_bytes, - &per_packet_overhead_bytes) != 0) { - LogError("GetRecommendedBufferSize", idx); - return -1; - } - if (idx == 0) { - min_of_buffer_size_bytes_ = buffer_size_in_bytes; - min_of_max_num_packets_ = max_num_packets; - per_packet_overhead_bytes_ = per_packet_overhead_bytes; - } else { - min_of_buffer_size_bytes_ = std::min(min_of_buffer_size_bytes_, - buffer_size_in_bytes); - min_of_max_num_packets_ = std::min(min_of_max_num_packets_, - max_num_packets); - } - if (neteq_packet_buffer_[idx] != NULL) { - free(neteq_packet_buffer_[idx]); - neteq_packet_buffer_[idx] = NULL; - } - - neteq_packet_buffer_[idx] = (int16_t *) malloc(buffer_size_in_bytes); - if (neteq_packet_buffer_[idx] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AllocatePacketBufferByIdxSafe: NetEq Initialization error: " - "could not allocate memory for NetEq Packet Buffer"); - return -1; - } - if (WebRtcNetEQ_AssignBuffer(inst_[idx], max_num_packets, - neteq_packet_buffer_[idx], - buffer_size_in_bytes) != 0) { - if (neteq_packet_buffer_[idx] != NULL) { - free(neteq_packet_buffer_[idx]); - neteq_packet_buffer_[idx] = NULL; - } - LogError("AssignBuffer", idx); - return -1; - } - return 0; -} - -int32_t ACMNetEQ::SetAVTPlayout(const bool enable) { - CriticalSectionScoped lock(neteq_crit_sect_); - if (avt_playout_ != enable) { - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetAVTPlayout: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_SetAVTPlayout(inst_[idx], (enable) ? 1 : 0) < 0) { - LogError("SetAVTPlayout", idx); - return -1; - } - } - } - avt_playout_ = enable; - return 0; -} - -bool ACMNetEQ::avt_playout() const { - CriticalSectionScoped lock(neteq_crit_sect_); - return avt_playout_; -} - -int32_t ACMNetEQ::CurrentSampFreqHz() const { - CriticalSectionScoped lock(neteq_crit_sect_); - if (!is_initialized_[0]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "CurrentSampFreqHz: NetEq is not initialized."); - return -1; - } - return (int32_t)(1000 * current_samp_freq_khz_); -} - -int32_t ACMNetEQ::SetPlayoutMode(const AudioPlayoutMode mode) { - CriticalSectionScoped lock(neteq_crit_sect_); - if (playout_mode_ == mode) - return 0; - - enum WebRtcNetEQPlayoutMode playout_mode = kPlayoutOff; - enum WebRtcNetEQBGNMode background_noise_mode = kBGNOn; - switch (mode) { - case voice: - playout_mode = kPlayoutOn; - background_noise_mode = kBGNOn; - break; - case fax: - playout_mode = kPlayoutFax; - WebRtcNetEQ_GetBGNMode(inst_[0], &background_noise_mode); // No change. - break; - case streaming: - playout_mode = kPlayoutStreaming; - background_noise_mode = kBGNOff; - break; - case off: - playout_mode = kPlayoutOff; - background_noise_mode = kBGNOff; - break; - } - - int err = 0; - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetPlayoutMode: NetEq is not initialized."); - return -1; - } - - if (WebRtcNetEQ_SetPlayoutMode(inst_[idx], playout_mode) < 0) { - LogError("SetPlayoutMode", idx); - err = -1; - } - - if (WebRtcNetEQ_SetBGNMode(inst_[idx], kBGNOff) < 0) { - LogError("SetPlayoutMode::SetBGNMode", idx); - err = -1; - } - } - if (err == 0) - playout_mode_ = mode; - return err; -} - -AudioPlayoutMode ACMNetEQ::playout_mode() const { - CriticalSectionScoped lock(neteq_crit_sect_); - return playout_mode_; -} - -int32_t ACMNetEQ::NetworkStatistics( - ACMNetworkStatistics* statistics) const { - WebRtcNetEQ_NetworkStatistics stats; - CriticalSectionScoped lock(neteq_crit_sect_); - if (!is_initialized_[0]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "NetworkStatistics: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_GetNetworkStatistics(inst_[0], &stats) == 0) { - statistics->currentAccelerateRate = stats.currentAccelerateRate; - statistics->currentBufferSize = stats.currentBufferSize; - statistics->jitterPeaksFound = (stats.jitterPeaksFound > 0); - statistics->currentDiscardRate = stats.currentDiscardRate; - statistics->currentExpandRate = stats.currentExpandRate; - statistics->currentPacketLossRate = stats.currentPacketLossRate; - statistics->currentPreemptiveRate = stats.currentPreemptiveRate; - statistics->preferredBufferSize = stats.preferredBufferSize; - statistics->clockDriftPPM = stats.clockDriftPPM; - statistics->addedSamples = stats.addedSamples; - } else { - LogError("getNetworkStatistics", 0); - return -1; - } - const int kArrayLen = 100; - int waiting_times[kArrayLen]; - int waiting_times_len = WebRtcNetEQ_GetRawFrameWaitingTimes(inst_[0], - kArrayLen, - waiting_times); - if (waiting_times_len > 0) { - std::vector<int> waiting_times_vec(waiting_times, - waiting_times + waiting_times_len); - std::sort(waiting_times_vec.begin(), waiting_times_vec.end()); - size_t size = waiting_times_vec.size(); - assert(size == static_cast<size_t>(waiting_times_len)); - if (size % 2 == 0) { - statistics->medianWaitingTimeMs = (waiting_times_vec[size / 2 - 1] + - waiting_times_vec[size / 2]) / 2; - } else { - statistics->medianWaitingTimeMs = waiting_times_vec[size / 2]; - } - statistics->minWaitingTimeMs = waiting_times_vec.front(); - statistics->maxWaitingTimeMs = waiting_times_vec.back(); - double sum = 0; - for (size_t i = 0; i < size; ++i) { - sum += waiting_times_vec[i]; - } - statistics->meanWaitingTimeMs = static_cast<int>(sum / size); - } else if (waiting_times_len == 0) { - statistics->meanWaitingTimeMs = -1; - statistics->medianWaitingTimeMs = -1; - statistics->minWaitingTimeMs = -1; - statistics->maxWaitingTimeMs = -1; - } else { - LogError("getRawFrameWaitingTimes", 0); - return -1; - } - return 0; -} - -// Should only be called in AV-sync mode. -int ACMNetEQ::RecIn(const WebRtcRTPHeader& rtp_info, - uint32_t receive_timestamp) { - assert(av_sync_); - - // Translate to NetEq structure. - WebRtcNetEQ_RTPInfo neteq_rtpinfo; - neteq_rtpinfo.payloadType = rtp_info.header.payloadType; - neteq_rtpinfo.sequenceNumber = rtp_info.header.sequenceNumber; - neteq_rtpinfo.timeStamp = rtp_info.header.timestamp; - neteq_rtpinfo.SSRC = rtp_info.header.ssrc; - neteq_rtpinfo.markerBit = rtp_info.header.markerBit; - - CriticalSectionScoped lock(neteq_crit_sect_); - - // Master should be initialized. - assert(is_initialized_[0]); - - // Push into Master. - int status = WebRtcNetEQ_RecInSyncRTP(inst_[0], &neteq_rtpinfo, - receive_timestamp); - if (status < 0) { - LogError("RecInSyncRTP", 0); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecIn (sync): NetEq, error in pushing in Master"); - return -1; - } - - // If the received stream is stereo, insert a sync payload into slave. - if (rtp_info.type.Audio.channel == 2) { - // Slave should be initialized. - assert(is_initialized_[1]); - - // PUSH into Slave - status = WebRtcNetEQ_RecInSyncRTP(inst_[1], &neteq_rtpinfo, - receive_timestamp); - if (status < 0) { - LogError("RecInRTPStruct", 1); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecIn (sync): NetEq, error in pushing in Slave"); - return -1; - } - } - return status; -} - -int32_t ACMNetEQ::RecIn(const uint8_t* incoming_payload, - const int32_t length_payload, - const WebRtcRTPHeader& rtp_info, - uint32_t receive_timestamp) { - int16_t payload_length = static_cast<int16_t>(length_payload); - - // Translate to NetEq structure. - WebRtcNetEQ_RTPInfo neteq_rtpinfo; - neteq_rtpinfo.payloadType = rtp_info.header.payloadType; - neteq_rtpinfo.sequenceNumber = rtp_info.header.sequenceNumber; - neteq_rtpinfo.timeStamp = rtp_info.header.timestamp; - neteq_rtpinfo.SSRC = rtp_info.header.ssrc; - neteq_rtpinfo.markerBit = rtp_info.header.markerBit; - - CriticalSectionScoped lock(neteq_crit_sect_); - - int status; - // In case of stereo payload, first half of the data should be pushed into - // master, and the second half into slave. - if (rtp_info.type.Audio.channel == 2) { - payload_length = payload_length / 2; - } - - // Check that master is initialized. - if (!is_initialized_[0]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecIn: NetEq is not initialized."); - return -1; - } - // Push into Master. - status = WebRtcNetEQ_RecInRTPStruct(inst_[0], &neteq_rtpinfo, - incoming_payload, payload_length, - receive_timestamp); - if (status < 0) { - LogError("RecInRTPStruct", 0); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecIn: NetEq, error in pushing in Master"); - return -1; - } - - // If the received stream is stereo, insert second half of paket into slave. - if (rtp_info.type.Audio.channel == 2) { - if (!is_initialized_[1]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecIn: NetEq is not initialized."); - return -1; - } - // Push into Slave. - status = WebRtcNetEQ_RecInRTPStruct(inst_[1], &neteq_rtpinfo, - &incoming_payload[payload_length], - payload_length, receive_timestamp); - if (status < 0) { - LogError("RecInRTPStruct", 1); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecIn: NetEq, error in pushing in Slave"); - return -1; - } - } - - return 0; -} - -int32_t ACMNetEQ::RecOut(AudioFrame& audio_frame) { - enum WebRtcNetEQOutputType type; - int16_t payload_len_sample; - enum WebRtcNetEQOutputType type_master; - enum WebRtcNetEQOutputType type_slave; - - int16_t payload_len_sample_slave; - - CriticalSectionScoped lockNetEq(neteq_crit_sect_); - - if (!received_stereo_) { - if (!is_initialized_[0]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecOut: NetEq is not initialized."); - return -1; - } - { - WriteLockScoped lockCodec(*decode_lock_); - if (WebRtcNetEQ_RecOut(inst_[0], &(audio_frame.data_[0]), - &payload_len_sample) != 0) { - LogError("RecOut", 0); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecOut: NetEq, error in pulling out for mono case"); - // Check for errors that can be recovered from: - // RECOUT_ERROR_SAMPLEUNDERRUN = 2003 - int error_code = WebRtcNetEQ_GetErrorCode(inst_[0]); - if (error_code != 2003) { - // Cannot recover; return an error - return -1; - } - } - } - WebRtcNetEQ_GetSpeechOutputType(inst_[0], &type); - audio_frame.num_channels_ = 1; - } else { - if (!is_initialized_[0] || !is_initialized_[1]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecOut: NetEq is not initialized."); - return -1; - } - int16_t payload_master[480]; - int16_t payload_slave[480]; - { - WriteLockScoped lockCodec(*decode_lock_); - if (WebRtcNetEQ_RecOutMasterSlave(inst_[0], payload_master, - &payload_len_sample, master_slave_info_, - 1) != 0) { - LogError("RecOutMasterSlave", 0); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecOut: NetEq, error in pulling out for master"); - - // Check for errors that can be recovered from: - // RECOUT_ERROR_SAMPLEUNDERRUN = 2003 - int error_code = WebRtcNetEQ_GetErrorCode(inst_[0]); - if (error_code != 2003) { - // Cannot recover; return an error - return -1; - } - } - if (WebRtcNetEQ_RecOutMasterSlave(inst_[1], payload_slave, - &payload_len_sample_slave, - master_slave_info_, 0) != 0) { - LogError("RecOutMasterSlave", 1); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RecOut: NetEq, error in pulling out for slave"); - - // Check for errors that can be recovered from: - // RECOUT_ERROR_SAMPLEUNDERRUN = 2003 - int error_code = WebRtcNetEQ_GetErrorCode(inst_[1]); - if (error_code != 2003) { - // Cannot recover; return an error - return -1; - } - } - } - - if (payload_len_sample != payload_len_sample_slave) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "RecOut: mismatch between the lenght of the decoded audio " - "by Master (%d samples) and Slave (%d samples).", - payload_len_sample, payload_len_sample_slave); - if (payload_len_sample > payload_len_sample_slave) { - memset(&payload_slave[payload_len_sample_slave], 0, - (payload_len_sample - payload_len_sample_slave) * - sizeof(int16_t)); - } - } - - for (int16_t n = 0; n < payload_len_sample; n++) { - audio_frame.data_[n << 1] = payload_master[n]; - audio_frame.data_[(n << 1) + 1] = payload_slave[n]; - } - audio_frame.num_channels_ = 2; - - WebRtcNetEQ_GetSpeechOutputType(inst_[0], &type_master); - WebRtcNetEQ_GetSpeechOutputType(inst_[1], &type_slave); - if ((type_master == kOutputNormal) || (type_slave == kOutputNormal)) { - type = kOutputNormal; - } else { - type = type_master; - } - } - - audio_frame.samples_per_channel_ = - static_cast<uint16_t>(payload_len_sample); - // NetEq always returns 10 ms of audio. - current_samp_freq_khz_ = - static_cast<float>(audio_frame.samples_per_channel_) / 10.0f; - audio_frame.sample_rate_hz_ = audio_frame.samples_per_channel_ * 100; - if (vad_status_) { - if (type == kOutputVADPassive) { - audio_frame.vad_activity_ = AudioFrame::kVadPassive; - audio_frame.speech_type_ = AudioFrame::kNormalSpeech; - } else if (type == kOutputNormal) { - audio_frame.vad_activity_ = AudioFrame::kVadActive; - audio_frame.speech_type_ = AudioFrame::kNormalSpeech; - } else if (type == kOutputPLC) { - audio_frame.vad_activity_ = previous_audio_activity_; - audio_frame.speech_type_ = AudioFrame::kPLC; - } else if (type == kOutputCNG) { - audio_frame.vad_activity_ = AudioFrame::kVadPassive; - audio_frame.speech_type_ = AudioFrame::kCNG; - } else { - audio_frame.vad_activity_ = AudioFrame::kVadPassive; - audio_frame.speech_type_ = AudioFrame::kPLCCNG; - } - } else { - // Always return kVadUnknown when receive VAD is inactive - audio_frame.vad_activity_ = AudioFrame::kVadUnknown; - if (type == kOutputNormal) { - audio_frame.speech_type_ = AudioFrame::kNormalSpeech; - } else if (type == kOutputPLC) { - audio_frame.speech_type_ = AudioFrame::kPLC; - } else if (type == kOutputPLCtoCNG) { - audio_frame.speech_type_ = AudioFrame::kPLCCNG; - } else if (type == kOutputCNG) { - audio_frame.speech_type_ = AudioFrame::kCNG; - } else { - // type is kOutputVADPassive which - // we don't expect to get if vad_status_ is false - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "RecOut: NetEq returned kVadPassive while vad_status_ is " - "false."); - audio_frame.vad_activity_ = AudioFrame::kVadUnknown; - audio_frame.speech_type_ = AudioFrame::kNormalSpeech; - } - } - previous_audio_activity_ = audio_frame.vad_activity_; - - WebRtcNetEQ_ProcessingActivity processing_stats; - WebRtcNetEQ_GetProcessingActivity(inst_[0], &processing_stats); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "ACM::RecOut accelerate_bgn=%d accelerate_normal=%d" - " expand_bgn=%d expand_normal=%d" - " preemptive_bgn=%d preemptive_normal=%d" - " merge_bgn=%d merge_normal=%d", - processing_stats.accelerate_bgn_samples, - processing_stats.accelerate_normal_samples, - processing_stats.expand_bgn_sampels, - processing_stats.expand_normal_samples, - processing_stats.preemptive_expand_bgn_samples, - processing_stats.preemptive_expand_normal_samples, - processing_stats.merge_expand_bgn_samples, - processing_stats.merge_expand_normal_samples); - return 0; -} - -// When ACMGenericCodec has set the codec specific parameters in codec_def -// it calls AddCodec() to add the new codec to the NetEQ database. -int32_t ACMNetEQ::AddCodec(WebRtcNetEQ_CodecDef* codec_def, - bool to_master) { - if (codec_def == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ACMNetEQ::AddCodec: error, codec_def is NULL"); - return -1; - } - CriticalSectionScoped lock(neteq_crit_sect_); - - int16_t idx; - if (to_master) { - idx = 0; - } else { - idx = 1; - } - - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ACMNetEQ::AddCodec: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_CodecDbAdd(inst_[idx], codec_def) < 0) { - LogError("CodecDB_Add", idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ACMNetEQ::AddCodec: NetEq, error in adding codec"); - return -1; - } else { - return 0; - } -} - -// Creates a Word16 RTP packet out of a Word8 payload and an rtp info struct. -// Must be byte order safe. -void ACMNetEQ::RTPPack(int16_t* rtp_packet, const int8_t* payload, - const int32_t payload_length_bytes, - const WebRtcRTPHeader& rtp_info) { - int32_t idx = 0; - WEBRTC_SPL_SET_BYTE(rtp_packet, (int8_t) 0x80, idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, rtp_info.header.payloadType, idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.sequenceNumber), 1), - idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.sequenceNumber), 0), - idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.timestamp), 3), - idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.timestamp), 2), - idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.timestamp), 1), - idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.timestamp), 0), - idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, - WEBRTC_SPL_GET_BYTE(&(rtp_info.header.ssrc), 3), idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, WEBRTC_SPL_GET_BYTE(&(rtp_info.header.ssrc), - 2), idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, WEBRTC_SPL_GET_BYTE(&(rtp_info.header.ssrc), - 1), idx); - idx++; - WEBRTC_SPL_SET_BYTE(rtp_packet, WEBRTC_SPL_GET_BYTE(&(rtp_info.header.ssrc), - 0), idx); - idx++; - for (int16_t i = 0; i < payload_length_bytes; i++) { - WEBRTC_SPL_SET_BYTE(rtp_packet, payload[i], idx); - idx++; - } - if (payload_length_bytes & 1) { - // Our 16 bits buffer is one byte too large, set that - // last byte to zero. - WEBRTC_SPL_SET_BYTE(rtp_packet, 0x0, idx); - } -} - -int16_t ACMNetEQ::EnableVAD() { - CriticalSectionScoped lock(neteq_crit_sect_); - if (vad_status_) { - return 0; - } - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetVADStatus: NetEq is not initialized."); - return -1; - } - // VAD was off and we have to turn it on - if (EnableVADByIdxSafe(idx) < 0) { - return -1; - } - - // Set previous VAD status to PASSIVE - previous_audio_activity_ = AudioFrame::kVadPassive; - } - vad_status_ = true; - return 0; -} - -ACMVADMode ACMNetEQ::vad_mode() const { - CriticalSectionScoped lock(neteq_crit_sect_); - return vad_mode_; -} - -int16_t ACMNetEQ::SetVADMode(const ACMVADMode mode) { - CriticalSectionScoped lock(neteq_crit_sect_); - if ((mode < VADNormal) || (mode > VADVeryAggr)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetVADMode: NetEq error: could not set VAD mode, mode is not " - "supported"); - return -1; - } else { - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetVADMode: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_SetVADMode(inst_[idx], mode) < 0) { - LogError("SetVADmode", idx); - return -1; - } - } - vad_mode_ = mode; - return 0; - } -} - -int32_t ACMNetEQ::FlushBuffers() { - CriticalSectionScoped lock(neteq_crit_sect_); - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "FlushBuffers: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_FlushBuffers(inst_[idx]) < 0) { - LogError("FlushBuffers", idx); - return -1; - } - } - return 0; -} - -int16_t ACMNetEQ::RemoveCodec(WebRtcNetEQDecoder codec_idx, - bool is_stereo) { - // sanity check - if ((codec_idx <= kDecoderReservedStart) || - (codec_idx >= kDecoderReservedEnd)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RemoveCodec: NetEq error: could not Remove Codec, codec " - "index out of range"); - return -1; - } - CriticalSectionScoped lock(neteq_crit_sect_); - if (!is_initialized_[0]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RemoveCodec: NetEq is not initialized."); - return -1; - } - - if (WebRtcNetEQ_CodecDbRemove(inst_[0], codec_idx) < 0) { - LogError("CodecDB_Remove", 0); - return -1; - } - - if (is_stereo) { - if (WebRtcNetEQ_CodecDbRemove(inst_[1], codec_idx) < 0) { - LogError("CodecDB_Remove", 1); - return -1; - } - } - - return 0; -} - -int16_t ACMNetEQ::SetBackgroundNoiseMode( - const ACMBackgroundNoiseMode mode) { - CriticalSectionScoped lock(neteq_crit_sect_); - for (int16_t idx = 0; idx < num_slaves_ + 1; idx++) { - if (!is_initialized_[idx]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetBackgroundNoiseMode: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_SetBGNMode(inst_[idx], (WebRtcNetEQBGNMode) mode) < 0) { - LogError("SetBGNMode", idx); - return -1; - } - } - return 0; -} - -int16_t ACMNetEQ::BackgroundNoiseMode(ACMBackgroundNoiseMode& mode) { - WebRtcNetEQBGNMode my_mode; - CriticalSectionScoped lock(neteq_crit_sect_); - if (!is_initialized_[0]) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "BackgroundNoiseMode: NetEq is not initialized."); - return -1; - } - if (WebRtcNetEQ_GetBGNMode(inst_[0], &my_mode) < 0) { - LogError("WebRtcNetEQ_GetBGNMode", 0); - return -1; - } else { - mode = (ACMBackgroundNoiseMode) my_mode; - } - return 0; -} - -void ACMNetEQ::set_id(int32_t id) { - CriticalSectionScoped lock(neteq_crit_sect_); - id_ = id; -} - -void ACMNetEQ::LogError(const char* neteq_func_name, - const int16_t idx) const { - char error_name[NETEQ_ERR_MSG_LEN_BYTE]; - char my_func_name[50]; - int neteq_error_code = WebRtcNetEQ_GetErrorCode(inst_[idx]); - WebRtcNetEQ_GetErrorName(neteq_error_code, error_name, - NETEQ_ERR_MSG_LEN_BYTE - 1); - strncpy(my_func_name, neteq_func_name, 49); - error_name[NETEQ_ERR_MSG_LEN_BYTE - 1] = '\0'; - my_func_name[49] = '\0'; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "NetEq-%d Error in function %s, error-code: %d, error-string: " - " %s", idx, my_func_name, neteq_error_code, error_name); -} - -int32_t ACMNetEQ::PlayoutTimestamp(uint32_t& timestamp) { - CriticalSectionScoped lock(neteq_crit_sect_); - if (WebRtcNetEQ_GetSpeechTimeStamp(inst_[0], ×tamp) < 0) { - LogError("GetSpeechTimeStamp", 0); - return -1; - } else { - return 0; - } -} - -void ACMNetEQ::RemoveSlaves() { - CriticalSectionScoped lock(neteq_crit_sect_); - RemoveSlavesSafe(); -} - -void ACMNetEQ::RemoveSlavesSafe() { - for (int i = 1; i < num_slaves_ + 1; i++) { - RemoveNetEQSafe(i); - } - - if (master_slave_info_ != NULL) { - free(master_slave_info_); - master_slave_info_ = NULL; - } - num_slaves_ = 0; -} - -void ACMNetEQ::RemoveNetEQSafe(int index) { - if (inst_mem_[index] != NULL) { - free(inst_mem_[index]); - inst_mem_[index] = NULL; - inst_[index] = NULL; - } - if (neteq_packet_buffer_[index] != NULL) { - free(neteq_packet_buffer_[index]); - neteq_packet_buffer_[index] = NULL; - } - if (ptr_vadinst_[index] != NULL) { - WebRtcVad_Free(ptr_vadinst_[index]); - ptr_vadinst_[index] = NULL; - } -} - -int16_t ACMNetEQ::AddSlave(const WebRtcNetEQDecoder* used_codecs, - int16_t num_codecs) { - CriticalSectionScoped lock(neteq_crit_sect_); - const int16_t slave_idx = 1; - if (num_slaves_ < 1) { - // initialize the receiver, this also sets up VAD. - if (InitByIdxSafe(slave_idx) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AddSlave: AddSlave Failed, Could not Initialize"); - return -1; - } - - // Allocate buffer. - if (AllocatePacketBufferByIdxSafe(used_codecs, num_codecs, - slave_idx) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AddSlave: AddSlave Failed, Could not Allocate Packet " - "Buffer"); - return -1; - } - - if (master_slave_info_ != NULL) { - free(master_slave_info_); - master_slave_info_ = NULL; - } - int ms_info_size = WebRtcNetEQ_GetMasterSlaveInfoSize(); - master_slave_info_ = malloc(ms_info_size); - - if (master_slave_info_ == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AddSlave: AddSlave Failed, Could not Allocate memory for " - "Master-Slave Info"); - return -1; - } - - // We accept this as initialized NetEQ, the rest is to synchronize - // Slave with Master. - num_slaves_ = 1; - is_initialized_[slave_idx] = true; - - // Set AVT - if (WebRtcNetEQ_SetAVTPlayout(inst_[slave_idx], - (avt_playout_) ? 1 : 0) < 0) { - LogError("SetAVTPlayout", slave_idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AddSlave: AddSlave Failed, Could not set AVT playout."); - return -1; - } - - // Set Background Noise - WebRtcNetEQBGNMode current_mode; - if (WebRtcNetEQ_GetBGNMode(inst_[0], ¤t_mode) < 0) { - LogError("GetBGNMode", 0); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AAddSlave: AddSlave Failed, Could not Get BGN form " - "Master."); - return -1; - } - - if (WebRtcNetEQ_SetBGNMode(inst_[slave_idx], - (WebRtcNetEQBGNMode) current_mode) < 0) { - LogError("SetBGNMode", slave_idx); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AddSlave: AddSlave Failed, Could not set BGN mode."); - return -1; - } - - enum WebRtcNetEQPlayoutMode playout_mode = kPlayoutOff; - switch (playout_mode_) { - case voice: - playout_mode = kPlayoutOn; - break; - case fax: - playout_mode = kPlayoutFax; - break; - case streaming: - playout_mode = kPlayoutStreaming; - break; - case off: - playout_mode = kPlayoutOff; - break; - } - if (WebRtcNetEQ_SetPlayoutMode(inst_[slave_idx], playout_mode) < 0) { - LogError("SetPlayoutMode", 1); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "AddSlave: AddSlave Failed, Could not Set Playout Mode."); - return -1; - } - - // Set AV-sync for the slave. - WebRtcNetEQ_EnableAVSync(inst_[slave_idx], av_sync_ ? 1 : 0); - - // Set minimum delay. - if (minimum_delay_ms_ > 0) - WebRtcNetEQ_SetMinimumDelay(inst_[slave_idx], minimum_delay_ms_); - - // Set maximum delay. - if (maximum_delay_ms_ > 0) - WebRtcNetEQ_SetMaximumDelay(inst_[slave_idx], maximum_delay_ms_); - } - - return 0; -} - -void ACMNetEQ::set_received_stereo(bool received_stereo) { - CriticalSectionScoped lock(neteq_crit_sect_); - received_stereo_ = received_stereo; -} - -uint8_t ACMNetEQ::num_slaves() { - CriticalSectionScoped lock(neteq_crit_sect_); - return num_slaves_; -} - -void ACMNetEQ::EnableAVSync(bool enable) { - CriticalSectionScoped lock(neteq_crit_sect_); - av_sync_ = enable; - for (int i = 0; i < num_slaves_ + 1; ++i) { - assert(is_initialized_[i]); - WebRtcNetEQ_EnableAVSync(inst_[i], enable ? 1 : 0); - } -} - -int ACMNetEQ::SetMinimumDelay(int minimum_delay_ms) { - CriticalSectionScoped lock(neteq_crit_sect_); - for (int i = 0; i < num_slaves_ + 1; ++i) { - assert(is_initialized_[i]); - if (WebRtcNetEQ_SetMinimumDelay(inst_[i], minimum_delay_ms) < 0) - return -1; - } - minimum_delay_ms_ = minimum_delay_ms; - return 0; -} - -int ACMNetEQ::SetMaximumDelay(int maximum_delay_ms) { - CriticalSectionScoped lock(neteq_crit_sect_); - for (int i = 0; i < num_slaves_ + 1; ++i) { - assert(is_initialized_[i]); - if (WebRtcNetEQ_SetMaximumDelay(inst_[i], maximum_delay_ms) < 0) - return -1; - } - maximum_delay_ms_ = maximum_delay_ms; - return 0; -} - -int ACMNetEQ::LeastRequiredDelayMs() const { - CriticalSectionScoped lock(neteq_crit_sect_); - assert(is_initialized_[0]); - - // Sufficient to query the master. - return WebRtcNetEQ_GetRequiredDelayMs(inst_[0]); -} - -bool ACMNetEQ::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const { - CriticalSectionScoped lock(neteq_crit_sect_); - if (WebRtcNetEQ_DecodedRtpInfo(inst_[0], sequence_number, timestamp) < 0) - return false; - return true; -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq.h deleted file mode 100644 index e52ddc79571..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq.h +++ /dev/null @@ -1,399 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ - -#include "webrtc/common_audio/vad/include/webrtc_vad.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -class CriticalSectionWrapper; -class RWLockWrapper; -struct CodecInst; - -namespace acm1 { - -#define MAX_NUM_SLAVE_NETEQ 1 - -class ACMNetEQ { - public: - enum JitterBuffer { - kMasterJb = 0, - kSlaveJb = 1 - }; - - // Constructor of the class - ACMNetEQ(); - - // Destructor of the class. - ~ACMNetEQ(); - - // - // Init() - // Allocates memory for NetEQ and VAD and initializes them. - // - // Return value : 0 if ok. - // -1 if NetEQ or VAD returned an error or - // if out of memory. - // - int32_t Init(); - - // - // RecIn() - // Gives the payload to NetEQ. - // - // Input: - // - incoming_payload : Incoming audio payload. - // - length_payload : Length of incoming audio payload. - // - rtp_info : RTP header for the incoming payload containing - // information about payload type, sequence number, - // timestamp, SSRC and marker bit. - // - receive_timestamp : received timestamp. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int32_t RecIn(const uint8_t* incoming_payload, - const int32_t length_payload, - const WebRtcRTPHeader& rtp_info, - uint32_t receive_timestamp); - - // - // RecIn() - // Insert a sync payload to NetEq. Should only be called if |av_sync_| is - // enabled; - // - // Input: - // - rtp_info : RTP header for the incoming payload containing - // information about payload type, sequence number, - // timestamp, SSRC and marker bit. - // - receive_timestamp : received timestamp. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int RecIn(const WebRtcRTPHeader& rtp_info, uint32_t receive_timestamp); - - // - // RecOut() - // Asks NetEQ for 10 ms of decoded audio. - // - // Input: - // -audio_frame : an audio frame were output data and - // associated parameters are written to. - // - // Return value : 0 if ok. - // -1 if NetEQ returned an error. - // - int32_t RecOut(AudioFrame& audio_frame); - - // - // AddCodec() - // Adds a new codec to the NetEQ codec database. - // - // Input: - // - codec_def : The codec to be added. - // - to_master : true if the codec has to be added to Master - // NetEq, otherwise will be added to the Slave - // NetEQ. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int32_t AddCodec(WebRtcNetEQ_CodecDef *codec_def, - bool to_master = true); - - // - // AllocatePacketBuffer() - // Allocates the NetEQ packet buffer. - // - // Input: - // - used_codecs : An array of the codecs to be used by NetEQ. - // - num_codecs : Number of codecs in used_codecs. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int32_t AllocatePacketBuffer(const WebRtcNetEQDecoder* used_codecs, - int16_t num_codecs); - - // - // SetAVTPlayout() - // Enable/disable playout of AVT payloads. - // - // Input: - // - enable : Enable if true, disable if false. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int32_t SetAVTPlayout(const bool enable); - - // - // AVTPlayout() - // Get the current AVT playout state. - // - // Return value : True if AVT playout is enabled. - // False if AVT playout is disabled. - // - bool avt_playout() const; - - // - // CurrentSampFreqHz() - // Get the current sampling frequency in Hz. - // - // Return value : Sampling frequency in Hz. - // - int32_t CurrentSampFreqHz() const; - - // - // SetPlayoutMode() - // Sets the playout mode to voice or fax. - // - // Input: - // - mode : The playout mode to be used, voice, - // fax, or streaming. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int32_t SetPlayoutMode(const AudioPlayoutMode mode); - - // - // PlayoutMode() - // Get the current playout mode. - // - // Return value : The current playout mode. - // - AudioPlayoutMode playout_mode() const; - - // - // NetworkStatistics() - // Get the current network statistics from NetEQ. - // - // Output: - // - statistics : The current network statistics. - // - // Return value : 0 if ok. - // <0 if NetEQ returned an error. - // - int32_t NetworkStatistics(ACMNetworkStatistics* statistics) const; - - // - // VADMode() - // Get the current VAD Mode. - // - // Return value : The current VAD mode. - // - ACMVADMode vad_mode() const; - - // - // SetVADMode() - // Set the VAD mode. - // - // Input: - // - mode : The new VAD mode. - // - // Return value : 0 if ok. - // -1 if an error occurred. - // - int16_t SetVADMode(const ACMVADMode mode); - - // - // DecodeLock() - // Get the decode lock used to protect decoder instances while decoding. - // - // Return value : Pointer to the decode lock. - // - RWLockWrapper* DecodeLock() const { - return decode_lock_; - } - - // - // FlushBuffers() - // Flushes the NetEQ packet and speech buffers. - // - // Return value : 0 if ok. - // -1 if NetEQ returned an error. - // - int32_t FlushBuffers(); - - // - // RemoveCodec() - // Removes a codec from the NetEQ codec database. - // - // Input: - // - codec_idx : Codec to be removed. - // - // Return value : 0 if ok. - // -1 if an error occurred. - // - int16_t RemoveCodec(WebRtcNetEQDecoder codec_idx, - bool is_stereo = false); - - // - // SetBackgroundNoiseMode() - // Set the mode of the background noise. - // - // Input: - // - mode : an enumerator specifying the mode of the - // background noise. - // - // Return value : 0 if succeeded, - // -1 if failed to set the mode. - // - int16_t SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode); - - // - // BackgroundNoiseMode() - // return the mode of the background noise. - // - // Return value : The mode of background noise. - // - int16_t BackgroundNoiseMode(ACMBackgroundNoiseMode& mode); - - void set_id(int32_t id); - - int32_t PlayoutTimestamp(uint32_t& timestamp); - - void set_received_stereo(bool received_stereo); - - uint8_t num_slaves(); - - // Delete all slaves. - void RemoveSlaves(); - - int16_t AddSlave(const WebRtcNetEQDecoder* used_codecs, - int16_t num_codecs); - - void BufferSpec(int& num_packets, int& size_bytes, int& overhead_bytes) { - num_packets = min_of_max_num_packets_; - size_bytes = min_of_buffer_size_bytes_; - overhead_bytes = per_packet_overhead_bytes_; - } - - // - // Set AV-sync mode. - // - void EnableAVSync(bool enable); - - // - // Get sequence number and timestamp of the last decoded RTP. - // - bool DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const; - - // - // Set a minimum delay in NetEq. Unless channel condition dictates a longer - // delay, the given delay is maintained by NetEq. - // - int SetMinimumDelay(int minimum_delay_ms); - - // - // Set a maximum delay in NetEq. - // - int SetMaximumDelay(int maximum_delay_ms); - - // - // The shortest latency, in milliseconds, required by jitter buffer. This - // is computed based on inter-arrival times and playout mode of NetEq. The - // actual delay is the maximum of least-required-delay and the minimum-delay - // specified by SetMinumumPlayoutDelay() API. - // - int LeastRequiredDelayMs() const ; - - private: - // - // RTPPack() - // Creates a Word16 RTP packet out of the payload data in Word16 and - // a WebRtcRTPHeader. - // - // Input: - // - payload : Payload to be packetized. - // - payload_length_bytes : Length of the payload in bytes. - // - rtp_info : RTP header structure. - // - // Output: - // - rtp_packet : The RTP packet. - // - static void RTPPack(int16_t* rtp_packet, const int8_t* payload, - const int32_t payload_length_bytes, - const WebRtcRTPHeader& rtp_info); - - void LogError(const char* neteq_func_name, const int16_t idx) const; - - int16_t InitByIdxSafe(const int16_t idx); - - // - // EnableVAD() - // Enable VAD. - // - // Return value : 0 if ok. - // -1 if an error occurred. - // - int16_t EnableVAD(); - - int16_t EnableVADByIdxSafe(const int16_t idx); - - int16_t AllocatePacketBufferByIdxSafe( - const WebRtcNetEQDecoder* used_codecs, - int16_t num_codecs, - const int16_t idx); - - // Delete the NetEQ corresponding to |index|. - void RemoveNetEQSafe(int index); - - void RemoveSlavesSafe(); - - void* inst_[MAX_NUM_SLAVE_NETEQ + 1]; - void* inst_mem_[MAX_NUM_SLAVE_NETEQ + 1]; - - int16_t* neteq_packet_buffer_[MAX_NUM_SLAVE_NETEQ + 1]; - - int32_t id_; - float current_samp_freq_khz_; - bool avt_playout_; - AudioPlayoutMode playout_mode_; - CriticalSectionWrapper* neteq_crit_sect_; - - WebRtcVadInst* ptr_vadinst_[MAX_NUM_SLAVE_NETEQ + 1]; - - bool vad_status_; - ACMVADMode vad_mode_; - RWLockWrapper* decode_lock_; - bool is_initialized_[MAX_NUM_SLAVE_NETEQ + 1]; - uint8_t num_slaves_; - bool received_stereo_; - void* master_slave_info_; - AudioFrame::VADActivity previous_audio_activity_; - - CriticalSectionWrapper* callback_crit_sect_; - // Minimum of "max number of packets," among all NetEq instances. - int min_of_max_num_packets_; - // Minimum of buffer-size among all NetEq instances. - int min_of_buffer_size_bytes_; - int per_packet_overhead_bytes_; - - // Keep track of AV-sync. Just used to set the slave when a slave is added. - bool av_sync_; - - int minimum_delay_ms_; - int maximum_delay_ms_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc deleted file mode 100644 index 8b973ba230f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_neteq_unittest.cc +++ /dev/null @@ -1,153 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// This file contains unit tests for ACM's NetEQ wrapper (class ACMNetEQ). - -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" - -#include <stdlib.h> - -#include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -namespace acm1 { - -class AcmNetEqTest : public ::testing::Test { - protected: - static const size_t kMaxPayloadLen = 5760; // 60 ms, 48 kHz, 16 bit samples. - static const int kPcm16WbPayloadType = 94; - AcmNetEqTest() {} - virtual void SetUp(); - virtual void TearDown() {} - - void InsertZeroPacket(uint16_t sequence_number, - uint32_t timestamp, - uint8_t payload_type, - uint32_t ssrc, - bool marker_bit, - size_t len_payload_bytes); - void PullData(int expected_num_samples); - - ACMNetEQ neteq_; -}; - -void AcmNetEqTest::SetUp() { - ASSERT_EQ(0, neteq_.Init()); - ASSERT_EQ(0, neteq_.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs)); - WebRtcNetEQ_CodecDef codec_def; - SET_CODEC_PAR(codec_def, kDecoderPCM16Bwb, kPcm16WbPayloadType, NULL, 16000); - SET_PCM16B_WB_FUNCTIONS(codec_def); - ASSERT_EQ(0, neteq_.AddCodec(&codec_def, true)); -} - -void AcmNetEqTest::InsertZeroPacket(uint16_t sequence_number, - uint32_t timestamp, - uint8_t payload_type, - uint32_t ssrc, - bool marker_bit, - size_t len_payload_bytes) { - ASSERT_TRUE(len_payload_bytes <= kMaxPayloadLen); - uint16_t payload[kMaxPayloadLen] = {0}; - WebRtcRTPHeader rtp_header; - rtp_header.header.sequenceNumber = sequence_number; - rtp_header.header.timestamp = timestamp; - rtp_header.header.ssrc = ssrc; - rtp_header.header.payloadType = payload_type; - rtp_header.header.markerBit = marker_bit; - rtp_header.type.Audio.channel = 1; - // Receive timestamp can be set to send timestamp in this test. - ASSERT_EQ(0, neteq_.RecIn(reinterpret_cast<uint8_t*>(payload), - len_payload_bytes, rtp_header, timestamp)); -} - -void AcmNetEqTest::PullData(int expected_num_samples) { - AudioFrame out_frame; - ASSERT_EQ(0, neteq_.RecOut(out_frame)); - ASSERT_EQ(expected_num_samples, out_frame.samples_per_channel_); -} - -TEST_F(AcmNetEqTest, NetworkStatistics) { - // Use fax mode to avoid time-scaling. This is to simplify the testing of - // packet waiting times in the packet buffer. - neteq_.SetPlayoutMode(fax); - // Insert 31 dummy packets at once. Each packet contains 10 ms 16 kHz audio. - int num_frames = 30; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - int i, j; - for (i = 0; i < num_frames; ++i) { - InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false, - kPayloadBytes); - } - // Pull out data once. - PullData(kSamples); - // Insert one more packet (to produce different mean and median). - i = num_frames; - InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false, - kPayloadBytes); - // Pull out all data. - for (j = 1; j < num_frames + 1; ++j) { - PullData(kSamples); - } - - ACMNetworkStatistics stats; - ASSERT_EQ(0, neteq_.NetworkStatistics(&stats)); - EXPECT_EQ(0, stats.currentBufferSize); - EXPECT_EQ(0, stats.preferredBufferSize); - EXPECT_FALSE(stats.jitterPeaksFound); - EXPECT_EQ(0, stats.currentPacketLossRate); - EXPECT_EQ(0, stats.currentDiscardRate); - EXPECT_EQ(0, stats.currentExpandRate); - EXPECT_EQ(0, stats.currentPreemptiveRate); - EXPECT_EQ(0, stats.currentAccelerateRate); - EXPECT_EQ(-916, stats.clockDriftPPM); // Initial value is slightly off. - EXPECT_EQ(300, stats.maxWaitingTimeMs); - EXPECT_EQ(10, stats.minWaitingTimeMs); - EXPECT_EQ(159, stats.meanWaitingTimeMs); - EXPECT_EQ(160, stats.medianWaitingTimeMs); -} - -TEST_F(AcmNetEqTest, TestZeroLengthWaitingTimesVector) { - // Insert one packet. - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - int i = 0; - InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false, - kPayloadBytes); - // Do not pull out any data. - - ACMNetworkStatistics stats; - ASSERT_EQ(0, neteq_.NetworkStatistics(&stats)); - EXPECT_EQ(0, stats.currentBufferSize); - EXPECT_EQ(0, stats.preferredBufferSize); - EXPECT_FALSE(stats.jitterPeaksFound); - EXPECT_EQ(0, stats.currentPacketLossRate); - EXPECT_EQ(0, stats.currentDiscardRate); - EXPECT_EQ(0, stats.currentExpandRate); - EXPECT_EQ(0, stats.currentPreemptiveRate); - EXPECT_EQ(0, stats.currentAccelerateRate); - EXPECT_EQ(-916, stats.clockDriftPPM); // Initial value is slightly off. - EXPECT_EQ(-1, stats.minWaitingTimeMs); - EXPECT_EQ(-1, stats.maxWaitingTimeMs); - EXPECT_EQ(-1, stats.meanWaitingTimeMs); - EXPECT_EQ(-1, stats.medianWaitingTimeMs); -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc deleted file mode 100644 index 413f3715fc6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc +++ /dev/null @@ -1,319 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_opus.h" - -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_OPUS -#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_OPUS - -ACMOpus::ACMOpus(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - sample_freq_(0), - bitrate_(0), - channels_(1) { - return; -} - -ACMOpus::~ACMOpus() { - return; -} - -int16_t ACMOpus::InternalEncode(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMOpus::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMOpus::InternalInitDecoder(WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMOpus::CreateInstance(void) { - return NULL; -} - -int16_t ACMOpus::InternalCreateEncoder() { - return -1; -} - -void ACMOpus::DestructEncoderSafe() { - return; -} - -int16_t ACMOpus::InternalCreateDecoder() { - return -1; -} - -void ACMOpus::DestructDecoderSafe() { - return; -} - -void ACMOpus::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -int16_t ACMOpus::SetBitRateSafe(const int32_t /*rate*/) { - return -1; -} - -bool ACMOpus::IsTrueStereoCodec() { - return true; -} - -void ACMOpus::SplitStereoPacket(uint8_t* /*payload*/, - int32_t* /*payload_length*/) {} - -#else //===================== Actual Implementation ======================= - -ACMOpus::ACMOpus(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - sample_freq_(32000), // Default sampling frequency. - bitrate_(20000), // Default bit-rate. - channels_(1) { // Default mono - codec_id_ = codec_id; - - // Opus has internal DTX, but we don't use it for now. - has_internal_dtx_ = false; - - if (codec_id_ != ACMCodecDB::kOpus) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Wrong codec id for Opus."); - sample_freq_ = -1; - bitrate_ = -1; - } - return; -} - -ACMOpus::~ACMOpus() { - if (encoder_inst_ptr_ != NULL) { - WebRtcOpus_EncoderFree(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcOpus_DecoderFree(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMOpus::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - // Call Encoder. - *bitstream_len_byte = WebRtcOpus_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], - frame_len_smpl_, - MAX_PAYLOAD_SIZE_BYTE, bitstream); - // Check for error reported from encoder. - if (*bitstream_len_byte < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "InternalEncode: Encode error for Opus"); - *bitstream_len_byte = 0; - return -1; - } - - // Increment the read index. This tells the caller how far - // we have gone forward in reading the audio buffer. - in_audio_ix_read_ += frame_len_smpl_ * channels_; - - return *bitstream_len_byte; -} - -int16_t ACMOpus::DecodeSafe(uint8_t* bitstream, int16_t bitstream_len_byte, - int16_t* audio, int16_t* audio_samples, - int8_t* speech_type) { - return 0; -} - -int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { - int16_t ret; - if (encoder_inst_ptr_ != NULL) { - WebRtcOpus_EncoderFree(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - ret = WebRtcOpus_EncoderCreate(&encoder_inst_ptr_, - codec_params->codec_inst.channels); - // Store number of channels. - channels_ = codec_params->codec_inst.channels; - - if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Encoder creation failed for Opus"); - return ret; - } - ret = WebRtcOpus_SetBitRate(encoder_inst_ptr_, - codec_params->codec_inst.rate); - if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Setting initial bitrate failed for Opus"); - return ret; - } - - // Store bitrate. - bitrate_ = codec_params->codec_inst.rate; - - return 0; -} - -int16_t ACMOpus::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { - if (decoder_inst_ptr_ == NULL) { - if (WebRtcOpus_DecoderCreate(&decoder_inst_ptr_, - codec_params->codec_inst.channels) < 0) { - return -1; - } - } - - // Number of channels in decoder should match the number in |codec_params|. - assert(codec_params->codec_inst.channels == - WebRtcOpus_DecoderChannels(decoder_inst_ptr_)); - - if (WebRtcOpus_DecoderInit(decoder_inst_ptr_) < 0) { - return -1; - } - if (WebRtcOpus_DecoderInitSlave(decoder_inst_ptr_) < 0) { - return -1; - } - return 0; -} - -int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "CodeDef: Decoder uninitialized for Opus"); - return -1; - } - - // Fill up the structure by calling "SET_CODEC_PAR" & "SET_OPUS_FUNCTION." - // Then call NetEQ to add the codec to its database. - // TODO(tlegrand): Decoder is registered in NetEQ as a 32 kHz decoder, which - // is true until we have a full 48 kHz system, and remove the downsampling - // in the Opus decoder wrapper. - SET_CODEC_PAR(codec_def, kDecoderOpus, codec_inst.pltype, - decoder_inst_ptr_, 32000); - - // If this is the master of NetEQ, regular decoder will be added, otherwise - // the slave decoder will be used. - if (is_master_) { - SET_OPUS_FUNCTIONS(codec_def); - } else { - SET_OPUSSLAVE_FUNCTIONS(codec_def); - } - - return 0; -} - -ACMGenericCodec* ACMOpus::CreateInstance(void) { - return NULL; -} - -int16_t ACMOpus::InternalCreateEncoder() { - // Real encoder will be created in InternalInitEncoder. - return 0; -} - -void ACMOpus::DestructEncoderSafe() { - if (encoder_inst_ptr_) { - WebRtcOpus_EncoderFree(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } -} - -int16_t ACMOpus::InternalCreateDecoder() { - // Real decoder will be created in InternalInitDecoder - return 0; -} - -void ACMOpus::DestructDecoderSafe() { - decoder_initialized_ = false; - if (decoder_inst_ptr_) { - WebRtcOpus_DecoderFree(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } -} - -void ACMOpus::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcOpus_EncoderFree(reinterpret_cast<OpusEncInst*>(ptr_inst)); - } - return; -} - -int16_t ACMOpus::SetBitRateSafe(const int32_t rate) { - if (rate < 6000 || rate > 510000) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "SetBitRateSafe: Invalid rate Opus"); - return -1; - } - - bitrate_ = rate; - - // Ask the encoder for the new rate. - if (WebRtcOpus_SetBitRate(encoder_inst_ptr_, bitrate_) >= 0) { - encoder_params_.codec_inst.rate = bitrate_; - return 0; - } - - return -1; -} - -bool ACMOpus::IsTrueStereoCodec() { - return true; -} - -// Copy the stereo packet so that NetEq will insert into both master and slave. -void ACMOpus::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { - // Check for valid inputs. - assert(payload != NULL); - assert(*payload_length > 0); - - // Duplicate the payload. - memcpy(&payload[*payload_length], &payload[0], - sizeof(uint8_t) * (*payload_length)); - // Double the size of the packet. - *payload_length *= 2; -} - -#endif // WEBRTC_CODEC_OPUS - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.h deleted file mode 100644 index 1e586ff41ad..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.h +++ /dev/null @@ -1,78 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ - -#include "webrtc/common_audio/resampler/include/resampler.h" -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -struct WebRtcOpusEncInst; -struct WebRtcOpusDecInst; - -namespace webrtc { - -namespace acm1 { - -class ACMOpus : public ACMGenericCodec { - public: - explicit ACMOpus(int16_t codec_id); - virtual ~ACMOpus(); - - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual int16_t SetBitRateSafe(const int32_t rate) OVERRIDE; - - virtual bool IsTrueStereoCodec() OVERRIDE; - - virtual void SplitStereoPacket(uint8_t* payload, - int32_t* payload_length) OVERRIDE; - - WebRtcOpusEncInst* encoder_inst_ptr_; - WebRtcOpusDecInst* decoder_inst_ptr_; - uint16_t sample_freq_; - uint32_t bitrate_; - int channels_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc deleted file mode 100644 index 6fe12f757ca..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc +++ /dev/null @@ -1,251 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_pcm16b.h" - -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_PCM16 -#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_PCM16 - -ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) { - return; -} - -ACMPCM16B::~ACMPCM16B() { - return; -} - -int16_t ACMPCM16B::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMPCM16B::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMPCM16B::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMPCM16B::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMPCM16B::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMPCM16B::CreateInstance(void) { - return NULL; -} - -int16_t ACMPCM16B::InternalCreateEncoder() { - return -1; -} - -int16_t ACMPCM16B::InternalCreateDecoder() { - return -1; -} - -void ACMPCM16B::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -void ACMPCM16B::DestructEncoderSafe() { - return; -} - -void ACMPCM16B::DestructDecoderSafe() { - return; -} - -void ACMPCM16B::SplitStereoPacket(uint8_t* /*payload*/, - int32_t* /*payload_length*/) { -} - -#else //===================== Actual Implementation ======================= -ACMPCM16B::ACMPCM16B(int16_t codec_id) { - codec_id_ = codec_id; - sampling_freq_hz_ = ACMCodecDB::CodecFreq(codec_id_); -} - -ACMPCM16B::~ACMPCM16B() { - return; -} - -int16_t ACMPCM16B::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - *bitstream_len_byte = WebRtcPcm16b_Encode(&in_audio_[in_audio_ix_read_], - frame_len_smpl_ * num_channels_, - bitstream); - // Increment the read index to tell the caller that how far - // we have gone forward in reading the audio buffer. - in_audio_ix_read_ += frame_len_smpl_ * num_channels_; - return *bitstream_len_byte; -} - -int16_t ACMPCM16B::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMPCM16B::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, PCM has no instance. - return 0; -} - -int16_t ACMPCM16B::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, PCM has no instance. - return 0; -} - -int32_t ACMPCM16B::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - // Fill up the structure by calling "SET_CODEC_PAR" & "SET_PCMU_FUNCTION". - // Then call NetEQ to add the codec to it's database. - if (codec_inst.channels == 1) { - switch (sampling_freq_hz_) { - case 8000: { - SET_CODEC_PAR(codec_def, kDecoderPCM16B, codec_inst.pltype, NULL, 8000); - SET_PCM16B_FUNCTIONS(codec_def); - break; - } - case 16000: { - SET_CODEC_PAR(codec_def, kDecoderPCM16Bwb, codec_inst.pltype, NULL, - 16000); - SET_PCM16B_WB_FUNCTIONS(codec_def); - break; - } - case 32000: { - SET_CODEC_PAR(codec_def, kDecoderPCM16Bswb32kHz, codec_inst.pltype, - NULL, 32000); - SET_PCM16B_SWB32_FUNCTIONS(codec_def); - break; - } - default: { - return -1; - } - } - } else { - switch (sampling_freq_hz_) { - case 8000: { - SET_CODEC_PAR(codec_def, kDecoderPCM16B_2ch, codec_inst.pltype, NULL, - 8000); - SET_PCM16B_FUNCTIONS(codec_def); - break; - } - case 16000: { - SET_CODEC_PAR(codec_def, kDecoderPCM16Bwb_2ch, codec_inst.pltype, - NULL, 16000); - SET_PCM16B_WB_FUNCTIONS(codec_def); - break; - } - case 32000: { - SET_CODEC_PAR(codec_def, kDecoderPCM16Bswb32kHz_2ch, codec_inst.pltype, - NULL, 32000); - SET_PCM16B_SWB32_FUNCTIONS(codec_def); - break; - } - default: { - return -1; - } - } - } - return 0; -} - -ACMGenericCodec* ACMPCM16B::CreateInstance(void) { - return NULL; -} - -int16_t ACMPCM16B::InternalCreateEncoder() { - // PCM has no instance. - return 0; -} - -int16_t ACMPCM16B::InternalCreateDecoder() { - // PCM has no instance. - return 0; -} - -void ACMPCM16B::InternalDestructEncoderInst(void* /* ptr_inst */) { - // PCM has no instance. - return; -} - -void ACMPCM16B::DestructEncoderSafe() { - // PCM has no instance. - encoder_exist_ = false; - encoder_initialized_ = false; - return; -} - -void ACMPCM16B::DestructDecoderSafe() { - // PCM has no instance. - decoder_exist_ = false; - decoder_initialized_ = false; - return; -} - -// Split the stereo packet and place left and right channel after each other -// in the payload vector. -void ACMPCM16B::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { - uint8_t right_byte_msb; - uint8_t right_byte_lsb; - - // Check for valid inputs. - assert(payload != NULL); - assert(*payload_length > 0); - - // Move two bytes representing right channel each loop, and place it at the - // end of the bytestream vector. After looping the data is reordered to: - // l1 l2 l3 l4 ... l(N-1) lN r1 r2 r3 r4 ... r(N-1) r(N), - // where N is the total number of samples. - - for (int i = 0; i < *payload_length / 2; i += 2) { - right_byte_msb = payload[i + 2]; - right_byte_lsb = payload[i + 3]; - memmove(&payload[i + 2], &payload[i + 4], *payload_length - i - 4); - payload[*payload_length - 2] = right_byte_msb; - payload[*payload_length - 1] = right_byte_lsb; - } -} -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.h deleted file mode 100644 index a97589b57a9..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.h +++ /dev/null @@ -1,67 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -namespace webrtc { - -namespace acm1 { - -class ACMPCM16B : public ACMGenericCodec { - public: - explicit ACMPCM16B(int16_t codec_id); - virtual ~ACMPCM16B(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual void SplitStereoPacket(uint8_t* payload, - int32_t* payload_length) OVERRIDE; - - int32_t sampling_freq_hz_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc deleted file mode 100644 index 9e5514a9e9e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_pcma.h" - -#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -// Codec interface - -namespace webrtc { - -namespace acm1 { - -ACMPCMA::ACMPCMA(int16_t codec_id) { - codec_id_ = codec_id; -} - -ACMPCMA::~ACMPCMA() { - return; -} - -int16_t ACMPCMA::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - *bitstream_len_byte = WebRtcG711_EncodeA(NULL, &in_audio_[in_audio_ix_read_], - frame_len_smpl_ * num_channels_, - (int16_t*) bitstream); - // Increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer. - in_audio_ix_read_ += frame_len_smpl_ * num_channels_; - return *bitstream_len_byte; -} - -int16_t ACMPCMA::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMPCMA::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, PCM has no instance. - return 0; -} - -int16_t ACMPCMA::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, PCM has no instance. - return 0; -} - -int32_t ACMPCMA::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_PCMA_FUNCTION." - // Then call NetEQ to add the codec to it's database. - if (codec_inst.channels == 1) { - // Mono mode. - SET_CODEC_PAR(codec_def, kDecoderPCMa, codec_inst.pltype, NULL, 8000); - } else { - // Stereo mode. - SET_CODEC_PAR(codec_def, kDecoderPCMa_2ch, codec_inst.pltype, NULL, 8000); - } - SET_PCMA_FUNCTIONS(codec_def); - return 0; -} - -ACMGenericCodec* ACMPCMA::CreateInstance(void) { - return NULL; -} - -int16_t ACMPCMA::InternalCreateEncoder() { - // PCM has no instance. - return 0; -} - -int16_t ACMPCMA::InternalCreateDecoder() { - // PCM has no instance. - return 0; -} - -void ACMPCMA::InternalDestructEncoderInst(void* /* ptr_inst */) { - // PCM has no instance. - return; -} - -void ACMPCMA::DestructEncoderSafe() { - // PCM has no instance. - return; -} - -void ACMPCMA::DestructDecoderSafe() { - // PCM has no instance. - decoder_initialized_ = false; - decoder_exist_ = false; - return; -} - -// Split the stereo packet and place left and right channel after each other -// in the payload vector. -void ACMPCMA::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { - uint8_t right_byte; - - // Check for valid inputs. - assert(payload != NULL); - assert(*payload_length > 0); - - // Move one bytes representing right channel each loop, and place it at the - // end of the bytestream vector. After looping the data is reordered to: - // l1 l2 l3 l4 ... l(N-1) lN r1 r2 r3 r4 ... r(N-1) r(N), - // where N is the total number of samples. - for (int i = 0; i < *payload_length / 2; i++) { - right_byte = payload[i + 1]; - memmove(&payload[i + 1], &payload[i + 2], *payload_length - i - 2); - payload[*payload_length - 1] = right_byte; - } -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcma.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcma.h deleted file mode 100644 index cb506eaa6e9..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcma.h +++ /dev/null @@ -1,65 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -namespace webrtc { - -namespace acm1 { - -class ACMPCMA : public ACMGenericCodec { - public: - explicit ACMPCMA(int16_t codec_id); - virtual ~ACMPCMA(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual void SplitStereoPacket(uint8_t* payload, - int32_t* payload_length) OVERRIDE; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc deleted file mode 100644 index 6f4eb27aa7a..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc +++ /dev/null @@ -1,136 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_pcmu.h" - -#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -// Codec interface - -namespace webrtc { - -namespace acm1 { - -ACMPCMU::ACMPCMU(int16_t codec_id) { - codec_id_ = codec_id; -} - -ACMPCMU::~ACMPCMU() { - return; -} - -int16_t ACMPCMU::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - *bitstream_len_byte = WebRtcG711_EncodeU(NULL, &in_audio_[in_audio_ix_read_], - frame_len_smpl_ * num_channels_, - (int16_t*)bitstream); - // Increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer. - in_audio_ix_read_ += frame_len_smpl_ * num_channels_; - return *bitstream_len_byte; -} - -int16_t ACMPCMU::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMPCMU::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, PCM has no instance. - return 0; -} - -int16_t ACMPCMU::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, PCM has no instance. - return 0; -} - -int32_t ACMPCMU::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION." - // Then call NetEQ to add the codec to it's database. - if (codec_inst.channels == 1) { - // Mono mode. - SET_CODEC_PAR(codec_def, kDecoderPCMu, codec_inst.pltype, NULL, 8000); - } else { - // Stereo mode. - SET_CODEC_PAR(codec_def, kDecoderPCMu_2ch, codec_inst.pltype, NULL, 8000); - } - SET_PCMU_FUNCTIONS(codec_def); - return 0; -} - -ACMGenericCodec* ACMPCMU::CreateInstance(void) { - return NULL; -} - -int16_t ACMPCMU::InternalCreateEncoder() { - // PCM has no instance. - return 0; -} - -int16_t ACMPCMU::InternalCreateDecoder() { - // PCM has no instance. - return 0; -} - -void ACMPCMU::InternalDestructEncoderInst(void* /* ptr_inst */) { - // PCM has no instance. - return; -} - -void ACMPCMU::DestructEncoderSafe() { - // PCM has no instance. - encoder_exist_ = false; - encoder_initialized_ = false; - return; -} - -void ACMPCMU::DestructDecoderSafe() { - // PCM has no instance. - decoder_initialized_ = false; - decoder_exist_ = false; - return; -} - -// Split the stereo packet and place left and right channel after each other -// in the payload vector. -void ACMPCMU::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) { - uint8_t right_byte; - - // Check for valid inputs. - assert(payload != NULL); - assert(*payload_length > 0); - - // Move one bytes representing right channel each loop, and place it at the - // end of the bytestream vector. After looping the data is reordered to: - // l1 l2 l3 l4 ... l(N-1) lN r1 r2 r3 r4 ... r(N-1) r(N), - // where N is the total number of samples. - for (int i = 0; i < *payload_length / 2; i++) { - right_byte = payload[i + 1]; - memmove(&payload[i + 1], &payload[i + 2], *payload_length - i - 2); - payload[*payload_length - 1] = right_byte; - } -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.h deleted file mode 100644 index ea401d59c96..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.h +++ /dev/null @@ -1,65 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -namespace webrtc { - -namespace acm1 { - -class ACMPCMU : public ACMGenericCodec { - public: - explicit ACMPCMU(int16_t codec_id); - virtual ~ACMPCMU(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; - - virtual void SplitStereoPacket(uint8_t* payload, - int32_t* payload_length) OVERRIDE; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_red.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_red.cc deleted file mode 100644 index 0d8134c171a..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_red.cc +++ /dev/null @@ -1,108 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_red.h" - -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -namespace acm1 { - -ACMRED::ACMRED(int16_t codec_id) { - codec_id_ = codec_id; -} - -ACMRED::~ACMRED() { - return; -} - -int16_t ACMRED::InternalEncode(uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - // RED is never used as an encoder - // RED has no instance - return 0; -} - -int16_t ACMRED::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMRED::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, - // RED has no instance - return 0; -} - -int16_t ACMRED::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - // This codec does not need initialization, - // RED has no instance - return 0; -} - -int32_t ACMRED::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - // Todo: - // log error - return -1; - } - - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION." - // Then call NetEQ to add the codec to it's - // database. - SET_CODEC_PAR((codec_def), kDecoderRED, codec_inst.pltype, NULL, 8000); - SET_RED_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMRED::CreateInstance(void) { - return NULL; -} - -int16_t ACMRED::InternalCreateEncoder() { - // RED has no instance - return 0; -} - -int16_t ACMRED::InternalCreateDecoder() { - // RED has no instance - return 0; -} - -void ACMRED::InternalDestructEncoderInst(void* /* ptr_inst */) { - // RED has no instance - return; -} - -void ACMRED::DestructEncoderSafe() { - // RED has no instance - return; -} - -void ACMRED::DestructDecoderSafe() { - // RED has no instance - return; -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_red.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_red.h deleted file mode 100644 index ede18b5218e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_red.h +++ /dev/null @@ -1,62 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -namespace webrtc { - -namespace acm1 { - -class ACMRED : public ACMGenericCodec { - public: - explicit ACMRED(int16_t codec_id); - virtual ~ACMRED(); - - // for FEC - virtual ACMGenericCodec* CreateInstance(void) OVERRIDE; - - virtual int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) OVERRIDE; - - virtual int16_t InternalInitEncoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - virtual int16_t InternalInitDecoder( - WebRtcACMCodecParams* codec_params) OVERRIDE; - - protected: - virtual int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type) OVERRIDE; - - virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) OVERRIDE; - - virtual void DestructEncoderSafe() OVERRIDE; - - virtual void DestructDecoderSafe() OVERRIDE; - - virtual int16_t InternalCreateEncoder() OVERRIDE; - - virtual int16_t InternalCreateDecoder() OVERRIDE; - - virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc deleted file mode 100644 index 50ddab1d8b9..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc +++ /dev/null @@ -1,63 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_resampler.h" - -#include <string.h> - -#include "webrtc/common_audio/resampler/include/push_resampler.h" -#include "webrtc/system_wrappers/interface/logging.h" - -namespace webrtc { - -namespace acm1 { - -ACMResampler::ACMResampler() { -} - -ACMResampler::~ACMResampler() { -} - -int16_t ACMResampler::Resample10Msec(const int16_t* in_audio, - int32_t in_freq_hz, - int16_t* out_audio, - int32_t out_freq_hz, - uint8_t num_audio_channels) { - if (in_freq_hz == out_freq_hz) { - size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100); - memcpy(out_audio, in_audio, length * sizeof(int16_t)); - return static_cast<int16_t>(in_freq_hz / 100); - } - - // |max_length| is the maximum number of samples for 10ms at 48kHz. - // TODO(turajs): is this actually the capacity of the |out_audio| buffer? - int max_length = 480 * num_audio_channels; - int in_length = in_freq_hz / 100 * num_audio_channels; - - if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz, - num_audio_channels) != 0) { - LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz, - num_audio_channels); - return -1; - } - - int out_length = resampler_.Resample(in_audio, in_length, out_audio, - max_length); - if (out_length == -1) { - LOG_FERR4(LS_ERROR, Resample, in_audio, in_length, out_audio, max_length); - return -1; - } - - return out_length / num_audio_channels; -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.h deleted file mode 100644 index b50e722c443..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ - -#include "webrtc/common_audio/resampler/include/push_resampler.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -namespace acm1 { - -class ACMResampler { - public: - ACMResampler(); - ~ACMResampler(); - - int16_t Resample10Msec(const int16_t* in_audio, - const int32_t in_freq_hz, - int16_t* out_audio, - const int32_t out_freq_hz, - uint8_t num_audio_channels); - - private: - PushResampler resampler_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc deleted file mode 100644 index 1567929d860..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc +++ /dev/null @@ -1,471 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/acm_speex.h" - -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/system_wrappers/interface/trace.h" - -#ifdef WEBRTC_CODEC_SPEEX -// NOTE! Speex is not included in the open-source package. Modify this file or -// your codec API to match the function calls and names of used Speex API file. -#include "speex_interface.h" -#endif - -namespace webrtc { - -namespace acm1 { - -#ifndef WEBRTC_CODEC_SPEEX -ACMSPEEX::ACMSPEEX(int16_t /* codec_id */) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL), - compl_mode_(0), - vbr_enabled_(false), - encoding_rate_(-1), - sampling_frequency_(-1), - samples_in_20ms_audio_(-1) { - return; -} - -ACMSPEEX::~ACMSPEEX() { - return; -} - -int16_t ACMSPEEX::InternalEncode( - uint8_t* /* bitstream */, - int16_t* /* bitstream_len_byte */) { - return -1; -} - -int16_t ACMSPEEX::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return -1; -} - -int16_t ACMSPEEX::EnableDTX() { - return -1; -} - -int16_t ACMSPEEX::DisableDTX() { - return -1; -} - -int16_t ACMSPEEX::InternalInitEncoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int16_t ACMSPEEX::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - return -1; -} - -int32_t ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, - const CodecInst& /* codec_inst */) { - return -1; -} - -ACMGenericCodec* ACMSPEEX::CreateInstance(void) { - return NULL; -} - -int16_t ACMSPEEX::InternalCreateEncoder() { - return -1; -} - -void ACMSPEEX::DestructEncoderSafe() { - return; -} - -int16_t ACMSPEEX::InternalCreateDecoder() { - return -1; -} - -void ACMSPEEX::DestructDecoderSafe() { - return; -} - -int16_t ACMSPEEX::SetBitRateSafe(const int32_t /* rate */) { - return -1; -} - -void ACMSPEEX::InternalDestructEncoderInst(void* /* ptr_inst */) { - return; -} - -#ifdef UNUSEDSPEEX -int16_t ACMSPEEX::EnableVBR() { - return -1; -} - -int16_t ACMSPEEX::DisableVBR() { - return -1; -} - -int16_t ACMSPEEX::SetComplMode(int16_t mode) { - return -1; -} -#endif - -#else //===================== Actual Implementation ======================= - -ACMSPEEX::ACMSPEEX(int16_t codec_id) - : encoder_inst_ptr_(NULL), - decoder_inst_ptr_(NULL) { - codec_id_ = codec_id; - - // Set sampling frequency, frame size and rate Speex - if (codec_id_ == ACMCodecDB::kSPEEX8) { - sampling_frequency_ = 8000; - samples_in_20ms_audio_ = 160; - encoding_rate_ = 11000; - } else if (codec_id_ == ACMCodecDB::kSPEEX16) { - sampling_frequency_ = 16000; - samples_in_20ms_audio_ = 320; - encoding_rate_ = 22000; - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Wrong codec id for Speex."); - - sampling_frequency_ = -1; - samples_in_20ms_audio_ = -1; - encoding_rate_ = -1; - } - - has_internal_dtx_ = true; - dtx_enabled_ = false; - vbr_enabled_ = false; - compl_mode_ = 3; // default complexity value - - return; -} - -ACMSPEEX::~ACMSPEEX() { - if (encoder_inst_ptr_ != NULL) { - WebRtcSpeex_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - if (decoder_inst_ptr_ != NULL) { - WebRtcSpeex_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - return; -} - -int16_t ACMSPEEX::InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte) { - int16_t status; - int16_t num_encoded_samples = 0; - int16_t n = 0; - - while (num_encoded_samples < frame_len_smpl_) { - status = WebRtcSpeex_Encode(encoder_inst_ptr_, - &in_audio_[in_audio_ix_read_], encoding_rate_); - - // increment the read index this tell the caller that how far - // we have gone forward in reading the audio buffer - in_audio_ix_read_ += samples_in_20ms_audio_; - num_encoded_samples += samples_in_20ms_audio_; - - if (status < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Error in Speex encoder"); - return status; - } - - // Update VAD, if internal DTX is used - if (has_internal_dtx_ && dtx_enabled_) { - vad_label_[n++] = status; - vad_label_[n++] = status; - } - - if (status == 0) { - // This frame is detected as inactive. We need send whatever - // encoded so far. - *bitstream_len_byte = WebRtcSpeex_GetBitstream(encoder_inst_ptr_, - (int16_t*)bitstream); - return *bitstream_len_byte; - } - } - - *bitstream_len_byte = WebRtcSpeex_GetBitstream(encoder_inst_ptr_, - (int16_t*)bitstream); - return *bitstream_len_byte; -} - -int16_t ACMSPEEX::DecodeSafe(uint8_t* /* bitstream */, - int16_t /* bitstream_len_byte */, - int16_t* /* audio */, - int16_t* /* audio_samples */, - int8_t* /* speech_type */) { - return 0; -} - -int16_t ACMSPEEX::EnableDTX() { - if (dtx_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // enable DTX - if (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, (vbr_enabled_ ? 1 : 0), - compl_mode_, 1) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Cannot enable DTX for Speex"); - return -1; - } - dtx_enabled_ = true; - return 0; - } else { - return -1; - } - - return 0; -} - -int16_t ACMSPEEX::DisableDTX() { - if (!dtx_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // disable DTX - if (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, (vbr_enabled_ ? 1 : 0), - compl_mode_, 0) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Cannot disable DTX for Speex"); - return -1; - } - dtx_enabled_ = false; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } - - return 0; -} - -int16_t ACMSPEEX::InternalInitEncoder( - WebRtcACMCodecParams* codec_params) { - // sanity check - if (encoder_inst_ptr_ == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Cannot initialize Speex encoder, instance does not exist"); - return -1; - } - - int16_t status = SetBitRateSafe((codec_params->codecInstant).rate); - status += - (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, vbr_enabled_, compl_mode_, - ((codec_params->enable_dtx) ? 1 : 0)) < 0) ? - -1 : 0; - - if (status >= 0) { - return 0; - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Error in initialization of Speex encoder"); - return -1; - } -} - -int16_t ACMSPEEX::InternalInitDecoder( - WebRtcACMCodecParams* /* codec_params */) { - int16_t status; - - // sanity check - if (decoder_inst_ptr_ == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Cannot initialize Speex decoder, instance does not exist"); - return -1; - } - status = ((WebRtcSpeex_DecoderInit(decoder_inst_ptr_) < 0) ? -1 : 0); - - if (status >= 0) { - return 0; - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Error in initialization of Speex decoder"); - return -1; - } -} - -int32_t ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst) { - if (!decoder_initialized_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Error, Speex decoder is not initialized"); - return -1; - } - - // Fill up the structure by calling - // "SET_CODEC_PAR" & "SET_SPEEX_FUNCTION." - // Then call NetEQ to add the codec to its - // database. - - switch (sampling_frequency_) { - case 8000: { - SET_CODEC_PAR((codec_def), kDecoderSPEEX_8, codec_inst.pltype, - decoder_inst_ptr_, 8000); - break; - } - case 16000: { - SET_CODEC_PAR((codec_def), kDecoderSPEEX_16, codec_inst.pltype, - decoder_inst_ptr_, 16000); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Unsupported sampling frequency for Speex"); - - return -1; - } - } - - SET_SPEEX_FUNCTIONS((codec_def)); - return 0; -} - -ACMGenericCodec* ACMSPEEX::CreateInstance(void) { - return NULL; -} - -int16_t ACMSPEEX::InternalCreateEncoder() { - return WebRtcSpeex_CreateEnc(&encoder_inst_ptr_, sampling_frequency_); -} - -void ACMSPEEX::DestructEncoderSafe() { - if (encoder_inst_ptr_ != NULL) { - WebRtcSpeex_FreeEnc(encoder_inst_ptr_); - encoder_inst_ptr_ = NULL; - } - // there is no encoder set the following - encoder_exist_ = false; - encoder_initialized_ = false; - encoding_rate_ = 0; -} - -int16_t ACMSPEEX::InternalCreateDecoder() { - return WebRtcSpeex_CreateDec(&decoder_inst_ptr_, sampling_frequency_, 1); -} - -void ACMSPEEX::DestructDecoderSafe() { - if (decoder_inst_ptr_ != NULL) { - WebRtcSpeex_FreeDec(decoder_inst_ptr_); - decoder_inst_ptr_ = NULL; - } - // there is no encoder instance set the followings - decoder_exist_ = false; - decoder_initialized_ = false; -} - -int16_t ACMSPEEX::SetBitRateSafe(const int32_t rate) { - // Check if changed rate - if (rate == encoding_rate_) { - return 0; - } else if (rate > 2000) { - encoding_rate_ = rate; - encoder_params_.codecInstant.rate = rate; - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Unsupported encoding rate for Speex"); - - return -1; - } - - return 0; -} - -void ACMSPEEX::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - WebRtcSpeex_FreeEnc((SPEEX_encinst_t_*) ptr_inst); - } - return; -} - -#ifdef UNUSEDSPEEX - -// This API is currently not in use. If requested to be able to enable/disable -// VBR an ACM API need to be added. -int16_t ACMSPEEX::EnableVBR() { - if (vbr_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // enable Variable Bit Rate (VBR) - if (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, 1, compl_mode_, - (dtx_enabled_ ? 1 : 0)) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Cannot enable VBR mode for Speex"); - - return -1; - } - vbr_enabled_ = true; - return 0; - } else { - return -1; - } -} - -// This API is currently not in use. If requested to be able to enable/disable -// VBR an ACM API need to be added. -int16_t ACMSPEEX::DisableVBR() { - if (!vbr_enabled_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // disable DTX - if (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, 0, compl_mode_, - (dtx_enabled_ ? 1 : 0)) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Cannot disable DTX for Speex"); - - return -1; - } - vbr_enabled_ = false; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } -} - -// This API is currently not in use. If requested to be able to set complexity -// an ACM API need to be added. -int16_t ACMSPEEX::SetComplMode(int16_t mode) { - // Check if new mode - if (mode == compl_mode_) { - return 0; - } else if (encoder_exist_) { // check if encoder exist - // Set new mode - if (WebRtcSpeex_EncoderInit(encoder_inst_ptr_, 0, mode, - (dtx_enabled_ ? 1 : 0)) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, - "Error in complexity mode for Speex"); - return -1; - } - compl_mode_ = mode; - return 0; - } else { - // encoder doesn't exists, therefore disabling is harmless - return 0; - } -} - -#endif - -#endif - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_speex.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_speex.h deleted file mode 100644 index 762aea8d9c2..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_speex.h +++ /dev/null @@ -1,86 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ - -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" - -// forward declaration -struct SPEEX_encinst_t_; -struct SPEEX_decinst_t_; - -namespace webrtc { - -namespace acm1 { - -class ACMSPEEX : public ACMGenericCodec { - public: - explicit ACMSPEEX(int16_t codec_id); - ~ACMSPEEX(); - - // for FEC - ACMGenericCodec* CreateInstance(void); - - int16_t InternalEncode(uint8_t* bitstream, - int16_t* bitstream_len_byte); - - int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); - - int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params); - - protected: - int16_t DecodeSafe(uint8_t* bitstream, - int16_t bitstream_len_byte, - int16_t* audio, - int16_t* audio_samples, - int8_t* speech_type); - - int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, - const CodecInst& codec_inst); - - void DestructEncoderSafe(); - - void DestructDecoderSafe(); - - int16_t InternalCreateEncoder(); - - int16_t InternalCreateDecoder(); - - void InternalDestructEncoderInst(void* ptr_inst); - - int16_t SetBitRateSafe(const int32_t rate); - - int16_t EnableDTX(); - - int16_t DisableDTX(); - -#ifdef UNUSEDSPEEX - int16_t EnableVBR(); - - int16_t DisableVBR(); - - int16_t SetComplMode(int16_t mode); -#endif - - SPEEX_encinst_t_* encoder_inst_ptr_; - SPEEX_decinst_t_* decoder_inst_ptr_; - int16_t compl_mode_; - bool vbr_enabled_; - int32_t encoding_rate_; - int16_t sampling_frequency_; - uint16_t samples_in_20ms_audio_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi b/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi deleted file mode 100644 index a0389b03ef8..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi +++ /dev/null @@ -1,153 +0,0 @@ -# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -{ - 'variables': { - 'audio_coding_dependencies': [ - 'CNG', - 'G711', - 'G722', - 'iLBC', - 'iSAC', - 'iSACFix', - 'PCM16B', - 'NetEq', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', - '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', - ], - 'audio_coding_defines': [], - 'conditions': [ - ['include_opus==1', { - 'audio_coding_dependencies': ['webrtc_opus',], - 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',], - }], - ], - }, - 'targets': [ - { - 'target_name': 'audio_coding_module', - 'type': 'static_library', - 'defines': [ - '<@(audio_coding_defines)', - ], - 'dependencies': [ - '<@(audio_coding_dependencies)', - 'acm2', - ], - 'include_dirs': [ - '../interface', - '../../../interface', - '<(webrtc_root)', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - '../interface', - '../../../interface', - '<(webrtc_root)', - ], - }, - 'sources': [ - '../interface/audio_coding_module.h', - '../interface/audio_coding_module_typedefs.h', - 'acm_amr.cc', - 'acm_amr.h', - 'acm_amrwb.cc', - 'acm_amrwb.h', - 'acm_celt.cc', - 'acm_celt.h', - 'acm_cng.cc', - 'acm_cng.h', - 'acm_codec_database.cc', - 'acm_codec_database.h', - 'acm_dtmf_detection.cc', - 'acm_dtmf_detection.h', - 'acm_dtmf_playout.cc', - 'acm_dtmf_playout.h', - 'acm_g722.cc', - 'acm_g722.h', - 'acm_g7221.cc', - 'acm_g7221.h', - 'acm_g7221c.cc', - 'acm_g7221c.h', - 'acm_g729.cc', - 'acm_g729.h', - 'acm_g7291.cc', - 'acm_g7291.h', - 'acm_generic_codec.cc', - 'acm_generic_codec.h', - 'acm_gsmfr.cc', - 'acm_gsmfr.h', - 'acm_ilbc.cc', - 'acm_ilbc.h', - 'acm_isac.cc', - 'acm_isac.h', - 'acm_isac_macros.h', - 'acm_neteq.cc', - 'acm_neteq.h', - 'acm_opus.cc', - 'acm_opus.h', - 'acm_speex.cc', - 'acm_speex.h', - 'acm_pcm16b.cc', - 'acm_pcm16b.h', - 'acm_pcma.cc', - 'acm_pcma.h', - 'acm_pcmu.cc', - 'acm_pcmu.h', - 'acm_red.cc', - 'acm_red.h', - 'acm_resampler.cc', - 'acm_resampler.h', - 'audio_coding_module_impl.cc', - 'audio_coding_module_impl.h', - ], - }, - ], - 'conditions': [ - ['include_tests==1', { - 'targets': [ - { - 'target_name': 'delay_test', - 'type': 'executable', - 'dependencies': [ - 'audio_coding_module', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/test/test.gyp:test_support', - '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', - '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', - ], - 'sources': [ - '../test/delay_test.cc', - '../test/Channel.cc', - '../test/PCMFile.cc', - '../test/utility.cc', - ], - }, # delay_test - { - 'target_name': 'insert_packet_with_timing', - 'type': 'executable', - 'dependencies': [ - 'audio_coding_module', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/test/test.gyp:test_support', - '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', - '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', - ], - 'sources': [ - '../test/insert_packet_with_timing.cc', - '../test/Channel.cc', - '../test/PCMFile.cc', - ], - }, # delay_test - ], - }], - ], - 'includes': [ - '../acm2/audio_coding_module.gypi', - ], -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc deleted file mode 100644 index 556f530ecfd..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc +++ /dev/null @@ -1,3048 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h" - -#include <assert.h> -#include <stdlib.h> - -#include <algorithm> // For std::max. - -#include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" -#include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h" -#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" -#include "webrtc/modules/audio_coding/main/source/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/acm2/nack.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/logging.h" -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/tick_util.h" -#include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/system_wrappers/interface/trace_event.h" - -namespace webrtc { - -namespace acm1 { - -enum { - kACMToneEnd = 999 -}; - -// Maximum number of bytes in one packet (PCM16B, 20 ms packets, stereo). -enum { - kMaxPacketSize = 2560 -}; - -// Maximum number of payloads that can be packed in one RED payload. For -// regular FEC, we only pack two payloads. In case of dual-streaming, in worst -// case we might pack 3 payloads in one RED payload. -enum { - kNumFecFragmentationVectors = 2, - kMaxNumFragmentationVectors = 3 -}; - -static const uint32_t kMaskTimestamp = 0x03ffffff; -static const int kDefaultTimestampDiff = 960; // 20 ms @ 48 kHz. - -// If packet N is arrived all packets prior to N - |kNackThresholdPackets| which -// are not received are considered as lost, and appear in NACK list. -static const int kNackThresholdPackets = 2; - -namespace { - -bool IsCodecRED(const CodecInst* codec) { - return (STR_CASE_CMP(codec->plname, "RED") == 0); -} - -bool IsCodecRED(int index) { - return (IsCodecRED(&ACMCodecDB::database_[index])); -} - -bool IsCodecCN(const CodecInst* codec) { - return (STR_CASE_CMP(codec->plname, "CN") == 0); -} - -bool IsCodecCN(int index) { - return (IsCodecCN(&ACMCodecDB::database_[index])); -} - -// Stereo-to-mono can be used as in-place. -int DownMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) { - if (length_out_buff < frame.samples_per_channel_) { - return -1; - } - for (int n = 0; n < frame.samples_per_channel_; ++n) - out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1; - return 0; -} - -// Mono-to-stereo can be used as in-place. -int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) { - if (length_out_buff < frame.samples_per_channel_) { - return -1; - } - for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) { - out_buff[2 * n + 1] = frame.data_[n]; - out_buff[2 * n] = frame.data_[n]; - } - return 0; -} - -// Return 1 if timestamp t1 is less than timestamp t2, while compensating for -// wrap-around. -int TimestampLessThan(uint32_t t1, uint32_t t2) { - uint32_t kHalfFullRange = static_cast<uint32_t>(0xFFFFFFFF) / 2; - if (t1 == t2) { - return 0; - } else if (t1 < t2) { - if (t2 - t1 < kHalfFullRange) - return 1; - return 0; - } else { - if (t1 - t2 < kHalfFullRange) - return 0; - return 1; - } -} - -} // namespace - -AudioCodingModuleImpl::AudioCodingModuleImpl(const int32_t id, Clock* clock) - : packetization_callback_(NULL), - id_(id), - last_timestamp_(0xD87F3F9F), - last_in_timestamp_(0xD87F3F9F), - send_codec_inst_(), - cng_nb_pltype_(255), - cng_wb_pltype_(255), - cng_swb_pltype_(255), - cng_fb_pltype_(255), - red_pltype_(255), - vad_enabled_(false), - dtx_enabled_(false), - vad_mode_(VADNormal), - stereo_receive_registered_(false), - stereo_send_(false), - prev_received_channel_(0), - expected_channels_(1), - current_send_codec_idx_(-1), - current_receive_codec_idx_(-1), - send_codec_registered_(false), - acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vad_callback_(NULL), - last_recv_audio_codec_pltype_(255), - is_first_red_(true), - fec_enabled_(false), - last_fec_timestamp_(0), - receive_red_pltype_(255), - previous_pltype_(255), - dummy_rtp_header_(NULL), - recv_pl_frame_size_smpls_(0), - receiver_initialized_(false), - dtmf_detector_(NULL), - dtmf_callback_(NULL), - last_detected_tone_(kACMToneEnd), - callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - secondary_send_codec_inst_(), - initial_delay_ms_(0), - num_packets_accumulated_(0), - num_bytes_accumulated_(0), - accumulated_audio_ms_(0), - first_payload_received_(false), - last_incoming_send_timestamp_(0), - track_neteq_buffer_(false), - playout_ts_(0), - av_sync_(false), - last_timestamp_diff_(kDefaultTimestampDiff), - last_sequence_number_(0), - last_ssrc_(0), - last_packet_was_sync_(false), - clock_(clock), - nack_(), - nack_enabled_(false) { - - // Nullify send codec memory, set payload type and set codec name to - // invalid values. - const char no_name[] = "noCodecRegistered"; - strncpy(send_codec_inst_.plname, no_name, RTP_PAYLOAD_NAME_SIZE - 1); - send_codec_inst_.pltype = -1; - - strncpy(secondary_send_codec_inst_.plname, no_name, - RTP_PAYLOAD_NAME_SIZE - 1); - secondary_send_codec_inst_.pltype = -1; - - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - codecs_[i] = NULL; - registered_pltypes_[i] = -1; - stereo_receive_[i] = false; - slave_codecs_[i] = NULL; - mirror_codec_idx_[i] = -1; - } - - neteq_.set_id(id_); - - // Allocate memory for RED. - red_buffer_ = new uint8_t[MAX_PAYLOAD_SIZE_BYTE]; - - // TODO(turajs): This might not be exactly how this class is supposed to work. - // The external usage might be that |fragmentationVectorSize| has to match - // the allocated space for the member-arrays, while here, we allocate - // according to the maximum number of fragmentations and change - // |fragmentationVectorSize| on-the-fly based on actual number of - // fragmentations. However, due to copying to local variable before calling - // SendData, the RTP module receives a "valid" fragmentation, where allocated - // space matches |fragmentationVectorSize|, therefore, this should not cause - // any problem. A better approach is not using RTPFragmentationHeader as - // member variable, instead, use an ACM-specific structure to hold RED-related - // data. See module_common_type.h for the definition of - // RTPFragmentationHeader. - fragmentation_.VerifyAndAllocateFragmentationHeader( - kMaxNumFragmentationVectors); - - // Register the default payload type for RED and for CNG at sampling rates of - // 8, 16, 32 and 48 kHz. - for (int i = (ACMCodecDB::kNumCodecs - 1); i >= 0; i--) { - if (IsCodecRED(i)) { - red_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (IsCodecCN(i)) { - if (ACMCodecDB::database_[i].plfreq == 8000) { - cng_nb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (ACMCodecDB::database_[i].plfreq == 16000) { - cng_wb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (ACMCodecDB::database_[i].plfreq == 32000) { - cng_swb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (ACMCodecDB::database_[i].plfreq == 48000) { - cng_fb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } - } - } - - if (InitializeReceiverSafe() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot initialize receiver"); - } - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created"); -} - -AudioCodingModuleImpl::~AudioCodingModuleImpl() { - { - CriticalSectionScoped lock(acm_crit_sect_); - current_send_codec_idx_ = -1; - - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - if (codecs_[i] != NULL) { - // True stereo codecs share the same memory for master and - // slave, so slave codec need to be nullified here, since the - // memory will be deleted. - if (slave_codecs_[i] == codecs_[i]) { - slave_codecs_[i] = NULL; - } - - // Mirror index holds the address of the codec memory. - assert(mirror_codec_idx_[i] > -1); - if (codecs_[mirror_codec_idx_[i]] != NULL) { - delete codecs_[mirror_codec_idx_[i]]; - codecs_[mirror_codec_idx_[i]] = NULL; - } - - codecs_[i] = NULL; - } - - if (slave_codecs_[i] != NULL) { - // Delete memory for stereo usage of mono codecs. - assert(mirror_codec_idx_[i] > -1); - if (slave_codecs_[mirror_codec_idx_[i]] != NULL) { - delete slave_codecs_[mirror_codec_idx_[i]]; - slave_codecs_[mirror_codec_idx_[i]] = NULL; - } - slave_codecs_[i] = NULL; - } - } - - if (dtmf_detector_ != NULL) { - delete dtmf_detector_; - dtmf_detector_ = NULL; - } - if (dummy_rtp_header_ != NULL) { - delete dummy_rtp_header_; - dummy_rtp_header_ = NULL; - } - if (red_buffer_ != NULL) { - delete[] red_buffer_; - red_buffer_ = NULL; - } - } - - delete callback_crit_sect_; - callback_crit_sect_ = NULL; - - delete acm_crit_sect_; - acm_crit_sect_ = NULL; - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, - "Destroyed"); -} - -int32_t AudioCodingModuleImpl::ChangeUniqueId(const int32_t id) { - { - CriticalSectionScoped lock(acm_crit_sect_); - id_ = id; - - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - if (codecs_[i] != NULL) { - codecs_[i]->SetUniqueID(id); - } - } - } - - neteq_.set_id(id_); - return 0; -} - -// Returns the number of milliseconds until the module want a -// worker thread to call Process. -int32_t AudioCodingModuleImpl::TimeUntilNextProcess() { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("TimeUntilNextProcess")) { - return -1; - } - return codecs_[current_send_codec_idx_]->SamplesLeftToEncode() / - (send_codec_inst_.plfreq / 1000); -} - -int32_t AudioCodingModuleImpl::Process() { - bool dual_stream; - { - CriticalSectionScoped lock(acm_crit_sect_); - dual_stream = (secondary_encoder_.get() != NULL); - } - if (dual_stream) { - return ProcessDualStream(); - } - return ProcessSingleStream(); -} - -int AudioCodingModuleImpl::EncodeFragmentation(int fragmentation_index, - int payload_type, - uint32_t current_timestamp, - ACMGenericCodec* encoder, - uint8_t* stream) { - int16_t len_bytes = MAX_PAYLOAD_SIZE_BYTE; - uint32_t rtp_timestamp; - WebRtcACMEncodingType encoding_type; - if (encoder->Encode(stream, &len_bytes, &rtp_timestamp, &encoding_type) < 0) { - return -1; - } - assert(encoding_type == kActiveNormalEncoded); - assert(len_bytes > 0); - - fragmentation_.fragmentationLength[fragmentation_index] = len_bytes; - fragmentation_.fragmentationPlType[fragmentation_index] = payload_type; - fragmentation_.fragmentationTimeDiff[fragmentation_index] = - static_cast<uint16_t>(current_timestamp - rtp_timestamp); - fragmentation_.fragmentationVectorSize++; - return len_bytes; -} - -// Primary payloads are sent immediately, whereas a single secondary payload is -// buffered to be combined with "the next payload." -// Normally "the next payload" would be a primary payload. In case two -// consecutive secondary payloads are generated with no primary payload in -// between, then two secondary payloads are packed in one RED. -int AudioCodingModuleImpl::ProcessDualStream() { - uint8_t stream[kMaxNumFragmentationVectors * MAX_PAYLOAD_SIZE_BYTE]; - uint32_t current_timestamp; - int16_t length_bytes = 0; - RTPFragmentationHeader my_fragmentation; - - uint8_t my_red_payload_type; - - { - CriticalSectionScoped lock(acm_crit_sect_); - // Check if there is an encoder before. - if (!HaveValidEncoder("ProcessDualStream") || - secondary_encoder_.get() == NULL) { - return -1; - } - ACMGenericCodec* primary_encoder = codecs_[current_send_codec_idx_]; - // If primary encoder has a full frame of audio to generate payload. - bool primary_ready_to_encode = primary_encoder->HasFrameToEncode(); - // If the secondary encoder has a frame of audio to generate a payload. - bool secondary_ready_to_encode = secondary_encoder_->HasFrameToEncode(); - - if (!primary_ready_to_encode && !secondary_ready_to_encode) { - // Nothing to send. - return 0; - } - int len_bytes_previous_secondary = static_cast<int>( - fragmentation_.fragmentationLength[2]); - assert(len_bytes_previous_secondary <= MAX_PAYLOAD_SIZE_BYTE); - bool has_previous_payload = len_bytes_previous_secondary > 0; - - uint32_t primary_timestamp = primary_encoder->EarliestTimestamp(); - uint32_t secondary_timestamp = secondary_encoder_->EarliestTimestamp(); - - if (!has_previous_payload && !primary_ready_to_encode && - secondary_ready_to_encode) { - // Secondary payload will be the ONLY bit-stream. Encode by secondary - // encoder, store the payload, and return. No packet is sent. - int16_t len_bytes = MAX_PAYLOAD_SIZE_BYTE; - WebRtcACMEncodingType encoding_type; - if (secondary_encoder_->Encode(red_buffer_, &len_bytes, - &last_fec_timestamp_, - &encoding_type) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessDual(): Encoding of secondary encoder Failed"); - return -1; - } - assert(len_bytes > 0); - assert(encoding_type == kActiveNormalEncoded); - assert(len_bytes <= MAX_PAYLOAD_SIZE_BYTE); - fragmentation_.fragmentationLength[2] = len_bytes; - return 0; - } - - // Initialize with invalid but different values, so later can have sanity - // check if they are different. - int index_primary = -1; - int index_secondary = -2; - int index_previous_secondary = -3; - - if (primary_ready_to_encode) { - index_primary = secondary_ready_to_encode ? - TimestampLessThan(primary_timestamp, secondary_timestamp) : 0; - index_primary += has_previous_payload ? - TimestampLessThan(primary_timestamp, last_fec_timestamp_) : 0; - } - - if (secondary_ready_to_encode) { - // Timestamp of secondary payload can only be less than primary payload, - // but is always larger than the timestamp of previous secondary payload. - index_secondary = primary_ready_to_encode ? - (1 - TimestampLessThan(primary_timestamp, secondary_timestamp)) : 0; - } - - if (has_previous_payload) { - index_previous_secondary = primary_ready_to_encode ? - (1 - TimestampLessThan(primary_timestamp, last_fec_timestamp_)) : 0; - // If secondary is ready it always have a timestamp larger than previous - // secondary. So the index is either 0 or 1. - index_previous_secondary += secondary_ready_to_encode ? 1 : 0; - } - - // Indices must not be equal. - assert(index_primary != index_secondary); - assert(index_primary != index_previous_secondary); - assert(index_secondary != index_previous_secondary); - - // One of the payloads has to be at position zero. - assert(index_primary == 0 || index_secondary == 0 || - index_previous_secondary == 0); - - // Timestamp of the RED payload. - if (index_primary == 0) { - current_timestamp = primary_timestamp; - } else if (index_secondary == 0) { - current_timestamp = secondary_timestamp; - } else { - current_timestamp = last_fec_timestamp_; - } - - fragmentation_.fragmentationVectorSize = 0; - if (has_previous_payload) { - assert(index_previous_secondary >= 0 && - index_previous_secondary < kMaxNumFragmentationVectors); - assert(len_bytes_previous_secondary <= MAX_PAYLOAD_SIZE_BYTE); - memcpy(&stream[index_previous_secondary * MAX_PAYLOAD_SIZE_BYTE], - red_buffer_, sizeof(stream[0]) * len_bytes_previous_secondary); - fragmentation_.fragmentationLength[index_previous_secondary] = - len_bytes_previous_secondary; - fragmentation_.fragmentationPlType[index_previous_secondary] = - secondary_send_codec_inst_.pltype; - fragmentation_.fragmentationTimeDiff[index_previous_secondary] = - static_cast<uint16_t>(current_timestamp - last_fec_timestamp_); - fragmentation_.fragmentationVectorSize++; - } - - if (primary_ready_to_encode) { - assert(index_primary >= 0 && index_primary < kMaxNumFragmentationVectors); - int i = index_primary * MAX_PAYLOAD_SIZE_BYTE; - if (EncodeFragmentation(index_primary, send_codec_inst_.pltype, - current_timestamp, primary_encoder, - &stream[i]) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessDualStream(): Encoding of primary encoder Failed"); - return -1; - } - } - - if (secondary_ready_to_encode) { - assert(index_secondary >= 0 && - index_secondary < kMaxNumFragmentationVectors - 1); - int i = index_secondary * MAX_PAYLOAD_SIZE_BYTE; - if (EncodeFragmentation(index_secondary, - secondary_send_codec_inst_.pltype, - current_timestamp, secondary_encoder_.get(), - &stream[i]) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessDualStream(): Encoding of secondary encoder " - "Failed"); - return -1; - } - } - // Copy to local variable, as it will be used outside the ACM lock. - my_fragmentation.CopyFrom(fragmentation_); - my_red_payload_type = red_pltype_; - length_bytes = 0; - for (int n = 0; n < fragmentation_.fragmentationVectorSize; n++) { - length_bytes += fragmentation_.fragmentationLength[n]; - } - } - - { - CriticalSectionScoped lock(callback_crit_sect_); - if (packetization_callback_ != NULL) { - // Callback with payload data, including redundant data (FEC/RED). - if (packetization_callback_->SendData(kAudioFrameSpeech, - my_red_payload_type, - current_timestamp, stream, - length_bytes, - &my_fragmentation) < 0) { - return -1; - } - } - } - - { - CriticalSectionScoped lock(acm_crit_sect_); - // Now that data is sent, clean up fragmentation. - ResetFragmentation(0); - } - return 0; -} - -// Process any pending tasks such as timeouts. -int AudioCodingModuleImpl::ProcessSingleStream() { - // Make room for 1 RED payload. - uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE]; - int16_t length_bytes = 2 * MAX_PAYLOAD_SIZE_BYTE; - int16_t red_length_bytes = length_bytes; - uint32_t rtp_timestamp; - int16_t status; - WebRtcACMEncodingType encoding_type; - FrameType frame_type = kAudioFrameSpeech; - uint8_t current_payload_type = 0; - bool has_data_to_send = false; - bool fec_active = false; - RTPFragmentationHeader my_fragmentation; - - // Keep the scope of the ACM critical section limited. - { - CriticalSectionScoped lock(acm_crit_sect_); - // Check if there is an encoder before. - if (!HaveValidEncoder("ProcessSingleStream")) { - return -1; - } - status = codecs_[current_send_codec_idx_]->Encode(stream, &length_bytes, - &rtp_timestamp, - &encoding_type); - if (status < 0) { - // Encode failed. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessSingleStream(): Encoding Failed"); - length_bytes = 0; - return -1; - } else if (status == 0) { - // Not enough data. - return 0; - } else { - switch (encoding_type) { - case kNoEncoding: { - current_payload_type = previous_pltype_; - frame_type = kFrameEmpty; - length_bytes = 0; - break; - } - case kActiveNormalEncoded: - case kPassiveNormalEncoded: { - current_payload_type = static_cast<uint8_t>(send_codec_inst_.pltype); - frame_type = kAudioFrameSpeech; - break; - } - case kPassiveDTXNB: { - current_payload_type = cng_nb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - case kPassiveDTXWB: { - current_payload_type = cng_wb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - case kPassiveDTXSWB: { - current_payload_type = cng_swb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - case kPassiveDTXFB: { - current_payload_type = cng_fb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - } - has_data_to_send = true; - previous_pltype_ = current_payload_type; - - // Redundancy encode is done here. The two bitstreams packetized into - // one RTP packet and the fragmentation points are set. - // Only apply RED on speech data. - if ((fec_enabled_) && - ((encoding_type == kActiveNormalEncoded) || - (encoding_type == kPassiveNormalEncoded))) { - // FEC is enabled within this scope. - // - // Note that, a special solution exists for iSAC since it is the only - // codec for which GetRedPayload has a non-empty implementation. - // - // Summary of the FEC scheme below (use iSAC as example): - // - // 1st (is_first_red_ is true) encoded iSAC frame (primary #1) => - // - call GetRedPayload() and store redundancy for packet #1 in - // second fragment of RED buffer (old data) - // - drop the primary iSAC frame - // - don't call SendData - // 2nd (is_first_red_ is false) encoded iSAC frame (primary #2) => - // - store primary #2 in 1st fragment of RED buffer and send the - // combined packet - // - the transmitted packet contains primary #2 (new) and - // reduncancy for packet #1 (old) - // - call GetRed_Payload() and store redundancy for packet #2 in - // second fragment of RED buffer - // - // ... - // - // Nth encoded iSAC frame (primary #N) => - // - store primary #N in 1st fragment of RED buffer and send the - // combined packet - // - the transmitted packet contains primary #N (new) and - // reduncancy for packet #(N-1) (old) - // - call GetRedPayload() and store redundancy for packet #N in - // second fragment of RED buffer - // - // For all other codecs, GetRedPayload does nothing and returns -1 => - // redundant data is only a copy. - // - // First combined packet contains : #2 (new) and #1 (old) - // Second combined packet contains: #3 (new) and #2 (old) - // Third combined packet contains : #4 (new) and #3 (old) - // - // Hence, even if every second packet is dropped, perfect - // reconstruction is possible. - fec_active = true; - - has_data_to_send = false; - // Skip the following part for the first packet in a RED session. - if (!is_first_red_) { - // Rearrange stream such that FEC packets are included. - // Replace stream now that we have stored current stream. - memcpy(stream + fragmentation_.fragmentationOffset[1], red_buffer_, - fragmentation_.fragmentationLength[1]); - // Update the fragmentation time difference vector, in number of - // timestamps. - uint16_t time_since_last = static_cast<uint16_t>(rtp_timestamp - - last_fec_timestamp_); - - // Update fragmentation vectors. - fragmentation_.fragmentationPlType[1] = - fragmentation_.fragmentationPlType[0]; - fragmentation_.fragmentationTimeDiff[1] = time_since_last; - has_data_to_send = true; - } - - // Insert new packet length. - fragmentation_.fragmentationLength[0] = length_bytes; - - // Insert new packet payload type. - fragmentation_.fragmentationPlType[0] = current_payload_type; - last_fec_timestamp_ = rtp_timestamp; - - // Can be modified by the GetRedPayload() call if iSAC is utilized. - red_length_bytes = length_bytes; - - // A fragmentation header is provided => packetization according to - // RFC 2198 (RTP Payload for Redundant Audio Data) will be used. - // First fragment is the current data (new). - // Second fragment is the previous data (old). - length_bytes = static_cast<int16_t>( - fragmentation_.fragmentationLength[0] + - fragmentation_.fragmentationLength[1]); - - // Get, and store, redundant data from the encoder based on the recently - // encoded frame. - // NOTE - only iSAC contains an implementation; all other codecs does - // nothing and returns -1. - if (codecs_[current_send_codec_idx_]->GetRedPayload( - red_buffer_, - &red_length_bytes) == -1) { - // The codec was not iSAC => use current encoder output as redundant - // data instead (trivial FEC scheme). - memcpy(red_buffer_, stream, red_length_bytes); - } - - is_first_red_ = false; - // Update payload type with RED payload type. - current_payload_type = red_pltype_; - // We have packed 2 payloads. - fragmentation_.fragmentationVectorSize = kNumFecFragmentationVectors; - - // Copy to local variable, as it will be used outside ACM lock. - my_fragmentation.CopyFrom(fragmentation_); - // Store RED length. - fragmentation_.fragmentationLength[1] = red_length_bytes; - } - } - } - - if (has_data_to_send) { - CriticalSectionScoped lock(callback_crit_sect_); - - if (packetization_callback_ != NULL) { - if (fec_active) { - // Callback with payload data, including redundant data (FEC/RED). - packetization_callback_->SendData(frame_type, current_payload_type, - rtp_timestamp, stream, - length_bytes, - &my_fragmentation); - } else { - // Callback with payload data. - packetization_callback_->SendData(frame_type, current_payload_type, - rtp_timestamp, stream, - length_bytes, NULL); - } - } - - if (vad_callback_ != NULL) { - // Callback with VAD decision. - vad_callback_->InFrameType(static_cast<int16_t>(encoding_type)); - } - } - return length_bytes; -} - -///////////////////////////////////////// -// Sender -// - -// Initialize send codec. -int32_t AudioCodingModuleImpl::InitializeSender() { - CriticalSectionScoped lock(acm_crit_sect_); - - // Start with invalid values. - send_codec_registered_ = false; - current_send_codec_idx_ = -1; - send_codec_inst_.plname[0] = '\0'; - - // Delete all encoders to start fresh. - for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if (codecs_[id] != NULL) { - codecs_[id]->DestructEncoder(); - } - } - - // Initialize FEC/RED. - is_first_red_ = true; - if (fec_enabled_ || secondary_encoder_.get() != NULL) { - if (red_buffer_ != NULL) { - memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); - } - if (fec_enabled_) { - ResetFragmentation(kNumFecFragmentationVectors); - } else { - ResetFragmentation(0); - } - } - - return 0; -} - -int32_t AudioCodingModuleImpl::ResetEncoder() { - CriticalSectionScoped lock(acm_crit_sect_); - if (!HaveValidEncoder("ResetEncoder")) { - return -1; - } - return codecs_[current_send_codec_idx_]->ResetEncoder(); -} - -void AudioCodingModuleImpl::UnregisterSendCodec() { - CriticalSectionScoped lock(acm_crit_sect_); - send_codec_registered_ = false; - current_send_codec_idx_ = -1; - // If send Codec is unregistered then remove the secondary codec as well. - if (secondary_encoder_.get() != NULL) - secondary_encoder_.reset(); - return; -} - -ACMGenericCodec* AudioCodingModuleImpl::CreateCodec(const CodecInst& codec) { - ACMGenericCodec* my_codec = NULL; - - my_codec = ACMCodecDB::CreateCodecInstance(&codec); - if (my_codec == NULL) { - // Error, could not create the codec. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ACMCodecDB::CreateCodecInstance() failed in CreateCodec()"); - return my_codec; - } - my_codec->SetUniqueID(id_); - my_codec->SetNetEqDecodeLock(neteq_.DecodeLock()); - - return my_codec; -} - -// Check if the given codec is a valid to be registered as send codec. -static int IsValidSendCodec(const CodecInst& send_codec, - bool is_primary_encoder, - int acm_id, - int* mirror_id) { - if ((send_codec.channels != 1) && (send_codec.channels != 2)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "Wrong number of channels (%d, only mono and stereo are " - "supported) for %s encoder", send_codec.channels, - is_primary_encoder ? "primary" : "secondary"); - return -1; - } - - int codec_id = ACMCodecDB::CodecNumber(&send_codec, mirror_id); - if (codec_id < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "Invalid settings for the send codec."); - return -1; - } - - // TODO(tlegrand): Remove this check. Already taken care of in - // ACMCodecDB::CodecNumber(). - // Check if the payload-type is valid - if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "Invalid payload-type %d for %s.", send_codec.pltype, - send_codec.plname); - return -1; - } - - // Telephone-event cannot be a send codec. - if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "telephone-event cannot be a send codec"); - *mirror_id = -1; - return -1; - } - - if (ACMCodecDB::codec_settings_[codec_id].channel_support - < send_codec.channels) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "%d number of channels not supportedn for %s.", - send_codec.channels, send_codec.plname); - *mirror_id = -1; - return -1; - } - - if (!is_primary_encoder) { - // If registering the secondary encoder, then RED and CN are not valid - // choices as encoder. - if (IsCodecRED(&send_codec)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "RED cannot be secondary codec"); - *mirror_id = -1; - return -1; - } - - if (IsCodecCN(&send_codec)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "DTX cannot be secondary codec"); - *mirror_id = -1; - return -1; - } - } - return codec_id; -} - -int AudioCodingModuleImpl::RegisterSecondarySendCodec( - const CodecInst& send_codec) { - CriticalSectionScoped lock(acm_crit_sect_); - if (!send_codec_registered_) { - return -1; - } - // Primary and Secondary codecs should have the same sampling rates. - if (send_codec.plfreq != send_codec_inst_.plfreq) { - return -1; - } - int mirror_id; - int codec_id = IsValidSendCodec(send_codec, false, id_, &mirror_id); - if (codec_id < 0) { - return -1; - } - ACMGenericCodec* encoder = CreateCodec(send_codec); - WebRtcACMCodecParams codec_params; - // Initialize the codec before registering. For secondary codec VAD & DTX are - // disabled. - memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst)); - codec_params.enable_vad = false; - codec_params.enable_dtx = false; - codec_params.vad_mode = VADNormal; - // Force initialization. - if (encoder->InitEncoder(&codec_params, true) < 0) { - // Could not initialize, therefore cannot be registered. - delete encoder; - return -1; - } - secondary_encoder_.reset(encoder); - memcpy(&secondary_send_codec_inst_, &send_codec, sizeof(send_codec)); - - // Disable VAD & DTX. - SetVADSafe(false, false, VADNormal); - - // Cleaning. - if (red_buffer_) { - memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); - } - ResetFragmentation(0); - return 0; -} - -void AudioCodingModuleImpl::UnregisterSecondarySendCodec() { - CriticalSectionScoped lock(acm_crit_sect_); - if (secondary_encoder_.get() == NULL) { - return; - } - secondary_encoder_.reset(); - ResetFragmentation(0); -} - -int AudioCodingModuleImpl::SecondarySendCodec( - CodecInst* secondary_codec) const { - CriticalSectionScoped lock(acm_crit_sect_); - if (secondary_encoder_.get() == NULL) { - return -1; - } - memcpy(secondary_codec, &secondary_send_codec_inst_, - sizeof(secondary_send_codec_inst_)); - return 0; -} - -// Can be called multiple times for Codec, CNG, RED. -int32_t AudioCodingModuleImpl::RegisterSendCodec( - const CodecInst& send_codec) { - int mirror_id; - int codec_id = IsValidSendCodec(send_codec, true, id_, &mirror_id); - - CriticalSectionScoped lock(acm_crit_sect_); - - // Check for reported errors from function IsValidSendCodec(). - if (codec_id < 0) { - if (!send_codec_registered_) { - // This values has to be NULL if there is no codec registered. - current_send_codec_idx_ = -1; - } - return -1; - } - - // RED can be registered with other payload type. If not registered a default - // payload type is used. - if (IsCodecRED(&send_codec)) { - // TODO(tlegrand): Remove this check. Already taken care of in - // ACMCodecDB::CodecNumber(). - // Check if the payload-type is valid - if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid payload-type %d for %s.", send_codec.pltype, - send_codec.plname); - return -1; - } - // Set RED payload type. - red_pltype_ = static_cast<uint8_t>(send_codec.pltype); - return 0; - } - - // CNG can be registered with other payload type. If not registered the - // default payload types from codec database will be used. - if (IsCodecCN(&send_codec)) { - // CNG is registered. - switch (send_codec.plfreq) { - case 8000: { - cng_nb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - case 16000: { - cng_wb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - case 32000: { - cng_swb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - case 48000: { - cng_fb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RegisterSendCodec() failed, invalid frequency for CNG " - "registration"); - return -1; - } - } - return 0; - } - - // Set Stereo, and make sure VAD and DTX is turned off. - if (send_codec.channels == 2) { - stereo_send_ = true; - if (vad_enabled_ || dtx_enabled_) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "VAD/DTX is turned off, not supported when sending stereo."); - } - vad_enabled_ = false; - dtx_enabled_ = false; - } else { - stereo_send_ = false; - } - - // Check if the codec is already registered as send codec. - bool is_send_codec; - if (send_codec_registered_) { - int send_codec_mirror_id; - int send_codec_id = ACMCodecDB::CodecNumber(&send_codec_inst_, - &send_codec_mirror_id); - assert(send_codec_id >= 0); - is_send_codec = (send_codec_id == codec_id) || - (mirror_id == send_codec_mirror_id); - } else { - is_send_codec = false; - } - - // If there is secondary codec registered and the new send codec has a - // sampling rate different than that of secondary codec, then unregister the - // secondary codec. - if (secondary_encoder_.get() != NULL && - secondary_send_codec_inst_.plfreq != send_codec.plfreq) { - secondary_encoder_.reset(); - ResetFragmentation(0); - } - - // If new codec, or new settings, register. - if (!is_send_codec) { - if (codecs_[mirror_id] == NULL) { - codecs_[mirror_id] = CreateCodec(send_codec); - if (codecs_[mirror_id] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Create the codec"); - return -1; - } - mirror_codec_idx_[mirror_id] = mirror_id; - } - - if (mirror_id != codec_id) { - codecs_[codec_id] = codecs_[mirror_id]; - mirror_codec_idx_[codec_id] = mirror_id; - } - - ACMGenericCodec* codec_ptr = codecs_[codec_id]; - WebRtcACMCodecParams codec_params; - - memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst)); - codec_params.enable_vad = vad_enabled_; - codec_params.enable_dtx = dtx_enabled_; - codec_params.vad_mode = vad_mode_; - // Force initialization. - if (codec_ptr->InitEncoder(&codec_params, true) < 0) { - // Could not initialize the encoder. - - // Check if already have a registered codec. - // Depending on that different messages are logged. - if (!send_codec_registered_) { - current_send_codec_idx_ = -1; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Initialize the encoder No Encoder is registered"); - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Initialize the encoder, continue encoding with " - "the previously registered codec"); - } - return -1; - } - - // Update states. - dtx_enabled_ = codec_params.enable_dtx; - vad_enabled_ = codec_params.enable_vad; - vad_mode_ = codec_params.vad_mode; - - // Everything is fine so we can replace the previous codec with this one. - if (send_codec_registered_) { - // If we change codec we start fresh with FEC. - // This is not strictly required by the standard. - is_first_red_ = true; - - codec_ptr->SetVAD(&dtx_enabled_, &vad_enabled_, &vad_mode_); - } - - current_send_codec_idx_ = codec_id; - send_codec_registered_ = true; - memcpy(&send_codec_inst_, &send_codec, sizeof(CodecInst)); - previous_pltype_ = send_codec_inst_.pltype; - return 0; - } else { - // If codec is the same as already registered check if any parameters - // has changed compared to the current values. - // If any parameter is valid then apply it and record. - bool force_init = false; - - if (mirror_id != codec_id) { - codecs_[codec_id] = codecs_[mirror_id]; - mirror_codec_idx_[codec_id] = mirror_id; - } - - // Check the payload type. - if (send_codec.pltype != send_codec_inst_.pltype) { - // At this point check if the given payload type is valid. - // Record it later when the sampling frequency is changed - // successfully. - if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Out of range payload type"); - return -1; - } - } - - // If there is a codec that ONE instance of codec supports multiple - // sampling frequencies, then we need to take care of it here. - // one such a codec is iSAC. Both WB and SWB are encoded and decoded - // with one iSAC instance. Therefore, we need to update the encoder - // frequency if required. - if (send_codec_inst_.plfreq != send_codec.plfreq) { - force_init = true; - - // If sampling frequency is changed we have to start fresh with RED. - is_first_red_ = true; - } - - // If packet size or number of channels has changed, we need to - // re-initialize the encoder. - if (send_codec_inst_.pacsize != send_codec.pacsize) { - force_init = true; - } - if (send_codec_inst_.channels != send_codec.channels) { - force_init = true; - } - - if (force_init) { - WebRtcACMCodecParams codec_params; - - memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst)); - codec_params.enable_vad = vad_enabled_; - codec_params.enable_dtx = dtx_enabled_; - codec_params.vad_mode = vad_mode_; - - // Force initialization. - if (codecs_[current_send_codec_idx_]->InitEncoder(&codec_params, - true) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Could not change the codec packet-size."); - return -1; - } - - send_codec_inst_.plfreq = send_codec.plfreq; - send_codec_inst_.pacsize = send_codec.pacsize; - send_codec_inst_.channels = send_codec.channels; - } - - // If the change of sampling frequency has been successful then - // we store the payload-type. - send_codec_inst_.pltype = send_codec.pltype; - - // Check if a change in Rate is required. - if (send_codec.rate != send_codec_inst_.rate) { - if (codecs_[codec_id]->SetBitRate(send_codec.rate) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Could not change the codec rate."); - return -1; - } - send_codec_inst_.rate = send_codec.rate; - } - previous_pltype_ = send_codec_inst_.pltype; - - return 0; - } -} - -// Get current send codec. -int32_t AudioCodingModuleImpl::SendCodec( - CodecInst* current_codec) const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendCodec()"); - CriticalSectionScoped lock(acm_crit_sect_); - - assert(current_codec); - if (!send_codec_registered_) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendCodec Failed, no codec is registered"); - - return -1; - } - WebRtcACMCodecParams encoder_param; - codecs_[current_send_codec_idx_]->EncoderParams(&encoder_param); - encoder_param.codec_inst.pltype = send_codec_inst_.pltype; - memcpy(current_codec, &(encoder_param.codec_inst), sizeof(CodecInst)); - - return 0; -} - -// Get current send frequency. -int32_t AudioCodingModuleImpl::SendFrequency() const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendFrequency()"); - CriticalSectionScoped lock(acm_crit_sect_); - - if (!send_codec_registered_) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendFrequency Failed, no codec is registered"); - - return -1; - } - - return send_codec_inst_.plfreq; -} - -// Get encode bitrate. -// Adaptive rate codecs return their current encode target rate, while other -// codecs return there longterm avarage or their fixed rate. -int32_t AudioCodingModuleImpl::SendBitrate() const { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!send_codec_registered_) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendBitrate Failed, no codec is registered"); - - return -1; - } - - WebRtcACMCodecParams encoder_param; - codecs_[current_send_codec_idx_]->EncoderParams(&encoder_param); - - return encoder_param.codec_inst.rate; -} - -// Set available bandwidth, inform the encoder about the estimated bandwidth -// received from the remote party. -int32_t AudioCodingModuleImpl::SetReceivedEstimatedBandwidth( - const int32_t bw) { - return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw); -} - -// Register a transport callback which will be called to deliver -// the encoded buffers. -int32_t AudioCodingModuleImpl::RegisterTransportCallback( - AudioPacketizationCallback* transport) { - CriticalSectionScoped lock(callback_crit_sect_); - packetization_callback_ = transport; - return 0; -} - -// Add 10MS of raw (PCM) audio data to the encoder. -int32_t AudioCodingModuleImpl::Add10MsData( - const AudioFrame& audio_frame) { - if (audio_frame.samples_per_channel_ <= 0) { - assert(false); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, payload length is negative or " - "zero"); - return -1; - } - - // Allow for 8, 16, 32 and 48kHz input audio. - if ((audio_frame.sample_rate_hz_ != 8000) - && (audio_frame.sample_rate_hz_ != 16000) - && (audio_frame.sample_rate_hz_ != 32000) - && (audio_frame.sample_rate_hz_ != 48000)) { - assert(false); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, input frequency not valid"); - return -1; - } - - // If the length and frequency matches. We currently just support raw PCM. - if ((audio_frame.sample_rate_hz_ / 100) - != audio_frame.samples_per_channel_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, input frequency and length doesn't" - " match"); - return -1; - } - - if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, invalid number of channels."); - return -1; - } - - CriticalSectionScoped lock(acm_crit_sect_); - // Do we have a codec registered? - if (!HaveValidEncoder("Add10MsData")) { - return -1; - } - - const AudioFrame* ptr_frame; - // Perform a resampling, also down-mix if it is required and can be - // performed before resampling (a down mix prior to resampling will take - // place if both primary and secondary encoders are mono and input is in - // stereo). - if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) { - return -1; - } - TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Audio", ptr_frame->timestamp_, - "now", clock_->TimeInMilliseconds()); - - // Check whether we need an up-mix or down-mix? - bool remix = ptr_frame->num_channels_ != send_codec_inst_.channels; - if (secondary_encoder_.get() != NULL) { - remix = remix || - (ptr_frame->num_channels_ != secondary_send_codec_inst_.channels); - } - - // If a re-mix is required (up or down), this buffer will store re-mixed - // version of the input. - int16_t buffer[WEBRTC_10MS_PCM_AUDIO]; - if (remix) { - if (ptr_frame->num_channels_ == 1) { - if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, buffer) < 0) - return -1; - } else { - if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, buffer) < 0) - return -1; - } - } - - // When adding data to encoders this pointer is pointing to an audio buffer - // with correct number of channels. - const int16_t* ptr_audio = ptr_frame->data_; - - // For pushing data to primary, point the |ptr_audio| to correct buffer. - if (send_codec_inst_.channels != ptr_frame->num_channels_) - ptr_audio = buffer; - - if (codecs_[current_send_codec_idx_]->Add10MsData( - ptr_frame->timestamp_, ptr_audio, ptr_frame->samples_per_channel_, - send_codec_inst_.channels) < 0) - return -1; - - if (secondary_encoder_.get() != NULL) { - // For pushing data to secondary, point the |ptr_audio| to correct buffer. - ptr_audio = ptr_frame->data_; - if (secondary_send_codec_inst_.channels != ptr_frame->num_channels_) - ptr_audio = buffer; - - if (secondary_encoder_->Add10MsData( - ptr_frame->timestamp_, ptr_audio, ptr_frame->samples_per_channel_, - secondary_send_codec_inst_.channels) < 0) - return -1; - } - - return 0; -} - -// Perform a resampling and down-mix if required. We down-mix only if -// encoder is mono and input is stereo. In case of dual-streaming, both -// encoders has to be mono for down-mix to take place. -// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing -// is required, |*ptr_out| points to |in_frame|. -int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, - const AudioFrame** ptr_out) { - // Primary and secondary (if exists) should have the same sampling rate. - assert((secondary_encoder_.get() != NULL) ? - secondary_send_codec_inst_.plfreq == send_codec_inst_.plfreq : true); - - bool resample = static_cast<int32_t>(in_frame.sample_rate_hz_) != - send_codec_inst_.plfreq; - - // This variable is true if primary codec and secondary codec (if exists) - // are both mono and input is stereo. - bool down_mix; - if (secondary_encoder_.get() != NULL) { - down_mix = (in_frame.num_channels_ == 2) && - (send_codec_inst_.channels == 1) && - (secondary_send_codec_inst_.channels == 1); - } else { - down_mix = (in_frame.num_channels_ == 2) && - (send_codec_inst_.channels == 1); - } - - if (!down_mix && !resample) { - // No pre-processing is required. - last_in_timestamp_ = in_frame.timestamp_; - last_timestamp_ = in_frame.timestamp_; - *ptr_out = &in_frame; - return 0; - } - - *ptr_out = &preprocess_frame_; - preprocess_frame_.num_channels_ = in_frame.num_channels_; - int16_t audio[WEBRTC_10MS_PCM_AUDIO]; - const int16_t* src_ptr_audio = in_frame.data_; - int16_t* dest_ptr_audio = preprocess_frame_.data_; - if (down_mix) { - // If a resampling is required the output of a down-mix is written into a - // local buffer, otherwise, it will be written to the output frame. - if (resample) - dest_ptr_audio = audio; - if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0) - return -1; - preprocess_frame_.num_channels_ = 1; - // Set the input of the resampler is the down-mixed signal. - src_ptr_audio = audio; - } - - preprocess_frame_.timestamp_ = in_frame.timestamp_; - preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_; - preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; - // If it is required, we have to do a resampling. - if (resample) { - // The result of the resampler is written to output frame. - dest_ptr_audio = preprocess_frame_.data_; - - uint32_t timestamp_diff; - - // Calculate the timestamp of this frame. - if (last_in_timestamp_ > in_frame.timestamp_) { - // A wrap around has happened. - timestamp_diff = (static_cast<uint32_t>(0xFFFFFFFF) - last_in_timestamp_) - + in_frame.timestamp_; - } else { - timestamp_diff = in_frame.timestamp_ - last_in_timestamp_; - } - preprocess_frame_.timestamp_ = last_timestamp_ + - static_cast<uint32_t>(timestamp_diff * - (static_cast<double>(send_codec_inst_.plfreq) / - static_cast<double>(in_frame.sample_rate_hz_))); - - preprocess_frame_.samples_per_channel_ = input_resampler_.Resample10Msec( - src_ptr_audio, in_frame.sample_rate_hz_, dest_ptr_audio, - send_codec_inst_.plfreq, preprocess_frame_.num_channels_); - - if (preprocess_frame_.samples_per_channel_ < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add 10 ms audio, resmapling failed"); - return -1; - } - preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq; - } - last_in_timestamp_ = in_frame.timestamp_; - last_timestamp_ = preprocess_frame_.timestamp_; - - return 0; -} - -///////////////////////////////////////// -// (FEC) Forward Error Correction -// - -bool AudioCodingModuleImpl::FECStatus() const { - CriticalSectionScoped lock(acm_crit_sect_); - return fec_enabled_; -} - -// Configure FEC status i.e on/off. -int32_t -AudioCodingModuleImpl::SetFECStatus( -#ifdef WEBRTC_CODEC_RED - const bool enable_fec) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (fec_enabled_ != enable_fec) { - // Reset the RED buffer. - memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); - - // Reset fragmentation buffers. - ResetFragmentation(kNumFecFragmentationVectors); - // Set fec_enabled_. - fec_enabled_ = enable_fec; - } - is_first_red_ = true; // Make sure we restart FEC. - return 0; -#else - const bool /* enable_fec */) { - fec_enabled_ = false; - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - " WEBRTC_CODEC_RED is undefined => fec_enabled_ = %d", - fec_enabled_); - return -1; -#endif -} - -///////////////////////////////////////// -// (VAD) Voice Activity Detection -// -int32_t AudioCodingModuleImpl::SetVAD(bool enable_dtx, bool enable_vad, - ACMVADMode mode) { - CriticalSectionScoped lock(acm_crit_sect_); - return SetVADSafe(enable_dtx, enable_vad, mode); -} - -int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx, bool enable_vad, - ACMVADMode mode) { - // Sanity check of the mode. - if ((mode != VADNormal) && (mode != VADLowBitrate) - && (mode != VADAggr) && (mode != VADVeryAggr)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid VAD Mode %d, no change is made to VAD/DTX status", - static_cast<int>(mode)); - return -1; - } - - // Check that the send codec is mono. We don't support VAD/DTX for stereo - // sending. - if ((enable_dtx || enable_vad) && stereo_send_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "VAD/DTX not supported for stereo sending."); - dtx_enabled_ = false; - vad_enabled_ = false; - vad_mode_ = mode; - return -1; - } - - // We don't support VAD/DTX when dual-streaming is enabled, i.e. - // secondary-encoder is registered. - if ((enable_dtx || enable_vad) && secondary_encoder_.get() != NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "VAD/DTX not supported when dual-streaming is enabled."); - dtx_enabled_ = false; - vad_enabled_ = false; - vad_mode_ = mode; - return -1; - } - - // Store VAD/DTX settings. Values can be changed in the call to "SetVAD" - // below. - dtx_enabled_ = enable_dtx; - vad_enabled_ = enable_vad; - vad_mode_ = mode; - - // If a send codec is registered, set VAD/DTX for the codec. - if (HaveValidEncoder("SetVAD")) { - if (codecs_[current_send_codec_idx_]->SetVAD(&dtx_enabled_, &vad_enabled_, - &vad_mode_) < 0) { - // SetVAD failed. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetVAD failed"); - dtx_enabled_ = false; - vad_enabled_ = false; - return -1; - } - } - - return 0; -} - -// Get VAD/DTX settings. -// TODO(tlegrand): Change this method to void. -int32_t AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled, - ACMVADMode* mode) const { - CriticalSectionScoped lock(acm_crit_sect_); - - *dtx_enabled = dtx_enabled_; - *vad_enabled = vad_enabled_; - *mode = vad_mode_; - - return 0; -} - -///////////////////////////////////////// -// Receiver -// - -int32_t AudioCodingModuleImpl::InitializeReceiver() { - CriticalSectionScoped lock(acm_crit_sect_); - return InitializeReceiverSafe(); -} - -// Initialize receiver, resets codec database etc. -int32_t AudioCodingModuleImpl::InitializeReceiverSafe() { - initial_delay_ms_ = 0; - num_packets_accumulated_ = 0; - num_bytes_accumulated_ = 0; - accumulated_audio_ms_ = 0; - first_payload_received_ = 0; - last_incoming_send_timestamp_ = 0; - track_neteq_buffer_ = false; - playout_ts_ = 0; - // If the receiver is already initialized then we want to destroy any - // existing decoders. After a call to this function, we should have a clean - // start-up. - if (receiver_initialized_) { - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (UnregisterReceiveCodecSafe(i) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitializeReceiver() failed, Could not unregister codec"); - return -1; - } - } - } - if (neteq_.Init() != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitializeReceiver() failed, Could not initialize NetEQ"); - return -1; - } - neteq_.set_id(id_); - if (neteq_.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs) != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "NetEQ cannot allocate_packet Buffer"); - return -1; - } - - // Register RED and CN. - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (IsCodecRED(i) || IsCodecCN(i)) { - if (RegisterRecCodecMSSafe(ACMCodecDB::database_[i], i, i, - ACMNetEQ::kMasterJb) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register master codec."); - return -1; - } - registered_pltypes_[i] = ACMCodecDB::database_[i].pltype; - } - } - - receiver_initialized_ = true; - return 0; -} - -// Reset the decoder state. -int32_t AudioCodingModuleImpl::ResetDecoder() { - CriticalSectionScoped lock(acm_crit_sect_); - - for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if ((codecs_[id] != NULL) && (registered_pltypes_[id] != -1)) { - if (codecs_[id]->ResetDecoder(registered_pltypes_[id]) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ResetDecoder failed:"); - return -1; - } - } - } - return neteq_.FlushBuffers(); -} - -// Get current receive frequency. -int32_t AudioCodingModuleImpl::ReceiveFrequency() const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "ReceiveFrequency()"); - WebRtcACMCodecParams codec_params; - - CriticalSectionScoped lock(acm_crit_sect_); - if (DecoderParamByPlType(last_recv_audio_codec_pltype_, codec_params) < 0) { - return neteq_.CurrentSampFreqHz(); - } else if (codec_params.codec_inst.plfreq == 48000) { - // TODO(tlegrand): Remove this option when we have full 48 kHz support. - return 32000; - } else { - return codec_params.codec_inst.plfreq; - } -} - -// Get current playout frequency. -int32_t AudioCodingModuleImpl::PlayoutFrequency() const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "PlayoutFrequency()"); - - CriticalSectionScoped lock(acm_crit_sect_); - - return neteq_.CurrentSampFreqHz(); -} - -// Register possible receive codecs, can be called multiple times, -// for codecs, CNG (NB, WB and SWB), DTMF, RED. -int32_t AudioCodingModuleImpl::RegisterReceiveCodec( - const CodecInst& receive_codec) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (receive_codec.channels > 2) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "More than 2 audio channel is not supported."); - return -1; - } - - int mirror_id; - int codec_id = ACMCodecDB::ReceiverCodecNumber(&receive_codec, &mirror_id); - - if (codec_id < 0 || codec_id >= ACMCodecDB::kNumCodecs) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Wrong codec params to be registered as receive codec"); - return -1; - } - // Check if the payload-type is valid. - if (!ACMCodecDB::ValidPayloadType(receive_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid payload-type %d for %s.", receive_codec.pltype, - receive_codec.plname); - return -1; - } - - if (!receiver_initialized_) { - if (InitializeReceiverSafe() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot initialize reciver, so failed registering a codec."); - return -1; - } - } - - // If codec already registered, unregister. Except for CN where we only - // unregister if payload type is changing. - if ((registered_pltypes_[codec_id] == receive_codec.pltype) - && IsCodecCN(&receive_codec)) { - // Codec already registered as receiver with this payload type. Nothing - // to be done. - return 0; - } else if (registered_pltypes_[codec_id] != -1) { - if (UnregisterReceiveCodecSafe(codec_id) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register master codec."); - return -1; - } - } - - if (RegisterRecCodecMSSafe(receive_codec, codec_id, mirror_id, - ACMNetEQ::kMasterJb) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register master codec."); - return -1; - } - - // TODO(andrew): Refactor how the slave is initialized. Can we instead - // always start up a slave and pre-register CN and RED? We should be able - // to get rid of stereo_receive_registered_. - // http://code.google.com/p/webrtc/issues/detail?id=453 - - // Register stereo codecs with the slave, or, if we've had already seen a - // stereo codec, register CN or RED as a special case. - if (receive_codec.channels == 2 || - (stereo_receive_registered_ && (IsCodecCN(&receive_codec) || - IsCodecRED(&receive_codec)))) { - // TODO(andrew): refactor this block to combine with InitStereoSlave(). - - if (!stereo_receive_registered_) { - // This is the first time a stereo codec has been registered. Make - // some stereo preparations. - - // Add a stereo slave. - assert(neteq_.num_slaves() == 0); - if (neteq_.AddSlave(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add slave jitter buffer to NetEQ."); - return -1; - } - - // Register any existing CN or RED codecs with the slave and as stereo. - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (registered_pltypes_[i] != -1 && (IsCodecRED(i) || IsCodecCN(i))) { - stereo_receive_[i] = true; - - CodecInst codec; - memcpy(&codec, &ACMCodecDB::database_[i], sizeof(CodecInst)); - codec.pltype = registered_pltypes_[i]; - if (RegisterRecCodecMSSafe(codec, i, i, ACMNetEQ::kSlaveJb) < 0) { - WEBRTC_TRACE(kTraceError, kTraceAudioCoding, id_, - "Cannot register slave codec."); - return -1; - } - } - } - } - - if (RegisterRecCodecMSSafe(receive_codec, codec_id, mirror_id, - ACMNetEQ::kSlaveJb) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register slave codec."); - return -1; - } - - if (!stereo_receive_[codec_id] && - (last_recv_audio_codec_pltype_ == receive_codec.pltype)) { - // The last received payload type is the same as the one we are - // registering. Expected number of channels to receive is one (mono), - // but we are now registering the receiving codec as stereo (number of - // channels is 2). - // Set |last_recv_audio_coded_pltype_| to invalid value to trigger a - // flush in NetEq, and a reset of expected number of channels next time a - // packet is received in AudioCodingModuleImpl::IncomingPacket(). - last_recv_audio_codec_pltype_ = -1; - } - - stereo_receive_[codec_id] = true; - stereo_receive_registered_ = true; - } else { - if (last_recv_audio_codec_pltype_ == receive_codec.pltype && - expected_channels_ == 2) { - // The last received payload type is the same as the one we are - // registering. Expected number of channels to receive is two (stereo), - // but we are now registering the receiving codec as mono (number of - // channels is 1). - // Set |last_recv_audio_coded_pl_type_| to invalid value to trigger a - // flush in NetEq, and a reset of expected number of channels next time a - // packet is received in AudioCodingModuleImpl::IncomingPacket(). - last_recv_audio_codec_pltype_ = -1; - } - stereo_receive_[codec_id] = false; - } - - registered_pltypes_[codec_id] = receive_codec.pltype; - - if (IsCodecRED(&receive_codec)) { - receive_red_pltype_ = receive_codec.pltype; - } - return 0; -} - -int32_t AudioCodingModuleImpl::RegisterRecCodecMSSafe( - const CodecInst& receive_codec, int16_t codec_id, - int16_t mirror_id, ACMNetEQ::JitterBuffer jitter_buffer) { - ACMGenericCodec** codecs; - if (jitter_buffer == ACMNetEQ::kMasterJb) { - codecs = &codecs_[0]; - } else if (jitter_buffer == ACMNetEQ::kSlaveJb) { - codecs = &slave_codecs_[0]; - if (codecs_[codec_id]->IsTrueStereoCodec()) { - // True stereo codecs need to use the same codec memory - // for both master and slave. - slave_codecs_[mirror_id] = codecs_[mirror_id]; - mirror_codec_idx_[mirror_id] = mirror_id; - } - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RegisterReceiveCodecMSSafe failed, jitter_buffer is neither " - "master or slave "); - return -1; - } - - if (codecs[mirror_id] == NULL) { - codecs[mirror_id] = CreateCodec(receive_codec); - if (codecs[mirror_id] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot create codec to register as receive codec"); - return -1; - } - mirror_codec_idx_[mirror_id] = mirror_id; - } - if (mirror_id != codec_id) { - codecs[codec_id] = codecs[mirror_id]; - mirror_codec_idx_[codec_id] = mirror_id; - } - - codecs[codec_id]->SetIsMaster(jitter_buffer == ACMNetEQ::kMasterJb); - - int16_t status = 0; - WebRtcACMCodecParams codec_params; - memcpy(&(codec_params.codec_inst), &receive_codec, sizeof(CodecInst)); - codec_params.enable_vad = false; - codec_params.enable_dtx = false; - codec_params.vad_mode = VADNormal; - if (!codecs[codec_id]->DecoderInitialized()) { - // Force initialization. - status = codecs[codec_id]->InitDecoder(&codec_params, true); - if (status < 0) { - // Could not initialize the decoder, we don't want to - // continue if we could not initialize properly. - WEBRTC_TRACE( - webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "could not initialize the receive codec, codec not registered"); - - return -1; - } - } else if (mirror_id != codec_id) { - // Currently this only happens for iSAC. - // We have to store the decoder parameters. - codecs[codec_id]->SaveDecoderParam(&codec_params); - } - - if (codecs[codec_id]->RegisterInNetEq(&neteq_, receive_codec) != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Receive codec could not be registered in NetEQ"); - return -1; - } - // Guarantee that the same payload-type that is - // registered in NetEQ is stored in the codec. - codecs[codec_id]->SaveDecoderParam(&codec_params); - - return status; -} - -// Get current received codec. -int32_t AudioCodingModuleImpl::ReceiveCodec( - CodecInst* current_codec) const { - WebRtcACMCodecParams decoder_param; - CriticalSectionScoped lock(acm_crit_sect_); - - for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if (codecs_[id] != NULL) { - if (codecs_[id]->DecoderInitialized()) { - if (codecs_[id]->DecoderParams(&decoder_param, - last_recv_audio_codec_pltype_)) { - memcpy(current_codec, &decoder_param.codec_inst, - sizeof(CodecInst)); - return 0; - } - } - } - } - - // If we are here then we haven't found any codec. Set codec pltype to -1 to - // indicate that the structure is invalid and return -1. - current_codec->pltype = -1; - return -1; -} - -// Incoming packet from network parsed and ready for decode. -int32_t AudioCodingModuleImpl::IncomingPacket( - const uint8_t* incoming_payload, - const int32_t payload_length, - const WebRtcRTPHeader& rtp_info) { - WebRtcRTPHeader rtp_header; - - memcpy(&rtp_header, &rtp_info, sizeof(WebRtcRTPHeader)); - - if (payload_length < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "IncomingPacket() Error, payload-length cannot be negative"); - return -1; - } - - { - // Store the payload Type. This will be used to retrieve "received codec" - // and "received frequency." - CriticalSectionScoped lock(acm_crit_sect_); - - // Check there are packets missed between the last injected packet, and the - // latest received packet. If so and we are in AV-sync mode then we would - // like to fill the gap. Shouldn't be the first payload. - if (av_sync_ && first_payload_received_ && - rtp_info.header.sequenceNumber > last_sequence_number_ + 1) { - // If the last packet pushed was sync-packet account for all missing - // packets. Otherwise leave some room for PLC. - if (last_packet_was_sync_) { - while (rtp_info.header.sequenceNumber > last_sequence_number_ + 2) { - PushSyncPacketSafe(); - } - } else { - // Leave two packet room for NetEq perform PLC. - if (rtp_info.header.sequenceNumber > last_sequence_number_ + 3) { - last_sequence_number_ += 2; - last_incoming_send_timestamp_ += last_timestamp_diff_ * 2; - last_receive_timestamp_ += 2 * last_timestamp_diff_; - while (rtp_info.header.sequenceNumber > last_sequence_number_ + 1) - PushSyncPacketSafe(); - } - } - } - - uint8_t my_payload_type; - - // Check if this is an RED payload. - if (rtp_info.header.payloadType == receive_red_pltype_) { - // Get the primary payload-type. - my_payload_type = incoming_payload[0] & 0x7F; - } else { - my_payload_type = rtp_info.header.payloadType; - } - - // If payload is audio, check if received payload is different from - // previous. - if (!rtp_info.type.Audio.isCNG) { - // This is Audio not CNG. - - if (my_payload_type != last_recv_audio_codec_pltype_) { - // We detect a change in payload type. It is necessary for iSAC - // we are going to use ONE iSAC instance for decoding both WB and - // SWB payloads. If payload is changed there might be a need to reset - // sampling rate of decoder. depending what we have received "now". - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - if (registered_pltypes_[i] == my_payload_type) { - if (UpdateUponReceivingCodec(i) != 0) - return -1; - break; - } - } - // Codec is changed, there might be a jump in timestamp, therefore, - // we have to reset some variables that track NetEq buffer. - if (track_neteq_buffer_ || av_sync_) { - last_incoming_send_timestamp_ = rtp_info.header.timestamp; - } - - if (nack_enabled_) { - assert(nack_.get()); - // Codec is changed, reset NACK and update sampling rate. - nack_->Reset(); - nack_->UpdateSampleRate( - ACMCodecDB::database_[current_receive_codec_idx_].plfreq); - } - } - last_recv_audio_codec_pltype_ = my_payload_type; - } - - // Current timestamp based on the receiver sampling frequency. - last_receive_timestamp_ = NowTimestamp(current_receive_codec_idx_); - - if (nack_enabled_) { - assert(nack_.get()); - nack_->UpdateLastReceivedPacket(rtp_header.header.sequenceNumber, - rtp_header.header.timestamp); - } - } - - int per_neteq_payload_length = payload_length; - // Split the payload for stereo packets, so that first half of payload - // vector holds left channel, and second half holds right channel. - if (expected_channels_ == 2) { - if (!rtp_info.type.Audio.isCNG) { - // Create a new vector for the payload, maximum payload size. - int32_t length = payload_length; - uint8_t payload[kMaxPacketSize]; - assert(payload_length <= kMaxPacketSize); - memcpy(payload, incoming_payload, payload_length); - codecs_[current_receive_codec_idx_]->SplitStereoPacket(payload, &length); - rtp_header.type.Audio.channel = 2; - per_neteq_payload_length = length / 2; - // Insert packet into NetEQ. - if (neteq_.RecIn(payload, length, rtp_header, - last_receive_timestamp_) < 0) - return -1; - } else { - // If we receive a CNG packet while expecting stereo, we ignore the - // packet and continue. CNG is not supported for stereo. - return 0; - } - } else { - if (neteq_.RecIn(incoming_payload, payload_length, rtp_header, - last_receive_timestamp_) < 0) - return -1; - } - - { - CriticalSectionScoped lock(acm_crit_sect_); - - // Update buffering uses |last_incoming_send_timestamp_| so it should be - // before the next block. - if (track_neteq_buffer_) - UpdateBufferingSafe(rtp_header, per_neteq_payload_length); - - if (av_sync_) { - if (rtp_info.header.sequenceNumber == last_sequence_number_ + 1) { - last_timestamp_diff_ = rtp_info.header.timestamp - - last_incoming_send_timestamp_; - } - last_sequence_number_ = rtp_info.header.sequenceNumber; - last_ssrc_ = rtp_info.header.ssrc; - last_packet_was_sync_ = false; - } - - if (av_sync_ || track_neteq_buffer_) { - last_incoming_send_timestamp_ = rtp_info.header.timestamp; - } - - // Set the following regardless of tracking NetEq buffer or being in - // AV-sync mode. Only if the received packet is not CNG. - if (!rtp_info.type.Audio.isCNG) - first_payload_received_ = true; - } - return 0; -} - -int AudioCodingModuleImpl::UpdateUponReceivingCodec(int index) { - if (codecs_[index] == NULL) { - WEBRTC_TRACE(kTraceError, kTraceAudioCoding, id_, - "IncomingPacket() error: payload type found but " - "corresponding codec is NULL"); - return -1; - } - codecs_[index]->UpdateDecoderSampFreq(index); - neteq_.set_received_stereo(stereo_receive_[index]); - current_receive_codec_idx_ = index; - - // If we have a change in the expected number of channels, flush packet - // buffers in NetEQ. - if ((stereo_receive_[index] && (expected_channels_ == 1)) || - (!stereo_receive_[index] && (expected_channels_ == 2))) { - neteq_.FlushBuffers(); - codecs_[index]->ResetDecoder(registered_pltypes_[index]); - } - - if (stereo_receive_[index] && (expected_channels_ == 1)) { - // When switching from a mono to stereo codec reset the slave. - if (InitStereoSlave() != 0) - return -1; - } - - // Store number of channels we expect to receive for the current payload type. - if (stereo_receive_[index]) { - expected_channels_ = 2; - } else { - expected_channels_ = 1; - } - - // Reset previous received channel. - prev_received_channel_ = 0; - return 0; -} - -bool AudioCodingModuleImpl::IsCodecForSlave(int index) const { - return (registered_pltypes_[index] != -1 && stereo_receive_[index]); -} - -int AudioCodingModuleImpl::InitStereoSlave() { - neteq_.RemoveSlaves(); - - if (neteq_.AddSlave(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add slave jitter buffer to NetEQ."); - return -1; - } - - // Register all needed codecs with slave. - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (codecs_[i] != NULL && IsCodecForSlave(i)) { - WebRtcACMCodecParams decoder_params; - if (codecs_[i]->DecoderParams(&decoder_params, registered_pltypes_[i])) { - if (RegisterRecCodecMSSafe(decoder_params.codec_inst, - i, ACMCodecDB::MirrorID(i), - ACMNetEQ::kSlaveJb) < 0) { - WEBRTC_TRACE(kTraceError, kTraceAudioCoding, id_, - "Cannot register slave codec."); - return -1; - } - } - } - } - return 0; -} - -int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) { - { - CriticalSectionScoped lock(acm_crit_sect_); - // Don't let the extra delay modified while accumulating buffers in NetEq. - if (track_neteq_buffer_ && first_payload_received_) - return 0; - } - return neteq_.SetMinimumDelay(time_ms); -} - -int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) { - return neteq_.SetMaximumDelay(time_ms); -} - -// Get Dtmf playout status. -bool AudioCodingModuleImpl::DtmfPlayoutStatus() const { -#ifndef WEBRTC_CODEC_AVT - return false; -#else - return neteq_.avt_playout(); -#endif -} - -// Configure Dtmf playout status i.e on/off playout the incoming outband -// Dtmf tone. -int32_t AudioCodingModuleImpl::SetDtmfPlayoutStatus( -#ifndef WEBRTC_CODEC_AVT - const bool /* enable */) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "SetDtmfPlayoutStatus() failed: AVT is not supported."); - return -1; -#else - const bool enable) { - return neteq_.SetAVTPlayout(enable); -#endif -} - -// Estimate the Bandwidth based on the incoming stream, needed for one way -// audio where the RTCP send the BW estimate. -// This is also done in the RTP module. -int32_t AudioCodingModuleImpl::DecoderEstimatedBandwidth() const { - CodecInst codec; - int16_t codec_id = -1; - int pltype_wb; - int pltype_swb; - - // Get iSAC settings. - for (int id = 0; id < ACMCodecDB::kNumCodecs; id++) { - // Store codec settings for codec number "codeCntr" in the output struct. - ACMCodecDB::Codec(id, &codec); - - if (!STR_CASE_CMP(codec.plname, "isac")) { - codec_id = 1; - pltype_wb = codec.pltype; - - ACMCodecDB::Codec(id + 1, &codec); - pltype_swb = codec.pltype; - - break; - } - } - - if (codec_id < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "DecoderEstimatedBandwidth failed"); - return -1; - } - - if ((last_recv_audio_codec_pltype_ == pltype_wb) || - (last_recv_audio_codec_pltype_ == pltype_swb)) { - return codecs_[codec_id]->GetEstimatedBandwidth(); - } else { - return -1; - } -} - -// Set playout mode for: voice, fax, or streaming. -int32_t AudioCodingModuleImpl::SetPlayoutMode( - const AudioPlayoutMode mode) { - if ((mode != voice) && (mode != fax) && (mode != streaming) && - (mode != off)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid playout mode."); - return -1; - } - return neteq_.SetPlayoutMode(mode); -} - -// Get playout mode voice, fax. -AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const { - return neteq_.playout_mode(); -} - -// Get 10 milliseconds of raw audio data to play out. -// Automatic resample to the requested frequency. -int32_t AudioCodingModuleImpl::PlayoutData10Ms( - int32_t desired_freq_hz, AudioFrame* audio_frame) { - TRACE_EVENT_ASYNC_BEGIN0("webrtc", "ACM::PlayoutData10Ms", this); - bool stereo_mode; - - if (GetSilence(desired_freq_hz, audio_frame)) { - TRACE_EVENT_ASYNC_END1("webrtc", "ACM::PlayoutData10Ms", this, - "silence", true); - return 0; // Silence is generated, return. - } - - // RecOut always returns 10 ms. - if (neteq_.RecOut(audio_frame_) != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "PlayoutData failed, RecOut Failed"); - return -1; - } - int decoded_seq_num; - uint32_t decoded_timestamp; - bool update_nack = - neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp) && - nack_enabled_; // Update NACK only if it is enabled. - audio_frame->num_channels_ = audio_frame_.num_channels_; - audio_frame->vad_activity_ = audio_frame_.vad_activity_; - audio_frame->speech_type_ = audio_frame_.speech_type_; - - stereo_mode = (audio_frame_.num_channels_ > 1); - - // For stereo playout: - // Master and Slave samples are interleaved starting with Master. - const uint16_t receive_freq = - static_cast<uint16_t>(audio_frame_.sample_rate_hz_); - bool tone_detected = false; - int16_t last_detected_tone; - int16_t tone; - - // Limit the scope of ACM Critical section. - { - CriticalSectionScoped lock(acm_crit_sect_); - - // Update call statistics. - call_stats_.DecodedByNetEq(audio_frame->speech_type_); - - if (update_nack) { - assert(nack_.get()); - nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp); - } - - // If we are in AV-sync and have already received an audio packet, but the - // latest packet is too late, then insert sync packet. - if (av_sync_ && first_payload_received_ && - NowTimestamp(current_receive_codec_idx_) > 5 * last_timestamp_diff_ + - last_receive_timestamp_) { - if (!last_packet_was_sync_) { - // If the last packet inserted has been a regular packet Skip two - // packets to give room for PLC. - last_incoming_send_timestamp_ += 2 * last_timestamp_diff_; - last_sequence_number_ += 2; - last_receive_timestamp_ += 2 * last_timestamp_diff_; - } - - // One sync packet. - if (PushSyncPacketSafe() < 0) - return -1; - } - - if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) { - TRACE_EVENT_ASYNC_END2("webrtc", "ACM::PlayoutData10Ms", this, - "seqnum", decoded_seq_num, - "now", clock_->TimeInMilliseconds()); - // Resample payload_data. - int16_t temp_len = output_resampler_.Resample10Msec( - audio_frame_.data_, receive_freq, audio_frame->data_, - desired_freq_hz, audio_frame_.num_channels_); - - if (temp_len < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "PlayoutData failed, resampler failed"); - return -1; - } - - // Set the payload data length from the resampler. - audio_frame->samples_per_channel_ = static_cast<uint16_t>(temp_len); - // Set the sampling frequency. - audio_frame->sample_rate_hz_ = desired_freq_hz; - } else { - TRACE_EVENT_ASYNC_END2("webrtc", "ACM::PlayoutData10Ms", this, - "seqnum", decoded_seq_num, - "now", clock_->TimeInMilliseconds()); - memcpy(audio_frame->data_, audio_frame_.data_, - audio_frame_.samples_per_channel_ * audio_frame->num_channels_ - * sizeof(int16_t)); - // Set the payload length. - audio_frame->samples_per_channel_ = - audio_frame_.samples_per_channel_; - // Set the sampling frequency. - audio_frame->sample_rate_hz_ = receive_freq; - } - - // Tone detection done for master channel. - if (dtmf_detector_ != NULL) { - // Dtmf Detection. - if (audio_frame->sample_rate_hz_ == 8000) { - // Use audio_frame->data_ then Dtmf detector doesn't - // need resampling. - if (!stereo_mode) { - dtmf_detector_->Detect(audio_frame->data_, - audio_frame->samples_per_channel_, - audio_frame->sample_rate_hz_, tone_detected, - tone); - } else { - // We are in 8 kHz so the master channel needs only 80 samples. - int16_t master_channel[80]; - for (int n = 0; n < 80; n++) { - master_channel[n] = audio_frame->data_[n << 1]; - } - dtmf_detector_->Detect(master_channel, - audio_frame->samples_per_channel_, - audio_frame->sample_rate_hz_, tone_detected, - tone); - } - } else { - // Do the detection on the audio that we got from NetEQ (audio_frame_). - if (!stereo_mode) { - dtmf_detector_->Detect(audio_frame_.data_, - audio_frame_.samples_per_channel_, - receive_freq, tone_detected, tone); - } else { - int16_t master_channel[WEBRTC_10MS_PCM_AUDIO]; - for (int n = 0; n < audio_frame_.samples_per_channel_; n++) { - master_channel[n] = audio_frame_.data_[n << 1]; - } - dtmf_detector_->Detect(master_channel, - audio_frame_.samples_per_channel_, - receive_freq, tone_detected, tone); - } - } - } - - // We want to do this while we are in acm_crit_sect_. - // (Doesn't really need to initialize the following - // variable but Linux complains if we don't.) - last_detected_tone = kACMToneEnd; - if (tone_detected) { - last_detected_tone = last_detected_tone_; - last_detected_tone_ = tone; - } - } - - if (tone_detected) { - // We will deal with callback here, so enter callback critical section. - CriticalSectionScoped lock(callback_crit_sect_); - - if (dtmf_callback_ != NULL) { - if (tone != kACMToneEnd) { - // just a tone - dtmf_callback_->IncomingDtmf(static_cast<uint8_t>(tone), false); - } else if ((tone == kACMToneEnd) && (last_detected_tone != kACMToneEnd)) { - // The tone is "END" and the previously detected tone is - // not "END," so call fir an end. - dtmf_callback_->IncomingDtmf(static_cast<uint8_t>(last_detected_tone), - true); - } - } - } - - audio_frame->id_ = id_; - audio_frame->energy_ = -1; - audio_frame->timestamp_ = 0; - - return 0; -} - -///////////////////////////////////////// -// Statistics -// - -int32_t AudioCodingModuleImpl::NetworkStatistics( - ACMNetworkStatistics* statistics) { - int32_t status; - status = neteq_.NetworkStatistics(statistics); - return status; -} - -void AudioCodingModuleImpl::DestructEncoderInst(void* inst) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "DestructEncoderInst()"); - if (!HaveValidEncoder("DestructEncoderInst")) { - return; - } - - codecs_[current_send_codec_idx_]->DestructEncoderInst(inst); -} - -int16_t AudioCodingModuleImpl::AudioBuffer( - WebRtcACMAudioBuff& buffer) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "AudioBuffer()"); - if (!HaveValidEncoder("AudioBuffer")) { - return -1; - } - buffer.last_in_timestamp = last_in_timestamp_; - return codecs_[current_send_codec_idx_]->AudioBuffer(buffer); -} - -int16_t AudioCodingModuleImpl::SetAudioBuffer( - WebRtcACMAudioBuff& buffer) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "SetAudioBuffer()"); - if (!HaveValidEncoder("SetAudioBuffer")) { - return -1; - } - return codecs_[current_send_codec_idx_]->SetAudioBuffer(buffer); -} - -uint32_t AudioCodingModuleImpl::EarliestTimestamp() const { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "EarliestTimestamp()"); - if (!HaveValidEncoder("EarliestTimestamp")) { - return -1; - } - return codecs_[current_send_codec_idx_]->EarliestTimestamp(); -} - -int32_t AudioCodingModuleImpl::RegisterVADCallback( - ACMVADCallback* vad_callback) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "RegisterVADCallback()"); - CriticalSectionScoped lock(callback_crit_sect_); - vad_callback_ = vad_callback; - return 0; -} - -// TODO(turajs): Remove this API if it is not used. -// TODO(tlegrand): Modify this function to work for stereo, and add tests. -// TODO(turajs): Receive timestamp in this method is incremented by frame-size -// and does not reflect the true receive frame-size. Therefore, subsequent -// jitter computations are not accurate. -int32_t AudioCodingModuleImpl::IncomingPayload( - const uint8_t* incoming_payload, const int32_t payload_length, - const uint8_t payload_type, const uint32_t timestamp) { - if (payload_length < 0) { - // Log error in trace file. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "IncomingPacket() Error, payload-length cannot be negative"); - return -1; - } - - if (dummy_rtp_header_ == NULL) { - // This is the first time that we are using |dummy_rtp_header_| - // so we have to create it. - WebRtcACMCodecParams codec_params; - dummy_rtp_header_ = new WebRtcRTPHeader; - if (dummy_rtp_header_ == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "IncomingPayload() Error, out of memory"); - return -1; - } - dummy_rtp_header_->header.payloadType = payload_type; - // Don't matter in this case. - dummy_rtp_header_->header.ssrc = 0; - dummy_rtp_header_->header.markerBit = false; - // Start with random numbers. - dummy_rtp_header_->header.sequenceNumber = rand(); - dummy_rtp_header_->header.timestamp = - (static_cast<uint32_t>(rand()) << 16) + - static_cast<uint32_t>(rand()); - dummy_rtp_header_->type.Audio.channel = 1; - - if (DecoderParamByPlType(payload_type, codec_params) < 0) { - // We didn't find a codec with the given payload. - // Something is wrong we exit, but we delete |dummy_rtp_header_| - // and set it to NULL to start clean next time. - delete dummy_rtp_header_; - dummy_rtp_header_ = NULL; - return -1; - } - recv_pl_frame_size_smpls_ = codec_params.codec_inst.pacsize; - } - - if (payload_type != dummy_rtp_header_->header.payloadType) { - // Payload type has changed since the last time we might need to - // update the frame-size. - WebRtcACMCodecParams codec_params; - if (DecoderParamByPlType(payload_type, codec_params) < 0) { - // We didn't find a codec with the given payload. - return -1; - } - recv_pl_frame_size_smpls_ = codec_params.codec_inst.pacsize; - dummy_rtp_header_->header.payloadType = payload_type; - } - - if (timestamp > 0) { - dummy_rtp_header_->header.timestamp = timestamp; - } - - // Store the payload Type. this will be used to retrieve "received codec" - // and "received frequency." - last_recv_audio_codec_pltype_ = payload_type; - - last_receive_timestamp_ += recv_pl_frame_size_smpls_; - // Insert in NetEQ. - if (neteq_.RecIn(incoming_payload, payload_length, *dummy_rtp_header_, - last_receive_timestamp_) < 0) { - return -1; - } - - // Get ready for the next payload. - dummy_rtp_header_->header.sequenceNumber++; - dummy_rtp_header_->header.timestamp += recv_pl_frame_size_smpls_; - return 0; -} - -int16_t AudioCodingModuleImpl::DecoderParamByPlType( - const uint8_t payload_type, - WebRtcACMCodecParams& codec_params) const { - CriticalSectionScoped lock(acm_crit_sect_); - for (int16_t id = 0; id < ACMCodecDB::kMaxNumCodecs; - id++) { - if (codecs_[id] != NULL) { - if (codecs_[id]->DecoderInitialized()) { - if (codecs_[id]->DecoderParams(&codec_params, payload_type)) { - return 0; - } - } - } - } - // If we are here it means that we could not find a - // codec with that payload type. reset the values to - // not acceptable values and return -1. - codec_params.codec_inst.plname[0] = '\0'; - codec_params.codec_inst.pacsize = 0; - codec_params.codec_inst.rate = 0; - codec_params.codec_inst.pltype = -1; - return -1; -} - -int16_t AudioCodingModuleImpl::DecoderListIDByPlName( - const char* name, const uint16_t frequency) const { - WebRtcACMCodecParams codec_params; - CriticalSectionScoped lock(acm_crit_sect_); - for (int16_t id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if ((codecs_[id] != NULL)) { - if (codecs_[id]->DecoderInitialized()) { - assert(registered_pltypes_[id] >= 0); - assert(registered_pltypes_[id] <= 255); - codecs_[id]->DecoderParams( - &codec_params, static_cast<uint8_t>(registered_pltypes_[id])); - if (!STR_CASE_CMP(codec_params.codec_inst.plname, name)) { - // Check if the given sampling frequency matches. - // A zero sampling frequency means we matching the names - // is sufficient and we don't need to check for the - // frequencies. - // Currently it is only iSAC which has one name but two - // sampling frequencies. - if ((frequency == 0)|| - (codec_params.codec_inst.plfreq == frequency)) { - return id; - } - } - } - } - } - // If we are here it means that we could not find a - // codec with that payload type. return -1. - return -1; -} - -int32_t AudioCodingModuleImpl::LastEncodedTimestamp( - uint32_t& timestamp) const { - CriticalSectionScoped lock(acm_crit_sect_); - if (!HaveValidEncoder("LastEncodedTimestamp")) { - return -1; - } - timestamp = codecs_[current_send_codec_idx_]->LastEncodedTimestamp(); - return 0; -} - -int32_t AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc( - bool use_webrtc_dtx) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) { - WEBRTC_TRACE( - webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot replace codec internal DTX when no send codec is registered."); - return -1; - } - - int32_t res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX( - use_webrtc_dtx); - // Check if VAD is turned on, or if there is any error. - if (res == 1) { - vad_enabled_ = true; - } else if (res < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Failed to set ReplaceInternalDTXWithWebRtc(%d)", - use_webrtc_dtx); - return res; - } - - return 0; -} - -int32_t AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc( - bool* uses_webrtc_dtx) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc")) { - return -1; - } - if (codecs_[current_send_codec_idx_]->IsInternalDTXReplaced(uses_webrtc_dtx) - < 0) { - return -1; - } - return 0; -} - -int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("SetISACMaxRate")) { - return -1; - } - - return codecs_[current_send_codec_idx_]->SetISACMaxRate(max_bit_per_sec); -} - -int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("SetISACMaxPayloadSize")) { - return -1; - } - - return codecs_[current_send_codec_idx_]->SetISACMaxPayloadSize( - max_size_bytes); -} - -int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator( - int frame_size_ms, - int rate_bit_per_sec, - bool enforce_frame_size) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) { - return -1; - } - - return codecs_[current_send_codec_idx_]->ConfigISACBandwidthEstimator( - frame_size_ms, rate_bit_per_sec, enforce_frame_size); -} - -int32_t AudioCodingModuleImpl::PlayoutTimestamp( - uint32_t* timestamp) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "PlayoutTimestamp()"); - { - CriticalSectionScoped lock(acm_crit_sect_); - if (track_neteq_buffer_) { - *timestamp = playout_ts_; - return 0; - } - } - return neteq_.PlayoutTimestamp(*timestamp); -} - -bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const { - if ((!send_codec_registered_) || (current_send_codec_idx_ < 0) || - (current_send_codec_idx_ >= ACMCodecDB::kNumCodecs)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "%s failed: No send codec is registered.", caller_name); - return false; - } - if ((current_send_codec_idx_ < 0) || - (current_send_codec_idx_ >= ACMCodecDB::kNumCodecs)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "%s failed: Send codec index out of range.", caller_name); - return false; - } - if (codecs_[current_send_codec_idx_] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "%s failed: Send codec is NULL pointer.", caller_name); - return false; - } - return true; -} - -int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) { - CriticalSectionScoped lock(acm_crit_sect_); - int id; - - // Search through the list of registered payload types. - for (id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if (registered_pltypes_[id] == payload_type) { - // We have found the id registered with the payload type. - break; - } - } - - if (id >= ACMCodecDB::kNumCodecs) { - // Payload type was not registered. No need to unregister. - return 0; - } - - // Unregister the codec with the given payload type. - return UnregisterReceiveCodecSafe(id); -} - -int32_t AudioCodingModuleImpl::UnregisterReceiveCodecSafe( - const int16_t codec_id) { - const WebRtcNetEQDecoder *neteq_decoder = ACMCodecDB::NetEQDecoders(); - int16_t mirror_id = ACMCodecDB::MirrorID(codec_id); - bool stereo_receiver = false; - - if (codecs_[codec_id] != NULL) { - if (registered_pltypes_[codec_id] != -1) { - // Store stereo information for future use. - stereo_receiver = stereo_receive_[codec_id]; - - // Before deleting the decoder instance unregister from NetEQ. - if (neteq_.RemoveCodec(neteq_decoder[codec_id], - stereo_receive_[codec_id]) < 0) { - CodecInst codec; - ACMCodecDB::Codec(codec_id, &codec); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Unregistering %s-%d from NetEQ failed.", codec.plname, - codec.plfreq); - return -1; - } - - // CN is a special case for NetEQ, all three sampling frequencies - // are unregistered if one is deleted. - if (IsCodecCN(codec_id)) { - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (IsCodecCN(i)) { - stereo_receive_[i] = false; - registered_pltypes_[i] = -1; - } - } - } else { - if (codec_id == mirror_id) { - codecs_[codec_id]->DestructDecoder(); - if (stereo_receive_[codec_id]) { - slave_codecs_[codec_id]->DestructDecoder(); - stereo_receive_[codec_id] = false; - } - } - } - - // Check if this is the last registered stereo receive codec. - if (stereo_receiver) { - bool no_stereo = true; - - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (stereo_receive_[i]) { - // We still have stereo codecs registered. - no_stereo = false; - break; - } - } - - // If we don't have any stereo codecs left, change status. - if (no_stereo) { - neteq_.RemoveSlaves(); // No longer need the slave. - stereo_receive_registered_ = false; - } - } - } - } - - if (registered_pltypes_[codec_id] == receive_red_pltype_) { - // RED is going to be unregistered, set to an invalid value. - receive_red_pltype_ = 255; - } - registered_pltypes_[codec_id] = -1; - - return 0; -} - -int32_t AudioCodingModuleImpl::REDPayloadISAC( - const int32_t isac_rate, const int16_t isac_bw_estimate, - uint8_t* payload, int16_t* length_bytes) { - if (!HaveValidEncoder("EncodeData")) { - return -1; - } - int16_t status; - status = codecs_[current_send_codec_idx_]->REDPayloadISAC(isac_rate, - isac_bw_estimate, - payload, - length_bytes); - return status; -} - -void AudioCodingModuleImpl::ResetFragmentation(int vector_size) { - for (int n = 0; n < kMaxNumFragmentationVectors; n++) { - fragmentation_.fragmentationOffset[n] = n * MAX_PAYLOAD_SIZE_BYTE; - } - memset(fragmentation_.fragmentationLength, 0, kMaxNumFragmentationVectors * - sizeof(fragmentation_.fragmentationLength[0])); - memset(fragmentation_.fragmentationTimeDiff, 0, kMaxNumFragmentationVectors * - sizeof(fragmentation_.fragmentationTimeDiff[0])); - memset(fragmentation_.fragmentationPlType, 0, kMaxNumFragmentationVectors * - sizeof(fragmentation_.fragmentationPlType[0])); - fragmentation_.fragmentationVectorSize = - static_cast<uint16_t>(vector_size); -} - -// TODO(turajs): Add second parameter to enable/disable AV-sync. -int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) { - if (delay_ms < 0 || delay_ms > 10000) { - return -1; - } - - CriticalSectionScoped lock(acm_crit_sect_); - - // Receiver should be initialized before this call processed. - if (!receiver_initialized_) { - InitializeReceiverSafe(); - } - - if (first_payload_received_) { - // Too late for this API. Only works before a call is started. - return -1; - } - initial_delay_ms_ = delay_ms; - - // If initial delay is zero, NetEq buffer should not be tracked, also we - // don't want to be in AV-sync mode. - track_neteq_buffer_ = delay_ms > 0; - av_sync_ = delay_ms > 0; - - neteq_.EnableAVSync(av_sync_); - return neteq_.SetMinimumDelay(delay_ms); -} - -bool AudioCodingModuleImpl::GetSilence(int desired_sample_rate_hz, - AudioFrame* frame) { - CriticalSectionScoped lock(acm_crit_sect_); - if (initial_delay_ms_ == 0 || !track_neteq_buffer_) { - return false; - } - - if (accumulated_audio_ms_ >= initial_delay_ms_) { - // We have enough data stored that match our initial delay target. - track_neteq_buffer_ = false; - return false; - } - - // Record call to silence generator. - call_stats_.DecodedBySilenceGenerator(); - - // We stop accumulating packets, if the number of packets or the total size - // exceeds a threshold. - int max_num_packets; - int buffer_size_bytes; - int per_payload_overhead_bytes; - neteq_.BufferSpec(max_num_packets, buffer_size_bytes, - per_payload_overhead_bytes); - int total_bytes_accumulated = num_bytes_accumulated_ + - num_packets_accumulated_ * per_payload_overhead_bytes; - if (num_packets_accumulated_ > max_num_packets * 0.9 || - total_bytes_accumulated > buffer_size_bytes * 0.9) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "GetSilence: Initial delay couldn't be achieved." - " num_packets_accumulated=%d, total_bytes_accumulated=%d", - num_packets_accumulated_, num_bytes_accumulated_); - track_neteq_buffer_ = false; - return false; - } - - if (desired_sample_rate_hz > 0) { - frame->sample_rate_hz_ = desired_sample_rate_hz; - } else { - frame->sample_rate_hz_ = 0; - if (current_receive_codec_idx_ >= 0) { - frame->sample_rate_hz_ = - ACMCodecDB::database_[current_receive_codec_idx_].plfreq; - } else { - // No payload received yet, use the default sampling rate of NetEq. - frame->sample_rate_hz_ = neteq_.CurrentSampFreqHz(); - } - } - frame->num_channels_ = expected_channels_; - frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms. - frame->speech_type_ = AudioFrame::kCNG; - frame->vad_activity_ = AudioFrame::kVadPassive; - frame->energy_ = 0; - int samples = frame->samples_per_channel_ * frame->num_channels_; - memset(frame->data_, 0, samples * sizeof(int16_t)); - return true; -} - -// Must be called within the scope of ACM critical section. -int AudioCodingModuleImpl::PushSyncPacketSafe() { - assert(av_sync_); - last_sequence_number_++; - last_incoming_send_timestamp_ += last_timestamp_diff_; - last_receive_timestamp_ += last_timestamp_diff_; - - WebRtcRTPHeader rtp_info; - rtp_info.header.payloadType = last_recv_audio_codec_pltype_; - rtp_info.header.ssrc = last_ssrc_; - rtp_info.header.markerBit = false; - rtp_info.header.sequenceNumber = last_sequence_number_; - rtp_info.header.timestamp = last_incoming_send_timestamp_; - rtp_info.type.Audio.channel = stereo_receive_[current_receive_codec_idx_] ? - 2 : 1; - last_packet_was_sync_ = true; - int payload_len_bytes = neteq_.RecIn(rtp_info, last_receive_timestamp_); - - if (payload_len_bytes < 0) - return -1; - - // This is to account for sync packets inserted during the buffering phase. - if (track_neteq_buffer_) - UpdateBufferingSafe(rtp_info, payload_len_bytes); - - return 0; -} - -// Must be called within the scope of ACM critical section. -void AudioCodingModuleImpl::UpdateBufferingSafe(const WebRtcRTPHeader& rtp_info, - int payload_len_bytes) { - const int in_sample_rate_khz = - (ACMCodecDB::database_[current_receive_codec_idx_].plfreq / 1000); - if (first_payload_received_ && - rtp_info.header.timestamp > last_incoming_send_timestamp_ && - in_sample_rate_khz > 0) { - accumulated_audio_ms_ += (rtp_info.header.timestamp - - last_incoming_send_timestamp_) / in_sample_rate_khz; - } - - num_packets_accumulated_++; - num_bytes_accumulated_ += payload_len_bytes; - - playout_ts_ = static_cast<uint32_t>( - rtp_info.header.timestamp - static_cast<uint32_t>( - initial_delay_ms_ * in_sample_rate_khz)); -} - -uint32_t AudioCodingModuleImpl::NowTimestamp(int codec_id) { - // Down-cast the time to (32-6)-bit since we only care about - // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms. - // we masked 6 most significant bits of 32-bit so we don't lose resolution - // when do the following multiplication. - int sample_rate_khz = ACMCodecDB::database_[codec_id].plfreq / 1000; - const uint32_t now_in_ms = static_cast<uint32_t>( - clock_->TimeInMilliseconds() & kMaskTimestamp); - return static_cast<uint32_t>(sample_rate_khz * now_in_ms); -} - -std::vector<uint16_t> AudioCodingModuleImpl::GetNackList( - int round_trip_time_ms) const { - CriticalSectionScoped lock(acm_crit_sect_); - if (round_trip_time_ms < 0) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "GetNackList: round trip time cannot be negative." - " round_trip_time_ms=%d", round_trip_time_ms); - } - if (nack_enabled_ && round_trip_time_ms >= 0) { - assert(nack_.get()); - return nack_->GetNackList(round_trip_time_ms); - } - std::vector<uint16_t> empty_list; - return empty_list; -} - -int AudioCodingModuleImpl::LeastRequiredDelayMs() const { - return std::max(neteq_.LeastRequiredDelayMs(), initial_delay_ms_); -} - -int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) { - // Don't do anything if |max_nack_list_size| is out of range. - if (max_nack_list_size == 0 || - max_nack_list_size > acm2::Nack::kNackListSizeLimit) - return -1; - - CriticalSectionScoped lock(acm_crit_sect_); - if (!nack_enabled_) { - nack_.reset(acm2::Nack::Create(kNackThresholdPackets)); - nack_enabled_ = true; - - // Sampling rate might need to be updated if we change from disable to - // enable. Do it if the receive codec is valid. - if (current_receive_codec_idx_ >= 0) { - nack_->UpdateSampleRate( - ACMCodecDB::database_[current_receive_codec_idx_].plfreq); - } - } - return nack_->SetMaxNackListSize(max_nack_list_size); -} - -void AudioCodingModuleImpl::DisableNack() { - CriticalSectionScoped lock(acm_crit_sect_); - nack_.reset(); // Memory is released. - nack_enabled_ = false; -} - -const char* AudioCodingModuleImpl::Version() const { - return kLegacyAcmVersion; -} - -void AudioCodingModuleImpl::GetDecodingCallStatistics( - AudioDecodingCallStats* call_stats) const { - CriticalSectionScoped lock(acm_crit_sect_); - *call_stats = call_stats_.GetDecodingStatistics(); -} - -} // namespace acm1 - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h deleted file mode 100644 index f0b22f11465..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h +++ /dev/null @@ -1,455 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_ - -#include <vector> - -#include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/source/acm_neteq.h" -#include "webrtc/modules/audio_coding/main/source/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" -#include "webrtc/system_wrappers/interface/scoped_ptr.h" - -namespace webrtc { - -struct WebRtcACMAudioBuff; -struct WebRtcACMCodecParams; -class CriticalSectionWrapper; -class RWLockWrapper; -class Clock; - -namespace acm2 { -class Nack; -} - -namespace acm1 { - -class ACMDTMFDetection; -class ACMGenericCodec; - -class AudioCodingModuleImpl : public AudioCodingModule { - public: - AudioCodingModuleImpl(const int32_t id, Clock* clock); - ~AudioCodingModuleImpl(); - - virtual const char* Version() const; - - // Change the unique identifier of this object. - virtual int32_t ChangeUniqueId(const int32_t id); - - // Returns the number of milliseconds until the module want a worker thread - // to call Process. - int32_t TimeUntilNextProcess(); - - // Process any pending tasks such as timeouts. - int32_t Process(); - - ///////////////////////////////////////// - // Sender - // - - // Initialize send codec. - int32_t InitializeSender(); - - // Reset send codec. - int32_t ResetEncoder(); - - // Can be called multiple times for Codec, CNG, RED. - int32_t RegisterSendCodec(const CodecInst& send_codec); - - // Register Secondary codec for dual-streaming. Dual-streaming is activated - // right after the secondary codec is registered. - int RegisterSecondarySendCodec(const CodecInst& send_codec); - - // Unregister the secondary codec. Dual-streaming is deactivated right after - // deregistering secondary codec. - void UnregisterSecondarySendCodec(); - - // Get the secondary codec. - int SecondarySendCodec(CodecInst* secondary_codec) const; - - // Get current send codec. - int32_t SendCodec(CodecInst* current_codec) const; - - // Get current send frequency. - int32_t SendFrequency() const; - - // Get encode bit-rate. - // Adaptive rate codecs return their current encode target rate, while other - // codecs return there long-term average or their fixed rate. - int32_t SendBitrate() const; - - // Set available bandwidth, inform the encoder about the - // estimated bandwidth received from the remote party. - virtual int32_t SetReceivedEstimatedBandwidth(const int32_t bw); - - // Register a transport callback which will be - // called to deliver the encoded buffers. - int32_t RegisterTransportCallback(AudioPacketizationCallback* transport); - - // Add 10 ms of raw (PCM) audio data to the encoder. - int32_t Add10MsData(const AudioFrame& audio_frame); - - ///////////////////////////////////////// - // (FEC) Forward Error Correction - // - - // Configure FEC status i.e on/off. - int32_t SetFECStatus(const bool enable_fec); - - // Get FEC status. - bool FECStatus() const; - - ///////////////////////////////////////// - // (VAD) Voice Activity Detection - // and - // (CNG) Comfort Noise Generation - // - - int32_t SetVAD(bool enable_dtx = true, - bool enable_vad = false, - ACMVADMode mode = VADNormal); - - int32_t VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const; - - int32_t RegisterVADCallback(ACMVADCallback* vad_callback); - - ///////////////////////////////////////// - // Receiver - // - - // Initialize receiver, resets codec database etc. - int32_t InitializeReceiver(); - - // Reset the decoder state. - int32_t ResetDecoder(); - - // Get current receive frequency. - int32_t ReceiveFrequency() const; - - // Get current playout frequency. - int32_t PlayoutFrequency() const; - - // Register possible receive codecs, can be called multiple times, - // for codecs, CNG, DTMF, RED. - int32_t RegisterReceiveCodec(const CodecInst& receive_codec); - - // Get current received codec. - int32_t ReceiveCodec(CodecInst* current_codec) const; - - // Incoming packet from network parsed and ready for decode. - int32_t IncomingPacket(const uint8_t* incoming_payload, - const int32_t payload_length, - const WebRtcRTPHeader& rtp_info); - - // Incoming payloads, without rtp-info, the rtp-info will be created in ACM. - // One usage for this API is when pre-encoded files are pushed in ACM. - int32_t IncomingPayload(const uint8_t* incoming_payload, - const int32_t payload_length, - const uint8_t payload_type, - const uint32_t timestamp = 0); - - // NetEq minimum playout delay (used for lip-sync). The actual target delay - // is the max of |time_ms| and the required delay dictated by the channel. - int SetMinimumPlayoutDelay(int time_ms); - - // NetEq maximum playout delay. The actual target delay is the min of - // |time_ms| and the required delay dictated by the channel. - int SetMaximumPlayoutDelay(int time_ms); - - // The shortest latency, in milliseconds, required by jitter buffer. This - // is computed based on inter-arrival times and playout mode of NetEq. The - // actual delay is the maximum of least-required-delay and the minimum-delay - // specified by SetMinumumPlayoutDelay() API. - // - int LeastRequiredDelayMs() const ; - - // Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf - // tone. - int32_t SetDtmfPlayoutStatus(const bool enable); - - // Get Dtmf playout status. - bool DtmfPlayoutStatus() const; - - // Estimate the Bandwidth based on the incoming stream, needed - // for one way audio where the RTCP send the BW estimate. - // This is also done in the RTP module . - int32_t DecoderEstimatedBandwidth() const; - - // Set playout mode voice, fax. - int32_t SetPlayoutMode(const AudioPlayoutMode mode); - - // Get playout mode voice, fax. - AudioPlayoutMode PlayoutMode() const; - - // Get playout timestamp. - int32_t PlayoutTimestamp(uint32_t* timestamp); - - // Get 10 milliseconds of raw audio data to play out, and - // automatic resample to the requested frequency if > 0. - int32_t PlayoutData10Ms(int32_t desired_freq_hz, - AudioFrame* audio_frame); - - ///////////////////////////////////////// - // Statistics - // - - int32_t NetworkStatistics(ACMNetworkStatistics* statistics); - - void DestructEncoderInst(void* inst); - - int16_t AudioBuffer(WebRtcACMAudioBuff& buffer); - - // GET RED payload for iSAC. The method id called when 'this' ACM is - // the default ACM. - int32_t REDPayloadISAC(const int32_t isac_rate, - const int16_t isac_bw_estimate, - uint8_t* payload, - int16_t* length_bytes); - - int16_t SetAudioBuffer(WebRtcACMAudioBuff& buffer); - - uint32_t EarliestTimestamp() const; - - int32_t LastEncodedTimestamp(uint32_t& timestamp) const; - - int32_t ReplaceInternalDTXWithWebRtc(const bool use_webrtc_dtx); - - int32_t IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx); - - int SetISACMaxRate(int max_bit_per_sec); - - int SetISACMaxPayloadSize(int max_size_bytes); - - int32_t ConfigISACBandwidthEstimator( - int frame_size_ms, - int rate_bit_per_sec, - bool enforce_frame_size = false); - - int UnregisterReceiveCodec(uint8_t payload_type); - - std::vector<uint16_t> GetNackList(int round_trip_time_ms) const; - - protected: - void UnregisterSendCodec(); - - int32_t UnregisterReceiveCodecSafe(const int16_t id); - - ACMGenericCodec* CreateCodec(const CodecInst& codec); - - int16_t DecoderParamByPlType(const uint8_t payload_type, - WebRtcACMCodecParams& codec_params) const; - - int16_t DecoderListIDByPlName( - const char* name, const uint16_t frequency = 0) const; - - int32_t InitializeReceiverSafe(); - - bool HaveValidEncoder(const char* caller_name) const; - - int32_t RegisterRecCodecMSSafe(const CodecInst& receive_codec, - int16_t codec_id, - int16_t mirror_id, - ACMNetEQ::JitterBuffer jitter_buffer); - - // Set VAD/DTX status. This function does not acquire a lock, and it is - // created to be called only from inside a critical section. - int SetVADSafe(bool enable_dtx, bool enable_vad, ACMVADMode mode); - - // Process buffered audio when dual-streaming is not enabled (When RED is - // enabled still this function is used.) - int ProcessSingleStream(); - - // Process buffered audio when dual-streaming is enabled, i.e. secondary send - // codec is registered. - int ProcessDualStream(); - - // Preprocessing of input audio, including resampling and down-mixing if - // required, before pushing audio into encoder's buffer. - // - // in_frame: input audio-frame - // ptr_out: pointer to output audio_frame. If no preprocessing is required - // |ptr_out| will be pointing to |in_frame|, otherwise pointing to - // |preprocess_frame_|. - // - // Return value: - // -1: if encountering an error. - // 0: otherwise. - int PreprocessToAddData(const AudioFrame& in_frame, - const AudioFrame** ptr_out); - - // Set initial playout delay. - // -delay_ms: delay in millisecond. - // - // Return value: - // -1: if cannot set the delay. - // 0: if delay set successfully. - int SetInitialPlayoutDelay(int delay_ms); - - // Enable NACK and set the maximum size of the NACK list. - int EnableNack(size_t max_nack_list_size); - - // Disable NACK. - void DisableNack(); - - void GetDecodingCallStatistics(AudioDecodingCallStats* call_stats) const; - - private: - // Change required states after starting to receive the codec corresponding - // to |index|. - int UpdateUponReceivingCodec(int index); - - // Remove all slaves and initialize a stereo slave with required codecs - // from the master. - int InitStereoSlave(); - - // Returns true if the codec's |index| is registered with the master and - // is a stereo codec, RED or CN. - bool IsCodecForSlave(int index) const; - - int EncodeFragmentation(int fragmentation_index, int payload_type, - uint32_t current_timestamp, - ACMGenericCodec* encoder, - uint8_t* stream); - - void ResetFragmentation(int vector_size); - - bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame); - - // Push a synchronization packet into NetEq. Such packets result in a frame - // of zeros (not decoded by the corresponding decoder). The size of the frame - // is the same as last decoding. NetEq has a special payload for this. - // Call within the scope of ACM critical section. - int PushSyncPacketSafe(); - - // Update the parameters required in initial phase of buffering, when - // initial playout delay is requested. Call within the scope of ACM critical - // section. - void UpdateBufferingSafe(const WebRtcRTPHeader& rtp_info, - int payload_len_bytes); - - // - // Return the timestamp of current time, computed according to sampling rate - // of the codec identified by |codec_id|. - // - uint32_t NowTimestamp(int codec_id); - - AudioPacketizationCallback* packetization_callback_; - int32_t id_; - uint32_t last_timestamp_; - uint32_t last_in_timestamp_; - CodecInst send_codec_inst_; - uint8_t cng_nb_pltype_; - uint8_t cng_wb_pltype_; - uint8_t cng_swb_pltype_; - uint8_t cng_fb_pltype_; - uint8_t red_pltype_; - bool vad_enabled_; - bool dtx_enabled_; - ACMVADMode vad_mode_; - ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs]; - ACMGenericCodec* slave_codecs_[ACMCodecDB::kMaxNumCodecs]; - int16_t mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs]; - bool stereo_receive_[ACMCodecDB::kMaxNumCodecs]; - bool stereo_receive_registered_; - bool stereo_send_; - int prev_received_channel_; - int expected_channels_; - int32_t current_send_codec_idx_; - int current_receive_codec_idx_; - bool send_codec_registered_; - ACMResampler input_resampler_; - ACMResampler output_resampler_; - ACMNetEQ neteq_; - CriticalSectionWrapper* acm_crit_sect_; - ACMVADCallback* vad_callback_; - uint8_t last_recv_audio_codec_pltype_; - - // RED/FEC. - bool is_first_red_; - bool fec_enabled_; - // TODO(turajs): |red_buffer_| is allocated in constructor, why having them - // as pointers and not an array. If concerned about the memory, then make a - // set-up function to allocate them only when they are going to be used, i.e. - // FEC or Dual-streaming is enabled. - uint8_t* red_buffer_; - // TODO(turajs): we actually don't need |fragmentation_| as a member variable. - // It is sufficient to keep the length & payload type of previous payload in - // member variables. - RTPFragmentationHeader fragmentation_; - uint32_t last_fec_timestamp_; - // If no RED is registered as receive codec this - // will have an invalid value. - uint8_t receive_red_pltype_; - - // This is to keep track of CN instances where we can send DTMFs. - uint8_t previous_pltype_; - - // This keeps track of payload types associated with codecs_[]. - // We define it as signed variable and initialize with -1 to indicate - // unused elements. - int16_t registered_pltypes_[ACMCodecDB::kMaxNumCodecs]; - - // Used when payloads are pushed into ACM without any RTP info - // One example is when pre-encoded bit-stream is pushed from - // a file. - WebRtcRTPHeader* dummy_rtp_header_; - uint16_t recv_pl_frame_size_smpls_; - - bool receiver_initialized_; - ACMDTMFDetection* dtmf_detector_; - - AudioCodingFeedback* dtmf_callback_; - int16_t last_detected_tone_; - CriticalSectionWrapper* callback_crit_sect_; - - AudioFrame audio_frame_; - AudioFrame preprocess_frame_; - CodecInst secondary_send_codec_inst_; - scoped_ptr<ACMGenericCodec> secondary_encoder_; - - // Initial delay. - int initial_delay_ms_; - int num_packets_accumulated_; - int num_bytes_accumulated_; - int accumulated_audio_ms_; - int first_payload_received_; - uint32_t last_incoming_send_timestamp_; - bool track_neteq_buffer_; - uint32_t playout_ts_; - - // AV-sync is enabled. In AV-sync mode, sync packet pushed during long packet - // losses. - bool av_sync_; - - // Latest send timestamp difference of two consecutive packets. - uint32_t last_timestamp_diff_; - uint16_t last_sequence_number_; - uint32_t last_ssrc_; - bool last_packet_was_sync_; - int64_t last_receive_timestamp_; - - Clock* clock_; - scoped_ptr<acm2::Nack> nack_; - bool nack_enabled_; - - acm2::CallStatistics call_stats_; -}; - -} // namespace acm1 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/Android.mk b/chromium/third_party/webrtc/modules/audio_coding/neteq/Android.mk deleted file mode 100644 index 84267becf2c..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/Android.mk +++ /dev/null @@ -1,73 +0,0 @@ -# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -LOCAL_PATH := $(call my-dir) - -include $(CLEAR_VARS) - -include $(LOCAL_PATH)/../../../../android-webrtc.mk - -LOCAL_ARM_MODE := arm -LOCAL_MODULE_CLASS := STATIC_LIBRARIES -LOCAL_MODULE := libwebrtc_neteq -LOCAL_MODULE_TAGS := optional -LOCAL_SRC_FILES := \ - accelerate.c \ - automode.c \ - bgn_update.c \ - bufstats_decision.c \ - cng_internal.c \ - codec_db.c \ - correlator.c \ - dsp.c \ - dsp_helpfunctions.c \ - dtmf_buffer.c \ - dtmf_tonegen.c \ - expand.c \ - mcu_address_init.c \ - mcu_dsp_common.c \ - mcu_reset.c \ - merge.c \ - min_distortion.c \ - mix_voice_unvoice.c \ - mute_signal.c \ - normal.c \ - packet_buffer.c \ - peak_detection.c \ - preemptive_expand.c \ - random_vector.c \ - recin.c \ - recout.c \ - rtcp.c \ - rtp.c \ - set_fs.c \ - signal_mcu.c \ - split_and_insert.c \ - unmute_signal.c \ - webrtc_neteq.c - -# Flags passed to both C and C++ files. -LOCAL_CFLAGS := \ - $(MY_WEBRTC_COMMON_DEFS) \ - '-DNETEQ_VOICEENGINE_CODECS' - -LOCAL_C_INCLUDES := \ - $(LOCAL_PATH)/interface \ - $(LOCAL_PATH)/../codecs/cng/include \ - $(LOCAL_PATH)/../../.. \ - $(LOCAL_PATH)/../../../common_audio/signal_processing/include - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libdl \ - libstlport - -ifndef NDK_ROOT -include external/stlport/libstlport.mk -endif -include $(BUILD_STATIC_LIBRARY) diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/neteq/OWNERS index b5c79cef499..072e754998f 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/OWNERS +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/OWNERS @@ -1,3 +1,11 @@ henrik.lundin@webrtc.org tina.legrand@webrtc.org turaj@webrtc.org +minyue@webrtc.org + +per-file *.isolate=kjellander@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.c deleted file mode 100644 index a345a8fdcd2..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.c +++ /dev/null @@ -1,493 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the Accelerate algorithm that is used to reduce - * the delay by removing a part of the audio stream. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" -#include "neteq_error_codes.h" - -#define ACCELERATE_CORR_LEN 50 -#define ACCELERATE_MIN_LAG 10 -#define ACCELERATE_MAX_LAG 60 -#define ACCELERATE_DOWNSAMPLED_LEN (ACCELERATE_CORR_LEN + ACCELERATE_MAX_LAG) - -/* Scratch usage: - - Type Name size startpos endpos - int16_t pw16_downSampSpeech 110 0 109 - int32_t pw32_corr 2*50 110 209 - int16_t pw16_corr 50 0 49 - - Total: 110+2*50 - */ - -#define SCRATCH_PW16_DS_SPEECH 0 -#define SCRATCH_PW32_CORR ACCELERATE_DOWNSAMPLED_LEN -#define SCRATCH_PW16_CORR 0 - -/**************************************************************************** - * WebRtcNetEQ_Accelerate(...) - * - * This function tries to shorten the audio data by removing one or several - * pitch periods. The operation is only carried out if the correlation is - * strong or if the signal energy is very low. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector. - * - decoded : Pointer to newly decoded speech. - * - len : Length of decoded speech. - * - BGNonly : If non-zero, Accelerate will only remove the last - * DEFAULT_TIME_ADJUST seconds of the input. - * No signal matching is done. - * - * Output: - * - inst : Updated instance - * - outData : Pointer to a memory space where the output data - * should be stored - * - pw16_len : Number of samples written to outData. - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_Accelerate(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - const int16_t *pw16_decoded, int len, - int16_t *pw16_outData, int16_t *pw16_len, - int16_t BGNonly) -{ - -#ifdef SCRATCH - /* Use scratch memory for internal temporary vectors */ - int16_t *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH; - int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_PW32_CORR); - int16_t *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR; -#else - /* Allocate memory for temporary vectors */ - int16_t pw16_downSampSpeech[ACCELERATE_DOWNSAMPLED_LEN]; - int32_t pw32_corr[ACCELERATE_CORR_LEN]; - int16_t pw16_corr[ACCELERATE_CORR_LEN]; -#endif - int16_t w16_decodedMax = 0; - int16_t w16_tmp; - int16_t w16_tmp2; - int32_t w32_tmp; - int32_t w32_tmp2; - - const int16_t w16_startLag = ACCELERATE_MIN_LAG; - const int16_t w16_endLag = ACCELERATE_MAX_LAG; - const int16_t w16_corrLen = ACCELERATE_CORR_LEN; - const int16_t *pw16_vec1, *pw16_vec2; - int16_t *pw16_vectmp; - int16_t w16_inc, w16_startfact; - int16_t w16_bestIndex, w16_bestVal; - int16_t w16_VAD = 1; - int16_t fsMult; - int16_t fsMult120; - int32_t w32_en1, w32_en2, w32_cc; - int16_t w16_en1, w16_en2; - int16_t w16_en1Scale, w16_en2Scale; - int16_t w16_sqrtEn1En2; - int16_t w16_bestCorr = 0; - int ok; - -#ifdef NETEQ_STEREO - MasterSlaveInfo *msInfo = inst->msInfo; -#endif - - fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */ - - /* Pre-calculate common multiplication with fsMult */ - fsMult120 = (int16_t) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */ - - inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */ - - /* Sanity check for len variable; must be (almost) 30 ms - (120*fsMult + max(bestIndex)) */ - if (len < (int16_t) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult)) - { - /* Length of decoded data too short */ - inst->w16_mode = MODE_UNSUCCESS_ACCELERATE; - *pw16_len = len; - - /* simply move all data from decoded to outData */ - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return NETEQ_OTHER_ERROR; - } - - /***********************************/ - /* Special operations for BGN only */ - /***********************************/ - - /* Check if "background noise only" flag is set */ - if (BGNonly) - { - /* special operation for BGN only; simply remove a chunk of data */ - w16_bestIndex = DEFAULT_TIME_ADJUST * WEBRTC_SPL_LSHIFT_W16(fsMult, 3); /* X*fs/1000 */ - - /* Sanity check for bestIndex */ - if (w16_bestIndex > len) - { /* not good, do nothing instead */ - inst->w16_mode = MODE_UNSUCCESS_ACCELERATE; - *pw16_len = len; - - /* simply move all data from decoded to outData */ - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return NETEQ_OTHER_ERROR; - } - - /* set length parameter */ - *pw16_len = len - w16_bestIndex; /* we remove bestIndex samples */ - - /* copy to output */ - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, *pw16_len); - - /* set mode */ - inst->w16_mode = MODE_LOWEN_ACCELERATE; - - /* update statistics */ - inst->statInst.accelerateLength += w16_bestIndex; - /* Short-term activity statistics. */ - inst->activity_stats.accelerate_bgn_samples += w16_bestIndex; - - return 0; - } /* end of special code for BGN mode */ - -#ifdef NETEQ_STEREO - - /* Sanity for msInfo */ - if (msInfo == NULL) - { - /* this should not happen here */ - return MASTER_SLAVE_ERROR; - } - - if (msInfo->msMode != NETEQ_SLAVE) - { - /* Find correlation lag only for non-slave instances */ - -#endif - - /****************************************************************/ - /* Find the strongest correlation lag by downsampling to 4 kHz, */ - /* calculating correlation for downsampled signal and finding */ - /* the strongest correlation peak. */ - /****************************************************************/ - - /* find maximum absolute value */ - w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len); - - /* downsample the decoded speech to 4 kHz */ - ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech, - ACCELERATE_DOWNSAMPLED_LEN, 1 /* compensate delay*/); - if (ok != 0) - { - /* error */ - inst->w16_mode = MODE_UNSUCCESS_ACCELERATE; - *pw16_len = len; - /* simply move all data from decoded to outData */ - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - return NETEQ_OTHER_ERROR; - } - - /* - * Set scaling factor for cross correlation to protect against overflow - * (log2(50) => 6) - */ - w16_tmp = 6 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)); - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */ - WebRtcNetEQ_CrossCorr( - pw32_corr, &pw16_downSampSpeech[w16_endLag], - &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen, - (int16_t) (w16_endLag - w16_startLag), w16_tmp, -1); - - /* Normalize correlation to 14 bits and put in a int16_t vector */ - w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen); - w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp); - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_corrLen, pw32_corr, w16_tmp); - -#ifdef NETEQ_STEREO - } /* end if (msInfo->msMode != NETEQ_SLAVE) */ - - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)) - { - /* Find the strongest correlation peak by using the parabolic fit method */ - WebRtcNetEQ_PeakDetection(pw16_corr, (int16_t) w16_corrLen, 1, fsMult, - &w16_bestIndex, &w16_bestVal); - /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */ - - /* Compensate bestIndex for displaced starting position */ - w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1); - /* 20*fsMult <= bestIndex <= 119*fsMult */ - - msInfo->bestIndex = w16_bestIndex; - } - else if (msInfo->msMode == NETEQ_SLAVE) - { - if (msInfo->extraInfo == ACC_FAIL) - { - /* Master has signaled an unsuccessful accelerate */ - w16_bestIndex = 0; - } - else - { - /* Get best index from master */ - w16_bestIndex = msInfo->bestIndex; - } - } - else - { - /* Invalid mode */ - return MASTER_SLAVE_ERROR; - } - -#else /* NETEQ_STEREO */ - - /* Find the strongest correlation peak by using the parabolic fit method */ - WebRtcNetEQ_PeakDetection(pw16_corr, (int16_t) w16_corrLen, 1, fsMult, - &w16_bestIndex, &w16_bestVal); - /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */ - - /* Compensate bestIndex for displaced starting position */ - w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1); - /* 20*fsMult <= bestIndex <= 119*fsMult */ - -#endif /* NETEQ_STEREO */ - -#ifdef NETEQ_STEREO - - if (msInfo->msMode != NETEQ_SLAVE) - { - /* Calculate correlation only for non-slave instances */ - -#endif /* NETEQ_STEREO */ - - /*****************************************************/ - /* Calculate correlation bestCorr for the found lag. */ - /* Also do a simple VAD decision. */ - /*****************************************************/ - - /* - * Calculate scaling to ensure that bestIndex samples can be square-summed - * without overflowing - */ - w16_tmp = (31 - - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax))); - w16_tmp += (31 - WebRtcSpl_NormW32(w16_bestIndex)); - w16_tmp -= 31; - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - /* vec1 starts at 15 ms minus one pitch period */ - pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex]; - /* vec2 start at 15 ms */ - pw16_vec2 = &pw16_decoded[fsMult120]; - - /* Calculate energies for vec1 and vec2 */ - w32_en1 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1, - (int16_t*) pw16_vec1, w16_bestIndex, w16_tmp); - w32_en2 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec2, - (int16_t*) pw16_vec2, w16_bestIndex, w16_tmp); - - /* Calculate cross-correlation at the found lag */ - w32_cc = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1, (int16_t*) pw16_vec2, - w16_bestIndex, w16_tmp); - - /* Check VAD constraint - ((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */ - w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_en1 + w32_en2, 4); /* (en1+en2)/(2*8) */ - if (inst->BGNInst.w16_initialized == 1) - { - w32_tmp2 = inst->BGNInst.w32_energy; - } - else - { - /* if BGN parameters have not been estimated, use a fixed threshold */ - w32_tmp2 = 75000; - } - w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2); - w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2); - w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2); - w16_tmp2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2); - w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2); - - /* Scale w32_tmp properly before comparing with w32_tmp2 */ - /* (w16_tmp is scaling before energy calculation, thus 2*w16_tmp) */ - if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1)) - { - /* Cannot scale only w32_tmp, must scale w32_temp2 too */ - int16_t tempshift = WebRtcSpl_NormW32(w32_tmp); - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift); - w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2, - WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift); - } - else - { - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, - WEBRTC_SPL_LSHIFT_W32(w16_tmp,1)); - } - - if (w32_tmp <= w32_tmp2) /*((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */ - { - /* The signal seems to be passive speech */ - w16_VAD = 0; - w16_bestCorr = 0; /* Correlation does not matter */ - } - else - { - /* The signal is active speech */ - w16_VAD = 1; - - /* Calculate correlation (cc/sqrt(en1*en2)) */ - - /* Start with calculating scale values */ - w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1); - w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale); - w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2); - w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale); - - /* Make sure total scaling is even (to simplify scale factor after sqrt) */ - if ((w16_en1Scale + w16_en2Scale) & 1) - { - w16_en1Scale += 1; - } - - /* Convert energies to int16_t */ - w16_en1 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale); - w16_en2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale); - - /* Calculate energy product */ - w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2); - - /* Calculate square-root of energy product */ - w16_sqrtEn1En2 = (int16_t) WebRtcSpl_SqrtFloor(w32_tmp); - - /* Calculate cc/sqrt(en1*en2) in Q14 */ - w16_tmp = 14 - WEBRTC_SPL_RSHIFT_W16(w16_en1Scale+w16_en2Scale, 1); - w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp); - w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */ - w16_bestCorr = (int16_t) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2); - w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */ - } - -#ifdef NETEQ_STEREO - - } /* end if (msInfo->msMode != NETEQ_SLAVE) */ - -#endif /* NETEQ_STEREO */ - - /************************************************/ - /* Check accelerate criteria and remove samples */ - /************************************************/ - - /* Check for strong correlation (>0.9) or passive speech */ -#ifdef NETEQ_STEREO - if ((((w16_bestCorr > 14746) || (w16_VAD == 0)) && (msInfo->msMode != NETEQ_SLAVE)) - || ((msInfo->msMode == NETEQ_SLAVE) && (msInfo->extraInfo != ACC_FAIL))) -#else - if ((w16_bestCorr > 14746) || (w16_VAD == 0)) -#endif - { - /* Do accelerate operation by overlap add */ - - /* - * Calculate cross-fading slope so that the fading factor goes from - * 1 (16384 in Q14) to 0 in one pitch period (bestIndex). - */ - w16_inc = (int16_t) WebRtcSpl_DivW32W16((int32_t) 16384, - (int16_t) (w16_bestIndex + 1)); /* in Q14 */ - - /* Initiate fading factor */ - w16_startfact = 16384 - w16_inc; - - /* vec1 starts at 15 ms minus one pitch period */ - pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex]; - /* vec2 start at 15 ms */ - pw16_vec2 = &pw16_decoded[fsMult120]; - - /* Copy unmodified part [0 to 15 ms minus 1 pitch period] */ - w16_tmp = (fsMult120 - w16_bestIndex); - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, w16_tmp); - - /* Generate interpolated part of length bestIndex (1 pitch period) */ - pw16_vectmp = pw16_outData + w16_tmp; /* start of interpolation output */ - /* Reuse mixing function from Expand */ - WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (int16_t*) pw16_vec1, - (int16_t*) pw16_vec2, &w16_startfact, w16_inc, w16_bestIndex); - - /* Move the last part (also unmodified) */ - /* Take from decoded at 15 ms + 1 pitch period */ - pw16_vec2 = &pw16_decoded[fsMult120 + w16_bestIndex]; - WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[fsMult120], pw16_vec2, - (int16_t) (len - fsMult120 - w16_bestIndex)); - - /* Set the mode flag */ - if (w16_VAD) - { - inst->w16_mode = MODE_SUCCESS_ACCELERATE; - } - else - { - inst->w16_mode = MODE_LOWEN_ACCELERATE; - } - - /* Calculate resulting length = original length - pitch period */ - *pw16_len = len - w16_bestIndex; - - /* Update in-call statistics */ - inst->statInst.accelerateLength += w16_bestIndex; - /* Short-term activity statistics. */ - inst->activity_stats.accelarate_normal_samples += w16_bestIndex; - - return 0; - } - else - { - /* Accelerate not allowed */ - -#ifdef NETEQ_STEREO - /* Signal to slave(s) that this was unsuccessful */ - if (msInfo->msMode == NETEQ_MASTER) - { - msInfo->extraInfo = ACC_FAIL; - } -#endif - - /* Set mode flag to unsuccessful accelerate */ - inst->w16_mode = MODE_UNSUCCESS_ACCELERATE; - - /* Length is unmodified */ - *pw16_len = len; - - /* Simply move all data from decoded to outData */ - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return 0; - } -} - -#undef SCRATCH_PW16_DS_SPEECH -#undef SCRATCH_PW32_CORR -#undef SCRATCH_PW16_CORR diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/accelerate.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.cc index 88cfa4dad9b..6acd778a233 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/accelerate.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/accelerate.h" +#include "webrtc/modules/audio_coding/neteq/accelerate.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" @@ -78,4 +78,11 @@ Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch( } } +Accelerate* AccelerateFactory::Create( + int sample_rate_hz, + size_t num_channels, + const BackgroundNoise& background_noise) const { + return new Accelerate(sample_rate_hz, num_channels, background_noise); +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/accelerate.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.h index 83e3e384543..2da999326a3 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/accelerate.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/accelerate.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_ACCELERATE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_ACCELERATE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_ #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/time_stretch.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/time_stretch.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -64,5 +64,14 @@ class Accelerate : public TimeStretch { DISALLOW_COPY_AND_ASSIGN(Accelerate); }; +struct AccelerateFactory { + AccelerateFactory() {} + virtual ~AccelerateFactory() {} + + virtual Accelerate* Create(int sample_rate_hz, + size_t num_channels, + const BackgroundNoise& background_noise) const; +}; + } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_ACCELERATE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier.cc new file mode 100644 index 00000000000..cc4bc97c30c --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier.cc @@ -0,0 +1,71 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/audio_classifier.h" + +#include <assert.h> +#include <string.h> + +namespace webrtc { + +static const int kDefaultSampleRateHz = 48000; +static const int kDefaultFrameRateHz = 50; +static const int kDefaultFrameSizeSamples = + kDefaultSampleRateHz / kDefaultFrameRateHz; +static const float kDefaultThreshold = 0.5f; + +AudioClassifier::AudioClassifier() + : analysis_info_(), + is_music_(false), + music_probability_(0), + // This actually assigns the pointer to a static constant struct + // rather than creates a struct and |celt_mode_| does not need + // to be deleted. + celt_mode_(opus_custom_mode_create(kDefaultSampleRateHz, + kDefaultFrameSizeSamples, + NULL)), + analysis_state_() { + assert(celt_mode_); +} + +AudioClassifier::~AudioClassifier() {} + +bool AudioClassifier::Analysis(const int16_t* input, + int input_length, + int channels) { + // Must be 20 ms frames at 48 kHz sampling. + assert((input_length / channels) == kDefaultFrameSizeSamples); + + // Only mono or stereo are allowed. + assert(channels == 1 || channels == 2); + + // Call Opus' classifier, defined in + // "third_party/opus/src/src/analysis.h", with lsb_depth = 16. + // Also uses a down-mixing function downmix_int, defined in + // "third_party/opus/src/src/opus_private.h", with + // constants c1 = 0, and c2 = -2. + run_analysis(&analysis_state_, + celt_mode_, + input, + kDefaultFrameSizeSamples, + kDefaultFrameSizeSamples, + 0, + -2, + channels, + kDefaultSampleRateHz, + 16, + downmix_int, + &analysis_info_); + music_probability_ = analysis_info_.music_prob; + is_music_ = music_probability_ > kDefaultThreshold; + return is_music_; +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier.h new file mode 100644 index 00000000000..e7b7807dbed --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier.h @@ -0,0 +1,59 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_ + +#if defined(__cplusplus) +extern "C" { +#endif +#include "celt.h" +#include "analysis.h" +#include "opus_private.h" +#if defined(__cplusplus) +} +#endif + +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +// This class provides a speech/music classification and is a wrapper over the +// Opus classifier. It currently only supports 48 kHz mono or stereo with a +// frame size of 20 ms. + +class AudioClassifier { + public: + AudioClassifier(); + virtual ~AudioClassifier(); + + // Classifies one frame of audio data in input, + // input_length : must be channels * 960; + // channels : must be 1 (mono) or 2 (stereo). + bool Analysis(const int16_t* input, int input_length, int channels); + + // Gets the current classification : true = music, false = speech. + virtual bool is_music() const { return is_music_; } + + // Gets the current music probability. + float music_probability() const { return music_probability_; } + + private: + AnalysisInfo analysis_info_; + bool is_music_; + float music_probability_; + const CELTMode* celt_mode_; + TonalityAnalysisState analysis_state_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc new file mode 100644 index 00000000000..cf623ca0884 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc @@ -0,0 +1,75 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/audio_classifier.h" + +#include <math.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <string> + +#include "gtest/gtest.h" +#include "webrtc/test/testsupport/fileutils.h" + +namespace webrtc { + +static const size_t kFrameSize = 960; + +TEST(AudioClassifierTest, AllZeroInput) { + int16_t in_mono[kFrameSize] = {0}; + + // Test all-zero vectors and let the classifier converge from its default + // to the expected value. + AudioClassifier zero_classifier; + for (int i = 0; i < 100; ++i) { + zero_classifier.Analysis(in_mono, kFrameSize, 1); + } + EXPECT_TRUE(zero_classifier.is_music()); +} + +void RunAnalysisTest(const std::string& audio_filename, + const std::string& data_filename, + size_t channels) { + AudioClassifier classifier; + scoped_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]); + bool is_music_ref; + + FILE* audio_file = fopen(audio_filename.c_str(), "rb"); + ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename + << std::endl; + FILE* data_file = fopen(data_filename.c_str(), "rb"); + ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename + << std::endl; + while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) == + channels * kFrameSize) { + bool is_music = + classifier.Analysis(in.get(), channels * kFrameSize, channels); + EXPECT_EQ(is_music, classifier.is_music()); + ASSERT_EQ(1u, fread(&is_music_ref, sizeof(is_music_ref), 1, data_file)); + EXPECT_EQ(is_music_ref, is_music); + } + fclose(audio_file); + fclose(data_file); +} + +TEST(AudioClassifierTest, DoAnalysisMono) { + RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"), + test::ResourcePath("short_mixed_mono_48", "dat"), + 1); +} + +TEST(AudioClassifierTest, DoAnalysisStereo) { + RunAnalysisTest(test::ResourcePath("short_mixed_stereo_48", "pcm"), + test::ResourcePath("short_mixed_stereo_48", "dat"), + 2); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder.cc index 35422e3f9f5..f539bb2e1ed 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder.cc @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/audio_decoder_impl.h" +#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" namespace webrtc { @@ -41,6 +41,16 @@ int AudioDecoder::PacketDuration(const uint8_t* encoded, size_t encoded_len) { return kNotImplemented; } +int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const { + return kNotImplemented; +} + +bool AudioDecoder::PacketHasFec(const uint8_t* encoded, + size_t encoded_len) const { + return false; +} + NetEqDecoder AudioDecoder::codec_type() const { return codec_type_; } bool AudioDecoder::CodecSupported(NetEqDecoder codec_type) { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc index 5296a1bd0f9..6c7269a35fe 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/audio_decoder_impl.h" +#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" #include <assert.h> #include <string.h> // memmove @@ -458,6 +458,19 @@ int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len, return ret; } +int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded, + size_t encoded_len, int16_t* decoded, + SpeechType* speech_type) { + int16_t temp_type = 1; // Default is speech. + int16_t ret = WebRtcOpus_DecodeFec(static_cast<OpusDecInst*>(state_), encoded, + static_cast<int16_t>(encoded_len), decoded, + &temp_type); + if (ret > 0) + ret *= static_cast<int16_t>(channels_); // Return total number of samples. + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + int AudioDecoderOpus::Init() { return WebRtcOpus_DecoderInitNew(static_cast<OpusDecInst*>(state_)); } @@ -467,6 +480,18 @@ int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, return WebRtcOpus_DurationEst(static_cast<OpusDecInst*>(state_), encoded, static_cast<int>(encoded_len)); } + +int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const { + return WebRtcOpus_FecDurationEst(encoded, static_cast<int>(encoded_len)); +} + +bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded, + size_t encoded_len) const { + int fec; + fec = WebRtcOpus_PacketHasFec(encoded, static_cast<int>(encoded_len)); + return (fec == 1); +} #endif AudioDecoderCng::AudioDecoderCng(enum NetEqDecoder type) diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h index aa35db78082..265d660bd79 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_DECODER_IMPL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_DECODER_IMPL_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_ #include <assert.h> @@ -18,8 +18,8 @@ // selection is made in the gypi file instead of in engine_configurations.h. #include "webrtc/engine_configurations.h" #endif -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -236,8 +236,13 @@ class AudioDecoderOpus : public AudioDecoder { virtual ~AudioDecoderOpus(); virtual int Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type); + virtual int DecodeRedundant(const uint8_t* encoded, size_t encoded_len, + int16_t* decoded, SpeechType* speech_type); virtual int Init(); virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len); + virtual int PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const; + virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; private: DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus); @@ -268,4 +273,4 @@ class AudioDecoderCng : public AudioDecoder { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_DECODER_IMPL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc index dbd9d121f4a..f82644cbc26 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/audio_decoder_impl.h" +#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" #include <assert.h> #include <stdlib.h> diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_unittests.isolate b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate index bb57e74b34f..bb57e74b34f 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_unittests.isolate +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc index baa912c860e..5a208a6972a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include <assert.h> @@ -74,8 +74,7 @@ void AudioMultiVector::PushBackInterleaved(const int16_t* append_this, return; } size_t length_per_channel = length / num_channels_; - int16_t* temp_array = - new int16_t[length_per_channel]; // Intermediate storage. + int16_t* temp_array = new int16_t[length_per_channel]; // Temporary storage. for (size_t channel = 0; channel < num_channels_; ++channel) { // Copy elements to |temp_array|. // Set |source_ptr| to first element of this channel. diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.h index 2d0a7494912..908de936d54 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_MULTI_VECTOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_MULTI_VECTOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_ #include <string.h> // Access to size_t. #include <vector> -#include "webrtc/modules/audio_coding/neteq4/audio_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -131,4 +131,4 @@ class AudioMultiVector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_MULTI_VECTOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc index be05a8260f9..94760385264 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include <assert.h> #include <stdlib.h> diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_vector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.cc index cbd46163068..d9fb4e58c24 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_vector.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/audio_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_vector.h" #include <assert.h> diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_vector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.h index 66bd518a806..f8aabdb8e8e 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_vector.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_VECTOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_VECTOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_ #include <string.h> // Access to size_t. -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" @@ -117,4 +117,4 @@ class AudioVector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_AUDIO_VECTOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_vector_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc index de5aac2d955..50da1fb46c4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/audio_vector_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/audio_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_vector.h" #include <assert.h> #include <stdlib.h> diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/automode.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/automode.c deleted file mode 100644 index 4dbd81ed665..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/automode.c +++ /dev/null @@ -1,783 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the implementation of automatic buffer level optimization. - */ - -#include "automode.h" - -#include <assert.h> - -#include "signal_processing_library.h" - -#include "neteq_defines.h" - -#ifdef NETEQ_DELAY_LOGGING -/* special code for offline delay logging */ -#include <stdio.h> -#include "delay_logging.h" - -extern FILE *delay_fid2; /* file pointer to delay log file */ -#endif /* NETEQ_DELAY_LOGGING */ - -// These two functions are copied from module_common_types.h, but adapted for C. -int WebRtcNetEQ_IsNewerSequenceNumber(uint16_t sequence_number, - uint16_t prev_sequence_number) { - return sequence_number != prev_sequence_number && - ((uint16_t) (sequence_number - prev_sequence_number)) < 0x8000; -} - -int WebRtcNetEQ_IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) { - return timestamp != prev_timestamp && - ((uint32_t) (timestamp - prev_timestamp)) < 0x80000000; -} - -int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen, - uint16_t seqNumber, uint32_t timeStamp, - int32_t fsHz, int mdCodec, int streamingMode) -{ - uint32_t timeIat; /* inter-arrival time */ - int i; - int32_t tempsum = 0; /* temp summation */ - int32_t tempvar; /* temporary variable */ - int retval = 0; /* return value */ - int16_t packetLenSamp; /* packet speech length in samples */ - - /****************/ - /* Sanity check */ - /****************/ - - if (maxBufLen <= 1 || fsHz <= 0) - { - /* maxBufLen must be at least 2 and fsHz must both be strictly positive */ - return -1; - } - - /****************************/ - /* Update packet statistics */ - /****************************/ - - /* Try calculating packet length from current and previous timestamps */ - if (!WebRtcNetEQ_IsNewerTimestamp(timeStamp, inst->lastTimeStamp) || - !WebRtcNetEQ_IsNewerSequenceNumber(seqNumber, inst->lastSeqNo)) - { - /* Wrong timestamp or sequence order; revert to backup plan */ - packetLenSamp = inst->packetSpeechLenSamp; /* use stored value */ - } - else - { - /* calculate timestamps per packet */ - packetLenSamp = (int16_t) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp, - seqNumber - inst->lastSeqNo); - } - - /* Check that the packet size is positive; if not, the statistics cannot be updated. */ - if (inst->firstPacketReceived && packetLenSamp > 0) - { /* packet size ok */ - - /* calculate inter-arrival time in integer packets (rounding down) */ - timeIat = WebRtcSpl_DivW32W16(inst->packetIatCountSamp, packetLenSamp); - - /* Special operations for streaming mode */ - if (streamingMode != 0) - { - /* - * Calculate IAT in Q8, including fractions of a packet (i.e., more accurate - * than timeIat). - */ - int16_t timeIatQ8 = (int16_t) WebRtcSpl_DivW32W16( - WEBRTC_SPL_LSHIFT_W32(inst->packetIatCountSamp, 8), packetLenSamp); - - /* - * Calculate cumulative sum iat with sequence number compensation (ideal arrival - * times makes this sum zero). - */ - inst->cSumIatQ8 += (timeIatQ8 - - WEBRTC_SPL_LSHIFT_W32(seqNumber - inst->lastSeqNo, 8)); - - /* subtract drift term */ - inst->cSumIatQ8 -= CSUM_IAT_DRIFT; - - /* ensure not negative */ - inst->cSumIatQ8 = WEBRTC_SPL_MAX(inst->cSumIatQ8, 0); - - /* remember max */ - if (inst->cSumIatQ8 > inst->maxCSumIatQ8) - { - inst->maxCSumIatQ8 = inst->cSumIatQ8; - inst->maxCSumUpdateTimer = 0; - } - - /* too long since the last maximum was observed; decrease max value */ - if (inst->maxCSumUpdateTimer > (uint32_t) WEBRTC_SPL_MUL_32_16(fsHz, - MAX_STREAMING_PEAK_PERIOD)) - { - inst->maxCSumIatQ8 -= 4; /* remove 1000*4/256 = 15.6 ms/s */ - } - } /* end of streaming mode */ - - /* check for discontinuous packet sequence and re-ordering */ - if (WebRtcNetEQ_IsNewerSequenceNumber(seqNumber, inst->lastSeqNo + 1)) - { - /* Compensate for gap in the sequence numbers. - * Reduce IAT with expected extra time due to lost packets, but ensure that - * the IAT is not negative. - */ - timeIat -= WEBRTC_SPL_MIN(timeIat, - (uint16_t) (seqNumber - (uint16_t) (inst->lastSeqNo + 1))); - } - else if (!WebRtcNetEQ_IsNewerSequenceNumber(seqNumber, inst->lastSeqNo)) - { - /* compensate for re-ordering */ - timeIat += (uint16_t) (inst->lastSeqNo + 1 - seqNumber); - } - - /* saturate IAT at maximum value */ - timeIat = WEBRTC_SPL_MIN( timeIat, MAX_IAT ); - - /* update iatProb = forgetting_factor * iatProb for all elements */ - for (i = 0; i <= MAX_IAT; i++) - { - int32_t tempHi, tempLo; /* Temporary variables */ - - /* - * Multiply iatProbFact (Q15) with iatProb (Q30) and right-shift 15 steps - * to come back to Q30. The operation is done in two steps: - */ - - /* - * 1) Multiply the high 16 bits (15 bits + sign) of iatProb. Shift iatProb - * 16 steps right to get the high 16 bits in a int16_t prior to - * multiplication, and left-shift with 1 afterwards to come back to - * Q30 = (Q15 * (Q30>>16)) << 1. - */ - tempHi = WEBRTC_SPL_MUL_16_16(inst->iatProbFact, - (int16_t) WEBRTC_SPL_RSHIFT_W32(inst->iatProb[i], 16)); - tempHi = WEBRTC_SPL_LSHIFT_W32(tempHi, 1); /* left-shift 1 step */ - - /* - * 2) Isolate and multiply the low 16 bits of iatProb. Right-shift 15 steps - * afterwards to come back to Q30 = (Q15 * Q30) >> 15. - */ - tempLo = inst->iatProb[i] & 0x0000FFFF; /* sift out the 16 low bits */ - tempLo = WEBRTC_SPL_MUL_16_U16(inst->iatProbFact, - (uint16_t) tempLo); - tempLo = WEBRTC_SPL_RSHIFT_W32(tempLo, 15); - - /* Finally, add the high and low parts */ - inst->iatProb[i] = tempHi + tempLo; - - /* Sum all vector elements while we are at it... */ - tempsum += inst->iatProb[i]; - } - - /* - * Increase the probability for the currently observed inter-arrival time - * with 1 - iatProbFact. The factor is in Q15, iatProb in Q30; - * hence, left-shift 15 steps to obtain result in Q30. - */ - inst->iatProb[timeIat] += (32768 - inst->iatProbFact) << 15; - - tempsum += (32768 - inst->iatProbFact) << 15; /* add to vector sum */ - - /* - * Update iatProbFact (changes only during the first seconds after reset) - * The factor converges to IAT_PROB_FACT. - */ - inst->iatProbFact += (IAT_PROB_FACT - inst->iatProbFact + 3) >> 2; - - /* iatProb should sum up to 1 (in Q30). */ - tempsum -= 1 << 30; /* should be zero */ - - /* Check if it does, correct if it doesn't. */ - if (tempsum > 0) - { - /* tempsum too large => decrease a few values in the beginning */ - i = 0; - while (i <= MAX_IAT && tempsum > 0) - { - /* Remove iatProb[i] / 16 from iatProb, but not more than tempsum */ - tempvar = WEBRTC_SPL_MIN(tempsum, inst->iatProb[i] >> 4); - inst->iatProb[i++] -= tempvar; - tempsum -= tempvar; - } - } - else if (tempsum < 0) - { - /* tempsum too small => increase a few values in the beginning */ - i = 0; - while (i <= MAX_IAT && tempsum < 0) - { - /* Add iatProb[i] / 16 to iatProb, but not more than tempsum */ - tempvar = WEBRTC_SPL_MIN(-tempsum, inst->iatProb[i] >> 4); - inst->iatProb[i++] += tempvar; - tempsum += tempvar; - } - } - - /* Calculate optimal buffer level based on updated statistics */ - tempvar = (int32_t) WebRtcNetEQ_CalcOptimalBufLvl(inst, fsHz, mdCodec, timeIat, - streamingMode); - if (tempvar > 0) - { - int high_lim_delay; - /* Convert the minimum delay from milliseconds to packets in Q8. - * |fsHz| is sampling rate in Hertz, and |packetLenSamp| - * is the number of samples per packet (according to the last - * decoding). - */ - int32_t minimum_delay_q8 = ((inst->minimum_delay_ms * - (fsHz / 1000)) << 8) / packetLenSamp; - - int32_t maximum_delay_q8 = ((inst->maximum_delay_ms * - (fsHz / 1000)) << 8) / packetLenSamp; - - inst->optBufLevel = tempvar; - - if (streamingMode != 0) - { - inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel, - inst->maxCSumIatQ8); - } - - /* The required delay. */ - inst->required_delay_q8 = inst->optBufLevel; - - // Maintain the target delay. - inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel, - minimum_delay_q8); - - if (maximum_delay_q8 > 0) { - // Make sure that max is at least one packet length. - maximum_delay_q8 = WEBRTC_SPL_MAX(maximum_delay_q8, (1 << 8)); - inst->optBufLevel = WEBRTC_SPL_MIN(inst->optBufLevel, - maximum_delay_q8); - } - /*********/ - /* Limit */ - /*********/ - - /* Subtract extra delay from maxBufLen */ - if (inst->extraDelayMs > 0 && inst->packetSpeechLenSamp > 0) - { - maxBufLen -= inst->extraDelayMs / inst->packetSpeechLenSamp * fsHz / 1000; - maxBufLen = WEBRTC_SPL_MAX(maxBufLen, 1); // sanity: at least one packet - } - - maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */ - - /* Enforce upper limit; 75% of maxBufLen */ - /* 1/2 + 1/4 = 75% */ - high_lim_delay = (maxBufLen >> 1) + (maxBufLen >> 2); - inst->optBufLevel = WEBRTC_SPL_MIN(inst->optBufLevel, - high_lim_delay); - inst->required_delay_q8 = WEBRTC_SPL_MIN(inst->required_delay_q8, - high_lim_delay); - } - else - { - retval = (int) tempvar; - } - - } /* end if */ - - /*******************************/ - /* Update post-call statistics */ - /*******************************/ - - /* Calculate inter-arrival time in ms = packetIatCountSamp / (fsHz / 1000) */ - timeIat = WEBRTC_SPL_UDIV( - WEBRTC_SPL_UMUL_32_16(inst->packetIatCountSamp, (int16_t) 1000), - (uint32_t) fsHz); - - /* Increase counter corresponding to current inter-arrival time */ - if (timeIat > 2000) - { - inst->countIAT2000ms++; - } - else if (timeIat > 1000) - { - inst->countIAT1000ms++; - } - else if (timeIat > 500) - { - inst->countIAT500ms++; - } - - if (timeIat > inst->longestIATms) - { - /* update maximum value */ - inst->longestIATms = timeIat; - } - - /***********************************/ - /* Prepare for next packet arrival */ - /***********************************/ - - inst->packetIatCountSamp = 0; /* reset inter-arrival time counter */ - - inst->lastSeqNo = seqNumber; /* remember current sequence number */ - - inst->lastTimeStamp = timeStamp; /* remember current timestamp */ - - inst->firstPacketReceived = 1; - - return retval; -} - - -int16_t WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, int32_t fsHz, - int mdCodec, uint32_t timeIatPkts, - int streamingMode) -{ - - int32_t sum1 = 1 << 30; /* assign to 1 in Q30 */ - int16_t B; - uint16_t Bopt; - int i; - int32_t betaInv; /* optimization parameter */ - -#ifdef NETEQ_DELAY_LOGGING - /* special code for offline delay logging */ - int temp_var; -#endif - - /****************/ - /* Sanity check */ - /****************/ - - if (fsHz <= 0) - { - /* fsHz must be strictly positive */ - return -1; - } - - /***********************************************/ - /* Get betaInv parameter based on playout mode */ - /***********************************************/ - - if (streamingMode) - { - /* streaming (listen-only) mode */ - betaInv = AUTOMODE_STREAMING_BETA_INV_Q30; - } - else - { - /* normal mode */ - betaInv = AUTOMODE_BETA_INV_Q30; - } - - /*******************************************************************/ - /* Calculate optimal buffer level without considering jitter peaks */ - /*******************************************************************/ - - /* - * Find the B for which the probability of observing an inter-arrival time larger - * than or equal to B is less than or equal to betaInv. - */ - B = 0; /* start from the beginning of iatProb */ - sum1 -= inst->iatProb[B]; /* ensure that optimal level is not less than 1 */ - - do - { - /* - * Subtract the probabilities one by one until the sum is no longer greater - * than betaInv. - */ - sum1 -= inst->iatProb[++B]; - } - while ((sum1 > betaInv) && (B < MAX_IAT)); - - Bopt = B; /* This is our primary value for the optimal buffer level Bopt */ - - if (mdCodec) - { - /* - * Use alternative cost function when multiple description codec is in use. - * Do not have to re-calculate all points, just back off a few steps from - * previous value of B. - */ - int32_t sum2 = sum1; /* copy sum1 */ - - while ((sum2 <= betaInv + inst->iatProb[Bopt]) && (Bopt > 0)) - { - /* Go backwards in the sum until the modified cost function solution is found */ - sum2 += inst->iatProb[Bopt--]; - } - - Bopt++; /* This is the optimal level when using an MD codec */ - - /* Now, Bopt and B can have different values. */ - } - -#ifdef NETEQ_DELAY_LOGGING - /* special code for offline delay logging */ - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF; - if (fwrite( &temp_var, sizeof(int), 1, delay_fid2 ) != 1) { - return -1; - } - temp_var = (int) (Bopt * inst->packetSpeechLenSamp); -#endif - - /******************************************************************/ - /* Make levelFiltFact adaptive: Larger B <=> larger levelFiltFact */ - /******************************************************************/ - - switch (B) - { - case 0: - case 1: - { - inst->levelFiltFact = 251; - break; - } - case 2: - case 3: - { - inst->levelFiltFact = 252; - break; - } - case 4: - case 5: - case 6: - case 7: - { - inst->levelFiltFact = 253; - break; - } - default: /* B > 7 */ - { - inst->levelFiltFact = 254; - break; - } - } - - /************************/ - /* Peak mode operations */ - /************************/ - - /* Compare current IAT with peak threshold - * - * If IAT > optimal level + threshold (+1 for MD codecs) - * or if IAT > 2 * optimal level (note: optimal level is in Q8): - */ - if (timeIatPkts > (uint32_t) (Bopt + inst->peakThresholdPkt + (mdCodec != 0)) - || timeIatPkts > (uint32_t) WEBRTC_SPL_LSHIFT_U16(Bopt, 1)) - { - /* A peak is observed */ - - if (inst->peakIndex == -1) - { - /* this is the first peak; prepare for next peak */ - inst->peakIndex = 0; - /* set the mode-disable counter */ - inst->peakModeDisabled = WEBRTC_SPL_LSHIFT_W16(1, NUM_PEAKS_REQUIRED-2); - } - else if (inst->peakIatCountSamp - <= - (uint32_t) WEBRTC_SPL_MUL_32_16(fsHz, MAX_PEAK_PERIOD)) - { - /* This is not the first peak and the period time is valid */ - - /* store time elapsed since last peak */ - inst->peakPeriodSamp[inst->peakIndex] = inst->peakIatCountSamp; - - /* saturate height to 16 bits */ - inst->peakHeightPkt[inst->peakIndex] - = - (int16_t) WEBRTC_SPL_MIN(timeIatPkts, WEBRTC_SPL_WORD16_MAX); - - /* increment peakIndex and wrap/modulo */ - inst->peakIndex = (inst->peakIndex + 1) & PEAK_INDEX_MASK; - - /* process peak vectors */ - inst->curPeakHeight = 0; - inst->curPeakPeriod = 0; - - for (i = 0; i < NUM_PEAKS; i++) - { - /* Find maximum of peak heights and peak periods */ - inst->curPeakHeight - = WEBRTC_SPL_MAX(inst->curPeakHeight, inst->peakHeightPkt[i]); - inst->curPeakPeriod - = WEBRTC_SPL_MAX(inst->curPeakPeriod, inst->peakPeriodSamp[i]); - - } - - inst->peakModeDisabled >>= 1; /* decrease mode-disable "counter" */ - - } - else if (inst->peakIatCountSamp > (uint32_t) WEBRTC_SPL_MUL_32_16(fsHz, - WEBRTC_SPL_LSHIFT_W16(MAX_PEAK_PERIOD, 1))) - { - /* - * More than 2 * MAX_PEAK_PERIOD has elapsed since last peak; - * too long time => reset peak statistics - */ - inst->curPeakHeight = 0; - inst->curPeakPeriod = 0; - for (i = 0; i < NUM_PEAKS; i++) - { - inst->peakHeightPkt[i] = 0; - inst->peakPeriodSamp[i] = 0; - } - - inst->peakIndex = -1; /* Next peak is first peak */ - inst->peakIatCountSamp = 0; - } - - inst->peakIatCountSamp = 0; /* Reset peak interval timer */ - } /* end if peak is observed */ - - /* Evaluate peak mode conditions */ - - /* - * If not disabled (enough peaks have been observed) and - * time since last peak is less than two peak periods. - */ - inst->peakFound = 0; - if ((!inst->peakModeDisabled) && (inst->peakIatCountSamp - <= WEBRTC_SPL_LSHIFT_W32(inst->curPeakPeriod , 1))) - { - /* Engage peak mode */ - inst->peakFound = 1; - /* Set optimal buffer level to curPeakHeight (if it's not already larger) */ - Bopt = WEBRTC_SPL_MAX(Bopt, inst->curPeakHeight); - -#ifdef NETEQ_DELAY_LOGGING - /* special code for offline delay logging */ - temp_var = (int) -(Bopt * inst->packetSpeechLenSamp); -#endif - } - - /* Scale Bopt to Q8 */ - Bopt = WEBRTC_SPL_LSHIFT_U16(Bopt,8); - -#ifdef NETEQ_DELAY_LOGGING - /* special code for offline delay logging */ - if (fwrite( &temp_var, sizeof(int), 1, delay_fid2 ) != 1) { - return -1; - } -#endif - - /* Sanity check: Bopt must be strictly positive */ - if (Bopt <= 0) - { - Bopt = WEBRTC_SPL_LSHIFT_W16(1, 8); /* 1 in Q8 */ - } - - return Bopt; /* return value in Q8 */ -} - - -int WebRtcNetEQ_BufferLevelFilter(int32_t curSizeMs8, AutomodeInst_t *inst, - int sampPerCall, int16_t fsMult) -{ - - int16_t curSizeFrames; - - /****************/ - /* Sanity check */ - /****************/ - - if (sampPerCall <= 0 || fsMult <= 0) - { - /* sampPerCall and fsMult must both be strictly positive */ - return -1; - } - - /* Check if packet size has been detected */ - if (inst->packetSpeechLenSamp > 0) - { - /* - * Current buffer level in packet lengths - * = (curSizeMs8 * fsMult) / packetSpeechLenSamp - */ - curSizeFrames = (int16_t) WebRtcSpl_DivW32W16( - WEBRTC_SPL_MUL_32_16(curSizeMs8, fsMult), inst->packetSpeechLenSamp); - } - else - { - curSizeFrames = 0; - } - - /* Filter buffer level */ - if (inst->levelFiltFact > 0) /* check that filter factor is set */ - { - /* Filter: - * buffLevelFilt = levelFiltFact * buffLevelFilt - * + (1-levelFiltFact) * curSizeFrames - * - * levelFiltFact is in Q8 - */ - inst->buffLevelFilt = ((inst->levelFiltFact * inst->buffLevelFilt) >> 8) + - (256 - inst->levelFiltFact) * curSizeFrames; - } - - /* Account for time-scale operations (accelerate and pre-emptive expand) */ - if (inst->prevTimeScale) - { - /* - * Time-scaling has been performed since last filter update. - * Subtract the sampleMemory from buffLevelFilt after converting sampleMemory - * from samples to packets in Q8. Make sure that the filtered value is - * non-negative. - */ - inst->buffLevelFilt = WEBRTC_SPL_MAX( inst->buffLevelFilt - - WebRtcSpl_DivW32W16( - WEBRTC_SPL_LSHIFT_W32(inst->sampleMemory, 8), /* sampleMemory in Q8 */ - inst->packetSpeechLenSamp ), /* divide by packetSpeechLenSamp */ - 0); - - /* - * Reset flag and set timescaleHoldOff timer to prevent further time-scaling - * for some time. - */ - inst->prevTimeScale = 0; - inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT; - } - - /* Update time counters and HoldOff timer */ - inst->packetIatCountSamp += sampPerCall; /* packet inter-arrival time */ - inst->peakIatCountSamp += sampPerCall; /* peak inter-arrival time */ - inst->timescaleHoldOff >>= 1; /* time-scaling limiter */ - inst->maxCSumUpdateTimer += sampPerCall; /* cumulative-sum timer */ - - return 0; - -} - - -int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, int16_t newLenSamp, - int32_t fsHz) -{ - - /* Sanity check for newLenSamp and fsHz */ - if (newLenSamp <= 0 || fsHz <= 0) - { - return -1; - } - - inst->packetSpeechLenSamp = newLenSamp; /* Store packet size in instance */ - - /* Make NetEQ wait for first regular packet before starting the timer */ - inst->lastPackCNGorDTMF = 1; - - inst->packetIatCountSamp = 0; /* Reset packet time counter */ - - /* - * Calculate peak threshold from packet size. The threshold is defined as - * the (fractional) number of packets that corresponds to PEAK_HEIGHT - * (in Q8 seconds). That is, threshold = PEAK_HEIGHT/256 * fsHz / packLen. - */ - inst->peakThresholdPkt = (uint16_t) WebRtcSpl_DivW32W16ResW16( - WEBRTC_SPL_MUL_16_16_RSFT(PEAK_HEIGHT, - (int16_t) WEBRTC_SPL_RSHIFT_W32(fsHz, 6), 2), inst->packetSpeechLenSamp); - - return 0; -} - - -int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets) -{ - - int i; - uint16_t tempprob = 0x4002; /* 16384 + 2 = 100000000000010 binary; */ - - /* Sanity check for maxBufLenPackets */ - if (maxBufLenPackets <= 1) - { - /* Invalid value; set to 10 instead (arbitary small number) */ - maxBufLenPackets = 10; - } - - /* Reset filtered buffer level */ - inst->buffLevelFilt = 0; - - /* Reset packet size to unknown */ - inst->packetSpeechLenSamp = 0; - - /* - * Flag that last packet was special payload, so that automode will treat the next speech - * payload as the first payload received. - */ - inst->lastPackCNGorDTMF = 1; - - /* Reset peak detection parameters */ - inst->peakModeDisabled = 1; /* disable peak mode */ - inst->peakIatCountSamp = 0; - inst->peakIndex = -1; /* indicates that no peak is registered */ - inst->curPeakHeight = 0; - inst->curPeakPeriod = 0; - for (i = 0; i < NUM_PEAKS; i++) - { - inst->peakHeightPkt[i] = 0; - inst->peakPeriodSamp[i] = 0; - } - - /* - * Set the iatProb PDF vector to an exponentially decaying distribution - * iatProb[i] = 0.5^(i+1), i = 0, 1, 2, ... - * iatProb is in Q30. - */ - for (i = 0; i <= MAX_IAT; i++) - { - /* iatProb[i] = 0.5^(i+1) = iatProb[i-1] / 2 */ - tempprob = WEBRTC_SPL_RSHIFT_U16(tempprob, 1); - /* store in PDF vector */ - inst->iatProb[i] = WEBRTC_SPL_LSHIFT_W32((int32_t) tempprob, 16); - } - - /* - * Calculate the optimal buffer level corresponding to the initial PDF. - * No need to call WebRtcNetEQ_CalcOptimalBufLvl() since we have just hard-coded - * all the variables that the buffer level depends on => we know the result - */ - inst->optBufLevel = WEBRTC_SPL_MIN(4, - (maxBufLenPackets >> 1) + (maxBufLenPackets >> 1)); /* 75% of maxBufLenPackets */ - inst->required_delay_q8 = inst->optBufLevel; - inst->levelFiltFact = 253; - - /* - * Reset the iat update forgetting factor to 0 to make the impact of the first - * incoming packets greater. - */ - inst->iatProbFact = 0; - - /* Reset packet inter-arrival time counter */ - inst->packetIatCountSamp = 0; - - /* Clear time-scaling related variables */ - inst->prevTimeScale = 0; - inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT; /* don't allow time-scaling immediately */ - - inst->cSumIatQ8 = 0; - inst->maxCSumIatQ8 = 0; - - return 0; -} - -int32_t WebRtcNetEQ_AverageIAT(const AutomodeInst_t *inst) { - int i; - int32_t sum_q24 = 0; - assert(inst); - for (i = 0; i <= MAX_IAT; ++i) { - /* Shift 6 to fit worst case: 2^30 * 64. */ - sum_q24 += (inst->iatProb[i] >> 6) * i; - } - /* Subtract the nominal inter-arrival time 1 = 2^24 in Q24. */ - sum_q24 -= (1 << 24); - /* - * Multiply with 1000000 / 2^24 = 15625 / 2^18 to get in parts-per-million. - * Shift 7 to Q17 first, then multiply with 15625 and shift another 11. - */ - return ((sum_q24 >> 7) * 15625) >> 11; -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/automode.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/automode.h deleted file mode 100644 index c5dd829b834..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/automode.h +++ /dev/null @@ -1,274 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the functionality for automatic buffer level optimization. - */ - -#ifndef AUTOMODE_H -#define AUTOMODE_H - -#include "typedefs.h" - -/*************/ -/* Constants */ -/*************/ - -/* The beta parameter defines the trade-off between delay and underrun probability. */ -/* It is defined through its inverse in Q30 */ -#define AUTOMODE_BETA_INV_Q30 53687091 /* 1/20 in Q30 */ -#define AUTOMODE_STREAMING_BETA_INV_Q30 536871 /* 1/2000 in Q30 */ - -/* Forgetting factor for the inter-arrival time statistics */ -#define IAT_PROB_FACT 32745 /* 0.9993 in Q15 */ - -/* Maximum inter-arrival time to register (in "packet-times") */ -#define MAX_IAT 64 -#define PEAK_HEIGHT 20 /* 0.08s in Q8 */ - -/* The value (1<<5) sets maximum accelerate "speed" to about 100 ms/s */ -#define AUTOMODE_TIMESCALE_LIMIT (1<<5) - -/* Peak mode related parameters */ -/* Number of peaks in peak vector; must be a power of 2 */ -#define NUM_PEAKS 8 - -/* Must be NUM_PEAKS-1 */ -#define PEAK_INDEX_MASK 0x0007 - -/* Longest accepted peak distance */ -#define MAX_PEAK_PERIOD 10 -#define MAX_STREAMING_PEAK_PERIOD 600 /* 10 minutes */ - -/* Number of peaks required before peak mode can be engaged */ -#define NUM_PEAKS_REQUIRED 3 - -/* Drift term for cumulative sum */ -#define CSUM_IAT_DRIFT 2 - -/*******************/ -/* Automode struct */ -/*******************/ - -/* The automode struct is a sub-struct of the - bufstats-struct (BufstatsInst_t). */ - -typedef struct -{ - - /* Filtered current buffer level */ - uint16_t levelFiltFact; /* filter forgetting factor in Q8 */ - int buffLevelFilt; /* filtered buffer level in Q8 */ - - /* Inter-arrival time (iat) statistics */ - int32_t iatProb[MAX_IAT + 1]; /* iat probabilities in Q30 */ - int16_t iatProbFact; /* iat forgetting factor in Q15 */ - uint32_t packetIatCountSamp; /* time (in timestamps) elapsed since last - packet arrival, based on RecOut calls */ - int optBufLevel; /* current optimal buffer level in Q8 */ - - /* Packet related information */ - int16_t packetSpeechLenSamp; /* speech samples per incoming packet */ - int16_t lastPackCNGorDTMF; /* indicates that the last received packet - contained special information */ - uint16_t lastSeqNo; /* sequence number for last packet received */ - uint32_t lastTimeStamp; /* timestamp for the last packet received */ - int firstPacketReceived; /* set to zero implicitly when the instance is - filled with zeros */ - int32_t sampleMemory; /* memory position for keeping track of how many - samples we cut during expand */ - int16_t prevTimeScale; /* indicates that the last mode was an accelerate - or pre-emptive expand operation */ - uint32_t timescaleHoldOff; /* counter that is shifted one step right each - RecOut call; time-scaling allowed when it has - reached 0 */ - int16_t extraDelayMs; /* extra delay for sync with video */ - - int minimum_delay_ms; /* Desired delay, NetEq maintains this amount of - delay unless jitter statistics suggests a higher value. */ - int maximum_delay_ms; /* Max desired delay, NetEq will not go above this - amount of delay even if jitter statistics suggests a higher value. */ - - int required_delay_q8; /* Smallest delay required. This is computed - according to inter-arrival time and playout mode. It has the same unit - as |optBufLevel|. */ - - /* Peak-detection */ - /* vector with the latest peak periods (peak spacing in samples) */ - uint32_t peakPeriodSamp[NUM_PEAKS]; - /* vector with the latest peak heights (in packets) */ - int16_t peakHeightPkt[NUM_PEAKS]; - int16_t peakIndex; /* index for the vectors peakPeriodSamp and peakHeightPkt; - -1 if still waiting for first peak */ - uint16_t peakThresholdPkt; /* definition of peak (in packets); - calculated from PEAK_HEIGHT */ - uint32_t peakIatCountSamp; /* samples elapsed since last peak was observed */ - uint32_t curPeakPeriod; /* current maximum of peakPeriodSamp vector */ - int16_t curPeakHeight; /* derived from peakHeightPkt vector; - used as optimal buffer level in peak mode */ - int16_t peakModeDisabled; /* ==0 if peak mode can be engaged; >0 if not */ - uint16_t peakFound; /* 1 if peaks are detected and extra delay is applied; - * 0 otherwise. */ - - /* Post-call statistics */ - uint32_t countIAT500ms; /* number of times we got small network outage */ - uint32_t countIAT1000ms; /* number of times we got medium network outage */ - uint32_t countIAT2000ms; /* number of times we got large network outage */ - uint32_t longestIATms; /* mSec duration of longest network outage */ - - int16_t cSumIatQ8; /* cumulative sum of inter-arrival times */ - int16_t maxCSumIatQ8; /* max cumulative sum IAT */ - uint32_t maxCSumUpdateTimer;/* time elapsed since maximum was observed */ -} AutomodeInst_t; - -/*************/ -/* Functions */ -/*************/ - -/**************************************************************************** - * WebRtcNetEQ_UpdateIatStatistics(...) - * - * Update the packet inter-arrival time statistics when a new packet arrives. - * This function should be called for every arriving packet, with some - * exceptions when using DTX/VAD and DTMF. A new optimal buffer level is - * calculated after the update. - * - * Input: - * - inst : Automode instance - * - maxBufLen : Maximum number of packets the buffer can hold - * - seqNumber : RTP sequence number of incoming packet - * - timeStamp : RTP timestamp of incoming packet - * - fsHz : Sample rate in Hz - * - mdCodec : Non-zero if the current codec is a multiple- - * description codec - * - streamingMode : A non-zero value will increase jitter robustness (and delay) - * - * Output: - * - inst : Updated automode instance - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen, - uint16_t seqNumber, uint32_t timeStamp, - int32_t fsHz, int mdCodec, int streamingMode); - -/**************************************************************************** - * WebRtcNetEQ_CalcOptimalBufLvl(...) - * - * Calculate the optimal buffer level based on packet inter-arrival time - * statistics. - * - * Input: - * - inst : Automode instance - * - fsHz : Sample rate in Hz - * - mdCodec : Non-zero if the current codec is a multiple- - * description codec - * - timeIatPkts : Currently observed inter-arrival time in packets - * - streamingMode : A non-zero value will increase jitter robustness (and delay) - * - * Output: - * - inst : Updated automode instance - * - * Return value : >0 - Optimal buffer level - * <0 - Error - */ - -int16_t WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, int32_t fsHz, - int mdCodec, uint32_t timeIatPkts, - int streamingMode); - -/**************************************************************************** - * WebRtcNetEQ_BufferLevelFilter(...) - * - * Update filtered buffer level. The function must be called once for each - * RecOut call, since the timing of automode hinges on counters that are - * updated by this function. - * - * Input: - * - curSizeMs8 : Total length of unused speech data in packet buffer - * and sync buffer, in ms * 8 - * - inst : Automode instance - * - sampPerCall : Number of samples per RecOut call - * - fsMult : Sample rate in Hz divided by 8000 - * - * Output: - * - inst : Updated automode instance - * - * Return value : 0 - Ok - * : <0 - Error - */ - -int WebRtcNetEQ_BufferLevelFilter(int32_t curSizeMs8, AutomodeInst_t *inst, - int sampPerCall, int16_t fsMult); - -/**************************************************************************** - * WebRtcNetEQ_SetPacketSpeechLen(...) - * - * Provide the number of speech samples extracted from a packet to the - * automode instance. Several of the calculations within automode depend - * on knowing the packet size. - * - * - * Input: - * - inst : Automode instance - * - newLenSamp : Number of samples per RecOut call - * - fsHz : Sample rate in Hz - * - * Output: - * - inst : Updated automode instance - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, int16_t newLenSamp, - int32_t fsHz); - -/**************************************************************************** - * WebRtcNetEQ_ResetAutomode(...) - * - * Reset the automode instance. - * - * - * Input: - * - inst : Automode instance - * - maxBufLenPackets : Maximum number of packets that the packet - * buffer can hold (>1) - * - * Output: - * - inst : Updated automode instance - * - * Return value : 0 - Ok - */ - -int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets); - -/**************************************************************************** - * WebRtcNetEQ_AverageIAT(...) - * - * Calculate the average inter-arrival time based on current statistics. - * The average is expressed in parts per million relative the nominal. That is, - * if the average inter-arrival time is equal to the nominal frame time, - * the return value is zero. A positive value corresponds to packet spacing - * being too large, while a negative value means that the packets arrive with - * less spacing than expected. - * - * - * Input: - * - inst : Automode instance. - * - * Return value : Average relative inter-arrival time in samples. - */ - -int32_t WebRtcNetEQ_AverageIAT(const AutomodeInst_t *inst); - -#endif /* AUTOMODE_H */ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/background_noise.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.cc index 2dfb3c1f392..e00c4f65e5c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/background_noise.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" #include <assert.h> #include <string.h> // memcpy @@ -16,8 +16,8 @@ #include <algorithm> // min, max #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/post_decode_vad.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/background_noise.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.h index ac5446bf7f6..8fb310ea8b6 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/background_noise.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_BACKGROUND_NOISE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_BACKGROUND_NOISE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_ #include <string.h> // size_t -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" @@ -126,7 +126,7 @@ class BackgroundNoise { int32_t residual_energy); size_t num_channels_; - scoped_array<ChannelParameters> channel_parameters_; + scoped_ptr<ChannelParameters[]> channel_parameters_; bool initialized_; NetEqBackgroundNoiseMode mode_; @@ -134,4 +134,4 @@ class BackgroundNoise { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_BACKGROUND_NOISE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/background_noise_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise_unittest.cc index eb7b9fa1edd..0aee62c97be 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/background_noise_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise_unittest.cc @@ -10,7 +10,7 @@ // Unit tests for BackgroundNoise class. -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" #include "gtest/gtest.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/bgn_update.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/bgn_update.c deleted file mode 100644 index 4d660ff55b6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/bgn_update.c +++ /dev/null @@ -1,247 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the function for updating the background noise estimate. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" - -/* Scratch usage: - Designed for BGN_LPC_ORDER <= 10 - - Type Name size startpos endpos - int32_t pw32_autoCorr 22 0 21 (Length (BGN_LPC_ORDER + 1)*2) - int16_t pw16_tempVec 10 22 31 (Length BGN_LPC_ORDER) - int16_t pw16_rc 10 32 41 (Length BGN_LPC_ORDER) - int16_t pw16_outVec 74 0 73 (Length BGN_LPC_ORDER + 64) - - Total: 74 - */ - -#if (BGN_LPC_ORDER > 10) && (defined SCRATCH) -#error BGN_LPC_ORDER is too large for current scratch memory allocation -#endif - -#define SCRATCH_PW32_AUTO_CORR 0 -#define SCRATCH_PW16_TEMP_VEC 22 -#define SCRATCH_PW16_RC 32 -#define SCRATCH_PW16_OUT_VEC 0 - -#define NETEQFIX_BGNFRAQINCQ16 229 /* 0.0035 in Q16 */ - -/**************************************************************************** - * WebRtcNetEQ_BGNUpdate(...) - * - * This function updates the background noise parameter estimates. - * - * Input: - * - inst : NetEQ instance, where the speech history is stored. - * - scratchPtr : Pointer to scratch vector. - * - * Output: - * - inst : Updated information about the BGN characteristics. - * - * Return value : No return value - */ - -void WebRtcNetEQ_BGNUpdate( -#ifdef SCRATCH - DSPInst_t *inst, int16_t *pw16_scratchPtr -#else - DSPInst_t *inst -#endif -) -{ - const int16_t w16_vecLen = 256; - BGNInst_t *BGN_Inst = &(inst->BGNInst); -#ifdef SCRATCH - int32_t *pw32_autoCorr = (int32_t*) (pw16_scratchPtr + SCRATCH_PW32_AUTO_CORR); - int16_t *pw16_tempVec = pw16_scratchPtr + SCRATCH_PW16_TEMP_VEC; - int16_t *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC; - int16_t *pw16_outVec = pw16_scratchPtr + SCRATCH_PW16_OUT_VEC; -#else - int32_t pw32_autoCorr[BGN_LPC_ORDER + 1]; - int16_t pw16_tempVec[BGN_LPC_ORDER]; - int16_t pw16_outVec[BGN_LPC_ORDER + 64]; - int16_t pw16_rc[BGN_LPC_ORDER]; -#endif - int16_t pw16_A[BGN_LPC_ORDER + 1]; - int32_t w32_tmp; - int16_t *pw16_vec; - int16_t w16_maxSample; - int16_t w16_tmp, w16_tmp2; - int16_t w16_enSampleShift; - int32_t w32_en, w32_enBGN; - int32_t w32_enUpdateThreashold; - int16_t stability; - - pw16_vec = inst->pw16_speechHistory + inst->w16_speechHistoryLen - w16_vecLen; - -#ifdef NETEQ_VAD - if( !inst->VADInst.VADEnabled /* we are not using post-decode VAD */ - || inst->VADInst.VADDecision == 0 ) - { /* ... or, post-decode VAD says passive speaker */ -#endif /* NETEQ_VAD */ - - /*Insert zeros to guarantee that boundary values do not distort autocorrelation */ - WEBRTC_SPL_MEMCPY_W16(pw16_tempVec, pw16_vec - BGN_LPC_ORDER, BGN_LPC_ORDER); - WebRtcSpl_MemSetW16(pw16_vec - BGN_LPC_ORDER, 0, BGN_LPC_ORDER); - - w16_maxSample = WebRtcSpl_MaxAbsValueW16(pw16_vec, w16_vecLen); - w16_tmp = 8 /* log2(w16_veclen) = 8 */ - - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_maxSample, w16_maxSample)); - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - WebRtcNetEQ_CrossCorr(pw32_autoCorr, pw16_vec, pw16_vec, w16_vecLen, BGN_LPC_ORDER + 1, - w16_tmp, -1); - - /* Copy back data */ - WEBRTC_SPL_MEMCPY_W16(pw16_vec - BGN_LPC_ORDER, pw16_tempVec, BGN_LPC_ORDER); - - w16_enSampleShift = 8 - w16_tmp; /* Number of shifts to get energy/sample */ - /* pw32_autoCorr[0]>>w16_enSampleShift */ - w32_en = WEBRTC_SPL_RSHIFT_W32(pw32_autoCorr[0], w16_enSampleShift); - if ((w32_en < BGN_Inst->w32_energyUpdate -#ifdef NETEQ_VAD - /* post-decode VAD disabled and w32_en sufficiently low */ - && !inst->VADInst.VADEnabled) - /* ... or, post-decode VAD says passive speaker */ - || (inst->VADInst.VADEnabled && inst->VADInst.VADDecision == 0) -#else - ) /* just close the extra parenthesis */ -#endif /* NETEQ_VAD */ - ) - { - /* Generate LPC coefficients */ - if (pw32_autoCorr[0] > 0) - { - /* regardless of whether the filter is actually updated or not, - update energy threshold levels, since we have in fact observed - a low energy signal */ - if (w32_en < BGN_Inst->w32_energyUpdate) - { - /* Never get under 1.0 in average sample energy */ - BGN_Inst->w32_energyUpdate = WEBRTC_SPL_MAX(w32_en, 1); - BGN_Inst->w32_energyUpdateLow = 0; - } - - stability = WebRtcSpl_LevinsonDurbin(pw32_autoCorr, pw16_A, pw16_rc, BGN_LPC_ORDER); - /* Only update BGN if filter is stable */ - if (stability != 1) - { - return; - } - } - else - { - /* Do not update */ - return; - } - /* Generate the CNG gain factor by looking at the energy of the residual */ - WebRtcSpl_FilterMAFastQ12(pw16_vec + w16_vecLen - 64, pw16_outVec, pw16_A, - BGN_LPC_ORDER + 1, 64); - w32_enBGN = WebRtcNetEQ_DotW16W16(pw16_outVec, pw16_outVec, 64, 0); - /* Dot product should never overflow since it is BGN and residual! */ - - /* - * Check spectral flatness - * Comparing the residual variance with the input signal variance tells - * if the spectrum is flat or not. - * (20*w32_enBGN) >= (w32_en<<6) - * Also ensure that the energy is non-zero. - */ - if ((WEBRTC_SPL_MUL_32_16(w32_enBGN, 20) >= WEBRTC_SPL_LSHIFT_W32(w32_en, 6)) - && (w32_en > 0)) - { - /* spectrum is flat enough; save filter parameters */ - - WEBRTC_SPL_MEMCPY_W16(BGN_Inst->pw16_filter, pw16_A, BGN_LPC_ORDER+1); - WEBRTC_SPL_MEMCPY_W16(BGN_Inst->pw16_filterState, - pw16_vec + w16_vecLen - BGN_LPC_ORDER, BGN_LPC_ORDER); - - /* Save energy level */ - BGN_Inst->w32_energy = WEBRTC_SPL_MAX(w32_en, 1); - - /* Update energy threshold levels */ - /* Never get under 1.0 in average sample energy */ - BGN_Inst->w32_energyUpdate = WEBRTC_SPL_MAX(w32_en, 1); - BGN_Inst->w32_energyUpdateLow = 0; - - /* Normalize w32_enBGN to 29 or 30 bits before sqrt */ - w16_tmp2 = WebRtcSpl_NormW32(w32_enBGN) - 1; - if (w16_tmp2 & 0x1) - { - w16_tmp2 -= 1; /* Even number of shifts required */ - } - w32_enBGN = WEBRTC_SPL_SHIFT_W32(w32_enBGN, w16_tmp2); - - /* Calculate scale and shift factor */ - BGN_Inst->w16_scale = (int16_t) WebRtcSpl_SqrtFloor(w32_enBGN); - BGN_Inst->w16_scaleShift = 13 + ((6 + w16_tmp2) >> 1); /* RANDN table is in Q13, */ - /* 6=log2(64) */ - - BGN_Inst->w16_initialized = 1; - } - - } - else - { - /* - * Will only happen if post-decode VAD is disabled and w32_en is not low enough. - * Increase the threshold for update so that it increases by a factor 4 in four - * seconds. - * energy = energy * 1.0035 - */ - w32_tmp = WEBRTC_SPL_MUL_16_16_RSFT(NETEQFIX_BGNFRAQINCQ16, - BGN_Inst->w32_energyUpdateLow, 16); - w32_tmp += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16, - (int16_t)(BGN_Inst->w32_energyUpdate & 0xFF)); - w32_tmp += (WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16, - (int16_t)((BGN_Inst->w32_energyUpdate>>8) & 0xFF)) << 8); - BGN_Inst->w32_energyUpdateLow += w32_tmp; - - BGN_Inst->w32_energyUpdate += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16, - (int16_t)(BGN_Inst->w32_energyUpdate>>16)); - BGN_Inst->w32_energyUpdate += BGN_Inst->w32_energyUpdateLow >> 16; - BGN_Inst->w32_energyUpdateLow = (BGN_Inst->w32_energyUpdateLow & 0x0FFFF); - - /* Update maximum energy */ - /* Decrease by a factor 1/1024 each time */ - BGN_Inst->w32_energyMax = BGN_Inst->w32_energyMax - (BGN_Inst->w32_energyMax >> 10); - if (w32_en > BGN_Inst->w32_energyMax) - { - BGN_Inst->w32_energyMax = w32_en; - } - - /* Set update level to at the minimum 60.21dB lower then the maximum energy */ - w32_enUpdateThreashold = (BGN_Inst->w32_energyMax + 524288) >> 20; - if (w32_enUpdateThreashold > BGN_Inst->w32_energyUpdate) - { - BGN_Inst->w32_energyUpdate = w32_enUpdateThreashold; - } - } - -#ifdef NETEQ_VAD -} /* closing initial if-statement */ -#endif /* NETEQ_VAD */ - - return; -} - -#undef SCRATCH_PW32_AUTO_CORR -#undef SCRATCH_PW16_TEMP_VEC -#undef SCRATCH_PW16_RC -#undef SCRATCH_PW16_OUT_VEC - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc index 70b49310660..0388b195024 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" #include <algorithm> // Provide access to std::max. diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter.h index 282ab7a2221..48f7f564c98 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_BUFFER_LEVEL_FILTER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_BUFFER_LEVEL_FILTER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_ -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" namespace webrtc { @@ -44,4 +44,4 @@ class BufferLevelFilter { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_BUFFER_LEVEL_FILTER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc index ddaf08d1d38..9589099d4a0 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc @@ -10,7 +10,7 @@ // Unit tests for BufferLevelFilter class. -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" #include <math.h> // Access to pow function. diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_stats.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_stats.h deleted file mode 100644 index 722f477ea04..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_stats.h +++ /dev/null @@ -1,95 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Calculates and stores the packet buffer statistics. - */ - -#ifndef BUFFER_STATS_H -#define BUFFER_STATS_H - -#include "automode.h" -#include "webrtc_neteq.h" /* to define enum WebRtcNetEQPlayoutMode */ - -/* NetEQ related decisions */ -#define BUFSTATS_DO_NORMAL 0 -#define BUFSTATS_DO_ACCELERATE 1 -#define BUFSTATS_DO_MERGE 2 -#define BUFSTATS_DO_EXPAND 3 -#define BUFSTAT_REINIT 4 -#define BUFSTATS_DO_RFC3389CNG_PACKET 5 -#define BUFSTATS_DO_RFC3389CNG_NOPACKET 6 -#define BUFSTATS_DO_INTERNAL_CNG_NOPACKET 7 -#define BUFSTATS_DO_PREEMPTIVE_EXPAND 8 -#define BUFSTAT_REINIT_DECODER 9 -#define BUFSTATS_DO_DTMF_ONLY 10 -/* Decisions related to when NetEQ is switched off (or in FAX mode) */ -#define BUFSTATS_DO_ALTERNATIVE_PLC 11 -#define BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS 12 -#define BUFSTATS_DO_AUDIO_REPETITION 13 -#define BUFSTATS_DO_AUDIO_REPETITION_INC_TS 14 - -/* Reinit decoder states after this number of expands (upon arrival of new packet) */ -#define REINIT_AFTER_EXPANDS 100 - -/* Wait no longer than this number of RecOut calls before using an "early" packet */ -#define MAX_WAIT_FOR_PACKET 10 - -/* CNG modes */ -#define CNG_OFF 0 -#define CNG_RFC3389_ON 1 -#define CNG_INTERNAL_ON 2 - -typedef struct -{ - - /* store statistical data here */ - int16_t w16_cngOn; /* remember if CNG is interrupted by other event (e.g. DTMF) */ - int16_t w16_noExpand; - int32_t uw32_CNGplayedTS; - - /* VQmon data */ - uint16_t avgDelayMsQ8; - int16_t maxDelayMs; - - AutomodeInst_t Automode_inst; - -} BufstatsInst_t; - -/**************************************************************************** - * WebRtcNetEQ_BufstatsDecision() - * - * Gives a decision about what action that is currently desired - * - * - * Input: - * inst: The bufstat instance - * cur_size: Current buffer size in ms in Q3 domain - * targetTS: The desired timestamp to start playout from - * availableTS: The closest future value available in buffer - * noPacket 1 if no packet is available, makes availableTS undefined - * prevPlayMode mode of last NetEq playout - * timestampsPerCall number of timestamp for 10ms - * - * Output: - * Returns: A decision, as defined above (see top of file) - * - */ - -uint16_t WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, int16_t frameSize, - int32_t cur_size, uint32_t targetTS, - uint32_t availableTS, int noPacket, - int cngPacket, int prevPlayMode, - enum WebRtcNetEQPlayoutMode playoutMode, - int timestampsPerCall, int NoOfExpandCalls, - int16_t fs_mult, - int16_t lastModeBGNonly, int playDtmf); - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c deleted file mode 100644 index 352e0507746..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c +++ /dev/null @@ -1,427 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the function where the main decision logic for buffer level - * adaptation happens. - */ - -#include "buffer_stats.h" - -#include <assert.h> - -#include "signal_processing_library.h" - -#include "automode.h" -#include "neteq_defines.h" -#include "neteq_error_codes.h" -#include "webrtc_neteq.h" - -#define NETEQ_BUFSTAT_20MS_Q7 2560 /* = 20 ms in Q7 */ - -uint16_t WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, int16_t frameSize, - int32_t cur_size, uint32_t targetTS, - uint32_t availableTS, int noPacket, - int cngPacket, int prevPlayMode, - enum WebRtcNetEQPlayoutMode playoutMode, - int timestampsPerCall, int NoOfExpandCalls, - int16_t fs_mult, - int16_t lastModeBGNonly, int playDtmf) -{ - - int currentDelayMs; - int32_t currSizeSamples = cur_size; - int extraDelayPacketsQ8 = 0; - - /* Avoid overflow if the buffer size should be really large (cur_size is limited 256ms) */ - int32_t curr_sizeQ7 = WEBRTC_SPL_LSHIFT_W32(cur_size, 4); - int level_limit_hi, level_limit_lo; - - inst->Automode_inst.prevTimeScale &= (prevPlayMode == MODE_SUCCESS_ACCELERATE - || prevPlayMode == MODE_LOWEN_ACCELERATE || prevPlayMode == MODE_SUCCESS_PREEMPTIVE - || prevPlayMode == MODE_LOWEN_PREEMPTIVE); - - if ((prevPlayMode != MODE_RFC3389CNG) && (prevPlayMode != MODE_CODEC_INTERNAL_CNG)) - { - /* - * Do not update buffer history if currently playing CNG - * since it will bias the filtered buffer level. - */ - WebRtcNetEQ_BufferLevelFilter(cur_size, &(inst->Automode_inst), timestampsPerCall, - fs_mult); - } - else - { - /* only update time counters */ - inst->Automode_inst.packetIatCountSamp += timestampsPerCall; /* packet inter-arrival time */ - inst->Automode_inst.peakIatCountSamp += timestampsPerCall; /* peak inter-arrival time */ - inst->Automode_inst.timescaleHoldOff >>= 1; /* time-scaling limiter */ - } - cur_size = WEBRTC_SPL_MIN(curr_sizeQ7, WEBRTC_SPL_WORD16_MAX); - - /* Calculate VQmon related variables */ - /* avgDelay = avgDelay*(511/512) + currentDelay*(1/512) (sample ms delay in Q8) */ - inst->avgDelayMsQ8 = (int16_t) (WEBRTC_SPL_MUL_16_16_RSFT(inst->avgDelayMsQ8,511,9) - + (cur_size >> 9)); - - /* Update maximum delay if needed */ - currentDelayMs = (curr_sizeQ7 >> 7); - if (currentDelayMs > inst->maxDelayMs) - { - inst->maxDelayMs = currentDelayMs; - } - - /* NetEQ is on with normal or steaming mode */ - if (playoutMode == kPlayoutOn || playoutMode == kPlayoutStreaming) - { - /* Guard for errors, so that it should not get stuck in error mode */ - if (prevPlayMode == MODE_ERROR) - { - if (noPacket) - { - return BUFSTATS_DO_EXPAND; - } - else - { - return BUFSTAT_REINIT; - } - } - - if (prevPlayMode != MODE_EXPAND && prevPlayMode != MODE_FADE_TO_BGN) - { - inst->w16_noExpand = 1; - } - else - { - inst->w16_noExpand = 0; - } - - if (cngPacket) - { - /* signed difference between wanted and available TS */ - int32_t diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS; - int32_t optimal_level_samp = (inst->Automode_inst.optBufLevel * - inst->Automode_inst.packetSpeechLenSamp) >> 8; - int32_t excess_waiting_time_samp = -diffTS - optimal_level_samp; - - if (excess_waiting_time_samp > optimal_level_samp / 2) - { - /* The waiting time for this packet will be longer than 1.5 - * times the wanted buffer delay. Advance the clock to cut - * waiting time down to the optimal. - */ - inst->uw32_CNGplayedTS += excess_waiting_time_samp; - diffTS += excess_waiting_time_samp; - } - - if ((diffTS) < 0 && (prevPlayMode == MODE_RFC3389CNG)) - { - /* Not time to play this packet yet. Wait another round before using this - * packet. Keep on playing CNG from previous CNG parameters. */ - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - - /* otherwise, go for the CNG packet now */ - return BUFSTATS_DO_RFC3389CNG_PACKET; - } - - /*Check for expand/cng */ - if (noPacket) - { - if (inst->w16_cngOn == CNG_RFC3389_ON) - { - /* keep on playing CNG */ - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - else if (inst->w16_cngOn == CNG_INTERNAL_ON) - { - /* keep on playing internal CNG */ - return BUFSTATS_DO_INTERNAL_CNG_NOPACKET; - } - else if (playDtmf == 1) - { - /* we have not audio data, but can play DTMF */ - return BUFSTATS_DO_DTMF_ONLY; - } - else - { - /* nothing to play => do Expand */ - return BUFSTATS_DO_EXPAND; - } - } - - /* - * If the expand period was very long, reset NetEQ since it is likely that the - * sender was restarted. - */ - if (NoOfExpandCalls > REINIT_AFTER_EXPANDS) return BUFSTAT_REINIT_DECODER; - - /* Calculate extra delay in Q8 packets */ - if (inst->Automode_inst.extraDelayMs > 0 && inst->Automode_inst.packetSpeechLenSamp - > 0) - { - - /* (extra delay in samples in Q8) */ - extraDelayPacketsQ8 = - ((inst->Automode_inst.extraDelayMs * 8 * fs_mult) << 8) / - inst->Automode_inst.packetSpeechLenSamp; - } - - /* Check if needed packet is available */ - if (targetTS == availableTS) - { - - /* If last mode was not expand, and there is no DTMF to play */ - if (inst->w16_noExpand == 1 && playDtmf == 0) - { - /* If so check for accelerate */ - - level_limit_lo = ((inst->Automode_inst.optBufLevel) >> 1) /* 50 % */ - + ((inst->Automode_inst.optBufLevel) >> 2); /* ... + 25% = 75% */ - - /* set upper limit to optBufLevel, but make sure that window is at least 20ms */ - level_limit_hi = WEBRTC_SPL_MAX(inst->Automode_inst.optBufLevel, - level_limit_lo + - WebRtcSpl_DivW32W16ResW16((WEBRTC_SPL_MUL(20*8, fs_mult) << 8), - inst->Automode_inst.packetSpeechLenSamp)); - - /* if extra delay is non-zero, add it */ - if (extraDelayPacketsQ8 > 0) - { - level_limit_hi += extraDelayPacketsQ8; - level_limit_lo += extraDelayPacketsQ8; - } - - if (((inst->Automode_inst.buffLevelFilt >= level_limit_hi) && - (inst->Automode_inst.timescaleHoldOff == 0)) || - (inst->Automode_inst.buffLevelFilt >= level_limit_hi << 2)) - { - /* - * Buffer level higher than limit and time-scaling allowed, - * OR buffer level _really_ high. - */ - return BUFSTATS_DO_ACCELERATE; - } - else if ((inst->Automode_inst.buffLevelFilt < level_limit_lo) - && (inst->Automode_inst.timescaleHoldOff == 0)) - { - return BUFSTATS_DO_PREEMPTIVE_EXPAND; - } - } - return BUFSTATS_DO_NORMAL; - } - - /* Check for Merge */ - else if (availableTS > targetTS) - { - - /* Check that we do not play a packet "too early" */ - if ((prevPlayMode == MODE_EXPAND) - && (availableTS - targetTS - < (uint32_t) WEBRTC_SPL_MUL_16_16((int16_t)timestampsPerCall, - (int16_t)REINIT_AFTER_EXPANDS)) - && (NoOfExpandCalls < MAX_WAIT_FOR_PACKET) - && (availableTS - > targetTS - + WEBRTC_SPL_MUL_16_16((int16_t)timestampsPerCall, - (int16_t)NoOfExpandCalls)) - && (inst->Automode_inst.buffLevelFilt <= inst->Automode_inst.optBufLevel - + extraDelayPacketsQ8)) - { - if (playDtmf == 1) - { - /* we still have DTMF to play, so do not perform expand */ - return BUFSTATS_DO_DTMF_ONLY; - } - else - { - /* nothing to play */ - return BUFSTATS_DO_EXPAND; - } - } - - /* If previous was CNG period or BGNonly then no merge is needed */ - if ((prevPlayMode == MODE_RFC3389CNG) || (prevPlayMode == MODE_CODEC_INTERNAL_CNG) - || lastModeBGNonly) - { - /* - * Keep the same delay as before the CNG (or maximum 70 ms in buffer as safety - * precaution), but make sure that the number of samples in buffer is no - * higher than 4 times the optimal level. - */ - int32_t diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS; - int val = ((inst->Automode_inst.optBufLevel + - extraDelayPacketsQ8) * - inst->Automode_inst.packetSpeechLenSamp) >> 6; - if (diffTS >= 0 || val < currSizeSamples) - { - /* it is time to play this new packet */ - return BUFSTATS_DO_NORMAL; - } - else - { - /* it is too early to play this new packet => keep on playing CNG */ - if (prevPlayMode == MODE_RFC3389CNG) - { - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - else if (prevPlayMode == MODE_CODEC_INTERNAL_CNG) - { - return BUFSTATS_DO_INTERNAL_CNG_NOPACKET; - } - else if (playDtmf == 1) - { - /* we have not audio data, but can play DTMF */ - return BUFSTATS_DO_DTMF_ONLY; - } - else /* lastModeBGNonly */ - { - /* signal expand, but this will result in BGN again */ - return BUFSTATS_DO_EXPAND; - } - } - } - - /* Do not merge unless we have done a Expand before (for complexity reasons) */ - if ((inst->w16_noExpand == 0) || ((frameSize < timestampsPerCall) && (cur_size - > NETEQ_BUFSTAT_20MS_Q7))) - { - return BUFSTATS_DO_MERGE; - } - else if (playDtmf == 1) - { - /* play DTMF instead of expand */ - return BUFSTATS_DO_DTMF_ONLY; - } - else - { - return BUFSTATS_DO_EXPAND; - } - } - } - else - { /* kPlayoutOff or kPlayoutFax */ - if (cngPacket) - { - if (((int32_t) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0) - { - /* time to play this packet now */ - return BUFSTATS_DO_RFC3389CNG_PACKET; - } - else - { - /* wait before playing this packet */ - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - } - if (noPacket) - { - /* - * No packet => - * 1. If in CNG mode play as usual - * 2. Otherwise use other method to generate data and hold TS value - */ - if (inst->w16_cngOn == CNG_RFC3389_ON) - { - /* keep on playing CNG */ - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - else if (inst->w16_cngOn == CNG_INTERNAL_ON) - { - /* keep on playing internal CNG */ - return BUFSTATS_DO_INTERNAL_CNG_NOPACKET; - } - else - { - /* nothing to play => invent some data to play out */ - if (playoutMode == kPlayoutOff) - { - return BUFSTATS_DO_ALTERNATIVE_PLC; - } - else if (playoutMode == kPlayoutFax) - { - return BUFSTATS_DO_AUDIO_REPETITION; - } - else - { - /* UNDEFINED, should not get here... */ - assert(0); - return BUFSTAT_REINIT; - } - } - } - else if (targetTS == availableTS) - { - return BUFSTATS_DO_NORMAL; - } - else - { - if (((int32_t) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0) - { - return BUFSTATS_DO_NORMAL; - } - else if (playoutMode == kPlayoutOff) - { - /* - * If currently playing CNG, continue with that. Don't increase TS - * since uw32_CNGplayedTS will be increased. - */ - if (inst->w16_cngOn == CNG_RFC3389_ON) - { - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - else if (inst->w16_cngOn == CNG_INTERNAL_ON) - { - return BUFSTATS_DO_INTERNAL_CNG_NOPACKET; - } - else - { - /* - * Otherwise, do PLC and increase TS while waiting for the time to - * play this packet. - */ - return BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS; - } - } - else if (playoutMode == kPlayoutFax) - { - /* - * If currently playing CNG, continue with that don't increase TS since - * uw32_CNGplayedTS will be increased. - */ - if (inst->w16_cngOn == CNG_RFC3389_ON) - { - return BUFSTATS_DO_RFC3389CNG_NOPACKET; - } - else if (inst->w16_cngOn == CNG_INTERNAL_ON) - { - return BUFSTATS_DO_INTERNAL_CNG_NOPACKET; - } - else - { - /* - * Otherwise, do audio repetition and increase TS while waiting for the - * time to play this packet. - */ - return BUFSTATS_DO_AUDIO_REPETITION_INC_TS; - } - } - else - { - /* UNDEFINED, should not get here... */ - assert(0); - return BUFSTAT_REINIT; - } - } - } - /* We should not get here (but sometimes we do anyway...) */ - return BUFSTAT_REINIT; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/cng_internal.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/cng_internal.c deleted file mode 100644 index cb4878fee57..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/cng_internal.c +++ /dev/null @@ -1,155 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the function for obtaining comfort noise from noise parameters - * according to IETF RFC 3389. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" -#include "webrtc_cng.h" - -#include "dsp_helpfunctions.h" -#include "neteq_error_codes.h" - -/**************************************************************************** - * WebRtcNetEQ_Cng(...) - * - * This function produces CNG according to RFC 3389. - * - * Input: - * - inst : NetEQ DSP instance - * - len : Number of samples to produce (max 640 or - * 640 - fsHz*5/8000 for first-time CNG, governed by - * the definition of WEBRTC_CNG_MAX_OUTSIZE_ORDER in - * webrtc_cng.h) - * - * Output: - * - pw16_outData : Output CNG - * - * Return value : 0 - Ok - * <0 - Error - */ - -#ifdef NETEQ_CNG_CODEC -/* Must compile NetEQ with CNG support to enable this function */ - -int WebRtcNetEQ_Cng(DSPInst_t *inst, int16_t *pw16_outData, int len) -{ - int16_t w16_winMute = 0; /* mixing factor for overlap data */ - int16_t w16_winUnMute = 0; /* mixing factor for comfort noise */ - int16_t w16_winMuteInc = 0; /* mixing factor increment (negative) */ - int16_t w16_winUnMuteInc = 0; /* mixing factor increment */ - int i; - - /* - * Check if last RecOut call was other than RFC3389, - * that is, this call is the first of a CNG period. - */ - if (inst->w16_mode != MODE_RFC3389CNG) - { - /* Reset generation and overlap slightly with old data */ - - /* Generate len samples + overlap */ - if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData, - (int16_t) (len + inst->ExpandInst.w16_overlap), 1) < 0) - { - /* error returned */ - return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst); - } - - /* Set windowing parameters depending on sample rate */ - if (inst->fs == 8000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_8KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_8KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_8KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC; -#ifdef NETEQ_WIDEBAND - } - else if (inst->fs == 16000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_16KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_16KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_16KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC; -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - } - else if (inst->fs == 32000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_32KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_32KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_32KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC; -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - } - else if (inst->fs == 48000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_48KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_48KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_48KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC; -#endif - } - else - { - /* Unsupported sample rate (should not be possible) */ - return NETEQ_OTHER_ERROR; - } - - /* Do overlap add between new vector and overlap */ - for (i = 0; i < inst->ExpandInst.w16_overlap; i++) - { - /* overlapVec[i] = WinMute * overlapVec[i] + WinUnMute * outData[i] */ - inst->ExpandInst.pw16_overlapVec[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32( - WEBRTC_SPL_MUL_16_16( - inst->ExpandInst.pw16_overlapVec[i], w16_winMute) + - WEBRTC_SPL_MUL_16_16(pw16_outData[i], w16_winUnMute) - + 16384, 15); /* shift with proper rounding */ - - w16_winMute += w16_winMuteInc; /* decrease mute factor (inc<0) */ - w16_winUnMute += w16_winUnMuteInc; /* increase unmute factor (inc>0) */ - - } - - /* - * Shift the contents of the outData buffer by overlap samples, since we - * already used these first samples in the overlapVec above - */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_outData+inst->ExpandInst.w16_overlap, len); - - } - else - { - /* This is a subsequent CNG call; no special overlap needed */ - - /* Generate len samples */ - if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData, (int16_t) len, 0) < 0) - { - /* error returned */ - return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst); - } - } - - return 0; - -} - -#endif /* NETEQ_CNG_CODEC */ - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db.c deleted file mode 100644 index bb34f5e58d3..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db.c +++ /dev/null @@ -1,782 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of the codec database. - */ - -#include "codec_db.h" - -#include <string.h> /* to define NULL */ - -#include "signal_processing_library.h" - -#include "neteq_error_codes.h" - -/* - * Resets the codec database. - */ - -int WebRtcNetEQ_DbReset(CodecDbInst_t *inst) -{ - int i; - - WebRtcSpl_MemSetW16((int16_t*) inst, 0, - sizeof(CodecDbInst_t) / sizeof(int16_t)); - - for (i = 0; i < NUM_TOTAL_CODECS; i++) - { - inst->position[i] = -1; - } - - for (i = 0; i < NUM_CODECS; i++) - { - inst->payloadType[i] = -1; - } - - for (i = 0; i < NUM_CNG_CODECS; i++) - { - inst->CNGpayloadType[i] = -1; - } - - return 0; -} - -/* - * Adds a new codec to the database. - */ - -int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec, - int16_t payloadType, FuncDecode funcDecode, - FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC, - FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt, - FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch, - FuncUpdBWEst funcUpdBWEst, FuncDurationEst funcDurationEst, - FuncGetErrorCode funcGetErrorCode, void* codec_state, - uint16_t codec_fs) -{ - - int temp; - int insertCNGcodec = 0, overwriteCNGcodec = 0, CNGpos = -1; - -#ifndef NETEQ_RED_CODEC - if (codec == kDecoderRED) - { - return CODEC_DB_UNSUPPORTED_CODEC; - } -#endif - if (((int) codec <= (int) kDecoderReservedStart) || ((int) codec - >= (int) kDecoderReservedEnd)) - { - return CODEC_DB_UNSUPPORTED_CODEC; - } - - if ((codec_fs != 8000) -#ifdef NETEQ_WIDEBAND - &&(codec_fs!=16000) -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - &&(codec_fs!=32000) -#endif -#if defined(NETEQ_48KHZ_WIDEBAND) || defined(NETEQ_OPUS_CODEC) - &&(codec_fs!=48000) -#endif - ) - { - return CODEC_DB_UNSUPPORTED_FS; - } - - /* Ensure that the codec type is supported */ - switch (codec) - { -#ifdef NETEQ_PCM16B_CODEC - case kDecoderPCM16B : - case kDecoderPCM16B_2ch : -#endif -#ifdef NETEQ_G711_CODEC - case kDecoderPCMu : - case kDecoderPCMa : - case kDecoderPCMu_2ch : - case kDecoderPCMa_2ch : -#endif -#ifdef NETEQ_ILBC_CODEC - case kDecoderILBC : -#endif -#ifdef NETEQ_ISAC_CODEC - case kDecoderISAC : -#endif -#ifdef NETEQ_ISAC_SWB_CODEC - case kDecoderISACswb : -#endif -#ifdef NETEQ_ISAC_FB_CODEC - case kDecoderISACfb : -#endif -#ifdef NETEQ_OPUS_CODEC - case kDecoderOpus : -#endif -#ifdef NETEQ_G722_CODEC - case kDecoderG722 : - case kDecoderG722_2ch : -#endif -#ifdef NETEQ_WIDEBAND - case kDecoderPCM16Bwb : - case kDecoderPCM16Bwb_2ch : -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - case kDecoderPCM16Bswb32kHz : - case kDecoderPCM16Bswb32kHz_2ch : -#endif -#ifdef NETEQ_CNG_CODEC - case kDecoderCNG : -#endif -#ifdef NETEQ_ATEVENT_DECODE - case kDecoderAVT : -#endif -#ifdef NETEQ_RED_CODEC - case kDecoderRED : -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - case kDecoderPCM16Bswb48kHz : -#endif -#ifdef NETEQ_ARBITRARY_CODEC - case kDecoderArbitrary: -#endif -#ifdef NETEQ_G729_CODEC - case kDecoderG729: -#endif -#ifdef NETEQ_G729_1_CODEC - case kDecoderG729_1 : -#endif -#ifdef NETEQ_G726_CODEC - case kDecoderG726_16 : - case kDecoderG726_24 : - case kDecoderG726_32 : - case kDecoderG726_40 : -#endif -#ifdef NETEQ_G722_1_CODEC - case kDecoderG722_1_16 : - case kDecoderG722_1_24 : - case kDecoderG722_1_32 : -#endif -#ifdef NETEQ_G722_1C_CODEC - case kDecoderG722_1C_24 : - case kDecoderG722_1C_32 : - case kDecoderG722_1C_48 : -#endif -#ifdef NETEQ_SPEEX_CODEC - case kDecoderSPEEX_8 : - case kDecoderSPEEX_16 : -#endif -#ifdef NETEQ_CELT_CODEC - case kDecoderCELT_32 : - case kDecoderCELT_32_2ch : -#endif -#ifdef NETEQ_GSMFR_CODEC - case kDecoderGSMFR : -#endif -#ifdef NETEQ_AMR_CODEC - case kDecoderAMR : -#endif -#ifdef NETEQ_AMRWB_CODEC - case kDecoderAMRWB : -#endif - { - /* If we end up here, the inserted codec is supported => Do nothing */ - break; - } - default: - { - /* If we get to this point, the inserted codec is not supported */ - return CODEC_DB_UNSUPPORTED_CODEC; - } - } - - /* Check to see if payload type is taken */ - if (WebRtcNetEQ_DbGetCodec(inst, payloadType) > 0) - { - return CODEC_DB_PAYLOAD_TAKEN; - } - - /* Special case for CNG codecs */ - if (codec == kDecoderCNG) - { - /* check if this is first CNG codec to be registered */ - if (WebRtcNetEQ_DbGetPayload(inst, codec) == CODEC_DB_NOT_EXIST2) - { - /* no other CNG codec found */ - insertCNGcodec = 1; - } - - /* find the appropriate insert position in CNG payload vector */ - switch (codec_fs) - { - case 8000: - CNGpos = 0; - /* - * The 8 kHz CNG payload type is the one associated with the regular codec DB - * should override any other setting. - * Overwrite if this isn't the first CNG - */ - overwriteCNGcodec = !insertCNGcodec; - break; -#ifdef NETEQ_WIDEBAND - case 16000: - CNGpos = 1; - break; -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - case 32000: - CNGpos = 2; - break; -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - case 48000: - CNGpos = 3; - break; -#endif - default: - /* If we get to this point, the inserted codec is not supported */ - return CODEC_DB_UNSUPPORTED_CODEC; - } - - /* insert CNG payload type */ - inst->CNGpayloadType[CNGpos] = payloadType; - - } - - if ((codec != kDecoderCNG) || (insertCNGcodec == 1) || (overwriteCNGcodec == 1)) - { - /* Check if we have reached the maximum numbers of simultaneous codecs */ - if (inst->nrOfCodecs == NUM_CODECS) return CODEC_DB_FULL; - - /* Check that codec has not already been initialized to DB => - remove it and reinitialize according to new spec */ - if ((inst->position[codec] != -1) && (overwriteCNGcodec != 1)) - { /* if registering multiple CNG codecs, don't remove, just overwrite */ - WebRtcNetEQ_DbRemove(inst, codec); - } - - if (overwriteCNGcodec == 1) - { - temp = inst->position[codec]; - } - else - { - temp = inst->nrOfCodecs; /* Store this codecs position */ - inst->position[codec] = temp; - inst->nrOfCodecs++; - } - - inst->payloadType[temp] = payloadType; - - /* Copy to database */ - inst->codec_state[temp] = codec_state; - inst->funcDecode[temp] = funcDecode; - inst->funcDecodeRCU[temp] = funcDecodeRCU; - inst->funcAddLatePkt[temp] = funcAddLatePkt; - inst->funcDecodeInit[temp] = funcDecodeInit; - inst->funcDecodePLC[temp] = funcDecodePLC; - inst->funcGetMDinfo[temp] = funcGetMDinfo; - inst->funcGetPitch[temp] = funcGetPitch; - inst->funcUpdBWEst[temp] = funcUpdBWEst; - inst->funcDurationEst[temp] = funcDurationEst; - inst->funcGetErrorCode[temp] = funcGetErrorCode; - inst->codec_fs[temp] = codec_fs; - - } - - return 0; -} - -/* - * Removes a codec from the database. - */ - -int WebRtcNetEQ_DbRemove(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec) -{ - int i; - int pos = -1; - -#ifndef NETEQ_RED_CODEC - if (codec == kDecoderRED) - { - return CODEC_DB_UNSUPPORTED_CODEC; - } -#endif - if (((int) codec <= (int) kDecoderReservedStart) || ((int) codec - >= (int) kDecoderReservedEnd)) - { - return CODEC_DB_UNSUPPORTED_CODEC; - } - - pos = inst->position[codec]; - if (pos == -1) - { - return CODEC_DB_NOT_EXIST4; - } - else - { - /* Remove this codec */ - inst->position[codec] = -1; - for (i = pos; i < (inst->nrOfCodecs - 1); i++) - { - inst->payloadType[i] = inst->payloadType[i + 1]; - inst->codec_state[i] = inst->codec_state[i + 1]; - inst->funcDecode[i] = inst->funcDecode[i + 1]; - inst->funcDecodeRCU[i] = inst->funcDecodeRCU[i + 1]; - inst->funcAddLatePkt[i] = inst->funcAddLatePkt[i + 1]; - inst->funcDecodeInit[i] = inst->funcDecodeInit[i + 1]; - inst->funcDecodePLC[i] = inst->funcDecodePLC[i + 1]; - inst->funcGetMDinfo[i] = inst->funcGetMDinfo[i + 1]; - inst->funcGetPitch[i] = inst->funcGetPitch[i + 1]; - inst->funcDurationEst[i] = inst->funcDurationEst[i + 1]; - inst->funcUpdBWEst[i] = inst->funcUpdBWEst[i + 1]; - inst->funcGetErrorCode[i] = inst->funcGetErrorCode[i + 1]; - inst->codec_fs[i] = inst->codec_fs[i + 1]; - } - inst->payloadType[i] = -1; - inst->codec_state[i] = NULL; - inst->funcDecode[i] = NULL; - inst->funcDecodeRCU[i] = NULL; - inst->funcAddLatePkt[i] = NULL; - inst->funcDecodeInit[i] = NULL; - inst->funcDecodePLC[i] = NULL; - inst->funcGetMDinfo[i] = NULL; - inst->funcGetPitch[i] = NULL; - inst->funcDurationEst[i] = NULL; - inst->funcUpdBWEst[i] = NULL; - inst->funcGetErrorCode[i] = NULL; - inst->codec_fs[i] = 0; - /* Move down all the codecs above this one */ - for (i = 0; i < NUM_TOTAL_CODECS; i++) - { - if (inst->position[i] >= pos) - { - inst->position[i] = inst->position[i] - 1; - } - } - inst->nrOfCodecs--; - - if (codec == kDecoderCNG) - { - /* also remove all registered CNG payload types */ - for (i = 0; i < NUM_CNG_CODECS; i++) - { - inst->CNGpayloadType[i] = -1; - } - } - } - return 0; -} - -/* - * Get the decoder function pointers for a codec. - */ - -int WebRtcNetEQ_DbGetPtrs(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec, - CodecFuncInst_t *ptr_inst) -{ - - int pos = inst->position[codec]; - if ((codec <= kDecoderReservedStart) || (codec >= kDecoderReservedEnd) || (codec - > NUM_TOTAL_CODECS)) - { - /* ERROR */ - pos = -1; - } - if (pos >= 0) - { - ptr_inst->codec_state = inst->codec_state[pos]; - ptr_inst->funcAddLatePkt = inst->funcAddLatePkt[pos]; - ptr_inst->funcDecode = inst->funcDecode[pos]; - ptr_inst->funcDecodeRCU = inst->funcDecodeRCU[pos]; - ptr_inst->funcDecodeInit = inst->funcDecodeInit[pos]; - ptr_inst->funcDecodePLC = inst->funcDecodePLC[pos]; - ptr_inst->funcGetMDinfo = inst->funcGetMDinfo[pos]; - ptr_inst->funcUpdBWEst = inst->funcUpdBWEst[pos]; - ptr_inst->funcGetErrorCode = inst->funcGetErrorCode[pos]; - ptr_inst->codec_fs = inst->codec_fs[pos]; - return 0; - } - else - { - WebRtcSpl_MemSetW16((int16_t*) ptr_inst, 0, - sizeof(CodecFuncInst_t) / sizeof(int16_t)); - return CODEC_DB_NOT_EXIST1; - } -} - -/* - * Returns payload number given a codec identifier. - */ - -int WebRtcNetEQ_DbGetPayload(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codecID) -{ - if (inst->position[codecID] == -1) - return CODEC_DB_NOT_EXIST2; - else - return (inst->payloadType[inst->position[codecID]]); - -} - -/* - * Returns codec identifier given a payload number. - * Returns -1 if the payload type does not exist. - */ - -int WebRtcNetEQ_DbGetCodec(const CodecDbInst_t *inst, int payloadType) -{ - int i, pos; - - for (i = 0; i < NUM_TOTAL_CODECS; i++) - { - pos = inst->position[i]; - if (pos != -1) - { - if (inst->payloadType[pos] == payloadType) return i; - } - } - - /* did not find payload type */ - /* check if it's a CNG codec */ - if (WebRtcNetEQ_DbIsCNGPayload(inst, payloadType)) - { - return kDecoderCNG; - } - - /* found no match */ - return CODEC_DB_NOT_EXIST3; -} - -/* - * Extracts the Payload Split information of the codec with the specified payloadType. - */ - -int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecID, - int codedsize) -{ - - switch (codecID) - { -#ifdef NETEQ_ISAC_CODEC - case kDecoderISAC: -#endif -#ifdef NETEQ_ISAC_SWB_CODEC - case kDecoderISACswb: -#endif -#ifdef NETEQ_ISAC_FB_CODEC - case kDecoderISACfb: -#endif -#ifdef NETEQ_OPUS_CODEC - case kDecoderOpus: -#endif -#ifdef NETEQ_ARBITRARY_CODEC - case kDecoderArbitrary: -#endif -#ifdef NETEQ_AMR_CODEC - case kDecoderAMR: -#endif -#ifdef NETEQ_AMRWB_CODEC - case kDecoderAMRWB: -#endif -#ifdef NETEQ_G726_CODEC - /* Treat G726 as non-splittable to simplify the implementation */ - case kDecoderG726_16: - case kDecoderG726_24: - case kDecoderG726_32: - case kDecoderG726_40: -#endif -#ifdef NETEQ_SPEEX_CODEC - case kDecoderSPEEX_8: - case kDecoderSPEEX_16: -#endif -#ifdef NETEQ_CELT_CODEC - case kDecoderCELT_32 : - case kDecoderCELT_32_2ch : -#endif -#ifdef NETEQ_G729_1_CODEC - case kDecoderG729_1: -#endif - { - /* These codecs' payloads are not splittable */ - inst->deltaBytes = NO_SPLIT; - return 0; - } - - /* - * Sample based coders are a special case. - * In this case, deltaTime signals the number of bytes per timestamp unit times 2 - * in log2 domain. - */ -#if (defined NETEQ_G711_CODEC) - case kDecoderPCMu: - case kDecoderPCMa: - case kDecoderPCMu_2ch: - case kDecoderPCMa_2ch: - { - inst->deltaBytes = -12; - inst->deltaTime = 1; - return 0; - } -#endif -#if (defined NETEQ_G722_CODEC) - case kDecoderG722: - case kDecoderG722_2ch: - { - inst->deltaBytes = -14; - inst->deltaTime = 0; - return 0; - } -#endif -#if (defined NETEQ_PCM16B_CODEC) - case kDecoderPCM16B: - case kDecoderPCM16B_2ch: - { - inst->deltaBytes = -12; - inst->deltaTime = 2; - return 0; - } -#endif -#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_WIDEBAND)) - case kDecoderPCM16Bwb: - case kDecoderPCM16Bwb_2ch: - { - inst->deltaBytes = -14; - inst->deltaTime = 2; - return 0; - } -#endif -#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_32KHZ_WIDEBAND)) - case kDecoderPCM16Bswb32kHz: - case kDecoderPCM16Bswb32kHz_2ch: - { - inst->deltaBytes = -18; - inst->deltaTime = 2; - return 0; - } -#endif -#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_48KHZ_WIDEBAND)) - case kDecoderPCM16Bswb48kHz: - { - inst->deltaBytes = -22; - inst->deltaTime = 2; - return 0; - } -#endif - - /* Splittable payloads */ -#ifdef NETEQ_G722_1_CODEC - case kDecoderG722_1_16: - { - inst->deltaBytes = 40; - inst->deltaTime = 320; - return 0; - } - case kDecoderG722_1_24: - { - inst->deltaBytes = 60; - inst->deltaTime = 320; - return 0; - } - case kDecoderG722_1_32: - { - inst->deltaBytes = 80; - inst->deltaTime = 320; - return 0; - } -#endif -#ifdef NETEQ_G722_1C_CODEC - case kDecoderG722_1C_24: - { - inst->deltaBytes = 60; - inst->deltaTime = 640; - return 0; - } - case kDecoderG722_1C_32: - { - inst->deltaBytes = 80; - inst->deltaTime = 640; - return 0; - } - case kDecoderG722_1C_48: - { - inst->deltaBytes = 120; - inst->deltaTime = 640; - return 0; - } -#endif -#ifdef NETEQ_G729_CODEC - case kDecoderG729: - { - inst->deltaBytes = 10; - inst->deltaTime = 80; - return 0; - } -#endif -#ifdef NETEQ_ILBC_CODEC - case kDecoderILBC: - { - /* Check for splitting of iLBC packets. - * If payload size is a multiple of 50 bytes it should be split into 30ms frames. - * If payload size is a multiple of 38 bytes it should be split into 20ms frames. - * Least common multiplier between 38 and 50 is 950, so the payload size must be less than - * 950 bytes in order to resolve the frames unambiguously. - * Currently max 12 frames in one bundle. - */ - switch (codedsize) - { - case 50: - case 100: - case 150: - case 200: - case 250: - case 300: - case 350: - case 400: - case 450: - case 500: - case 550: - case 600: - { - inst->deltaBytes = 50; - inst->deltaTime = 240; - break; - } - case 38: - case 76: - case 114: - case 152: - case 190: - case 228: - case 266: - case 304: - case 342: - case 380: - case 418: - case 456: - { - inst->deltaBytes = 38; - inst->deltaTime = 160; - break; - } - default: - { - return AMBIGUOUS_ILBC_FRAME_SIZE; /* Something not supported... */ - } - } - return 0; - } -#endif -#ifdef NETEQ_GSMFR_CODEC - case kDecoderGSMFR: - { - inst->deltaBytes = 33; - inst->deltaTime = 160; - return 0; - } -#endif - default: - { /*Unknown codec */ - inst->deltaBytes = NO_SPLIT; - return CODEC_DB_UNKNOWN_CODEC; - } - } /* end of switch */ -} - -/* - * Returns 1 if codec is multiple description, 0 otherwise. - * NOTE: This function is a stub, since there currently are no MD codecs. - */ -int WebRtcNetEQ_DbIsMDCodec(enum WebRtcNetEQDecoder codecID) -{ - if (0) /* Add test for MD codecs here */ - return 1; - else - return 0; -} - -/* - * Returns 1 if payload type is registered as a CNG codec, 0 otherwise - */ -int WebRtcNetEQ_DbIsCNGPayload(const CodecDbInst_t *inst, int payloadType) -{ -#ifdef NETEQ_CNG_CODEC - int i; - - for(i=0; i<NUM_CNG_CODECS; i++) - { - if( (inst->CNGpayloadType[i] != -1) && (inst->CNGpayloadType[i] == payloadType) ) - { - return 1; - } - } -#endif - - return 0; - -} - -/* - * Return the sample rate for the codec with the given payload type, 0 if error - */ -uint16_t WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType) -{ - int i; - CodecFuncInst_t codecInst; - - /* Sanity */ - if (inst == NULL) - { - /* return 0 Hz */ - return 0; - } - - /* Check among CNG payloads */ - for (i = 0; i < NUM_CNG_CODECS; i++) - { - if ((inst->CNGpayloadType[i] != -1) && (inst->CNGpayloadType[i] == payloadType)) - { - switch (i) - { -#ifdef NETEQ_WIDEBAND - case 1: - return 16000; -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - case 2: - return 32000; -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - case 3: - return 48000; -#endif - default: - return 8000; - } - } - } - - /* Not a CNG payload, check the other payloads */ - i = WebRtcNetEQ_DbGetCodec(inst, payloadType); - if (i >= 0) - { - if (WebRtcNetEQ_DbGetPtrs(inst, (enum WebRtcNetEQDecoder) i, &codecInst) != 0) - { - /* Unexpected error, return 0 Hz */ - return 0; - } - return codecInst.codec_fs; - } - - /* If we end up here, we got an error, return 0 Hz */ - return 0; - -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db.h deleted file mode 100644 index cc4b48e6f26..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db.h +++ /dev/null @@ -1,128 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Interface for the codec database. - */ - -#ifndef CODEC_DB_H -#define CODEC_DB_H - -#include "typedefs.h" - -#include "webrtc_neteq.h" -#include "codec_db_defines.h" -#include "neteq_defines.h" - -#if defined(NETEQ_48KHZ_WIDEBAND) - #define NUM_CNG_CODECS 4 -#elif defined(NETEQ_32KHZ_WIDEBAND) - #define NUM_CNG_CODECS 3 -#elif defined(NETEQ_WIDEBAND) - #define NUM_CNG_CODECS 2 -#else - #define NUM_CNG_CODECS 1 -#endif - -typedef struct -{ - - int16_t position[NUM_TOTAL_CODECS]; - int16_t nrOfCodecs; - - int16_t payloadType[NUM_CODECS]; - FuncDecode funcDecode[NUM_CODECS]; - FuncDecode funcDecodeRCU[NUM_CODECS]; - FuncDecodePLC funcDecodePLC[NUM_CODECS]; - FuncDecodeInit funcDecodeInit[NUM_CODECS]; - FuncAddLatePkt funcAddLatePkt[NUM_CODECS]; - FuncGetMDinfo funcGetMDinfo[NUM_CODECS]; - FuncGetPitchInfo funcGetPitch[NUM_CODECS]; - FuncUpdBWEst funcUpdBWEst[NUM_CODECS]; - FuncDurationEst funcDurationEst[NUM_CODECS]; - FuncGetErrorCode funcGetErrorCode[NUM_CODECS]; - void * codec_state[NUM_CODECS]; - uint16_t codec_fs[NUM_CODECS]; - int16_t CNGpayloadType[NUM_CNG_CODECS]; - -} CodecDbInst_t; - -#define NO_SPLIT -1 /* codec payload cannot be split */ - -typedef struct -{ - int16_t deltaBytes; - int16_t deltaTime; -} SplitInfo_t; - -/* - * Resets the codec database. - */ -int WebRtcNetEQ_DbReset(CodecDbInst_t *inst); - -/* - * Adds a new codec to the database. - */ -int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec, - int16_t payloadType, FuncDecode funcDecode, - FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC, - FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt, - FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch, - FuncUpdBWEst funcUpdBWEst, FuncDurationEst funcDurationEst, - FuncGetErrorCode funcGetErrorCode, void* codec_state, - uint16_t codec_fs); - -/* - * Removes a codec from the database. - */ -int WebRtcNetEQ_DbRemove(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec); - -/* - * Get the decoder function pointers for a codec. - */ -int WebRtcNetEQ_DbGetPtrs(CodecDbInst_t *inst, enum WebRtcNetEQDecoder, - CodecFuncInst_t *ptr_inst); - -/* - * Returns payload number given a codec identifier. - */ - -int WebRtcNetEQ_DbGetPayload(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codecID); - -/* - * Returns codec identifier given a payload number. - */ - -int WebRtcNetEQ_DbGetCodec(const CodecDbInst_t *inst, int payloadType); - -/* - * Extracts the Payload Split information of the codec with the specified payloadType. - */ - -int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecID, - int codedsize); - -/* - * Returns 1 if codec is multiple description type, 0 otherwise. - */ -int WebRtcNetEQ_DbIsMDCodec(enum WebRtcNetEQDecoder codecID); - -/* - * Returns 1 if payload type is registered as a CNG codec, 0 otherwise. - */ -int WebRtcNetEQ_DbIsCNGPayload(const CodecDbInst_t *inst, int payloadType); - -/* - * Return the sample rate for the codec with the given payload type, 0 if error. - */ -uint16_t WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType); - -#endif - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db_defines.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db_defines.h deleted file mode 100644 index d97306a333c..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/codec_db_defines.h +++ /dev/null @@ -1,97 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Some definitions related to the codec database. - */ - -#ifndef CODEC_DB_DEFINES_H -#define CODEC_DB_DEFINES_H - -#include "typedefs.h" - -#define NUM_CODECS 47 /* probably too large with the limited set of supported codecs*/ -#define NUM_TOTAL_CODECS kDecoderReservedEnd - -/* - * Pointer to decoder function. - */ -typedef int16_t (*FuncDecode)(void* state, int16_t* encoded, int16_t len, - int16_t* decoded, int16_t* speechType); - -/* - * Pointer to PLC function. - */ -typedef int16_t (*FuncDecodePLC)(void* state, int16_t* decodec, - int16_t frames); - -/* - * Pointer to decoder init function. - */ -typedef int16_t (*FuncDecodeInit)(void* state); - -/* - * Pointer to add late packet function. - */ -typedef int16_t - (*FuncAddLatePkt)(void* state, int16_t* encoded, int16_t len); - -/* - * Pointer to get MD infofunction. - */ -typedef int16_t (*FuncGetMDinfo)(void* state); - -/* - * Pointer to pitch info function. - * Return 0 for unvoiced, -1 if pitch not availiable. - */ -typedef int16_t (*FuncGetPitchInfo)(void* state, int16_t* encoded, - int16_t* length); - -/* - * Pointer to the update bandwidth estimate function - */ -typedef int16_t (*FuncUpdBWEst)(void* state, const uint16_t *encoded, - int32_t packet_size, - uint16_t rtp_seq_number, uint32_t send_ts, - uint32_t arr_ts); - -/* - * Pointer to the frame size estimate function. - * Returns the estimated number of samples in the packet. - */ -typedef int (*FuncDurationEst)(void* state, const uint8_t* payload, - int payload_length_bytes); - -/* - * Pointer to error code function - */ -typedef int16_t (*FuncGetErrorCode)(void* state); - -typedef struct CodecFuncInst_t_ -{ - - FuncDecode funcDecode; - FuncDecode funcDecodeRCU; - FuncDecodePLC funcDecodePLC; - FuncDecodeInit funcDecodeInit; - FuncAddLatePkt funcAddLatePkt; - FuncGetMDinfo funcGetMDinfo; - FuncUpdBWEst funcUpdBWEst; /* Currently in use for the ISAC family (without LC) only*/ - FuncDurationEst funcDurationEst; - FuncGetErrorCode funcGetErrorCode; - void * codec_state; - uint16_t codec_fs; - uint32_t timeStamp; - -} CodecFuncInst_t; - -#endif - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/comfort_noise.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.cc index 360767af193..31bb40c9275 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/comfort_noise.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.cc @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/comfort_noise.h" +#include "webrtc/modules/audio_coding/neteq/comfort_noise.h" #include <assert.h> #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/comfort_noise.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.h index 7e7c294ff0e..d4655962456 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/comfort_noise.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_COMFORT_NOISE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_COMFORT_NOISE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_ -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -70,4 +70,4 @@ class ComfortNoise { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_COMFORT_NOISE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/comfort_noise_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc index 0e849717125..6a1bbe0d143 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/comfort_noise_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc @@ -10,11 +10,11 @@ // Unit tests for ComfortNoise class. -#include "webrtc/modules/audio_coding/neteq4/comfort_noise.h" +#include "webrtc/modules/audio_coding/neteq/comfort_noise.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/correlator.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/correlator.c deleted file mode 100644 index 0a4404a432b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/correlator.c +++ /dev/null @@ -1,132 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" - -/* Scratch usage: - - Type Name size startpos endpos - int16_t pw16_corrVec 62 0 61 - int16_t pw16_data_ds 124 0 123 - int32_t pw32_corr 2*54 124 231 - - Total: 232 - */ - -#define SCRATCH_pw16_corrVec 0 -#define SCRATCH_pw16_data_ds 0 -#define SCRATCH_pw32_corr 124 - -#define NETEQ_CORRELATOR_DSVECLEN 124 /* 124 = 60 + 10 + 54 */ - -int16_t WebRtcNetEQ_Correlator(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_data, - int16_t w16_dataLen, - int16_t *pw16_corrOut, - int16_t *pw16_corrScale) -{ - int16_t w16_corrLen = 60; -#ifdef SCRATCH - int16_t *pw16_data_ds = pw16_scratchPtr + SCRATCH_pw16_corrVec; - int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_pw32_corr); - /* int16_t *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;*/ -#else - int16_t pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN]; - int32_t pw32_corr[54]; - /* int16_t pw16_corrVec[4+54+4];*/ -#endif - /* int16_t *pw16_corr=&pw16_corrVec[4];*/ - int16_t w16_maxVal; - int32_t w32_maxVal; - int16_t w16_normVal; - int16_t w16_normVal2; - /* int16_t w16_corrUpsLen;*/ - int16_t *pw16_B = NULL; - int16_t w16_Blen = 0; - int16_t w16_factor = 0; - - /* Set constants depending on frequency used */ - if (inst->fs == 8000) - { - w16_Blen = 3; - w16_factor = 2; - pw16_B = (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl; -#ifdef NETEQ_WIDEBAND - } - else if (inst->fs==16000) - { - w16_Blen = 5; - w16_factor = 4; - pw16_B = (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl; -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - } - else if (inst->fs==32000) - { - w16_Blen = 7; - w16_factor = 8; - pw16_B = (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl; -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - } - else /* if inst->fs==48000 */ - { - w16_Blen = 7; - w16_factor = 12; - pw16_B = (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl; -#endif - } - - /* Downsample data in order to work on a 4 kHz sampled signal */ - WebRtcSpl_DownsampleFast( - pw16_data + w16_dataLen - (NETEQ_CORRELATOR_DSVECLEN * w16_factor), - (int16_t) (NETEQ_CORRELATOR_DSVECLEN * w16_factor), pw16_data_ds, - NETEQ_CORRELATOR_DSVECLEN, pw16_B, w16_Blen, w16_factor, (int16_t) 0); - - /* Normalize downsampled vector to using entire 16 bit */ - w16_maxVal = WebRtcSpl_MaxAbsValueW16(pw16_data_ds, 124); - w16_normVal = 16 - WebRtcSpl_NormW32((int32_t) w16_maxVal); - WebRtcSpl_VectorBitShiftW16(pw16_data_ds, NETEQ_CORRELATOR_DSVECLEN, pw16_data_ds, - w16_normVal); - - /* Correlate from lag 10 to lag 60 (20..120 in NB and 40..240 in WB) */ - - WebRtcNetEQ_CrossCorr( - pw32_corr, &pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN - w16_corrLen], - &pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN - w16_corrLen - 10], 60, 54, - 6 /*maxValue... shifts*/, -1); - - /* - * Move data from w32 to w16 vector. - * Normalize downsampled vector to using all 14 bits - */ - w32_maxVal = WebRtcSpl_MaxAbsValueW32(pw32_corr, 54); - w16_normVal2 = 18 - WebRtcSpl_NormW32(w32_maxVal); - w16_normVal2 = WEBRTC_SPL_MAX(w16_normVal2, 0); - - WebRtcSpl_VectorBitShiftW32ToW16(pw16_corrOut, 54, pw32_corr, w16_normVal2); - - /* Total scale factor (right shifts) of correlation value */ - *pw16_corrScale = 2 * w16_normVal + 6 + w16_normVal2; - - return (50 + 1); -} - -#undef SCRATCH_pw16_corrVec -#undef SCRATCH_pw16_data_ds -#undef SCRATCH_pw32_corr - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.cc index 04b886a2e2e..5fb054c7850 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.cc @@ -8,17 +8,17 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/decision_logic.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic.h" #include <algorithm> -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" -#include "webrtc/modules/audio_coding/neteq4/decision_logic_fax.h" -#include "webrtc/modules/audio_coding/neteq4/decision_logic_normal.h" -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic_normal.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" #include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -145,8 +145,8 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer, reset_decoder); } -void DecisionLogic::ExpandDecision(bool is_expand_decision) { - if (is_expand_decision) { +void DecisionLogic::ExpandDecision(Operations operation) { + if (operation == kExpand) { num_consecutive_expands_++; } else { num_consecutive_expands_ = 0; diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.h index aca5ca40559..672ce939d41 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ -#include "webrtc/modules/audio_coding/neteq4/defines.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/defines.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -92,7 +92,7 @@ class DecisionLogic { // not. Note that this is necessary, since an expand decision can be changed // to kNormal in NetEqImpl::GetDecision if there is still enough data in the // sync buffer. - void ExpandDecision(bool is_expand_decision); + virtual void ExpandDecision(Operations operation); // Adds |value| to |sample_memory_|. void AddSampleMemory(int32_t value) { @@ -165,4 +165,4 @@ class DecisionLogic { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_fax.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc index 00c8bcf4a2b..08a4c4cb646 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_fax.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/decision_logic_fax.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h" #include <assert.h> #include <algorithm> -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_fax.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.h index 1ccd3524473..01a948fa429 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_fax.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_FAX_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_FAX_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_ -#include "webrtc/modules/audio_coding/neteq4/decision_logic.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -60,4 +60,4 @@ class DecisionLogicFax : public DecisionLogic { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_FAX_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc index 8688737c3ed..97a8843ae0a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc @@ -8,18 +8,18 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/decision_logic_normal.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic_normal.h" #include <assert.h> #include <algorithm> -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" #include "webrtc/modules/interface/module_common_types.h" namespace webrtc { @@ -44,7 +44,7 @@ Operations DecisionLogicNormal::GetDecisionSpecialized( uint32_t target_timestamp = sync_buffer.end_timestamp(); uint32_t available_timestamp = 0; - int is_cng_packet = 0; + bool is_cng_packet = false; if (packet_header) { available_timestamp = packet_header->timestamp; is_cng_packet = diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.h index 783b001fc3e..a339d160f29 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ -#include "webrtc/modules/audio_coding/neteq4/decision_logic.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -38,6 +38,10 @@ class DecisionLogicNormal : public DecisionLogic { virtual ~DecisionLogicNormal() {} protected: + static const int kAllowMergeWithoutExpandMs = 20; // 20 ms. + static const int kReinitAfterExpands = 100; + static const int kMaxWaitForPacket = 10; + // Returns the operation that should be done next. |sync_buffer| and |expand| // are provided for reference. |decoder_frame_length| is the number of samples // obtained from the last decoded frame. If there is a packet available, the @@ -54,31 +58,28 @@ class DecisionLogicNormal : public DecisionLogic { Modes prev_mode, bool play_dtmf, bool* reset_decoder); - private: - static const int kAllowMergeWithoutExpandMs = 20; // 20 ms. - static const int kReinitAfterExpands = 100; - static const int kMaxWaitForPacket = 10; + // Returns the operation to do given that the expected packet is not + // available, but a packet further into the future is at hand. + virtual Operations FuturePacketAvailable( + const SyncBuffer& sync_buffer, + const Expand& expand, + int decoder_frame_length, Modes prev_mode, + uint32_t target_timestamp, + uint32_t available_timestamp, + bool play_dtmf); - // Returns the operation given that the next available packet is a comfort - // noise payload (RFC 3389 only, not codec-internal). - Operations CngOperation(Modes prev_mode, uint32_t target_timestamp, - uint32_t available_timestamp); + // Returns the operation to do given that the expected packet is available. + virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf); // Returns the operation given that no packets are available (except maybe // a DTMF event, flagged by setting |play_dtmf| true). - Operations NoPacket(bool play_dtmf); + virtual Operations NoPacket(bool play_dtmf); - // Returns the operation to do given that the expected packet is available. - Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf); - - // Returns the operation to do given that the expected packet is not - // available, but a packet further into the future is at hand. - Operations FuturePacketAvailable(const SyncBuffer& sync_buffer, - const Expand& expand, - int decoder_frame_length, Modes prev_mode, - uint32_t target_timestamp, - uint32_t available_timestamp, - bool play_dtmf); + private: + // Returns the operation given that the next available packet is a comfort + // noise payload (RFC 3389 only, not codec-internal). + Operations CngOperation(Modes prev_mode, uint32_t target_timestamp, + uint32_t available_timestamp); // Checks if enough time has elapsed since the last successful timescale // operation was done (i.e., accelerate or preemptive expand). @@ -103,4 +104,4 @@ class DecisionLogicNormal : public DecisionLogic { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc index d596c0519a3..f9056a6cbae 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decision_logic_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc @@ -11,12 +11,12 @@ // Unit tests for DecisionLogic class and derived classes. #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/decision_logic.h" -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" namespace webrtc { @@ -24,7 +24,7 @@ TEST(DecisionLogic, CreateAndDestroy) { int fs_hz = 8000; int output_size_samples = fs_hz / 100; // Samples per 10 ms. DecoderDatabase decoder_database; - PacketBuffer packet_buffer(10, 1000); + PacketBuffer packet_buffer(10); DelayPeakDetector delay_peak_detector; DelayManager delay_manager(240, &delay_peak_detector); BufferLevelFilter buffer_level_filter; diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decoder_database.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.cc index 8d87519b25a..5049962b45b 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decoder_database.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.cc @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" #include <assert.h> #include <utility> // pair -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decoder_database.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.h index 9effd525ded..8a03f2123be 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decoder_database.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECODER_DATABASE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECODER_DATABASE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_ #include <map> +#include "webrtc/base/constructormagic.h" #include "webrtc/common_types.h" // NULL -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -155,4 +155,4 @@ class DecoderDatabase { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECODER_DATABASE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc index 76f5a099ecb..d0c6f5ae891 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" #include <assert.h> #include <stdlib.h> @@ -18,7 +18,7 @@ #include "gmock/gmock.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h" #include "webrtc/test/testsupport/gtest_disable.h" namespace webrtc { @@ -68,11 +68,11 @@ TEST(DecoderDatabase, GetRtpPayloadType) { db.GetRtpPayloadType(kDecoderISAC)); // iSAC is not registered. } -TEST(DecoderDatabase, DISABLED_ON_ANDROID(GetDecoder)) { +TEST(DecoderDatabase, GetDecoder) { DecoderDatabase db; const uint8_t kPayloadType = 0; EXPECT_EQ(DecoderDatabase::kOK, - db.RegisterPayload(kPayloadType, kDecoderILBC)); + db.RegisterPayload(kPayloadType, kDecoderPCM16B)); AudioDecoder* dec = db.GetDecoder(kPayloadType); ASSERT_TRUE(dec != NULL); } diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/defines.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/defines.h index b6f9eb2bc14..33d1bd9c3f0 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/defines.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/defines.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DEFINES_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DEFINES_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DEFINES_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DEFINES_H_ namespace webrtc { @@ -48,4 +48,4 @@ enum Modes { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DEFINES_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DEFINES_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_logging.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_logging.h deleted file mode 100644 index 04b1c401528..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_logging.h +++ /dev/null @@ -1,34 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Contains definitions for the delay logging functionality. Only used for debugging and - * tracing purposes. - */ - -#ifndef DELAY_LOGGING_H -#define DELAY_LOGGING_H - -#define NETEQ_DELAY_LOGGING_VERSION_STRING "2.0" - -#define NETEQ_DELAY_LOGGING_SIGNAL_RECIN 1 -#define NETEQ_DELAY_LOGGING_SIGNAL_FLUSH 2 -#define NETEQ_DELAY_LOGGING_SIGNAL_CLOCK 3 -#define NETEQ_DELAY_LOGGING_SIGNAL_EOF 4 -#define NETEQ_DELAY_LOGGING_SIGNAL_DECODE 5 -#define NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS 6 -#define NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO 7 -#define NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO 8 -#define NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO 9 -#define NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO 10 -#define NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF 11 -#define NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC 12 - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_manager.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.cc index e80b9de5142..a935561eff4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_manager.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" #include <assert.h> #include <math.h> @@ -16,7 +16,7 @@ #include <algorithm> // max, min #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/logging.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_manager.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.h index ed1e87b190d..96b5e19ebdc 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_manager.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DELAY_MANAGER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DELAY_MANAGER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_ #include <string.h> // Provide access to size_t. #include <vector> -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -161,4 +161,4 @@ class DelayManager { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DELAY_MANAGER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_manager_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc index 482a65c9a48..6f9733234d5 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_manager_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc @@ -10,13 +10,13 @@ // Unit tests for DelayManager class. -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" #include <math.h> #include "gmock/gmock.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc index fd5b9c08f0e..5996d7d197a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" #include <algorithm> // max diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.h index dfdd2537d7d..8bf6aba8b55 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DELAY_PEAK_DETECTOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DELAY_PEAK_DETECTOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_ #include <string.h> // size_t #include <list> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" namespace webrtc { @@ -73,4 +73,4 @@ class DelayPeakDetector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DELAY_PEAK_DETECTOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc index a3b48209cff..080309be014 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc @@ -10,7 +10,7 @@ // Unit tests for DelayPeakDetector class. -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" #include "gtest/gtest.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp.c deleted file mode 100644 index ea2fa87d501..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp.c +++ /dev/null @@ -1,532 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains some DSP initialization functions and - * constant table definitions. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "neteq_error_codes.h" - -/* Filter coefficients used when downsampling from the indicated - sample rates (8, 16, 32, 48 kHz) to 4 kHz. - Coefficients are in Q12. */ - -/* {0.3, 0.4, 0.3} */ -const int16_t WebRtcNetEQ_kDownsample8kHzTbl[] = { 1229, 1638, 1229 }; - -#ifdef NETEQ_WIDEBAND -/* {0.15, 0.2, 0.3, 0.2, 0.15} */ -const int16_t WebRtcNetEQ_kDownsample16kHzTbl[] = -{ 614, 819, 1229, 819, 614}; -#endif - -#ifdef NETEQ_32KHZ_WIDEBAND -/* {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425} */ -const int16_t WebRtcNetEQ_kDownsample32kHzTbl[] = -{ 584, 512, 625, 667, 625, 512, 584}; -#endif - -#ifdef NETEQ_48KHZ_WIDEBAND -/* {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487} */ -const int16_t WebRtcNetEQ_kDownsample48kHzTbl[] = -{ 1019, 390, 427, 440, 427, 390, 1019}; -#endif - -/* Constants used in expand function WebRtcNetEQ_Expand */ - -/* Q12: -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 */ -const int16_t WebRtcNetEQ_kMixFractionFuncTbl[4] = { -5179, 19931, -16422, 5776 }; - -/* Tabulated divisions to save complexity */ -/* 1049/{0, .., 6} */ -const int16_t WebRtcNetEQ_k1049div[7] = { 0, 1049, 524, 349, 262, 209, 174 }; - -/* 2097/{0, .., 6} */ -const int16_t WebRtcNetEQ_k2097div[7] = { 0, 2097, 1048, 699, 524, 419, 349 }; - -/* 5243/{0, .., 6} */ -const int16_t WebRtcNetEQ_k5243div[7] = { 0, 5243, 2621, 1747, 1310, 1048, 873 }; - -#ifdef WEBRTC_NETEQ_40BITACC_TEST -/* - * Run NetEQ with simulated 40-bit accumulator to run bit-exact to a DSP - * implementation where the main (spl and NetEQ) functions have been - * 40-bit optimized. For testing purposes. - */ - -/**************************************************************************** - * WebRtcNetEQ_40BitAccCrossCorr(...) - * - * Calculates the Cross correlation between two sequences seq1 and seq2. Seq1 - * is fixed and seq2 slides as the pointer is increased with step - * - * Input: - * - seq1 : First sequence (fixed throughout the correlation) - * - seq2 : Second sequence (slided step_seq2 for each - * new correlation) - * - dimSeq : Number of samples to use in the cross correlation. - * Should be no larger than 1024 to avoid overflow. - * - dimCrossCorr : Number of CrossCorrelations to calculate (start - * position for seq2 is updated for each new one) - * - rShift : Number of right shifts to use - * - step_seq2 : How many (positive or negative) steps the seq2 - * pointer should be updated for each new cross - * correlation value - * - * Output: - * - crossCorr : The cross correlation in Q-rShift - */ - -void WebRtcNetEQ_40BitAccCrossCorr(int32_t *crossCorr, - int16_t *seq1, - int16_t *seq2, - int16_t dimSeq, - int16_t dimCrossCorr, - int16_t rShift, - int16_t step_seq2) -{ - int i, j; - int16_t *seq1Ptr, *seq2Ptr; - int64_t acc; - - for (i = 0; i < dimCrossCorr; i++) - { - /* Set the pointer to the static vector, set the pointer to - the sliding vector and initialize crossCorr */ - seq1Ptr = seq1; - seq2Ptr = seq2 + (step_seq2 * i); - acc = 0; - - /* Perform the cross correlation */ - for (j = 0; j < dimSeq; j++) - { - acc += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr)); - seq1Ptr++; - seq2Ptr++; - } - - (*crossCorr) = (int32_t) (acc >> rShift); - crossCorr++; - } -} - -/**************************************************************************** - * WebRtcNetEQ_40BitAccDotW16W16(...) - * - * Calculates the dot product between two vectors (int16_t) - * - * Input: - * - vector1 : Vector 1 - * - vector2 : Vector 2 - * - len : Number of samples in vector - * Should be no larger than 1024 to avoid overflow. - * - scaling : The number of left shifts required to avoid overflow - * in the dot product - * Return value : The dot product - */ - -int32_t WebRtcNetEQ_40BitAccDotW16W16(int16_t *vector1, - int16_t *vector2, - int len, - int scaling) -{ - int32_t sum; - int i; - int64_t acc; - - acc = 0; - for (i = 0; i < len; i++) - { - acc += WEBRTC_SPL_MUL_16_16(*vector1++, *vector2++); - } - - sum = (int32_t) (acc >> scaling); - - return(sum); -} - -#endif /* WEBRTC_NETEQ_40BITACC_TEST */ - -/**************************************************************************** - * WebRtcNetEQ_DSPInit(...) - * - * Initializes DSP side of NetEQ. - * - * Input: - * - inst : NetEq DSP instance - * - fs : Initial sample rate (may change when decoding data) - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - * : non-zero - error - */ - -int WebRtcNetEQ_DSPInit(DSPInst_t *inst, uint16_t fs) -{ - - int res = 0; - int16_t fs_mult; - - /* Pointers and values to save before clearing the instance */ -#ifdef NETEQ_CNG_CODEC - void *savedPtr1 = inst->CNG_Codec_inst; -#endif - void *savedPtr2 = inst->pw16_readAddress; - void *savedPtr3 = inst->pw16_writeAddress; - void *savedPtr4 = inst->main_inst; -#ifdef NETEQ_VAD - void *savedVADptr = inst->VADInst.VADState; - VADInitFunction savedVADinit = inst->VADInst.initFunction; - VADSetmodeFunction savedVADsetmode = inst->VADInst.setmodeFunction; - VADFunction savedVADfunc = inst->VADInst.VADFunction; - int16_t savedVADEnabled = inst->VADInst.VADEnabled; - int savedVADMode = inst->VADInst.VADMode; -#endif /* NETEQ_VAD */ - DSPStats_t saveStats; - int16_t saveMsPerCall = inst->millisecondsPerCall; - enum BGNMode saveBgnMode = inst->BGNInst.bgnMode; -#ifdef NETEQ_STEREO - MasterSlaveInfo* saveMSinfo = inst->msInfo; -#endif - - /* copy contents of statInst to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveStats, &(inst->statInst), - sizeof(DSPStats_t)/sizeof(int16_t)); - - /* check that the sample rate is valid */ - if ((fs != 8000) -#ifdef NETEQ_WIDEBAND - &&(fs!=16000) -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - &&(fs!=32000) -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - &&(fs!=48000) -#endif - ) - { - /* invalid rate */ - return (CODEC_DB_UNSUPPORTED_FS); - } - - /* calcualte fs/8000 */ - fs_mult = WebRtcSpl_DivW32W16ResW16(fs, 8000); - - /* Set everything to zero since most variables should be zero at start */ - WebRtcSpl_MemSetW16((int16_t *) inst, 0, sizeof(DSPInst_t) / sizeof(int16_t)); - - /* Restore saved pointers */ -#ifdef NETEQ_CNG_CODEC - inst->CNG_Codec_inst = (CNG_dec_inst *)savedPtr1; -#endif - inst->pw16_readAddress = (int16_t *) savedPtr2; - inst->pw16_writeAddress = (int16_t *) savedPtr3; - inst->main_inst = savedPtr4; -#ifdef NETEQ_VAD - inst->VADInst.VADState = savedVADptr; - inst->VADInst.initFunction = savedVADinit; - inst->VADInst.setmodeFunction = savedVADsetmode; - inst->VADInst.VADFunction = savedVADfunc; - inst->VADInst.VADEnabled = savedVADEnabled; - inst->VADInst.VADMode = savedVADMode; -#endif /* NETEQ_VAD */ - - /* Initialize main part */ - inst->fs = fs; - inst->millisecondsPerCall = saveMsPerCall; - inst->timestampsPerCall = inst->millisecondsPerCall * 8 * fs_mult; - inst->ExpandInst.w16_overlap = 5 * fs_mult; - inst->endPosition = 565 * fs_mult; - inst->curPosition = inst->endPosition - inst->ExpandInst.w16_overlap; - inst->w16_seedInc = 1; - inst->uw16_seed = 777; - inst->w16_muteFactor = 16384; /* 1.0 in Q14 */ - inst->w16_frameLen = 3 * inst->timestampsPerCall; /* Dummy initialize to 30ms */ - - inst->w16_speechHistoryLen = 256 * fs_mult; - inst->pw16_speechHistory = &inst->speechBuffer[inst->endPosition - - inst->w16_speechHistoryLen]; - inst->ExpandInst.pw16_overlapVec = &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - - inst->ExpandInst.w16_overlap]); - - /* Reusage of memory in speechBuffer inside Expand */ - inst->ExpandInst.pw16_expVecs[0] = &inst->speechBuffer[0]; - inst->ExpandInst.pw16_expVecs[1] = &inst->speechBuffer[126 * fs_mult]; - inst->ExpandInst.pw16_arState = &inst->speechBuffer[2 * 126 * fs_mult]; - inst->ExpandInst.pw16_arFilter = &inst->speechBuffer[2 * 126 * fs_mult - + UNVOICED_LPC_ORDER]; - /* Ends at 2*126*fs_mult+UNVOICED_LPC_ORDER+(UNVOICED_LPC_ORDER+1) */ - - inst->ExpandInst.w16_expandMuteFactor = 16384; /* 1.0 in Q14 */ - - /* Initialize BGN part */ - inst->BGNInst.pw16_filter[0] = 4096; - inst->BGNInst.w16_scale = 20000; - inst->BGNInst.w16_scaleShift = 24; - inst->BGNInst.w32_energyUpdate = 500000; - inst->BGNInst.w32_energyUpdateLow = 0; - inst->BGNInst.w32_energy = 2500; - inst->BGNInst.w16_initialized = 0; - inst->BGNInst.bgnMode = saveBgnMode; - - /* Recreate statistics counters */WEBRTC_SPL_MEMCPY_W16(&(inst->statInst), &saveStats, - sizeof(DSPStats_t)/sizeof(int16_t)); - -#ifdef NETEQ_STEREO - /* Write back the pointer. */ - inst->msInfo = saveMSinfo; -#endif - -#ifdef NETEQ_CNG_CODEC - if (inst->CNG_Codec_inst!=NULL) - { - /* initialize comfort noise generator */ - res |= WebRtcCng_InitDec(inst->CNG_Codec_inst); - } -#endif - -#ifdef NETEQ_VAD - /* initialize PostDecode VAD instance - (don't bother checking for NULL instance, this is done inside init function) */ - res |= WebRtcNetEQ_InitVAD(&inst->VADInst, fs); -#endif /* NETEQ_VAD */ - - return (res); -} - -/**************************************************************************** - * WebRtcNetEQ_AddressInit(...) - * - * Initializes the shared-memory communication on the DSP side. - * - * Input: - * - inst : NetEQ DSP instance - * - data2McuAddress : Pointer to memory where DSP writes / MCU reads - * - data2DspAddress : Pointer to memory where MCU writes / DSP reads - * - mainInst : NetEQ main instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - */ - -int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress, - const void *data2DspAddress, const void *mainInst) -{ - - /* set shared-memory addresses in the DSP instance */ - inst->pw16_readAddress = (int16_t *) data2DspAddress; - inst->pw16_writeAddress = (int16_t *) data2McuAddress; - - /* set pointer to main NetEQ instance */ - inst->main_inst = (void *) mainInst; - - /* set output frame size to 10 ms = 80 samples in narrowband */ - inst->millisecondsPerCall = 10; - inst->timestampsPerCall = 80; - - return (0); - -} - -/**************************************************************************** - * NETEQDSP_clearInCallStats(...) - * - * Reset in-call statistics variables on DSP side. - * - * Input: - * - inst : NetEQ DSP instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - */ - -int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst) -{ - /* Reset statistics counters */ - inst->statInst.accelerateLength = 0; - inst->statInst.expandLength = 0; - inst->statInst.preemptiveLength = 0; - inst->statInst.addedSamples = 0; - return (0); -} - -/**************************************************************************** - * WebRtcNetEQ_ClearPostCallStats(...) - * - * Reset post-call statistics variables on DSP side. - * - * Input: - * - inst : NetEQ DSP instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - */ - -int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst) -{ - - /* Reset statistics counters */ - inst->statInst.expandedVoiceSamples = 0; - inst->statInst.expandedNoiseSamples = 0; - return (0); -} - -/**************************************************************************** - * WebRtcNetEQ_ClearActivityStats(...) - * - * Reset processing activity statistics. - * - * Input: - * - inst : NetEQ DSP instance - * - * Output: - * - inst : Updated instance - * - */ - -void WebRtcNetEQ_ClearActivityStats(DSPInst_t *inst) { - memset(&inst->activity_stats, 0, sizeof(ActivityStats)); -} - -#ifdef NETEQ_VAD - -/**************************************************************************** - * WebRtcNetEQ_InitVAD(...) - * - * Initializes post-decode VAD instance. - * - * Input: - * - VADinst : PostDecodeVAD instance - * - fs : Initial sample rate - * - * Output: - * - VADinst : Updated instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, uint16_t fs) -{ - - int res = 0; - - /* initially, disable the post-decode VAD */ - VADInst->VADEnabled = 0; - - if (VADInst->VADState != NULL /* if VAD state is provided */ - && VADInst->initFunction != NULL /* and all function ... */ - && VADInst->setmodeFunction != NULL /* ... pointers ... */ - && VADInst->VADFunction != NULL) /* ... are defined */ - { - res = VADInst->initFunction( VADInst->VADState ); /* call VAD init function */ - res |= WebRtcNetEQ_SetVADModeInternal( VADInst, VADInst->VADMode ); - - if (res!=0) - { - /* something is wrong; play it safe and set the VADstate to NULL */ - VADInst->VADState = NULL; - } - else if (fs<=16000) - { - /* enable VAD if NB or WB (VAD cannot handle SWB) */ - VADInst->VADEnabled = 1; - } - } - - /* reset SID/CNG interval counter */ - VADInst->SIDintervalCounter = 0; - - /* initialize with active-speaker decision */ - VADInst->VADDecision = 1; - - return(res); - -} - -/**************************************************************************** - * WebRtcNetEQ_SetVADModeInternal(...) - * - * Set the VAD mode in the VAD struct, and communicate it to the VAD instance - * if it exists. - * - * Input: - * - VADinst : PostDecodeVAD instance - * - mode : Mode number passed on to the VAD function - * - * Output: - * - VADinst : Updated instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_SetVADModeInternal(PostDecodeVAD_t *VADInst, int mode) -{ - - int res = 0; - - VADInst->VADMode = mode; - - if (VADInst->VADState != NULL) - { - /* call setmode function */ - res = VADInst->setmodeFunction(VADInst->VADState, mode); - } - - return(res); - -} - -#endif /* NETEQ_VAD */ - -/**************************************************************************** - * WebRtcNetEQ_FlushSpeechBuffer(...) - * - * Flush the speech buffer. - * - * Input: - * - inst : NetEq DSP instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - * : non-zero - error - */ - -int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst) -{ - int16_t fs_mult; - - /* calcualte fs/8000 */ - fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000); - - /* clear buffer */ - WebRtcSpl_MemSetW16(inst->speechBuffer, 0, SPEECH_BUF_SIZE); - inst->endPosition = 565 * fs_mult; - inst->curPosition = inst->endPosition - inst->ExpandInst.w16_overlap; - - return 0; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp.h deleted file mode 100644 index 9371938d5f6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp.h +++ /dev/null @@ -1,807 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains some DSP initialization functions, - * constant table definitions and other parameters. - * Also contains definitions of all DSP-side data structures. - */ - - -#ifndef DSP_H -#define DSP_H - -#include "typedefs.h" - -#include "webrtc_cng.h" - -#include "codec_db_defines.h" -#include "neteq_defines.h" -#include "neteq_statistics.h" - -#ifdef NETEQ_ATEVENT_DECODE -#include "dtmf_tonegen.h" -#endif - - - -/*****************************/ -/* Pre-processor definitions */ -/*****************************/ - -/* FSMULT is the sample rate divided by 8000 */ -#if defined(NETEQ_48KHZ_WIDEBAND) - #define FSMULT 6 -#elif defined(NETEQ_32KHZ_WIDEBAND) - #define FSMULT 4 -#elif defined(NETEQ_WIDEBAND) - #define FSMULT 2 -#else - #define FSMULT 1 -#endif - -/* Size of the speech buffer (or synchronization buffer). */ -/* 60 ms decoding + 10 ms syncbuff + 0.625ms lookahead */ -#define SPEECH_BUF_SIZE (565 * FSMULT) - -/* Misc definitions */ -#define BGN_LPC_ORDER (4 + FSMULT) /* 5, 6, 8, or 10 */ -#define UNVOICED_LPC_ORDER 6 -#define RANDVEC_NO_OF_SAMPLES 256 - -/* Number of milliseconds to remove/add during accelerate/pre-emptive expand - under BGNonly operation */ -#define DEFAULT_TIME_ADJUST 8 - -/* Number of RecOut calls without CNG/SID before re-enabling post-decode VAD */ -#define POST_DECODE_VAD_AUTO_ENABLE 3000 - -/* 8kHz windowing in Q15 (over 5 samples) */ -#define NETEQ_OVERLAP_WINMUTE_8KHZ_START 27307 -#define NETEQ_OVERLAP_WINMUTE_8KHZ_INC -5461 -#define NETEQ_OVERLAP_WINUNMUTE_8KHZ_START 5461 -#define NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC 5461 -/* 16kHz windowing in Q15 (over 10 samples) */ -#define NETEQ_OVERLAP_WINMUTE_16KHZ_START 29789 -#define NETEQ_OVERLAP_WINMUTE_16KHZ_INC -2979 -#define NETEQ_OVERLAP_WINUNMUTE_16KHZ_START 2979 -#define NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC 2979 -/* 32kHz windowing in Q15 (over 20 samples) */ -#define NETEQ_OVERLAP_WINMUTE_32KHZ_START 31208 -#define NETEQ_OVERLAP_WINMUTE_32KHZ_INC -1560 -#define NETEQ_OVERLAP_WINUNMUTE_32KHZ_START 1560 -#define NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC 1560 -/* 48kHz windowing in Q15 (over 30 samples) */ -#define NETEQ_OVERLAP_WINMUTE_48KHZ_START 31711 -#define NETEQ_OVERLAP_WINMUTE_48KHZ_INC -1057 -#define NETEQ_OVERLAP_WINUNMUTE_48KHZ_START 1057 -#define NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC 1057 - -/* Fade BGN towards zero after this many Expand calls */ -#define FADE_BGN_TIME 200 - - -/*******************/ -/* Constant tables */ -/*******************/ - -extern const int16_t WebRtcNetEQ_kDownsample8kHzTbl[]; -extern const int16_t WebRtcNetEQ_kDownsample16kHzTbl[]; -extern const int16_t WebRtcNetEQ_kDownsample32kHzTbl[]; -extern const int16_t WebRtcNetEQ_kDownsample48kHzTbl[]; -extern const int16_t WebRtcNetEQ_kRandnTbl[]; -extern const int16_t WebRtcNetEQ_kMixFractionFuncTbl[]; -extern const int16_t WebRtcNetEQ_k1049div[]; -extern const int16_t WebRtcNetEQ_k2097div[]; -extern const int16_t WebRtcNetEQ_k5243div[]; - - - -/************/ -/* Typedefs */ -/************/ - -enum BGNMode -{ - BGN_ON, /* default "normal" behavior with eternal noise */ - BGN_FADE, /* noise fades to zero after some time */ - BGN_OFF /* background noise is always zero */ -}; - -#ifdef NETEQ_STEREO -enum MasterSlaveMode -{ - NETEQ_MONO, /* stand-alone instance */ - NETEQ_MASTER, /* master instance in a spatial/stereo configuration */ - NETEQ_SLAVE /* slave instance in a spatial/stereo configuration */ -}; - -enum MasterSlaveExtraInfo -{ - NO_INFO, /* no info to convey */ - ACC_FAIL, /* signal that accelerate failed */ - PE_EXP_FAIL, /* signal that pre-emptive expand failed */ - DTMF_OVERDUB, /* signal that DTMF overdub is generated */ - DTMF_ONLY /* signal that DTMF only is played */ -}; -#endif - -/****************************/ -/* DSP-side data structures */ -/****************************/ - -/* Background noise (BGN) instance for storing BGN parameters - (sub-instance of NETEQDSP_inst) */ -typedef struct BGNInst_t_ -{ - - int32_t w32_energy; - int32_t w32_energyMax; - int32_t w32_energyUpdate; - int32_t w32_energyUpdateLow; - int16_t pw16_filterState[BGN_LPC_ORDER]; - int16_t pw16_filter[BGN_LPC_ORDER + 1]; - int16_t w16_mutefactor; - int16_t w16_scale; - int16_t w16_scaleShift; - int16_t w16_initialized; - enum BGNMode bgnMode; - -} BGNInst_t; - -/* Expansion instance (sub-instance of NETEQDSP_inst) */ -typedef struct ExpandInst_t_ -{ - - int16_t w16_overlap; /* Constant, 5 for NB and 10 for WB */ - int16_t w16_consecExp; /* Number of consecutive expand calls */ - int16_t *pw16_arFilter; /* length [UNVOICED_LPC_ORDER+1] */ - int16_t *pw16_arState; /* length [UNVOICED_LPC_ORDER] */ - int16_t w16_arGain; - int16_t w16_arGainScale; - int16_t w16_vFraction; /* Q14 */ - int16_t w16_currentVFraction; /* Q14 */ - int16_t *pw16_expVecs[2]; - int16_t w16_lags[3]; - int16_t w16_maxLag; - int16_t *pw16_overlapVec; /* last samples of speech history */ - int16_t w16_lagsDirection; - int16_t w16_lagsPosition; - int16_t w16_expandMuteFactor; /* Q14 */ - int16_t w16_stopMuting; - int16_t w16_onset; - int16_t w16_muteSlope; /* Q20 */ - -} ExpandInst_t; - -#ifdef NETEQ_VAD - -/* - * VAD function pointer types, replicating the typedefs in webrtc_neteq_internal.h. - * These function pointers match the definitions of WebRtc VAD functions WebRtcVad_Init, - * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in webrtc_vad.h. - */ -typedef int (*VADInitFunction)(void *VAD_inst); -typedef int (*VADSetmodeFunction)(void *VAD_inst, int mode); -typedef int (*VADFunction)(void *VAD_inst, int fs, int16_t *frame, - int frameLen); - -/* Post-decode VAD instance (sub-instance of NETEQDSP_inst) */ -typedef struct PostDecodeVAD_t_ -{ - - void *VADState; /* pointer to a VAD instance */ - - int16_t VADEnabled; /* 1 if enabled, 0 if disabled */ - int VADMode; /* mode parameter to pass to the VAD function */ - int VADDecision; /* 1 for active, 0 for passive */ - int16_t SIDintervalCounter; /* reset when decoding CNG/SID frame, - increment for each recout call */ - - /* Function pointers */ - VADInitFunction initFunction; /* VAD init function */ - VADSetmodeFunction setmodeFunction; /* VAD setmode function */ - VADFunction VADFunction; /* VAD function */ - -} PostDecodeVAD_t; - -#endif /* NETEQ_VAD */ - -#ifdef NETEQ_STEREO -#define MAX_MS_DECODES 10 - -typedef struct -{ - /* Stand-alone, master, or slave */ - enum MasterSlaveMode msMode; - - enum MasterSlaveExtraInfo extraInfo; - - uint16_t instruction; - int16_t distLag; - int16_t corrLag; - int16_t bestIndex; - - uint32_t endTimestamp; - uint16_t samplesLeftWithOverlap; - -} MasterSlaveInfo; -#endif - - -/* "Main" NetEQ DSP instance */ -typedef struct DSPInst_t_ -{ - - /* MCU/DSP Communication layer */ - int16_t *pw16_readAddress; - int16_t *pw16_writeAddress; - void *main_inst; - - /* Output frame size in ms and samples */ - int16_t millisecondsPerCall; - int16_t timestampsPerCall; - - /* - * Example of speech buffer - * - * ----------------------------------------------------------- - * | History T-60 to T | Future | - * ----------------------------------------------------------- - * ^ ^ - * | | - * curPosition endPosition - * - * History is gradually shifted out to the left when inserting - * new data at the end. - */ - - int16_t speechBuffer[SPEECH_BUF_SIZE]; /* History/future speech buffer */ - int curPosition; /* Next sample to play */ - int endPosition; /* Position that ends future data */ - uint32_t endTimestamp; /* Timestamp value at end of future data */ - uint32_t videoSyncTimestamp; /* (Estimated) timestamp of the last - played sample (usually same as - endTimestamp-(endPosition-curPosition) - except during Expand and CNG) */ - uint16_t fs; /* sample rate in Hz */ - int16_t w16_frameLen; /* decoder frame length in samples */ - int16_t w16_mode; /* operation used during last RecOut call */ - int16_t w16_muteFactor; /* speech mute factor in Q14 */ - int16_t *pw16_speechHistory; /* beginning of speech history during Expand */ - int16_t w16_speechHistoryLen; /* 256 for NB and 512 for WB */ - - /* random noise seed parameters */ - int16_t w16_seedInc; - uint32_t uw16_seed; - - /* VQmon related variable */ - int16_t w16_concealedTS; - - /*****************/ - /* Sub-instances */ - /*****************/ - - /* Decoder data */ - CodecFuncInst_t codec_ptr_inst; - -#ifdef NETEQ_CNG_CODEC - /* CNG "decoder" instance */ - CNG_dec_inst *CNG_Codec_inst; -#endif /* NETEQ_CNG_CODEC */ - -#ifdef NETEQ_ATEVENT_DECODE - /* DTMF generator instance */ - dtmf_tone_inst_t DTMFInst; -#endif /* NETEQ_CNG_CODEC */ - -#ifdef NETEQ_VAD - /* Post-decode VAD instance */ - PostDecodeVAD_t VADInst; -#endif /* NETEQ_VAD */ - - /* Expand instance (defined above) */ - ExpandInst_t ExpandInst; - - /* Background noise instance (defined above) */ - BGNInst_t BGNInst; - - /* Internal statistics instance */ - DSPStats_t statInst; - - /* Internal instance for short-term processing activity. */ - ActivityStats activity_stats; - -#ifdef NETEQ_STEREO - /* Pointer to Master/Slave info */ - MasterSlaveInfo *msInfo; -#endif - -} DSPInst_t; - - -/*************************/ -/* Function declarations */ -/*************************/ - -/**************************************************************************** - * WebRtcNetEQ_DSPInit(...) - * - * Initializes DSP side of NetEQ. - * - * Input: - * - inst : NetEq DSP instance - * - fs : Initial sample rate (may change when decoding data) - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - * : non-zero - error - */ - -int WebRtcNetEQ_DSPInit(DSPInst_t *inst, uint16_t fs); - -/**************************************************************************** - * WebRtcNetEQ_AddressInit(...) - * - * Initializes the shared-memory communication on the DSP side. - * - * Input: - * - inst : NetEQ DSP instance - * - data2McuAddress : Pointer to memory where DSP writes / MCU reads - * - data2DspAddress : Pointer to memory where MCU writes / DSP reads - * - mainInst : NetEQ main instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - */ - -int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress, - const void *data2DspAddress, const void *mainInst); - -/**************************************************************************** - * WebRtcNetEQ_ClearInCallStats(...) - * - * Reset in-call statistics variables on DSP side. - * - * Input: - * - inst : NetEQ DSP instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - */ - -int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_ClearPostCallStats(...) - * - * Reset post-call statistics variables on DSP side. - * - * Input: - * - inst : NetEQ DSP instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - */ - -int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_ClearActivityStats(...) - * - * Reset processing activity statistics. - * - * Input: - * - inst : NetEQ DSP instance - * - * Output: - * - inst : Updated instance - * - */ - -void WebRtcNetEQ_ClearActivityStats(DSPInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_RecOutInternal(...) - * - * This function asks NetEQ for more speech/audio data. - * - * Input: - * - inst : NetEQ instance, i.e. the user that requests more - * speech/audio data. - * - outdata : Pointer to a memory space where the output data - * should be stored. - * - BGNonly : If non-zero, RecOut will only produce background - * noise. It will still draw packets from the packet - * buffer, but they will never be decoded. - * - av_sync : 1 if NetEQ is in AV-sync, 0 otherwise. - * - * Output: - * - inst : Updated user information - * - len : Number of samples that were outputted from NetEq - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, int16_t *pw16_outData, - int16_t *pw16_len, int16_t BGNonly, int av_sync); - -/**************************************************************************** - * WebRtcNetEQ_Normal(...) - * - * This function has the possibility to modify data that is played out in Normal - * mode, for example adjust the gain of the signal. The length of the signal - * can not be changed. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector - * - decoded : Pointer to vector of new data from decoder - * - len : Number of input samples - * - * Output: - * - inst : Updated user information - * - pw16_len : Pointer to varibale where the number of samples - * produced will be written - * - * Return value : >=0 - Number of samples written to outData - * -1 - Error - */ - -int WebRtcNetEQ_Normal(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_decoded, int16_t len, - int16_t *pw16_outData, int16_t *pw16_len); - -/**************************************************************************** - * WebRtcNetEQ_Expand(...) - * - * This function produces one "chunk" of expansion data (PLC audio). The - * lenght of the produced audio depends on the speech history. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector - * - BGNonly : If non-zero, Expand will only produce background - * noise. - * - pw16_len : Desired number of samples (only for BGN mode). - * - * Output: - * - inst : Updated user information - * - outdata : Pointer to a memory space where the output data - * should be stored - * - pw16_len : Number of samples that were outputted from NetEq - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_Expand(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_outData, int16_t *pw16_len, - int16_t BGNonly); - -/**************************************************************************** - * WebRtcNetEQ_GenerateBGN(...) - * - * This function generates and writes len samples of background noise to the - * output vector. The Expand function will be called repeteadly until the - * correct number of samples is produced. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector - * - len : Desired length of produced BGN. - * - * - * Output: - * - pw16_outData : Pointer to a memory space where the output data - * should be stored - * - * Return value : >=0 - Number of noise samples produced and written - * to output - * -1 - Error - */ - -int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_outData, int16_t len); - -/**************************************************************************** - * WebRtcNetEQ_PreEmptiveExpand(...) - * - * This function tries to extend the audio data by repeating one or several - * pitch periods. The operation is only carried out if the correlation is - * strong or if the signal energy is very low. The algorithm is the - * reciprocal of the Accelerate algorithm. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector. - * - decoded : Pointer to newly decoded speech. - * - len : Length of decoded speech. - * - oldDataLen : Length of the part of decoded that has already been played out. - * - BGNonly : If non-zero, Pre-emptive Expand will only copy - * the first DEFAULT_TIME_ADJUST seconds of the - * input and append to the end. No signal matching is - * done. - * - * Output: - * - inst : Updated instance - * - outData : Pointer to a memory space where the output data - * should be stored. The vector must be at least - * min(len + 120*fs/8000, NETEQ_MAX_OUTPUT_SIZE) - * elements long. - * - pw16_len : Number of samples written to outData. - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - const int16_t *pw16_decoded, int len, int oldDataLen, - int16_t *pw16_outData, int16_t *pw16_len, - int16_t BGNonly); - -/**************************************************************************** - * WebRtcNetEQ_Accelerate(...) - * - * This function tries to shorten the audio data by removing one or several - * pitch periods. The operation is only carried out if the correlation is - * strong or if the signal energy is very low. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector. - * - decoded : Pointer to newly decoded speech. - * - len : Length of decoded speech. - * - BGNonly : If non-zero, Accelerate will only remove the last - * DEFAULT_TIME_ADJUST seconds of the intput. - * No signal matching is done. - * - * - * Output: - * - inst : Updated instance - * - outData : Pointer to a memory space where the output data - * should be stored - * - pw16_len : Number of samples written to outData. - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_Accelerate(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - const int16_t *pw16_decoded, int len, - int16_t *pw16_outData, int16_t *pw16_len, - int16_t BGNonly); - -/**************************************************************************** - * WebRtcNetEQ_Merge(...) - * - * This function is used to merge new data from the decoder to the exisiting - * stream in the synchronization buffer. The merge operation is typically - * done after a packet loss, where the end of the expanded data does not - * fit naturally with the new decoded data. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector. - * - decoded : Pointer to new decoded speech. - * - len : Number of samples in pw16_decoded. - * - * - * Output: - * - inst : Updated user information - * - outData : Pointer to a memory space where the output data - * should be stored - * - pw16_len : Number of samples written to pw16_outData - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_Merge(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_decoded, int len, int16_t *pw16_outData, - int16_t *pw16_len); - -/**************************************************************************** - * WebRtcNetEQ_Cng(...) - * - * This function produces CNG according to RFC 3389 - * - * Input: - * - inst : NetEQ DSP instance - * - len : Number of samples to produce - * - * Output: - * - pw16_outData : Output CNG - * - * Return value : 0 - Ok - * <0 - Error - */ - -#ifdef NETEQ_CNG_CODEC -/* Must compile NetEQ with CNG support to enable this function */ - -int WebRtcNetEQ_Cng(DSPInst_t *inst, int16_t *pw16_outData, int len); - -#endif /* NETEQ_CNG_CODEC */ - -/**************************************************************************** - * WebRtcNetEQ_BGNUpdate(...) - * - * This function updates the background noise parameter estimates. - * - * Input: - * - inst : NetEQ instance, where the speech history is stored. - * - scratchPtr : Pointer to scratch vector. - * - * Output: - * - inst : Updated information about the BGN characteristics. - * - * Return value : No return value - */ - -void WebRtcNetEQ_BGNUpdate( -#ifdef SCRATCH - DSPInst_t *inst, int16_t *pw16_scratchPtr -#else - DSPInst_t *inst -#endif - ); - -#ifdef NETEQ_VAD -/* Functions used by post-decode VAD */ - -/**************************************************************************** - * WebRtcNetEQ_InitVAD(...) - * - * Initializes post-decode VAD instance. - * - * Input: - * - VADinst : PostDecodeVAD instance - * - fs : Initial sample rate - * - * Output: - * - VADinst : Updated instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, uint16_t fs); - -/**************************************************************************** - * WebRtcNetEQ_SetVADModeInternal(...) - * - * Set the VAD mode in the VAD struct, and communicate it to the VAD instance - * if it exists. - * - * Input: - * - VADinst : PostDecodeVAD instance - * - mode : Mode number passed on to the VAD function - * - * Output: - * - VADinst : Updated instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_SetVADModeInternal(PostDecodeVAD_t *VADInst, int mode); - -#endif /* NETEQ_VAD */ - -/**************************************************************************** - * WebRtcNetEQ_FlushSpeechBuffer(...) - * - * Flush the speech buffer. - * - * Input: - * - inst : NetEq DSP instance - * - * Output: - * - inst : Updated instance - * - * Return value : 0 - ok - * : non-zero - error - */ - -int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst); - -#ifndef WEBRTC_NETEQ_40BITACC_TEST - -#include "signal_processing_library.h" -/* Map to regular SPL functions */ -#define WebRtcNetEQ_CrossCorr WebRtcSpl_CrossCorrelation -#define WebRtcNetEQ_DotW16W16 WebRtcSpl_DotProductWithScale - -#else /* WEBRTC_NETEQ_40BITACC_TEST defined */ -/* Run NetEQ with simulated 40-bit accumulator to run bit-exact to a DSP - implementation where the main (splib and NetEQ) functions have been - 40-bit optimized. */ - -/* Map to special 40-bit optimized functions, defined below */ -#define WebRtcNetEQ_CrossCorr WebRtcNetEQ_40BitAccCrossCorr -#define WebRtcNetEQ_DotW16W16 WebRtcNetEQ_40BitAccDotW16W16 - -/**************************************************************************** - * WebRtcNetEQ_40BitAccCrossCorr(...) - * - * Calculates the Cross correlation between two sequences seq1 and seq2. Seq1 - * is fixed and seq2 slides as the pointer is increased with step - * - * Input: - * - seq1 : First sequence (fixed throughout the correlation) - * - seq2 : Second sequence (slided step_seq2 for each - * new correlation) - * - dimSeq : Number of samples to use in the cross correlation. - * Should be no larger than 1024 to avoid overflow. - * - dimCrossCorr : Number of CrossCorrelations to calculate (start - * position for seq2 is updated for each new one) - * - rShift : Number of right shifts to use - * - step_seq2 : How many (positive or negative) steps the seq2 - * pointer should be updated for each new cross - * correlation value - * - * Output: - * - crossCorr : The cross correlation in Q-rShift - */ - -void WebRtcNetEQ_40BitAccCrossCorr(int32_t *crossCorr, int16_t *seq1, - int16_t *seq2, int16_t dimSeq, - int16_t dimCrossCorr, int16_t rShift, - int16_t step_seq2); - -/**************************************************************************** - * WebRtcNetEQ_40BitAccDotW16W16(...) - * - * Calculates the dot product between two vectors (int16_t) - * - * Input: - * - vector1 : Vector 1 - * - vector2 : Vector 2 - * - len : Number of samples in vector - * Should be no larger than 1024 to avoid overflow. - * - scaling : The number of right shifts (after multiplication) - * required to avoid overflow in the dot product. - * Return value : The dot product - */ - -int32_t WebRtcNetEQ_40BitAccDotW16W16(int16_t *vector1, int16_t *vector2, - int len, int scaling); - -#endif /* WEBRTC_NETEQ_40BITACC_TEST */ - -#endif /* DSP_H */ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.cc index e1aa0e53de7..7451ae26f8e 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h" +#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" #include <assert.h> #include <string.h> // Access to memset. diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h index 60cd995d840..af4f4d6c88c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DSP_HELPER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DSP_HELPER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ #include <string.h> // Access to size_t. -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -133,4 +133,4 @@ class DspHelper { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DSP_HELPER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc index 852c2ec927c..cbceff61945 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h" +#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c deleted file mode 100644 index ef721d55997..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c +++ /dev/null @@ -1,120 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains some help functions that did not fit elsewhere. - */ - -#include "dsp_helpfunctions.h" - - -int16_t WebRtcNetEQ_CalcFsMult(uint16_t fsHz) -{ - switch (fsHz) - { - case 8000: - { - return 1; - } - case 16000: - { - return 2; - } - case 32000: - { - return 4; - } - case 48000: - { - return 6; - } - default: - { - return 1; - } - } -} - - -int WebRtcNetEQ_DownSampleTo4kHz(const int16_t *in, int inLen, uint16_t inFsHz, - int16_t *out, int outLen, int compensateDelay) -{ - int16_t *B; /* filter coefficients */ - int16_t Blen; /* number of coefficients */ - int16_t filterDelay; /* phase delay in samples */ - int16_t factor; /* conversion rate (inFsHz/8000) */ - int ok; - - /* Set constants depending on frequency used */ - /* NOTE: The phase delay values are wrong compared to the true phase delay - of the filters. However, the error is preserved (through the +1 term) - for consistency. */ - switch (inFsHz) - { - case 8000: - { - Blen = 3; - factor = 2; - B = (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl; - filterDelay = 1 + 1; - break; - } -#ifdef NETEQ_WIDEBAND - case 16000: - { - Blen = 5; - factor = 4; - B = (int16_t*) WebRtcNetEQ_kDownsample16kHzTbl; - filterDelay = 2 + 1; - break; - } -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - case 32000: - { - Blen = 7; - factor = 8; - B = (int16_t*) WebRtcNetEQ_kDownsample32kHzTbl; - filterDelay = 3 + 1; - break; - } -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - case 48000: - { - Blen = 7; - factor = 12; - B = (int16_t*) WebRtcNetEQ_kDownsample48kHzTbl; - filterDelay = 3 + 1; - break; - } -#endif - default: - { - /* unsupported or wrong sample rate */ - return -1; - } - } - - if (!compensateDelay) - { - /* disregard delay compensation */ - filterDelay = 0; - } - - ok = WebRtcSpl_DownsampleFast((int16_t*) &in[Blen - 1], - (int16_t) (inLen - (Blen - 1)), /* number of input samples */ - out, (int16_t) outLen, /* number of output samples to produce */ - B, Blen, factor, filterDelay); /* filter parameters */ - - return ok; /* return value is -1 if input signal is too short */ - -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.h deleted file mode 100644 index 11119f1b825..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.h +++ /dev/null @@ -1,220 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Various help functions used by the DSP functions. - */ - -#ifndef DSP_HELPFUNCTIONS_H -#define DSP_HELPFUNCTIONS_H - -#include "typedefs.h" - -#include "dsp.h" - -/**************************************************************************** - * WebRtcNetEQ_Correlator(...) - * - * Calculate signal correlation. - * - * Input: - * - inst : DSP instance - * - data : Speech history to do expand from (older history in data[-4..-1]) - * - dataLen : Length of data - * - * Output: - * - corrOut : CC of downsampled signal - * - corrScale : Scale factor for correlation (-Qdomain) - * - * Return value : Length of correlated data - */ - -int16_t WebRtcNetEQ_Correlator(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_data, int16_t w16_dataLen, - int16_t *pw16_corrOut, - int16_t *pw16_corrScale); - -/**************************************************************************** - * WebRtcNetEQ_PeakDetection(...) - * - * Peak detection with parabolic fit. - * - * Input: - * - data : Data sequence for peak detection - * - dataLen : Length of data - * - nmbPeaks : Number of peaks to detect - * - fs_mult : Sample rate multiplier - * - * Output: - * - corrIndex : Index of the peak - * - winner : Value of the peak - * - * Return value : 0 for ok - */ - -int16_t WebRtcNetEQ_PeakDetection(int16_t *pw16_data, int16_t w16_dataLen, - int16_t w16_nmbPeaks, int16_t fs_mult, - int16_t *pw16_corrIndex, - int16_t *pw16_winners); - -/**************************************************************************** - * WebRtcNetEQ_PrblFit(...) - * - * Three-point parbola fit. - * - * Input: - * - 3pts : Three input samples - * - fs_mult : Sample rate multiplier - * - * Output: - * - Ind : Index of the peak - * - outVal : Value of the peak - * - * Return value : 0 for ok - */ - -int16_t WebRtcNetEQ_PrblFit(int16_t *pw16_3pts, int16_t *pw16_Ind, - int16_t *pw16_outVal, int16_t fs_mult); - -/**************************************************************************** - * WebRtcNetEQ_MinDistortion(...) - * - * Find the lag that results in minimum distortion. - * - * Input: - * - data : Start of speech to perform distortion on, second vector is assumed - * to be data[-Lag] - * - minLag : Start lag - * - maxLag : End lag - * - len : Length to correlate - * - * Output: - * - dist : Distorion value - * - * Return value : Lag for minimum distortion - */ - -int16_t WebRtcNetEQ_MinDistortion(const int16_t *pw16_data, - int16_t w16_minLag, int16_t w16_maxLag, - int16_t len, int32_t *pw16_dist); - -/**************************************************************************** - * WebRtcNetEQ_RandomVec(...) - * - * Generate random vector. - * - * Input: - * - seed : Current seed (input/output) - * - len : Number of samples to generate - * - incVal : Jump step - * - * Output: - * - randVec : Generated random vector - */ - -void WebRtcNetEQ_RandomVec(uint32_t *w32_seed, int16_t *pw16_randVec, - int16_t w16_len, int16_t w16_incval); - -/**************************************************************************** - * WebRtcNetEQ_MixVoiceUnvoice(...) - * - * Mix voiced and unvoiced signal. - * - * Input: - * - voicedVec : Voiced input signal - * - unvoicedVec : Unvoiced input signal - * - current_vfraction : Current mixing factor - * - vfraction_change : Mixing factor change per sample - * - N : Number of samples - * - * Output: - * - outData : Mixed signal - */ - -void WebRtcNetEQ_MixVoiceUnvoice(int16_t *pw16_outData, int16_t *pw16_voicedVec, - int16_t *pw16_unvoicedVec, - int16_t *w16_current_vfraction, - int16_t w16_vfraction_change, int16_t N); - -/**************************************************************************** - * WebRtcNetEQ_UnmuteSignal(...) - * - * Gradually reduce attenuation. - * - * Input: - * - inVec : Input signal - * - startMuteFact : Starting attenuation - * - unmuteFact : Factor to "unmute" with (Q20) - * - N : Number of samples - * - * Output: - * - outVec : Output signal - */ - -void WebRtcNetEQ_UnmuteSignal(int16_t *pw16_inVec, int16_t *startMuteFact, - int16_t *pw16_outVec, int16_t unmuteFact, - int16_t N); - -/**************************************************************************** - * WebRtcNetEQ_MuteSignal(...) - * - * Gradually increase attenuation. - * - * Input: - * - inout : Input/output signal - * - muteSlope : Slope of muting - * - N : Number of samples - */ - -void WebRtcNetEQ_MuteSignal(int16_t *pw16_inout, int16_t muteSlope, - int16_t N); - -/**************************************************************************** - * WebRtcNetEQ_CalcFsMult(...) - * - * Calculate the sample rate divided by 8000. - * - * Input: - * - fsHz : Sample rate in Hz in {8000, 16000, 32000, 48000}. - * - * Return value : fsHz/8000 for the valid values, 1 for other inputs - */ - -int16_t WebRtcNetEQ_CalcFsMult(uint16_t fsHz); - -/**************************************************************************** - * WebRtcNetEQ_DownSampleTo4kHz(...) - * - * Lowpass filter and downsample a signal to 4 kHz sample rate. - * - * Input: - * - in : Input signal samples. - * - inLen : Number of input samples. - * - inFsHz : Input sample rate in Hz. - * - outLen : Desired number of samples in decimated signal. - * - compensateDelay : If non-zero, compensate for the phase delay of - * of the anti-alias filter. - * - * Output: - * - out : Output signal samples. - * - * Return value : 0 - Ok - * -1 - Error - * - */ - -int WebRtcNetEQ_DownSampleTo4kHz(const int16_t *in, int inLen, uint16_t inFsHz, - int16_t *out, int outLen, int compensateDelay); - -#endif - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c deleted file mode 100644 index 1788635c7df..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c +++ /dev/null @@ -1,232 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of packet buffer for DTMF messages. - */ - -#include "dtmf_buffer.h" - -#include "typedefs.h" /* to define endianness */ -#include "signal_processing_library.h" - -#include "neteq_error_codes.h" - - -#ifdef NETEQ_ATEVENT_DECODE - -int16_t WebRtcNetEQ_DtmfRemoveEvent(dtmf_inst_t *DTMFdec_inst) -{ - - int i; - for (i = 0; i < 3; i++) - { - DTMFdec_inst->EventQueue[i] = DTMFdec_inst->EventQueue[i + 1]; - DTMFdec_inst->EventQueueVolume[i] = DTMFdec_inst->EventQueueVolume[i + 1]; - DTMFdec_inst->EventQueueEnded[i] = DTMFdec_inst->EventQueueEnded[i + 1]; - DTMFdec_inst->EventQueueStartTime[i] = DTMFdec_inst->EventQueueStartTime[i + 1]; - DTMFdec_inst->EventQueueEndTime[i] = DTMFdec_inst->EventQueueEndTime[i + 1]; - } - DTMFdec_inst->EventBufferSize--; - DTMFdec_inst->EventQueue[3] = -1; - DTMFdec_inst->EventQueueVolume[3] = 0; - DTMFdec_inst->EventQueueEnded[3] = 0; - DTMFdec_inst->EventQueueStartTime[3] = 0; - DTMFdec_inst->EventQueueEndTime[3] = 0; - - return 0; -} - -int16_t WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, uint16_t fs, - int16_t MaxPLCtime) -{ - int i; - if (((fs != 8000) && (fs != 16000) && (fs != 32000) && (fs != 48000)) || (MaxPLCtime < 0)) - { - return DTMF_DEC_PARAMETER_ERROR; - } - if (fs == 8000) - DTMFdec_inst->framelen = 80; - else if (fs == 16000) - DTMFdec_inst->framelen = 160; - else if (fs == 32000) - DTMFdec_inst->framelen = 320; - else - /* fs == 48000 */ - DTMFdec_inst->framelen = 480; - - DTMFdec_inst->MaxPLCtime = MaxPLCtime; - DTMFdec_inst->CurrentPLCtime = 0; - DTMFdec_inst->EventBufferSize = 0; - for (i = 0; i < 4; i++) - { - DTMFdec_inst->EventQueue[i] = -1; - DTMFdec_inst->EventQueueVolume[i] = 0; - DTMFdec_inst->EventQueueEnded[i] = 0; - DTMFdec_inst->EventQueueStartTime[i] = 0; - DTMFdec_inst->EventQueueEndTime[i] = 0; - } - return 0; -} - -int16_t WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst, - const int16_t *encoded, int16_t len, - uint32_t timeStamp) -{ - - int i; - int16_t value; - const int16_t *EventStart; - int16_t endEvent; - int16_t Volume; - int16_t Duration; - int16_t position = -1; - - /* Extract event */ - if (len == 4) - { - EventStart = encoded; -#ifdef WEBRTC_ARCH_BIG_ENDIAN - value=((*EventStart)>>8); - endEvent=((*EventStart)&0x80)>>7; - Volume=((*EventStart)&0x3F); - Duration=EventStart[1]; -#else - value = ((*EventStart) & 0xFF); - endEvent = ((*EventStart) & 0x8000) >> 15; - Volume = ((*EventStart) & 0x3F00) >> 8; - Duration = (((((uint16_t) EventStart[1]) >> 8) & 0xFF) - | (((uint16_t) (EventStart[1] & 0xFF)) << 8)); -#endif - /* Only events between 0-15 are supported (DTMF tones) */ - if ((value < 0) || (value > 15)) - { - return 0; - } - - /* Discard all DTMF tones with really low volume (<-36dbm0) */ - if (Volume > 36) - { - return 0; - } - - /*Are there any unended events of the same type? */ - for (i = 0; i < DTMFdec_inst->EventBufferSize; i++) - { - /* Going through the whole queue even when we have found a match will - ensure that we add to the latest applicable event */ - if ((DTMFdec_inst->EventQueue[i] == value) && (!DTMFdec_inst->EventQueueEnded[i] - || endEvent)) position = i; - } - if (position > -1) - { - DTMFdec_inst->EventQueueVolume[position] = Volume; - if ((timeStamp + Duration) > DTMFdec_inst->EventQueueEndTime[position]) DTMFdec_inst->EventQueueEndTime[position] - = DTMFdec_inst->EventQueueStartTime[position] + Duration; - if (endEvent) DTMFdec_inst->EventQueueEnded[position] = 1; - } - else - { - if (DTMFdec_inst->EventBufferSize == MAX_DTMF_QUEUE_SIZE) - { /* Buffer full */ - /* Remove one event */ - DTMFdec_inst->EventBufferSize--; - } - /* Store data in the instance on a new position*/ - DTMFdec_inst->EventQueue[DTMFdec_inst->EventBufferSize] = value; - DTMFdec_inst->EventQueueVolume[DTMFdec_inst->EventBufferSize] = Volume; - DTMFdec_inst->EventQueueEnded[DTMFdec_inst->EventBufferSize] = endEvent; - DTMFdec_inst->EventQueueStartTime[DTMFdec_inst->EventBufferSize] = timeStamp; - DTMFdec_inst->EventQueueEndTime[DTMFdec_inst->EventBufferSize] = timeStamp - + Duration; - DTMFdec_inst->EventBufferSize++; - } - return 0; - } - return DTMF_INSERT_ERROR; -} - -int16_t WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, int16_t *event, - int16_t *volume, uint32_t currTimeStamp) -{ - - if (DTMFdec_inst->EventBufferSize < 1) return 0; /* No events to play */ - - /* We have events, is it time to play them? */ - if (currTimeStamp < DTMFdec_inst->EventQueueStartTime[0]) - { - /*No, just return zero */ - return 0; - } - - /* Continue on the event that is currently ongoing */ - *event = DTMFdec_inst->EventQueue[0]; - *volume = DTMFdec_inst->EventQueueVolume[0]; - - if (DTMFdec_inst->EventQueueEndTime[0] >= (currTimeStamp + DTMFdec_inst->framelen)) - { - - /* Still at least framLen to play */ - - DTMFdec_inst->CurrentPLCtime = 0; - if ((DTMFdec_inst->EventQueueEndTime[0] == (currTimeStamp + DTMFdec_inst->framelen)) - && (DTMFdec_inst->EventQueueEnded[0])) - { /* We are done */ - /*Remove the event from Queue*/ - WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst); - } - return DTMFdec_inst->framelen; - - } - else - { - if ((DTMFdec_inst->EventQueueEnded[0]) || (DTMFdec_inst->EventQueue[1] > -1)) - { - /* - * Less than frameLen to play and end of event or already received next event. - * Give our a whole frame size of audio to simplify things. - */ - - /*Remove the event from Queue*/ - WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst); - DTMFdec_inst->CurrentPLCtime = 0; - - return DTMFdec_inst->framelen; - - } - else - { - /* Less than frameLen to play and not end of event. */ - DTMFdec_inst->CurrentPLCtime = (int16_t) (currTimeStamp - - DTMFdec_inst->EventQueueEndTime[0]); - - if ((DTMFdec_inst->CurrentPLCtime > DTMFdec_inst->MaxPLCtime) - || (DTMFdec_inst->CurrentPLCtime < -DTMFdec_inst->MaxPLCtime)) - { - /*Remove the event from queue*/ - WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst); - DTMFdec_inst->CurrentPLCtime = 0; - } - - /* If we have a new event that it's time to play */ - if ((DTMFdec_inst->EventQueue[1] > -1) && (DTMFdec_inst->EventQueueStartTime[1] - >= (currTimeStamp + DTMFdec_inst->framelen))) - { - /*Remove the event from queue*/ - WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst); - DTMFdec_inst->CurrentPLCtime = 0; - } - - return DTMFdec_inst->framelen; - } - } -} - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc index 1c81ad940c4..91debee14e8 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/dtmf_buffer.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h" #include <assert.h> #include <algorithm> // max diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.h index 99c9e6a499b..5dd31c2d2e7 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -8,94 +8,109 @@ * be found in the AUTHORS file in the root of the source tree. */ -/* - * Packet buffer for DTMF messages. - */ - -#ifndef DTMF_BUFFER_H -#define DTMF_BUFFER_H - -#include "typedefs.h" - -#include "neteq_defines.h" - -/* Include this code only if ATEVENT (DTMF) is defined in */ -#ifdef NETEQ_ATEVENT_DECODE - -#define MAX_DTMF_QUEUE_SIZE 4 - -typedef struct dtmf_inst_t_ -{ - int16_t MaxPLCtime; - int16_t CurrentPLCtime; - int16_t EventQueue[MAX_DTMF_QUEUE_SIZE]; - int16_t EventQueueVolume[MAX_DTMF_QUEUE_SIZE]; - int16_t EventQueueEnded[MAX_DTMF_QUEUE_SIZE]; - uint32_t EventQueueStartTime[MAX_DTMF_QUEUE_SIZE]; - uint32_t EventQueueEndTime[MAX_DTMF_QUEUE_SIZE]; - int16_t EventBufferSize; - int16_t framelen; -} dtmf_inst_t; - -/**************************************************************************** - * WebRtcNetEQ_DtmfDecoderInit(...) - * - * This function initializes a DTMF instance. - * - * Input: - * - DTMF_decinst_t : DTMF instance - * - fs : The sample rate used for the DTMF - * - MaxPLCtime : Maximum length for a PLC before zeros should be inserted - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, uint16_t fs, - int16_t MaxPLCtime); - -/**************************************************************************** - * WebRtcNetEQ_DtmfInsertEvent(...) - * - * This function decodes a packet with DTMF frames. - * - * Input: - * - DTMFdec_inst : DTMF instance - * - encoded : Encoded DTMF frame(s) - * - len : Bytes in encoded vector - * - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst, - const int16_t *encoded, int16_t len, - uint32_t timeStamp); - -/**************************************************************************** - * WebRtcNetEQ_DtmfDecode(...) - * - * This function decodes a packet with DTMF frame(s). Output will be the - * event that should be played for next 10 ms. - * - * Input: - * - DTMFdec_inst : DTMF instance - * - currTimeStamp : The current playout timestamp - * - * Output: - * - event : Event number to be played - * - volume : Event volume to be played - * - * Return value : >0 - There is a event to be played - * 0 - No event to be played - * -1 - Error - */ - -int16_t WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, int16_t *event, - int16_t *volume, uint32_t currTimeStamp); - -#endif /* NETEQ_ATEVENT_DECODE */ - -#endif /* DTMF_BUFFER_H */ - +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_ + +#include <list> +#include <string> // size_t + +#include "webrtc/base/constructormagic.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct DtmfEvent { + uint32_t timestamp; + int event_no; + int volume; + int duration; + bool end_bit; + + // Constructors + DtmfEvent() + : timestamp(0), + event_no(0), + volume(0), + duration(0), + end_bit(false) { + } + DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end) + : timestamp(ts), + event_no(ev), + volume(vol), + duration(dur), + end_bit(end) { + } +}; + +// This is the buffer holding DTMF events while waiting for them to be played. +class DtmfBuffer { + public: + enum BufferReturnCodes { + kOK = 0, + kInvalidPointer, + kPayloadTooShort, + kInvalidEventParameters, + kInvalidSampleRate + }; + + // Set up the buffer for use at sample rate |fs_hz|. + explicit DtmfBuffer(int fs_hz) { + SetSampleRate(fs_hz); + } + + virtual ~DtmfBuffer() {} + + // Flushes the buffer. + virtual void Flush() { buffer_.clear(); } + + // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733) + // and write the parsed information into the struct |event|. Input variable + // |rtp_timestamp| is simply copied into the struct. + static int ParseEvent(uint32_t rtp_timestamp, + const uint8_t* payload, + int payload_length_bytes, + DtmfEvent* event); + + // Inserts |event| into the buffer. The method looks for a matching event and + // merges the two if a match is found. + virtual int InsertEvent(const DtmfEvent& event); + + // Checks if a DTMF event should be played at time |current_timestamp|. If so, + // the method returns true; otherwise false. The parameters of the event to + // play will be written to |event|. + virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event); + + // Number of events in the buffer. + virtual size_t Length() const { return buffer_.size(); } + + virtual bool Empty() const { return buffer_.empty(); } + + // Set a new sample rate. + virtual int SetSampleRate(int fs_hz); + + private: + typedef std::list<DtmfEvent> DtmfList; + + int max_extrapolation_samples_; + int frame_len_samples_; // TODO(hlundin): Remove this later. + + // Compares two events and returns true if they are the same. + static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b); + + // Merges |event| to the event pointed out by |it|. The method checks that + // the two events are the same (using the SameEvent method), and merges them + // if that was the case, returning true. If the events are not the same, false + // is returned. + bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event); + + // Method used by the sort algorithm to rank events in the buffer. + static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b); + + DtmfList buffer_; + + DISALLOW_COPY_AND_ASSIGN(DtmfBuffer); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc index 0b5ed65b8e0..83f981386b7 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/dtmf_buffer.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h" #ifdef WIN32 #include <winsock2.h> // ntohl() diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc index c85534e9b7f..34c615d70f6 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc @@ -28,7 +28,7 @@ // 852 Hz 7 8 9 14 // 941 Hz 10 0 11 15 -#include "webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h" #include <assert.h> diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h index e93f0b883f5..fc1e5e4ad3d 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_TONE_GENERATOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_TONE_GENERATOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_ -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -53,4 +53,4 @@ class DtmfToneGenerator { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_TONE_GENERATOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc index 37e8bbda96c..94f79dc3456 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc @@ -10,12 +10,12 @@ // Unit tests for DtmfToneGenerator class. -#include "webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h" #include <math.h> #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c deleted file mode 100644 index 8ea413c76f1..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c +++ /dev/null @@ -1,367 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the DTMF tone generator and its parameters. - * - * A sinusoid is generated using the recursive oscillator model - * - * y[n] = sin(w*n + phi) = 2*cos(w) * y[n-1] - y[n-2] - * = a * y[n-1] - y[n-2] - * - * initialized with - * y[-2] = 0 - * y[-1] = sin(w) - * - * A DTMF signal is a combination of two sinusoids, depending - * on which event is sent (i.e, which key is pressed). The following - * table maps each key (event codes in parentheses) into two tones: - * - * 1209 Hz 1336 Hz 1477 Hz 1633 Hz - * 697 Hz 1 (ev. 1) 2 (ev. 2) 3 (ev. 3) A (ev. 12) - * 770 Hz 4 (ev. 4) 5 (ev. 5) 6 (ev. 6) B (ev. 13) - * 852 Hz 7 (ev. 7) 8 (ev. 8) 9 (ev. 9) C (ev. 14) - * 941 Hz * (ev. 10) 0 (ev. 0) # (ev. 11) D (ev. 15) - * - * The two tones are added to form the DTMF signal. - * - */ - -#include "dtmf_tonegen.h" - -#include "signal_processing_library.h" - -#include "neteq_error_codes.h" - -#ifdef NETEQ_ATEVENT_DECODE -/* Must compile NetEQ with DTMF support to enable the functionality */ - -/*******************/ -/* Constant tables */ -/*******************/ - -/* - * All tables corresponding to the oscillator model are organized so that - * the coefficients for a specific frequency is found in the same position - * in every table. The positions for the tones follow this layout: - * - * dummyVector[8] = - * { - * 697 Hz, 770 Hz, 852 Hz, 941 Hz, - * 1209 Hz, 1336 Hz, 1477 Hz, 1633 Hz - * }; - */ - -/* - * Tables for the constant a = 2*cos(w) = 2*cos(2*pi*f/fs) - * in the oscillator model, for 8, 16, 32 and 48 kHz sample rate. - * Table values in Q14. - */ - -const int16_t WebRtcNetEQ_dtfm_aTbl8Khz[8] = -{ - 27980, 26956, 25701, 24219, - 19073, 16325, 13085, 9315 -}; - -#ifdef NETEQ_WIDEBAND -const int16_t WebRtcNetEQ_dtfm_aTbl16Khz[8]= -{ - 31548, 31281, 30951, 30556, - 29144, 28361, 27409, 26258 -}; -#endif - -#ifdef NETEQ_32KHZ_WIDEBAND -const int16_t WebRtcNetEQ_dtfm_aTbl32Khz[8]= -{ - 32462, 32394, 32311, 32210, - 31849, 31647, 31400, 31098 -}; -#endif - -#ifdef NETEQ_48KHZ_WIDEBAND -const int16_t WebRtcNetEQ_dtfm_aTbl48Khz[8]= -{ - 32632, 32602, 32564, 32520, - 32359, 32268, 32157, 32022 -}; -#endif - -/* - * Initialization values y[-1] = sin(w) = sin(2*pi*f/fs), for 8, 16, 32 and 48 kHz sample rate. - * Table values in Q14. - */ - -const int16_t WebRtcNetEQ_dtfm_yInitTab8Khz[8] = -{ - 8528, 9315, 10163, 11036, - 13323, 14206,15021, 15708 -}; - -#ifdef NETEQ_WIDEBAND -const int16_t WebRtcNetEQ_dtfm_yInitTab16Khz[8]= -{ - 4429, 4879, 5380, 5918, - 7490, 8207, 8979, 9801 -}; -#endif - -#ifdef NETEQ_32KHZ_WIDEBAND -const int16_t WebRtcNetEQ_dtfm_yInitTab32Khz[8]= -{ - 2235, 2468, 2728, 3010, - 3853, 4249, 4685, 5164 -}; -#endif - -#ifdef NETEQ_48KHZ_WIDEBAND -const int16_t WebRtcNetEQ_dtfm_yInitTab48Khz[8]= -{ - 1493, 1649, 1823, 2013, - 2582, 2851, 3148, 3476 -}; -#endif - -/* Volume in dBm0 from 0 to -63, where 0 is the first table entry. - Everything below -36 is discarded, wherefore the table stops at -36. - Table entries are in Q14. - */ - -const int16_t WebRtcNetEQ_dtfm_dBm0[37] = { 16141, 14386, 12821, 11427, 10184, 9077, 8090, - 7210, 6426, 5727, 5104, 4549, 4054, 3614, - 3221, 2870, 2558, 2280, 2032, 1811, 1614, - 1439, 1282, 1143, 1018, 908, 809, 721, 643, - 573, 510, 455, 405, 361, 322, 287, 256 }; - -/**************************************************************************** - * WebRtcNetEQ_DTMFGenerate(...) - * - * Generate 10 ms DTMF signal according to input parameters. - * - * Input: - * - DTMFdecInst : DTMF instance - * - value : DTMF event number (0-15) - * - volume : Volume of generated signal (0-36) - * Volume is given in negative dBm0, i.e., volume == 0 - * means 0 dBm0 while volume == 36 mean -36 dBm0. - * - sampFreq : Sample rate in Hz - * - * Output: - * - signal : Pointer to vector where DTMF signal is stored; - * Vector must be at least sampFreq/100 samples long. - * - DTMFdecInst : Updated DTMF instance - * - * Return value : >0 - Number of samples written to signal - * : <0 - error - */ - -int16_t WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, int16_t value, - int16_t volume, int16_t *signal, - uint16_t sampFreq, int16_t extFrameLen) -{ - const int16_t *aTbl; /* pointer to a-coefficient table */ - const int16_t *yInitTable; /* pointer to initialization value table */ - int16_t a1 = 0; /* a-coefficient for first tone (low tone) */ - int16_t a2 = 0; /* a-coefficient for second tone (high tone) */ - int i; - int frameLen; /* number of samples to generate */ - int lowIndex = 0; /* Default to avoid compiler warnings. */ - int highIndex = 4; /* Default to avoid compiler warnings. */ - int32_t tempVal; - int16_t tempValLow; - int16_t tempValHigh; - - /* Sanity check for volume */ - if ((volume < 0) || (volume > 36)) - { - return DTMF_DEC_PARAMETER_ERROR; - } - - /* Sanity check for extFrameLen */ - if (extFrameLen < -1) - { - return DTMF_DEC_PARAMETER_ERROR; - } - - /* Select oscillator coefficient tables based on sample rate */ - if (sampFreq == 8000) - { - aTbl = WebRtcNetEQ_dtfm_aTbl8Khz; - yInitTable = WebRtcNetEQ_dtfm_yInitTab8Khz; - frameLen = 80; -#ifdef NETEQ_WIDEBAND - } - else if (sampFreq == 16000) - { - aTbl = WebRtcNetEQ_dtfm_aTbl16Khz; - yInitTable = WebRtcNetEQ_dtfm_yInitTab16Khz; - frameLen = 160; -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - } - else if (sampFreq == 32000) - { - aTbl = WebRtcNetEQ_dtfm_aTbl32Khz; - yInitTable = WebRtcNetEQ_dtfm_yInitTab32Khz; - frameLen = 320; -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - } - else if (sampFreq == 48000) - { - aTbl = WebRtcNetEQ_dtfm_aTbl48Khz; - yInitTable = WebRtcNetEQ_dtfm_yInitTab48Khz; - frameLen = 480; -#endif - } - else - { - /* unsupported sample rate */ - return DTMF_GEN_UNKNOWN_SAMP_FREQ; - } - - if (extFrameLen >= 0) - { - frameLen = extFrameLen; - } - - /* select low frequency based on event value */ - switch (value) - { - case 1: - case 2: - case 3: - case 12: /* first row on keypad */ - { - lowIndex = 0; /* low frequency: 697 Hz */ - break; - } - case 4: - case 5: - case 6: - case 13: /* second row on keypad */ - { - lowIndex = 1; /* low frequency: 770 Hz */ - break; - } - case 7: - case 8: - case 9: - case 14: /* third row on keypad */ - { - lowIndex = 2; /* low frequency: 852 Hz */ - break; - } - case 0: - case 10: - case 11: - case 15: /* fourth row on keypad */ - { - lowIndex = 3; /* low frequency: 941 Hz */ - break; - } - default: - { - return DTMF_DEC_PARAMETER_ERROR; - } - } /* end switch */ - - /* select high frequency based on event value */ - switch (value) - { - case 1: - case 4: - case 7: - case 10: /* first column on keypad */ - { - highIndex = 4; /* high frequency: 1209 Hz */ - break; - } - case 2: - case 5: - case 8: - case 0: /* second column on keypad */ - { - highIndex = 5;/* high frequency: 1336 Hz */ - break; - } - case 3: - case 6: - case 9: - case 11: /* third column on keypad */ - { - highIndex = 6;/* high frequency: 1477 Hz */ - break; - } - case 12: - case 13: - case 14: - case 15: /* fourth column on keypad (special) */ - { - highIndex = 7;/* high frequency: 1633 Hz */ - break; - } - } /* end switch */ - - /* select coefficients based on results from switches above */ - a1 = aTbl[lowIndex]; /* coefficient for first (low) tone */ - a2 = aTbl[highIndex]; /* coefficient for second (high) tone */ - - if (DTMFdecInst->reinit) - { - /* set initial values for the recursive model */ - DTMFdecInst->oldOutputLow[0] = yInitTable[lowIndex]; - DTMFdecInst->oldOutputLow[1] = 0; - DTMFdecInst->oldOutputHigh[0] = yInitTable[highIndex]; - DTMFdecInst->oldOutputHigh[1] = 0; - - /* reset reinit flag */ - DTMFdecInst->reinit = 0; - } - - /* generate signal sample by sample */ - for (i = 0; i < frameLen; i++) - { - - /* Use rescursion formula y[n] = a*y[n-1] - y[n-2] */ - tempValLow - = (int16_t) (((WEBRTC_SPL_MUL_16_16(a1, DTMFdecInst->oldOutputLow[1]) - + 8192) >> 14) - DTMFdecInst->oldOutputLow[0]); - tempValHigh - = (int16_t) (((WEBRTC_SPL_MUL_16_16(a2, DTMFdecInst->oldOutputHigh[1]) - + 8192) >> 14) - DTMFdecInst->oldOutputHigh[0]); - - /* Update recursion memory */ - DTMFdecInst->oldOutputLow[0] = DTMFdecInst->oldOutputLow[1]; - DTMFdecInst->oldOutputLow[1] = tempValLow; - DTMFdecInst->oldOutputHigh[0] = DTMFdecInst->oldOutputHigh[1]; - DTMFdecInst->oldOutputHigh[1] = tempValHigh; - - /* scale high tone with 32768 (15 left shifts) - and low tone with 23171 (3dB lower than high tone) */ - tempVal = WEBRTC_SPL_MUL_16_16(DTMF_AMP_LOW, tempValLow) - + WEBRTC_SPL_LSHIFT_W32((int32_t)tempValHigh, 15); - - /* Norm the signal to Q14 (with proper rounding) */ - tempVal = (tempVal + 16384) >> 15; - - /* Scale the signal to correct dbM0 value */ - signal[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32( - (WEBRTC_SPL_MUL_16_16(tempVal, WebRtcNetEQ_dtfm_dBm0[volume]) - + 8192), 14); /* volume value is in Q14; use proper rounding */ - } - - return frameLen; - -} - -#endif /* NETEQ_ATEVENT_DECODE */ - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.h deleted file mode 100644 index 5f4489940c2..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.h +++ /dev/null @@ -1,73 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the DTMF tone generator function. - */ - -#ifndef DTMF_TONEGEN_H -#define DTMF_TONEGEN_H - -#include "typedefs.h" - -#include "neteq_defines.h" - -#ifdef NETEQ_ATEVENT_DECODE -/* Must compile NetEQ with DTMF support to enable the functionality */ - -#define DTMF_AMP_LOW 23171 /* 3 dB lower than the high frequency */ - -/* The DTMF generator struct (part of DSP main struct DSPInst_t) */ -typedef struct dtmf_tone_inst_t_ -{ - - int16_t reinit; /* non-zero if the oscillator model should - be reinitialized for next event */ - int16_t oldOutputLow[2]; /* oscillator recursion history (low tone) */ - int16_t oldOutputHigh[2]; /* oscillator recursion history (high tone) */ - - int lastDtmfSample; /* index to the first non-DTMF sample in the - speech history, if non-negative */ -}dtmf_tone_inst_t; - -/**************************************************************************** - * WebRtcNetEQ_DTMFGenerate(...) - * - * Generate 10 ms DTMF signal according to input parameters. - * - * Input: - * - DTMFdecInst : DTMF instance - * - value : DTMF event number (0-15) - * - volume : Volume of generated signal (0-36) - * Volume is given in negative dBm0, i.e., volume == 0 - * means 0 dBm0 while volume == 36 mean -36 dBm0. - * - sampFreq : Sample rate in Hz - * - * Output: - * - signal : Pointer to vector where DTMF signal is stored; - * Vector must be at least sampFreq/100 samples long. - * - DTMFdecInst : Updated DTMF instance - * - * Return value : >0 - Number of samples written to signal - * : <0 - Error - */ - -int16_t WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, - int16_t value, - int16_t volume, - int16_t *signal, - uint16_t sampFreq, - int16_t frameLen -); - -#endif /* NETEQ_ATEVENT_DECODE */ - -#endif /* DTMF_TONEGEN_H */ - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.c deleted file mode 100644 index 9959f9222d7..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.c +++ /dev/null @@ -1,1220 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This is the function to expand from the speech history, to produce concealment data or - * increasing delay. - */ - -#include "dsp.h" - -#include <assert.h> - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" -#include "neteq_error_codes.h" - -#define CHECK_NO_OF_CORRMAX 3 -#define DISTLEN 20 -#define LPCANALASYSLEN 160 - -/* Scratch usage: - - Type Name size startpos endpos - (First part of first expand) - int16_t pw16_bestCorrIndex 3 0 2 - int16_t pw16_bestCorr 3 3 5 - int16_t pw16_bestDistIndex 3 6 8 - int16_t pw16_bestDist 3 9 11 - int16_t pw16_corrVec 102*fs/8000 12 11+102*fs/8000 - func WebRtcNetEQ_Correlator 232 12+102*fs/8000 243+102*fs/8000 - - (Second part of first expand) - int32_t pw32_corr2 99*fs/8000+1 0 99*fs/8000 - int32_t pw32_autoCorr 2*7 0 13 - int16_t pw16_rc 6 14 19 - - Signal combination: - int16_t pw16_randVec 30+120*fs/8000 0 29+120*fs/8000 - int16_t pw16_scaledRandVec 125*fs/8000 30+120*fs/8000 29+245*fs/8000 - int16_t pw16_unvoicedVecSpace 10+125*fs/8000 30+245*fs/8000 39+370*fs/8000 - - Total: 40+370*fs/8000 (size depends on UNVOICED_LPC_ORDER and BGN_LPC_ORDER) - */ - -#if ((BGN_LPC_ORDER > 10) || (UNVOICED_LPC_ORDER > 10)) && (defined SCRATCH) -#error BGN_LPC_ORDER and/or BGN_LPC_ORDER are too large for current scratch memory allocation -#endif - -#define SCRATCH_PW16_BEST_CORR_INDEX 0 -#define SCRATCH_PW16_BEST_CORR 3 -#define SCRATCH_PW16_BEST_DIST_INDEX 6 -#define SCRATCH_PW16_BEST_DIST 9 -#define SCRATCH_PW16_CORR_VEC 12 -#define SCRATCH_PW16_CORR2 0 -#define SCRATCH_PW32_AUTO_CORR 0 -#define SCRATCH_PW16_RC 14 -#define SCRATCH_PW16_RAND_VEC 0 - -#if (defined(NETEQ_48KHZ_WIDEBAND)) -#define SCRATCH_NETEQDSP_CORRELATOR 624 -#define SCRATCH_PW16_SCALED_RAND_VEC 750 -#define SCRATCH_PW16_UNVOICED_VEC_SPACE 1500 -#elif (defined(NETEQ_32KHZ_WIDEBAND)) -#define SCRATCH_NETEQDSP_CORRELATOR 420 -#define SCRATCH_PW16_SCALED_RAND_VEC 510 -#define SCRATCH_PW16_UNVOICED_VEC_SPACE 1010 -#elif (defined(NETEQ_WIDEBAND)) -#define SCRATCH_NETEQDSP_CORRELATOR 216 -#define SCRATCH_PW16_SCALED_RAND_VEC 270 -#define SCRATCH_PW16_UNVOICED_VEC_SPACE 520 -#else /* NB */ -#define SCRATCH_NETEQDSP_CORRELATOR 114 -#define SCRATCH_PW16_SCALED_RAND_VEC 150 -#define SCRATCH_PW16_UNVOICED_VEC_SPACE 275 -#endif - -/**************************************************************************** - * WebRtcNetEQ_Expand(...) - * - * This function produces one "chunk" of expansion data (PLC audio). The - * length of the produced audio depends on the speech history. - * - * Input: - * - inst : DSP instance - * - scratchPtr : Pointer to scratch vector - * - outdata : Pointer to a memory space where the output data - * should be stored - * - BGNonly : If non-zero, "expand" will only produce background noise. - * - pw16_len : Desired number of samples (only for BGN mode). - * - * Output: - * - inst : Updated instance - * - pw16_len : Number of samples that were output from NetEq - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_Expand(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_outData, int16_t *pw16_len, - int16_t BGNonly) -{ - - int16_t fs_mult; - ExpandInst_t *ExpandState = &(inst->ExpandInst); - BGNInst_t *BGNState = &(inst->BGNInst); - int i; -#ifdef SCRATCH - int16_t *pw16_randVec = pw16_scratchPtr + SCRATCH_PW16_RAND_VEC; - int16_t *pw16_scaledRandVec = pw16_scratchPtr + SCRATCH_PW16_SCALED_RAND_VEC; - int16_t *pw16_unvoicedVecSpace = pw16_scratchPtr + SCRATCH_PW16_UNVOICED_VEC_SPACE; -#else - int16_t pw16_randVec[FSMULT * 120 + 30]; /* 150 for NB and 270 for WB */ - int16_t pw16_scaledRandVec[FSMULT * 125]; /* 125 for NB and 250 for WB */ - int16_t pw16_unvoicedVecSpace[BGN_LPC_ORDER + FSMULT * 125]; -#endif - /* 125 for NB and 250 for WB etc. Reuse pw16_outData[] for this vector */ - int16_t *pw16_voicedVecStorage = pw16_outData; - int16_t *pw16_voicedVec = &pw16_voicedVecStorage[ExpandState->w16_overlap]; - int16_t *pw16_unvoicedVec = pw16_unvoicedVecSpace + UNVOICED_LPC_ORDER; - int16_t *pw16_cngVec = pw16_unvoicedVecSpace + BGN_LPC_ORDER; - int16_t w16_expVecsLen, w16_lag = 0, w16_expVecPos; - int16_t w16_randLen; - int16_t w16_vfractionChange; /* in Q14 */ - int16_t w16_winMute = 0, w16_winMuteInc = 0, w16_winUnMute = 0, w16_winUnMuteInc = 0; - int32_t w32_tmp; - int16_t w16_tmp, w16_tmp2; - int16_t stability; - enum BGNMode bgnMode = inst->BGNInst.bgnMode; - - /* Pre-calculate common multiplications with fs_mult */ - int16_t fsMult4; - int16_t fsMult20; - int16_t fsMult120; - int16_t fsMultDistLen; - int16_t fsMultLPCAnalasysLen; - -#ifdef NETEQ_STEREO - MasterSlaveInfo *msInfo = inst->msInfo; -#endif - - /* fs is uint16_t (to hold fs=48000) */ - fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs); /* calculate fs/8000 */ - - /* Pre-calculate common multiplications with fs_mult */ - fsMult4 = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, 4); - fsMult20 = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, 20); - fsMult120 = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, 120); - fsMultDistLen = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, DISTLEN); - fsMultLPCAnalasysLen = (int16_t) WEBRTC_SPL_MUL_16_16(fs_mult, LPCANALASYSLEN); - - /* - * Perform all the initial setup if it's the first expansion. - * If background noise (BGN) only, this setup is not needed. - */ - if (ExpandState->w16_consecExp == 0 && !BGNonly) - { - /* Setup more variables */ -#ifdef SCRATCH - int32_t *pw32_autoCorr = (int32_t*) (pw16_scratchPtr - + SCRATCH_PW32_AUTO_CORR); - int16_t *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC; - int16_t *pw16_bestCorrIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR_INDEX; - int16_t *pw16_bestCorr = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR; - int16_t *pw16_bestDistIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST_INDEX; - int16_t *pw16_bestDist = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST; - int16_t *pw16_corrVec = pw16_scratchPtr + SCRATCH_PW16_CORR_VEC; - int32_t *pw32_corr2 = (int32_t*) (pw16_scratchPtr + SCRATCH_PW16_CORR2); -#else - int32_t pw32_autoCorr[UNVOICED_LPC_ORDER+1]; - int16_t pw16_rc[UNVOICED_LPC_ORDER]; - int16_t pw16_corrVec[FSMULT*102]; /* 102 for NB */ - int16_t pw16_bestCorrIndex[CHECK_NO_OF_CORRMAX]; - int16_t pw16_bestCorr[CHECK_NO_OF_CORRMAX]; - int16_t pw16_bestDistIndex[CHECK_NO_OF_CORRMAX]; - int16_t pw16_bestDist[CHECK_NO_OF_CORRMAX]; - int32_t pw32_corr2[(99*FSMULT)+1]; -#endif - int32_t pw32_bestDist[CHECK_NO_OF_CORRMAX]; - int16_t w16_ind = 0; - int16_t w16_corrVecLen; - int16_t w16_corrScale; - int16_t w16_distScale; - int16_t w16_indMin, w16_indMax; - int16_t w16_len; - int32_t w32_en1, w32_en2, w32_cc; - int16_t w16_en1Scale, w16_en2Scale; - int16_t w16_en1, w16_en2; - int32_t w32_en1_mul_en2; - int16_t w16_sqrt_en1en2; - int16_t w16_ccShiftL; - int16_t w16_bestcorr; /* Correlation in Q14 */ - int16_t *pw16_vec1, *pw16_vec2; - int16_t w16_factor; - int16_t w16_DistLag, w16_CorrLag, w16_diffLag; - int16_t w16_energyLen; - int16_t w16_slope; - int16_t w16_startInd; - int16_t w16_noOfcorr2; - int16_t w16_scale; - - /* Initialize some variables */ - ExpandState->w16_lagsDirection = 1; - ExpandState->w16_lagsPosition = -1; - ExpandState->w16_expandMuteFactor = 16384; /* Start from 1.0 (Q14) */ - BGNState->w16_mutefactor = 0; /* Start with 0 gain for BGN (value in Q14) */ - inst->w16_seedInc = 1; - -#ifdef NETEQ_STEREO - /* Sanity for msInfo */ - if (msInfo == NULL) - { - /* this should not happen here */ - return MASTER_SLAVE_ERROR; - } - - /* - * Do not calculate correlations for slave instance(s) - * unless lag info from master is corrupt - */ - if ((msInfo->msMode != NETEQ_SLAVE) - || ((msInfo->distLag <= 0) || (msInfo->corrLag <= 0))) - { -#endif - /* Calculate correlation vector in downsampled domain (4 kHz sample rate) */ - w16_corrVecLen = WebRtcNetEQ_Correlator(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQDSP_CORRELATOR, -#endif - inst->pw16_speechHistory, inst->w16_speechHistoryLen, pw16_corrVec, - &w16_corrScale); - - /* Find peaks in correlation vector using parabolic fit method */ - WebRtcNetEQ_PeakDetection(pw16_corrVec, w16_corrVecLen, CHECK_NO_OF_CORRMAX, fs_mult, - pw16_bestCorrIndex, pw16_bestCorr); - - /* - * Adjust peak locations; cross-correlation lags start at 2.5 ms - * (20*fs_mult samples) - */ - pw16_bestCorrIndex[0] += fsMult20; - pw16_bestCorrIndex[1] += fsMult20; - pw16_bestCorrIndex[2] += fsMult20; - - /* Calculate distortion around the 3 (CHECK_NO_OF_CORRMAX) best lags */ - w16_distScale = 0; - for (i = 0; i < CHECK_NO_OF_CORRMAX; i++) - { - w16_tmp = fsMult20; - w16_tmp2 = pw16_bestCorrIndex[i] - fsMult4; - w16_indMin = WEBRTC_SPL_MAX(w16_tmp, w16_tmp2); - w16_tmp = fsMult120 - 1; - w16_tmp2 = pw16_bestCorrIndex[i] + fsMult4; - w16_indMax = WEBRTC_SPL_MIN(w16_tmp, w16_tmp2); - - pw16_bestDistIndex[i] = WebRtcNetEQ_MinDistortion( - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - fsMultDistLen]), - w16_indMin, w16_indMax, fsMultDistLen, &pw32_bestDist[i]); - - w16_distScale - = WEBRTC_SPL_MAX(16 - WebRtcSpl_NormW32(pw32_bestDist[i]), w16_distScale); - - } - - /* Shift the distortion values to fit in int16_t */ - WebRtcSpl_VectorBitShiftW32ToW16(pw16_bestDist, CHECK_NO_OF_CORRMAX, pw32_bestDist, - w16_distScale); - - /* - * Find index of maximum criteria, where crit[i] = bestCorr[i])/(bestDist[i]) - * Do this by a cross multiplication. - */ - - w32_en1 = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[0],pw16_bestDist[1]); - w32_en2 = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[1],pw16_bestDist[0]); - if (w32_en1 >= w32_en2) - { - /* 0 wins over 1 */ - w32_en1 - = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[0], pw16_bestDist[2]); - w32_en2 - = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[2], pw16_bestDist[0]); - if (w32_en1 >= w32_en2) - { - /* 0 wins over 2 */ - w16_ind = 0; - } - else - { - /* 2 wins over 0 */ - w16_ind = 2; - } - } - else - { - /* 1 wins over 0 */ - w32_en1 - = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[1],pw16_bestDist[2]); - w32_en2 - = WEBRTC_SPL_MUL_16_16((int32_t) pw16_bestCorr[2],pw16_bestDist[1]); - if ((int32_t) w32_en1 >= (int32_t) w32_en2) - { - /* 1 wins over 2 */ - w16_ind = 1; - } - else - { - /* 2 wins over 1 */ - w16_ind = 2; - } - } - -#ifdef NETEQ_STEREO - } - - /* Store DistLag and CorrLag of the position with highest criteria */ - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO) - || ((msInfo->msMode == NETEQ_SLAVE) && (msInfo->distLag <= 0 || msInfo->corrLag - <= 0))) - { - /* lags not provided externally */ - w16_DistLag = pw16_bestDistIndex[w16_ind]; - w16_CorrLag = pw16_bestCorrIndex[w16_ind]; - if (msInfo->msMode == NETEQ_MASTER) - { - msInfo->distLag = w16_DistLag; - msInfo->corrLag = w16_CorrLag; - } - } - else if (msInfo->msMode == NETEQ_SLAVE) - { - /* lags provided externally (from master) */ - w16_DistLag = msInfo->distLag; - w16_CorrLag = msInfo->corrLag; - - /* sanity for lag values */ - if ((w16_DistLag <= 0) || (w16_CorrLag <= 0)) - { - return MASTER_SLAVE_ERROR; - } - } - else - { - /* Invalid mode */ - return MASTER_SLAVE_ERROR; - } -#else /* not NETEQ_STEREO */ - w16_DistLag = pw16_bestDistIndex[w16_ind]; - w16_CorrLag = pw16_bestCorrIndex[w16_ind]; -#endif - - ExpandState->w16_maxLag = WEBRTC_SPL_MAX(w16_DistLag, w16_CorrLag); - - /* Calculate the exact best correlation (in the range within CorrLag-DistLag) */ - w16_len = w16_DistLag + 10; - w16_len = WEBRTC_SPL_MIN(w16_len, fsMult120); - w16_len = WEBRTC_SPL_MAX(w16_len, 60 * fs_mult); - - w16_startInd = WEBRTC_SPL_MIN(w16_DistLag, w16_CorrLag); - w16_noOfcorr2 = WEBRTC_SPL_ABS_W16((w16_DistLag-w16_CorrLag)) + 1; - /* w16_noOfcorr2 maximum value is 99*fs_mult + 1 */ - - /* Calculate suitable scaling */ - w16_tmp - = WebRtcSpl_MaxAbsValueW16( - &inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd - - w16_noOfcorr2], - (int16_t) (w16_len + w16_startInd + w16_noOfcorr2 - 1)); - w16_corrScale = ((31 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_tmp, w16_tmp))) - + (31 - WebRtcSpl_NormW32(w16_len))) - 31; - w16_corrScale = WEBRTC_SPL_MAX(0, w16_corrScale); - - /* - * Perform the correlation, store in pw32_corr2 - */ - - WebRtcNetEQ_CrossCorr(pw32_corr2, - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]), - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd]), - w16_len, w16_noOfcorr2, w16_corrScale, -1); - - /* Find maximizing index */ - w16_ind = WebRtcSpl_MaxIndexW32(pw32_corr2, w16_noOfcorr2); - w32_cc = pw32_corr2[w16_ind]; /* this is maximum correlation */ - w16_ind = w16_ind + w16_startInd; /* correct index for start offset */ - - /* Calculate energies */ - w32_en1 = WebRtcNetEQ_DotW16W16( - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]), - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]), w16_len, - w16_corrScale); - w32_en2 = WebRtcNetEQ_DotW16W16( - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_ind]), - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_ind]), - w16_len, w16_corrScale); - - /* Calculate the correlation value w16_bestcorr */ - if ((w32_en1 > 0) && (w32_en2 > 0)) - { - w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1); - w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale); - w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2); - w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale); - /* Make sure total scaling is even (to simplify scale factor after sqrt) */ - if ((w16_en1Scale + w16_en2Scale) & 1) - { - /* if sum is odd */ - w16_en1Scale += 1; - } - w16_en1 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale); - w16_en2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale); - w32_en1_mul_en2 = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2); - w16_sqrt_en1en2 = (int16_t) WebRtcSpl_SqrtFloor(w32_en1_mul_en2); - - /* Calculate cc/sqrt(en1*en2) in Q14 */ - w16_ccShiftL = 14 - ((w16_en1Scale + w16_en2Scale) >> 1); - w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_ccShiftL); - w16_bestcorr = (int16_t) WebRtcSpl_DivW32W16(w32_cc, w16_sqrt_en1en2); - w16_bestcorr = WEBRTC_SPL_MIN(16384, w16_bestcorr); /* set maximum to 1.0 */ - - } - else - { - /* if either en1 or en2 is zero */ - w16_bestcorr = 0; - } - - /* - * Extract the two vectors, pw16_expVecs[0][] and pw16_expVecs[1][], - * from the SpeechHistory[] - */ - w16_expVecsLen = ExpandState->w16_maxLag + ExpandState->w16_overlap; - pw16_vec1 = &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_expVecsLen]); - pw16_vec2 = pw16_vec1 - w16_DistLag; - /* Normalize the second vector to the same energy as the first */ - w32_en1 = WebRtcNetEQ_DotW16W16(pw16_vec1, pw16_vec1, w16_expVecsLen, w16_corrScale); - w32_en2 = WebRtcNetEQ_DotW16W16(pw16_vec2, pw16_vec2, w16_expVecsLen, w16_corrScale); - - /* - * Confirm that energy factor sqrt(w32_en1/w32_en2) is within difference 0.5 - 2.0 - * w32_en1/w32_en2 within 0.25 - 4 - */ - if (((w32_en1 >> 2) < w32_en2) && ((w32_en1) > (w32_en2 >> 2))) - { - - /* Energy constraint fulfilled => use both vectors and scale them accordingly */ - w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2); - w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale); - w16_en1Scale = w16_en2Scale - 13; - - /* calculate w32_en1/w32_en2 in Q13 */ - w32_en1_mul_en2 = WebRtcSpl_DivW32W16( - WEBRTC_SPL_SHIFT_W32(w32_en1, -w16_en1Scale), - (int16_t) (WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale))); - - /* calculate factor in Q13 (sqrt of en1/en2 in Q26) */ - w16_factor = (int16_t) WebRtcSpl_SqrtFloor( - WEBRTC_SPL_LSHIFT_W32(w32_en1_mul_en2, 13)); - - /* Copy the two vectors and give them the same energy */ - - WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[0], pw16_vec1, w16_expVecsLen); - WebRtcSpl_AffineTransformVector(ExpandState->pw16_expVecs[1], pw16_vec2, - w16_factor, 4096, 13, w16_expVecsLen); - - } - else - { - /* Energy change constraint not fulfilled => only use last vector */ - - WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[0], pw16_vec1, w16_expVecsLen); - WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[1], ExpandState->pw16_expVecs[0], - w16_expVecsLen); - - /* Set the w16_factor since it is used by muting slope */ - if (((w32_en1 >> 2) < w32_en2) || (w32_en2 == 0)) - { - w16_factor = 4096; /* 0.5 in Q13*/ - } - else - { - w16_factor = 16384; /* 2.0 in Q13*/ - } - } - - /* Set the 3 lag values */ - w16_diffLag = w16_DistLag - w16_CorrLag; - if (w16_diffLag == 0) - { - /* DistLag and CorrLag are equal */ - ExpandState->w16_lags[0] = w16_DistLag; - ExpandState->w16_lags[1] = w16_DistLag; - ExpandState->w16_lags[2] = w16_DistLag; - } - else - { - /* DistLag and CorrLag are not equal; use different combinations of the two */ - ExpandState->w16_lags[0] = w16_DistLag; /* DistLag only */ - ExpandState->w16_lags[1] = ((w16_DistLag + w16_CorrLag) >> 1); /* 50/50 */ - /* Third lag, move one half-step towards CorrLag (in both cases) */ - if (w16_diffLag > 0) - { - ExpandState->w16_lags[2] = (w16_DistLag + w16_CorrLag - 1) >> 1; - } - else - { - ExpandState->w16_lags[2] = (w16_DistLag + w16_CorrLag + 1) >> 1; - } - } - - /************************************************* - * Calculate the LPC and the gain of the filters * - *************************************************/ - - /* Calculate scale value needed for autocorrelation */ - w16_tmp = WebRtcSpl_MaxAbsValueW16( - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - fsMultLPCAnalasysLen]), - fsMultLPCAnalasysLen); - - w16_tmp = 16 - WebRtcSpl_NormW32(w16_tmp); - w16_tmp = WEBRTC_SPL_MIN(w16_tmp,0); - w16_tmp = (w16_tmp << 1) + 7; - w16_tmp = WEBRTC_SPL_MAX(w16_tmp,0); - - /* set w16_ind to simplify the following expressions */ - w16_ind = inst->w16_speechHistoryLen - fsMultLPCAnalasysLen - UNVOICED_LPC_ORDER; - - /* store first UNVOICED_LPC_ORDER samples in pw16_rc */ - - WEBRTC_SPL_MEMCPY_W16(pw16_rc, &inst->pw16_speechHistory[w16_ind], UNVOICED_LPC_ORDER); - - /* set first samples to zero */ - WebRtcSpl_MemSetW16(&inst->pw16_speechHistory[w16_ind], 0, UNVOICED_LPC_ORDER); - - /* Calculate UNVOICED_LPC_ORDER+1 lags of the ACF */ - - WebRtcNetEQ_CrossCorr( - pw32_autoCorr, &(inst->pw16_speechHistory[w16_ind + UNVOICED_LPC_ORDER]), - &(inst->pw16_speechHistory[w16_ind + UNVOICED_LPC_ORDER]), fsMultLPCAnalasysLen, - UNVOICED_LPC_ORDER + 1, w16_tmp, -1); - - /* Recover the stored samples from pw16_rc */ - - WEBRTC_SPL_MEMCPY_W16(&inst->pw16_speechHistory[w16_ind], pw16_rc, UNVOICED_LPC_ORDER); - - if (pw32_autoCorr[0] > 0) - { /* check that variance is positive */ - - /* estimate AR filter parameters using Levinson-Durbin algorithm - (UNVOICED_LPC_ORDER+1 filter coefficients) */ - stability = WebRtcSpl_LevinsonDurbin(pw32_autoCorr, ExpandState->pw16_arFilter, - pw16_rc, UNVOICED_LPC_ORDER); - - /* Only update BGN if filter is stable */ - if (stability != 1) - { - /* Set first coefficient to 4096 (1.0 in Q12)*/ - ExpandState->pw16_arFilter[0] = 4096; - /* Set remaining UNVOICED_LPC_ORDER coefficients to zero */ - WebRtcSpl_MemSetW16(ExpandState->pw16_arFilter + 1, 0, UNVOICED_LPC_ORDER); - } - - } - - if (w16_DistLag < 40) - { - w16_energyLen = 2 * w16_DistLag; - } - else - { - w16_energyLen = w16_DistLag; - } - w16_randLen = w16_energyLen + 30; /* Startup part */ - - /* Extract a noise segment */ - if (w16_randLen <= RANDVEC_NO_OF_SAMPLES) - { - WEBRTC_SPL_MEMCPY_W16(pw16_randVec, - (int16_t*) WebRtcNetEQ_kRandnTbl, w16_randLen); - } - else - { /* only applies to SWB where length could be larger than 256 */ -#if FSMULT >= 2 /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */ - WEBRTC_SPL_MEMCPY_W16(pw16_randVec, (int16_t*) WebRtcNetEQ_kRandnTbl, - RANDVEC_NO_OF_SAMPLES); - inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1); - assert(w16_randLen <= FSMULT * 120 + 30); - WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES], - (int16_t) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc); -#else - assert(0); -#endif - } - - /* Set up state vector and calculate scale factor for unvoiced filtering */ - - WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_arState, - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - UNVOICED_LPC_ORDER]), - UNVOICED_LPC_ORDER); - WEBRTC_SPL_MEMCPY_W16(pw16_unvoicedVec - UNVOICED_LPC_ORDER, - &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - 128 - UNVOICED_LPC_ORDER]), - UNVOICED_LPC_ORDER); - WebRtcSpl_FilterMAFastQ12(&inst->pw16_speechHistory[inst->w16_speechHistoryLen - 128], - pw16_unvoicedVec, ExpandState->pw16_arFilter, UNVOICED_LPC_ORDER + 1, 128); - if (WebRtcSpl_MaxAbsValueW16(pw16_unvoicedVec, 128) > 4000) - { - w16_scale = 4; - } - else - { - w16_scale = 0; - } - w32_tmp = WebRtcNetEQ_DotW16W16(pw16_unvoicedVec, pw16_unvoicedVec, 128, w16_scale); - - /* Normalize w32_tmp to 28 or 29 bits to preserve sqrt() accuracy */ - w16_tmp = WebRtcSpl_NormW32(w32_tmp) - 3; - w16_tmp += ((w16_tmp & 0x1) ^ 0x1); /* Make sure we do an odd number of shifts since we - from earlier have 7 shifts from dividing with 128.*/ - w32_tmp = WEBRTC_SPL_SHIFT_W32(w32_tmp, w16_tmp); - w32_tmp = WebRtcSpl_SqrtFloor(w32_tmp); - ExpandState->w16_arGainScale = 13 + ((w16_tmp + 7 - w16_scale) >> 1); - ExpandState->w16_arGain = (int16_t) w32_tmp; - - /******************************************************************** - * Calculate vfraction from bestcorr * - * if (bestcorr>0.480665) * - * vfraction = ((bestcorr-0.4)/(1-0.4)).^2 * - * else vfraction = 0 * - * * - * approximation (coefficients in Q12): * - * if (x>0.480665) (y(x)<0.3) * - * y(x) = -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 * - * else y(x) = 0; * - ********************************************************************/ - - if (w16_bestcorr > 7875) - { - /* if x>0.480665 */ - int16_t w16_x1, w16_x2, w16_x3; - w16_x1 = w16_bestcorr; - w32_tmp = WEBRTC_SPL_MUL_16_16((int32_t) w16_x1, w16_x1); - w16_x2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14); - w32_tmp = WEBRTC_SPL_MUL_16_16(w16_x1, w16_x2); - w16_x3 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14); - w32_tmp - = (int32_t) WEBRTC_SPL_LSHIFT_W32((int32_t) WebRtcNetEQ_kMixFractionFuncTbl[0], 14); - w32_tmp - += (int32_t) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[1], w16_x1); - w32_tmp - += (int32_t) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[2], w16_x2); - w32_tmp - += (int32_t) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[3], w16_x3); - ExpandState->w16_vFraction = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 12); - ExpandState->w16_vFraction = WEBRTC_SPL_MIN(ExpandState->w16_vFraction, 16384); - ExpandState->w16_vFraction = WEBRTC_SPL_MAX(ExpandState->w16_vFraction, 0); - } - else - { - ExpandState->w16_vFraction = 0; - } - - /*********************************************************************** - * Calculate muting slope, reuse value from earlier scaling of ExpVecs * - ***********************************************************************/ - w16_slope = w16_factor; - - if (w16_slope > 12288) - { - /* w16_slope > 1.5 ? */ - /* Calculate (1-(1/slope))/w16_DistLag = (slope-1)/(w16_DistLag*slope) */ - w32_tmp = w16_slope - 8192; - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 12); /* Value in Q25 (13+12=25) */ - w16_tmp = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(w16_DistLag, - w16_slope, 8); /* Value in Q5 (13-8=5) */ - w16_tmp = (int16_t) WebRtcSpl_DivW32W16(w32_tmp, - w16_tmp); /* Res in Q20 (25-5=20) */ - - if (w16_slope > 14746) - { /* w16_slope > 1.8 ? */ - ExpandState->w16_muteSlope = (w16_tmp + 1) >> 1; - } - else - { - ExpandState->w16_muteSlope = (w16_tmp + 4) >> 3; - } - ExpandState->w16_onset = 1; - } - else if (ExpandState->w16_vFraction > 13107) - { - /* w16_vFraction > 0.8 ? */ - if (w16_slope > 8028) - { - /* w16_vFraction > 0.98 ? */ - ExpandState->w16_muteSlope = 0; - } - else - { - /* Calculate (1-slope)/w16_DistLag */ - w32_tmp = 8192 - w16_slope; - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */ - ExpandState->w16_muteSlope = (int16_t) WebRtcSpl_DivW32W16(w32_tmp, - w16_DistLag); /* Res in Q20 (20-0=20) */ - } - ExpandState->w16_onset = 0; - } - else - { - /* - * Use the minimum of 0.005 (0.9 on 50 samples in NB and the slope) - * and ((1-slope)/w16_DistLag) - */ - w32_tmp = 8192 - w16_slope; - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */ - w32_tmp = WEBRTC_SPL_MAX(w32_tmp, 0); - ExpandState->w16_muteSlope = (int16_t) WebRtcSpl_DivW32W16(w32_tmp, - w16_DistLag); /* Res in Q20 (20-0=20) */ - w16_tmp = WebRtcNetEQ_k5243div[fs_mult]; /* 0.005/fs_mult = 5243/fs_mult */ - ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(w16_tmp, ExpandState->w16_muteSlope); - ExpandState->w16_onset = 0; - } - } - else - { - /* This is not the first Expansion, parameters are already estimated. */ - - /* Extract a noise segment */ - if (BGNonly) /* If we should produce nothing but background noise */ - { - if (*pw16_len > 0) - { - /* - * Set length to input parameter length, but not more than length - * of pw16_randVec - */ - w16_lag = WEBRTC_SPL_MIN(*pw16_len, FSMULT * 120 + 30); - } - else - { - /* set length to 15 ms */ - w16_lag = fsMult120; - } - w16_randLen = w16_lag; - } - else - { - w16_randLen = ExpandState->w16_maxLag; - } - - if (w16_randLen <= RANDVEC_NO_OF_SAMPLES) - { - inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1); - WebRtcNetEQ_RandomVec(&inst->uw16_seed, pw16_randVec, w16_randLen, - inst->w16_seedInc); - } - else - { /* only applies to SWB where length could be larger than 256 */ -#if FSMULT >= 2 /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */ - inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1); - WebRtcNetEQ_RandomVec(&inst->uw16_seed, pw16_randVec, RANDVEC_NO_OF_SAMPLES, - inst->w16_seedInc); - inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1); - assert(w16_randLen <= FSMULT * 120 + 30); - WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES], - (int16_t) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc); -#else - assert(0); -#endif - } - } /* end if(first expand or BGNonly) ... else ... */ - - if (!BGNonly) /* Voiced and unvoiced parts not used if generating BGN only */ - { - - /************************************************* - * Generate signal * - *************************************************/ - - /* - * Voiced part - */ - - /* Linearly mute the use_vfraction value from 1 to vfraction */ - if (ExpandState->w16_consecExp == 0) - { - ExpandState->w16_currentVFraction = 16384; /* 1.0 in Q14 */ - } - - ExpandState->w16_lagsPosition = ExpandState->w16_lagsPosition - + ExpandState->w16_lagsDirection; - - /* Change direction if needed */ - if (ExpandState->w16_lagsPosition == 0) - { - ExpandState->w16_lagsDirection = 1; - } - if (ExpandState->w16_lagsPosition == 2) - { - ExpandState->w16_lagsDirection = -1; - } - - /* Generate a weighted vector with the selected lag */ - w16_expVecsLen = ExpandState->w16_maxLag + ExpandState->w16_overlap; - w16_lag = ExpandState->w16_lags[ExpandState->w16_lagsPosition]; - /* Copy lag+overlap data */ - w16_expVecPos = w16_expVecsLen - w16_lag - ExpandState->w16_overlap; - w16_tmp = w16_lag + ExpandState->w16_overlap; - if (ExpandState->w16_lagsPosition == 0) - { - WEBRTC_SPL_MEMCPY_W16(pw16_voicedVecStorage, - &(ExpandState->pw16_expVecs[0][w16_expVecPos]), w16_tmp); - } - else if (ExpandState->w16_lagsPosition == 1) - { - WebRtcSpl_ScaleAndAddVectorsWithRound(&ExpandState->pw16_expVecs[0][w16_expVecPos], 3, - &ExpandState->pw16_expVecs[1][w16_expVecPos], 1, 2, pw16_voicedVecStorage, - w16_tmp); - - } - else if (ExpandState->w16_lagsPosition == 2) - { - WebRtcSpl_ScaleAndAddVectorsWithRound(&ExpandState->pw16_expVecs[0][w16_expVecPos], 1, - &ExpandState->pw16_expVecs[1][w16_expVecPos], 1, 1, pw16_voicedVecStorage, - w16_tmp); - } - - if (inst->fs == 8000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_8KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_8KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_8KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC; -#ifdef NETEQ_WIDEBAND - } - else if (inst->fs == 16000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_16KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_16KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_16KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC; -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - } - else if (inst->fs == 32000) - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_32KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_32KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_32KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC; -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - } - else /* if (inst->fs==48000) */ - { - /* Windowing in Q15 */ - w16_winMute = NETEQ_OVERLAP_WINMUTE_48KHZ_START; - w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_48KHZ_INC; - w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_48KHZ_START; - w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC; -#endif - } - - /* Smooth the expanded if it has not been muted to or vfraction is larger than 0.5 */ - if ((ExpandState->w16_expandMuteFactor > 819) && (ExpandState->w16_currentVFraction - > 8192)) - { - for (i = 0; i < ExpandState->w16_overlap; i++) - { - /* Do overlap add between new vector and overlap */ - ExpandState->pw16_overlapVec[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32( - WEBRTC_SPL_MUL_16_16(ExpandState->pw16_overlapVec[i], w16_winMute) + - WEBRTC_SPL_MUL_16_16( - WEBRTC_SPL_MUL_16_16_RSFT(ExpandState->w16_expandMuteFactor, - pw16_voicedVecStorage[i], 14), w16_winUnMute) + 16384, 15); - w16_winMute += w16_winMuteInc; - w16_winUnMute += w16_winUnMuteInc; - } - } - else if (ExpandState->w16_expandMuteFactor == 0 -#ifdef NETEQ_STEREO - && msInfo->msMode == NETEQ_MONO /* only if mono mode is selected */ -#endif - ) - { - /* if ExpandState->w16_expandMuteFactor = 0 => all is CNG component - set the output length to 15ms (for best CNG production) */ - w16_tmp = fsMult120; - ExpandState->w16_maxLag = w16_tmp; - ExpandState->w16_lags[0] = w16_tmp; - ExpandState->w16_lags[1] = w16_tmp; - ExpandState->w16_lags[2] = w16_tmp; - } - - /* - * Unvoiced part - */ - - WEBRTC_SPL_MEMCPY_W16(pw16_unvoicedVec - UNVOICED_LPC_ORDER, - ExpandState->pw16_arState, - UNVOICED_LPC_ORDER); - if (ExpandState->w16_arGainScale > 0) - { - w32_tmp = ((int32_t) 1) << (ExpandState->w16_arGainScale - 1); - } - else - { - w32_tmp = 0; - } - - /* Note that shift value can be >16 which complicates things for some DSPs */ - WebRtcSpl_AffineTransformVector(pw16_scaledRandVec, pw16_randVec, - ExpandState->w16_arGain, w32_tmp, ExpandState->w16_arGainScale, w16_lag); - - WebRtcSpl_FilterARFastQ12(pw16_scaledRandVec, pw16_unvoicedVec, - ExpandState->pw16_arFilter, UNVOICED_LPC_ORDER + 1, w16_lag); - - WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_arState, - &(pw16_unvoicedVec[w16_lag - UNVOICED_LPC_ORDER]), - UNVOICED_LPC_ORDER); - - /* - * Voiced + Unvoiced - */ - - /* For lag = - <=31*fs_mult => go from 1 to 0 in about 8 ms - (>=31..<=63)*fs_mult => go from 1 to 0 in about 16 ms - >=64*fs_mult => go from 1 to 0 in about 32 ms - */ - w16_tmp = (31 - WebRtcSpl_NormW32(ExpandState->w16_maxLag)) - 5; /* getbits(w16_maxLag) -5 */ - w16_vfractionChange = (int16_t) WEBRTC_SPL_RSHIFT_W32(256, w16_tmp); - if (ExpandState->w16_stopMuting == 1) - { - w16_vfractionChange = 0; - } - - /* Create combined signal (unmuted) by shifting in more and more of unvoiced part */ - w16_tmp = 8 - w16_tmp; /* getbits(w16_vfractionChange) */ - w16_tmp = (ExpandState->w16_currentVFraction - ExpandState->w16_vFraction) >> w16_tmp; - w16_tmp = WEBRTC_SPL_MIN(w16_tmp, w16_lag); - WebRtcNetEQ_MixVoiceUnvoice(pw16_outData, pw16_voicedVec, pw16_unvoicedVec, - &ExpandState->w16_currentVFraction, w16_vfractionChange, w16_tmp); - - if (w16_tmp < w16_lag) - { - if (w16_vfractionChange != 0) - { - ExpandState->w16_currentVFraction = ExpandState->w16_vFraction; - } - w16_tmp2 = 16384 - ExpandState->w16_currentVFraction; - WebRtcSpl_ScaleAndAddVectorsWithRound(pw16_voicedVec + w16_tmp, - ExpandState->w16_currentVFraction, pw16_unvoicedVec + w16_tmp, w16_tmp2, 14, - pw16_outData + w16_tmp, (int16_t) (w16_lag - w16_tmp)); - } - - /* Select muting factor */ - if (ExpandState->w16_consecExp == 3) - { - /* 0.95 on 50 samples in NB (0.0010/fs_mult in Q20) */ - ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(ExpandState->w16_muteSlope, - WebRtcNetEQ_k1049div[fs_mult]); - } - if (ExpandState->w16_consecExp == 7) - { - /* 0.90 on 50 samples in NB (0.0020/fs_mult in Q20) */ - ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(ExpandState->w16_muteSlope, - WebRtcNetEQ_k2097div[fs_mult]); - } - - /* Mute segment according to slope value */ - if ((ExpandState->w16_consecExp != 0) || (ExpandState->w16_onset != 1)) - { - /* Mute to the previous level, then continue with the muting */ - WebRtcSpl_AffineTransformVector(pw16_outData, pw16_outData, - ExpandState->w16_expandMuteFactor, 8192, 14, w16_lag); - - if ((ExpandState->w16_stopMuting != 1)) - { - WebRtcNetEQ_MuteSignal(pw16_outData, ExpandState->w16_muteSlope, w16_lag); - - w16_tmp = 16384 - (int16_t) ((WEBRTC_SPL_MUL_16_16(w16_lag, - ExpandState->w16_muteSlope) + 8192) >> 6); /* 20-14 = 6 */ - w16_tmp = (int16_t) ((WEBRTC_SPL_MUL_16_16(w16_tmp, - ExpandState->w16_expandMuteFactor) + 8192) >> 14); - - /* Guard against getting stuck with very small (but sometimes audible) gain */ - if ((ExpandState->w16_consecExp > 3) && (w16_tmp - >= ExpandState->w16_expandMuteFactor)) - { - ExpandState->w16_expandMuteFactor = 0; - } - else - { - ExpandState->w16_expandMuteFactor = w16_tmp; - } - } - } - - } /* end if(!BGNonly) */ - - /* - * BGN - */ - - if (BGNState->w16_initialized == 1) - { - /* BGN parameters are initialized; use them */ - - WEBRTC_SPL_MEMCPY_W16(pw16_cngVec - BGN_LPC_ORDER, - BGNState->pw16_filterState, - BGN_LPC_ORDER); - - if (BGNState->w16_scaleShift > 1) - { - w32_tmp = ((int32_t) 1) << (BGNState->w16_scaleShift - 1); - } - else - { - w32_tmp = 0; - } - - /* Scale random vector to correct energy level */ - /* Note that shift value can be >16 which complicates things for some DSPs */ - WebRtcSpl_AffineTransformVector(pw16_scaledRandVec, pw16_randVec, - BGNState->w16_scale, w32_tmp, BGNState->w16_scaleShift, w16_lag); - - WebRtcSpl_FilterARFastQ12(pw16_scaledRandVec, pw16_cngVec, BGNState->pw16_filter, - BGN_LPC_ORDER + 1, w16_lag); - - WEBRTC_SPL_MEMCPY_W16(BGNState->pw16_filterState, - &(pw16_cngVec[w16_lag-BGN_LPC_ORDER]), - BGN_LPC_ORDER); - - /* Unmute the insertion of background noise */ - - if (bgnMode == BGN_FADE && ExpandState->w16_consecExp >= FADE_BGN_TIME - && BGNState->w16_mutefactor > 0) - { - /* fade BGN to zero */ - /* calculate muting slope, approx 2^18/fsHz */ - int16_t muteFactor; - if (fs_mult == 1) - { - muteFactor = -32; - } - else if (fs_mult == 2) - { - muteFactor = -16; - } - else if (fs_mult == 4) - { - muteFactor = -8; - } - else - { - muteFactor = -5; - } - /* use UnmuteSignal function with negative slope */ - WebRtcNetEQ_UnmuteSignal(pw16_cngVec, &BGNState->w16_mutefactor, /* In Q14 */ - pw16_cngVec, muteFactor, /* In Q20 */ - w16_lag); - } - else if (BGNState->w16_mutefactor < 16384 && !BGNonly) - { - /* if (w16_mutefactor < 1) and not BGN only (since then we use no muting) */ - - /* - * If BGN_OFF, or if BNG_FADE has started fading, - * mutefactor should not be increased. - */ - if (ExpandState->w16_stopMuting != 1 && bgnMode != BGN_OFF && !(bgnMode - == BGN_FADE && ExpandState->w16_consecExp >= FADE_BGN_TIME)) - { - WebRtcNetEQ_UnmuteSignal(pw16_cngVec, &BGNState->w16_mutefactor, /* In Q14 */ - pw16_cngVec, ExpandState->w16_muteSlope, /* In Q20 */ - w16_lag); - } - else - { - /* BGN_ON and stop muting, or - * BGN_OFF (mute factor is always 0), or - * BGN_FADE has reached 0 */ - WebRtcSpl_AffineTransformVector(pw16_cngVec, pw16_cngVec, - BGNState->w16_mutefactor, 8192, 14, w16_lag); - } - } - } - else - { - /* BGN parameters have not been initialized; use zero noise */ - WebRtcSpl_MemSetW16(pw16_cngVec, 0, w16_lag); - } - - if (BGNonly) - { - /* Copy BGN to outdata */ - for (i = 0; i < w16_lag; i++) - { - pw16_outData[i] = pw16_cngVec[i]; - } - } - else - { - /* Add CNG vector to the Voiced + Unvoiced vectors */ - for (i = 0; i < w16_lag; i++) - { - pw16_outData[i] = pw16_outData[i] + pw16_cngVec[i]; - } - - /* increase call number */ - ExpandState->w16_consecExp = ExpandState->w16_consecExp + 1; - if (ExpandState->w16_consecExp < 0) /* Guard against overflow */ - ExpandState->w16_consecExp = FADE_BGN_TIME; /* "Arbitrary" large num of expands */ - } - - inst->w16_mode = MODE_EXPAND; - *pw16_len = w16_lag; - - /* Update in-call and post-call statistics */ - if (ExpandState->w16_stopMuting != 1 || BGNonly) - { - /* - * Only do this if StopMuting != 1 or if explicitly BGNonly, otherwise Expand is - * called from Merge or Normal and special measures must be taken. - */ - inst->statInst.expandLength += (uint32_t) *pw16_len; - if (ExpandState->w16_expandMuteFactor == 0 || BGNonly) - { - /* Only noise expansion */ - inst->statInst.expandedNoiseSamples += *pw16_len; - /* Short-term activity statistics. */ - inst->activity_stats.expand_bgn_samples += *pw16_len; - } - else - { - /* Voice expand (note: not necessarily _voiced_) */ - inst->statInst.expandedVoiceSamples += *pw16_len; - /* Short-term activity statistics. */ - inst->activity_stats.expand_normal_samples += *pw16_len; - } - } - - return 0; -} - -/**************************************************************************** - * WebRtcNetEQ_GenerateBGN(...) - * - * This function generates and writes len samples of background noise to the - * output vector. The Expand function will be called repeatedly until the - * correct number of samples is produced. - * - * Input: - * - inst : NetEq instance, i.e. the user that requests more - * speech/audio data - * - scratchPtr : Pointer to scratch vector - * - len : Desired length of produced BGN. - * - * - * Output: - * - pw16_outData : Pointer to a memory space where the output data - * should be stored - * - * Return value : >=0 - Number of noise samples produced and written - * to output - * -1 - Error - */ - -int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_outData, int16_t len) -{ - - int16_t pos = 0; - int16_t tempLen = len; - - while (tempLen > 0) - { - /* while we still need more noise samples, call Expand to obtain background noise */ - WebRtcNetEQ_Expand(inst, -#ifdef SCRATCH - pw16_scratchPtr, -#endif - &pw16_outData[pos], &tempLen, 1 /*BGNonly*/); - - pos += tempLen; /* we got this many samples */ - tempLen = len - pos; /* this is the number of samples we still need */ - } - - return pos; -} - -#undef SCRATCH_PW16_BEST_CORR_INDEX -#undef SCRATCH_PW16_BEST_CORR -#undef SCRATCH_PW16_BEST_DIST_INDEX -#undef SCRATCH_PW16_BEST_DIST -#undef SCRATCH_PW16_CORR_VEC -#undef SCRATCH_PW16_CORR2 -#undef SCRATCH_PW32_AUTO_CORR -#undef SCRATCH_PW16_RC -#undef SCRATCH_PW16_RAND_VEC -#undef SCRATCH_NETEQDSP_CORRELATOR -#undef SCRATCH_PW16_SCALED_RAND_VEC -#undef SCRATCH_PW16_UNVOICED_VEC_SPACE - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/expand.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.cc index 73f2ef85a56..14a77982234 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/expand.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/expand.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" #include <assert.h> #include <string.h> // memset @@ -17,10 +17,10 @@ #include <limits> // numeric_limits<T> #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h" -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" +#include "webrtc/modules/audio_coding/neteq/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { @@ -56,20 +56,9 @@ int Expand::Process(AudioMultiVector* output) { // This is not the first expansion, parameters are already estimated. // Extract a noise segment. int16_t rand_length = max_lag_; - // TODO(hlundin): This if-statement should not be needed. Should be just - // as good to generate all of the vector in one call in either case. - if (rand_length <= RandomVector::kRandomTableSize) { - random_vector_->IncreaseSeedIncrement(2); - random_vector_->Generate(rand_length, random_vector); - } else { - // This only applies to SWB where length could be larger than 256. - assert(rand_length <= kMaxSampleRate / 8000 * 120 + 30); - random_vector_->IncreaseSeedIncrement(2); - random_vector_->Generate(RandomVector::kRandomTableSize, random_vector); - random_vector_->IncreaseSeedIncrement(2); - random_vector_->Generate(rand_length - RandomVector::kRandomTableSize, - &random_vector[RandomVector::kRandomTableSize]); - } + // This only applies to SWB where length could be larger than 256. + assert(rand_length <= kMaxSampleRate / 8000 * 120 + 30); + GenerateRandomVector(2, rand_length, random_vector); } @@ -262,82 +251,12 @@ int Expand::Process(AudioMultiVector* output) { } // Background noise part. - // TODO(hlundin): Move to separate method? In BackgroundNoise class? - if (background_noise_->initialized()) { - // Use background noise parameters. - memcpy(noise_vector - kNoiseLpcOrder, - background_noise_->FilterState(channel_ix), - sizeof(int16_t) * kNoiseLpcOrder); - - if (background_noise_->ScaleShift(channel_ix) > 1) { - add_constant = 1 << (background_noise_->ScaleShift(channel_ix) - 1); - } else { - add_constant = 0; - } - - // Scale random vector to correct energy level. - WebRtcSpl_AffineTransformVector( - scaled_random_vector, random_vector, - background_noise_->Scale(channel_ix), add_constant, - background_noise_->ScaleShift(channel_ix), - static_cast<int>(current_lag)); - - WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_vector, - background_noise_->Filter(channel_ix), - kNoiseLpcOrder + 1, - static_cast<int>(current_lag)); - - background_noise_->SetFilterState( - channel_ix, - &(noise_vector[current_lag - kNoiseLpcOrder]), - kNoiseLpcOrder); - - // Unmute the background noise. - int16_t bgn_mute_factor = background_noise_->MuteFactor(channel_ix); - NetEqBackgroundNoiseMode bgn_mode = background_noise_->mode(); - if (bgn_mode == kBgnFade && - consecutive_expands_ >= kMaxConsecutiveExpands && - bgn_mute_factor > 0) { - // Fade BGN to zero. - // Calculate muting slope, approximately -2^18 / fs_hz. - int16_t mute_slope; - if (fs_hz_ == 8000) { - mute_slope = -32; - } else if (fs_hz_ == 16000) { - mute_slope = -16; - } else if (fs_hz_ == 32000) { - mute_slope = -8; - } else { - mute_slope = -5; - } - // Use UnmuteSignal function with negative slope. - // |bgn_mute_factor| is in Q14. |mute_slope| is in Q20. - DspHelper::UnmuteSignal(noise_vector, current_lag, &bgn_mute_factor, - mute_slope, noise_vector); - } else if (bgn_mute_factor < 16384) { - // If mode is kBgnOff, or if kBgnFade has started fading, - // Use regular |mute_slope|. - if (!stop_muting_ && bgn_mode != kBgnOff && - !(bgn_mode == kBgnFade && - consecutive_expands_ >= kMaxConsecutiveExpands)) { - DspHelper::UnmuteSignal(noise_vector, static_cast<int>(current_lag), - &bgn_mute_factor, parameters.mute_slope, - noise_vector); - } else { - // kBgnOn and stop muting, or - // kBgnOff (mute factor is always 0), or - // kBgnFade has reached 0. - WebRtcSpl_AffineTransformVector(noise_vector, noise_vector, - bgn_mute_factor, 8192, 14, - static_cast<int>(current_lag)); - } - } - // Update mute_factor in BackgroundNoise class. - background_noise_->SetMuteFactor(channel_ix, bgn_mute_factor); - } else { - // BGN parameters have not been initialized; use zero noise. - memset(noise_vector, 0, sizeof(int16_t) * current_lag); - } + GenerateBackgroundNoise(random_vector, + channel_ix, + channel_parameters_[channel_ix].mute_slope, + TooManyExpands(), + current_lag, + unvoiced_array_memory); // Add background noise to the combined voiced-unvoiced signal. for (size_t i = 0; i < current_lag; i++) { @@ -353,11 +272,8 @@ int Expand::Process(AudioMultiVector* output) { } // Increase call number and cap it. - ++consecutive_expands_; - if (consecutive_expands_ > kMaxConsecutiveExpands) { - consecutive_expands_ = kMaxConsecutiveExpands; - } - + consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ? + kMaxConsecutiveExpands : consecutive_expands_ + 1; return 0; } @@ -373,6 +289,24 @@ void Expand::SetParametersForMergeAfterExpand() { stop_muting_ = true; } +void Expand::InitializeForAnExpandPeriod() { + lag_index_direction_ = 1; + current_lag_index_ = -1; + stop_muting_ = false; + random_vector_->set_seed_increment(1); + consecutive_expands_ = 0; + for (size_t ix = 0; ix < num_channels_; ++ix) { + channel_parameters_[ix].current_voice_mix_factor = 16384; // 1.0 in Q14. + channel_parameters_[ix].mute_factor = 16384; // 1.0 in Q14. + // Start with 0 gain for background noise. + background_noise_->SetMuteFactor(ix, 0); + } +} + +bool Expand::TooManyExpands() { + return consecutive_expands_ >= kMaxConsecutiveExpands; +} + void Expand::AnalyzeSignal(int16_t* random_vector) { int32_t auto_correlation[kUnvoicedLpcOrder + 1]; int16_t reflection_coeff[kUnvoicedLpcOrder]; @@ -400,18 +334,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) { const int16_t* audio_history = &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length]; - // Initialize some member variables. - lag_index_direction_ = 1; - current_lag_index_ = -1; - stop_muting_ = false; - random_vector_->set_seed_increment(1); - consecutive_expands_ = 0; - for (size_t ix = 0; ix < num_channels_; ++ix) { - channel_parameters_[ix].current_voice_mix_factor = 16384; // 1.0 in Q14. - channel_parameters_[ix].mute_factor = 16384; // 1.0 in Q14. - // Start with 0 gain for background noise. - background_noise_->SetMuteFactor(ix, 0); - } + // Initialize. + InitializeForAnExpandPeriod(); // Calculate correlation in downsampled domain (4 kHz sample rate). int16_t correlation_scale; @@ -864,4 +788,117 @@ void Expand::UpdateLagIndex() { } } +Expand* ExpandFactory::Create(BackgroundNoise* background_noise, + SyncBuffer* sync_buffer, + RandomVector* random_vector, + int fs, + size_t num_channels) const { + return new Expand(background_noise, sync_buffer, random_vector, fs, + num_channels); +} + +// TODO(turajs): This can be moved to BackgroundNoise class. +void Expand::GenerateBackgroundNoise(int16_t* random_vector, + size_t channel, + int16_t mute_slope, + bool too_many_expands, + size_t num_noise_samples, + int16_t* buffer) { + static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder; + int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125]; + assert(kMaxSampleRate / 8000 * 125 >= (int)num_noise_samples); + int16_t* noise_samples = &buffer[kNoiseLpcOrder]; + if (background_noise_->initialized()) { + // Use background noise parameters. + memcpy(noise_samples - kNoiseLpcOrder, + background_noise_->FilterState(channel), + sizeof(int16_t) * kNoiseLpcOrder); + + int dc_offset = 0; + if (background_noise_->ScaleShift(channel) > 1) { + dc_offset = 1 << (background_noise_->ScaleShift(channel) - 1); + } + + // Scale random vector to correct energy level. + WebRtcSpl_AffineTransformVector( + scaled_random_vector, random_vector, + background_noise_->Scale(channel), dc_offset, + background_noise_->ScaleShift(channel), + static_cast<int>(num_noise_samples)); + + WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples, + background_noise_->Filter(channel), + kNoiseLpcOrder + 1, + static_cast<int>(num_noise_samples)); + + background_noise_->SetFilterState( + channel, + &(noise_samples[num_noise_samples - kNoiseLpcOrder]), + kNoiseLpcOrder); + + // Unmute the background noise. + int16_t bgn_mute_factor = background_noise_->MuteFactor(channel); + NetEqBackgroundNoiseMode bgn_mode = background_noise_->mode(); + if (bgn_mode == kBgnFade && too_many_expands && bgn_mute_factor > 0) { + // Fade BGN to zero. + // Calculate muting slope, approximately -2^18 / fs_hz. + int16_t mute_slope; + if (fs_hz_ == 8000) { + mute_slope = -32; + } else if (fs_hz_ == 16000) { + mute_slope = -16; + } else if (fs_hz_ == 32000) { + mute_slope = -8; + } else { + mute_slope = -5; + } + // Use UnmuteSignal function with negative slope. + // |bgn_mute_factor| is in Q14. |mute_slope| is in Q20. + DspHelper::UnmuteSignal(noise_samples, + num_noise_samples, + &bgn_mute_factor, + mute_slope, + noise_samples); + } else if (bgn_mute_factor < 16384) { + // If mode is kBgnOff, or if kBgnFade has started fading, + // Use regular |mute_slope|. + if (!stop_muting_ && bgn_mode != kBgnOff && + !(bgn_mode == kBgnFade && too_many_expands)) { + DspHelper::UnmuteSignal(noise_samples, + static_cast<int>(num_noise_samples), + &bgn_mute_factor, + mute_slope, + noise_samples); + } else { + // kBgnOn and stop muting, or + // kBgnOff (mute factor is always 0), or + // kBgnFade has reached 0. + WebRtcSpl_AffineTransformVector(noise_samples, noise_samples, + bgn_mute_factor, 8192, 14, + static_cast<int>(num_noise_samples)); + } + } + // Update mute_factor in BackgroundNoise class. + background_noise_->SetMuteFactor(channel, bgn_mute_factor); + } else { + // BGN parameters have not been initialized; use zero noise. + memset(noise_samples, 0, sizeof(int16_t) * num_noise_samples); + } +} + +void Expand::GenerateRandomVector(int seed_increment, + size_t length, + int16_t* random_vector) { + // TODO(turajs): According to hlundin The loop should not be needed. Should be + // just as good to generate all of the vector in one call. + size_t samples_generated = 0; + const size_t kMaxRandSamples = RandomVector::kRandomTableSize; + while(samples_generated < length) { + size_t rand_length = std::min(length - samples_generated, kMaxRandSamples); + random_vector_->IncreaseSeedIncrement(seed_increment); + random_vector_->Generate(rand_length, &random_vector[samples_generated]); + samples_generated += rand_length; + } +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/expand.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.h index 25ae61903a6..1acf951b980 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/expand.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_EXPAND_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_EXPAND_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_ #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" @@ -36,12 +36,13 @@ class Expand { RandomVector* random_vector, int fs, size_t num_channels) - : background_noise_(background_noise), + : random_vector_(random_vector), sync_buffer_(sync_buffer), - random_vector_(random_vector), first_expand_(true), fs_hz_(fs), num_channels_(num_channels), + consecutive_expands_(0), + background_noise_(background_noise), overlap_length_(5 * fs / 8000), lag_index_direction_(0), current_lag_index_(0), @@ -57,19 +58,19 @@ class Expand { virtual ~Expand() {} // Resets the object. - void Reset(); + virtual void Reset(); // The main method to produce concealment data. The data is appended to the // end of |output|. - int Process(AudioMultiVector* output); + virtual int Process(AudioMultiVector* output); // Prepare the object to do extra expansion during normal operation following // a period of expands. - void SetParametersForNormalAfterExpand(); + virtual void SetParametersForNormalAfterExpand(); // Prepare the object to do extra expansion during merge operation following // a period of expands. - void SetParametersForMergeAfterExpand(); + virtual void SetParametersForMergeAfterExpand(); // Sets the mute factor for |channel| to |value|. void SetMuteFactor(int16_t value, size_t channel) { @@ -84,9 +85,38 @@ class Expand { } // Accessors and mutators. - size_t overlap_length() const { return overlap_length_; } + virtual size_t overlap_length() const { return overlap_length_; } int16_t max_lag() const { return max_lag_; } + protected: + static const int kMaxConsecutiveExpands = 200; + void GenerateRandomVector(int seed_increment, + size_t length, + int16_t* random_vector); + + void GenerateBackgroundNoise(int16_t* random_vector, + size_t channel, + int16_t mute_slope, + bool too_many_expands, + size_t num_noise_samples, + int16_t* buffer); + + // Initializes member variables at the beginning of an expand period. + void InitializeForAnExpandPeriod(); + + bool TooManyExpands(); + + // Analyzes the signal history in |sync_buffer_|, and set up all parameters + // necessary to produce concealment data. + void AnalyzeSignal(int16_t* random_vector); + + RandomVector* random_vector_; + SyncBuffer* sync_buffer_; + bool first_expand_; + const int fs_hz_; + const size_t num_channels_; + int consecutive_expands_; + private: static const int kUnvoicedLpcOrder = 6; static const int kNumCorrelationCandidates = 3; @@ -94,7 +124,6 @@ class Expand { static const int kLpcAnalysisLength = 160; static const int kMaxSampleRate = 48000; static const int kNumLags = 3; - static const int kMaxConsecutiveExpands = 200; struct ChannelParameters { // Constructor. @@ -122,10 +151,6 @@ class Expand { int16_t mute_slope; /* Q20 */ }; - // Analyze the signal history in |sync_buffer_|, and set up all parameters - // necessary to produce concealment data. - void AnalyzeSignal(int16_t* random_vector); - // Calculate the auto-correlation of |input|, with length |input_length| // samples. The correlation is calculated from a downsampled version of // |input|, and is written to |output|. The scale factor is written to @@ -136,22 +161,27 @@ class Expand { void UpdateLagIndex(); BackgroundNoise* background_noise_; - SyncBuffer* sync_buffer_; - RandomVector* random_vector_; - bool first_expand_; - int fs_hz_; - size_t num_channels_; - size_t overlap_length_; - int consecutive_expands_; + const size_t overlap_length_; int16_t max_lag_; size_t expand_lags_[kNumLags]; int lag_index_direction_; int current_lag_index_; bool stop_muting_; - scoped_array<ChannelParameters> channel_parameters_; + scoped_ptr<ChannelParameters[]> channel_parameters_; DISALLOW_COPY_AND_ASSIGN(Expand); }; +struct ExpandFactory { + ExpandFactory() {} + virtual ~ExpandFactory() {} + + virtual Expand* Create(BackgroundNoise* background_noise, + SyncBuffer* sync_buffer, + RandomVector* random_vector, + int fs, + size_t num_channels) const; +}; + } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_EXPAND_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/expand_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand_unittest.cc index a63ed142f03..bd39f408ffa 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/expand_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand_unittest.cc @@ -10,12 +10,12 @@ // Unit tests for Expand class. -#include "webrtc/modules/audio_coding/neteq4/expand.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { @@ -28,6 +28,19 @@ TEST(Expand, CreateAndDestroy) { Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels); } +TEST(Expand, CreateUsingFactory) { + int fs = 8000; + size_t channels = 1; + BackgroundNoise bgn(channels); + SyncBuffer sync_buffer(1, 1000); + RandomVector random_vector; + ExpandFactory expand_factory; + Expand* expand = + expand_factory.Create(&bgn, &sync_buffer, &random_vector, fs, channels); + EXPECT_TRUE(expand != NULL); + delete expand; +} + // TODO(hlundin): Write more tests. } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h index f3bcc711f36..9a2fb8b4645 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_AUDIO_DECODER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_AUDIO_DECODER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_ #include <stdlib.h> // NULL -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -108,6 +108,17 @@ class AudioDecoder { // is available, or -1 in case of an error. virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len); + // Returns the duration in samples of the redandant payload in |encoded| which + // is |encoded_len| bytes long. Returns kNotImplemented if no duration + // estimate is available, or -1 in case of an error. + virtual int PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const; + + // Detects whether a packet has forward error correction. The packet is + // comprised of the samples in |encoded| which is |encoded_len| bytes long. + // Returns true if the packet has FEC and false otherwise. + virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; + virtual NetEqDecoder codec_type() const; // Returns the underlying decoder state. @@ -138,4 +149,4 @@ class AudioDecoder { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_AUDIO_DECODER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/interface/neteq.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/neteq.h index 617393093f3..c67ab12c6ce 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/interface/neteq.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/neteq.h @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_NETEQ_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_NETEQ_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_ #include <string.h> // Provide access to size_t. #include <vector> +#include "webrtc/base/constructormagic.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -67,6 +67,20 @@ enum NetEqBackgroundNoiseMode { // This is the interface class for NetEq. class NetEq { public: + struct Config { + Config() + : sample_rate_hz(16000), + enable_audio_classifier(false), + max_packets_in_buffer(50), + // |max_delay_ms| has the same effect as calling SetMaximumDelay(). + max_delay_ms(2000) {} + + int sample_rate_hz; // Initial vale. Will change with input data. + bool enable_audio_classifier; + int max_packets_in_buffer; + int max_delay_ms; + }; + enum ReturnCodes { kOK = 0, kFail = -1, @@ -98,17 +112,13 @@ class NetEq { kFrameSplitError, kRedundancySplitError, kPacketBufferCorruption, - kOversizePacket, kSyncPacketNotAccepted }; - static const int kMaxNumPacketsInBuffer = 240; // TODO(hlundin): Remove. - static const int kMaxBytesInBuffer = 113280; // TODO(hlundin): Remove. - - // Creates a new NetEq object, starting at the sample rate |sample_rate_hz|. - // (Note that it will still change the sample rate depending on what payloads - // are being inserted; |sample_rate_hz| is just for startup configuration.) - static NetEq* Create(int sample_rate_hz); + // Creates a new NetEq object, with parameters set in |config|. The |config| + // object will only have to be valid for the duration of the call to this + // method. + static NetEq* Create(const NetEq::Config& config); virtual ~NetEq() {} @@ -152,11 +162,10 @@ class NetEq { // Provides an externally created decoder object |decoder| to insert in the // decoder database. The decoder implements a decoder of type |codec| and - // associates it with |rtp_payload_type|. The decoder operates at the - // frequency |sample_rate_hz|. Returns kOK on success, kFail on failure. + // associates it with |rtp_payload_type|. Returns kOK on success, + // kFail on failure. virtual int RegisterExternalDecoder(AudioDecoder* decoder, enum NetEqDecoder codec, - int sample_rate_hz, uint8_t rtp_payload_type) = 0; // Removes |rtp_payload_type| from the codec database. Returns 0 on success, @@ -171,7 +180,8 @@ class NetEq { // Sets a maximum delay in milliseconds for packet buffer. The latency will // not exceed the given value, even required delay (given the channel - // conditions) is higher. + // conditions) is higher. Calling this method has the same effect as setting + // the |max_delay_ms| value in the NetEq::Config struct. virtual bool SetMaximumDelay(int delay_ms) = 0; // The smallest latency required. This is computed bases on inter-arrival @@ -218,8 +228,9 @@ class NetEq { // Disables post-decode VAD. virtual void DisableVad() = 0; - // Returns the RTP timestamp for the last sample delivered by GetAudio(). - virtual uint32_t PlayoutTimestamp() = 0; + // Gets the RTP timestamp for the last sample delivered by GetAudio(). + // Returns true if the RTP timestamp is valid, otherwise false. + virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0; // Not implemented. virtual int SetTargetNumberOfChannels() = 0; @@ -241,9 +252,7 @@ class NetEq { // Current usage of packet-buffer and it's limits. virtual void PacketBufferStatistics(int* current_num_packets, - int* max_num_packets, - int* current_memory_size_bytes, - int* max_memory_size_bytes) const = 0; + int* max_num_packets) const = 0; // Get sequence number and timestamp of the latest RTP. // This method is to facilitate NACK. @@ -264,4 +273,4 @@ class NetEq { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_NETEQ_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h deleted file mode 100644 index c2a01340b93..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h +++ /dev/null @@ -1,230 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This is the main API for NetEQ. Helper macros are located in webrtc_neteq_help_macros.h, - * while some internal API functions are found in webrtc_neteq_internal.h. - */ - -#include "typedefs.h" - -#ifndef WEBRTC_NETEQ_H -#define WEBRTC_NETEQ_H - -#ifdef __cplusplus -extern "C" -{ -#endif - -/********************************************************** - * Definitions - */ - -enum WebRtcNetEQDecoder -{ - kDecoderReservedStart, - kDecoderPCMu, - kDecoderPCMa, - kDecoderPCMu_2ch, - kDecoderPCMa_2ch, - kDecoderILBC, - kDecoderISAC, - kDecoderISACswb, - kDecoderISACfb, - kDecoderPCM16B, - kDecoderPCM16Bwb, - kDecoderPCM16Bswb32kHz, - kDecoderPCM16Bswb48kHz, - kDecoderPCM16B_2ch, - kDecoderPCM16Bwb_2ch, - kDecoderPCM16Bswb32kHz_2ch, - kDecoderG722, - kDecoderG722_2ch, - kDecoderRED, - kDecoderAVT, - kDecoderCNG, - kDecoderArbitrary, - kDecoderG729, - kDecoderG729_1, - kDecoderG726_16, - kDecoderG726_24, - kDecoderG726_32, - kDecoderG726_40, - kDecoderG722_1_16, - kDecoderG722_1_24, - kDecoderG722_1_32, - kDecoderG722_1C_24, - kDecoderG722_1C_32, - kDecoderG722_1C_48, - kDecoderOpus, - kDecoderSPEEX_8, - kDecoderSPEEX_16, - kDecoderCELT_32, - kDecoderCELT_32_2ch, - kDecoderGSMFR, - kDecoderAMR, - kDecoderAMRWB, - kDecoderReservedEnd -}; - -enum WebRtcNetEQNetworkType -{ - kUDPNormal, - kUDPVideoSync, - kTCPNormal, - kTCPLargeJitter, - kTCPXLargeJitter -}; - -enum WebRtcNetEQOutputType -{ - kOutputNormal, - kOutputPLC, - kOutputCNG, - kOutputPLCtoCNG, - kOutputVADPassive -}; - -enum WebRtcNetEQPlayoutMode -{ - kPlayoutOn, kPlayoutOff, kPlayoutFax, kPlayoutStreaming -}; - -/* Available modes for background noise (inserted after long expands) */ -enum WebRtcNetEQBGNMode -{ - kBGNOn, /* default "normal" behavior with eternal noise */ - kBGNFade, /* noise fades to zero after some time */ - kBGNOff -/* background noise is always zero */ -}; - -/************************************************* - * Definitions of decoder calls and the default - * API function calls for each codec - */ - -typedef int16_t (*WebRtcNetEQ_FuncDecode)(void* state, int16_t* encoded, - int16_t len, int16_t* decoded, - int16_t* speechType); -typedef int16_t (*WebRtcNetEQ_FuncDecodePLC)(void* state, int16_t* decoded, - int16_t frames); -typedef int16_t (*WebRtcNetEQ_FuncDecodeInit)(void* state); -typedef int16_t (*WebRtcNetEQ_FuncAddLatePkt)(void* state, int16_t* encoded, - int16_t len); -typedef int16_t (*WebRtcNetEQ_FuncGetMDinfo)(void* state); -typedef int16_t (*WebRtcNetEQ_FuncGetPitchInfo)(void* state, int16_t* encoded, - int16_t* length); -typedef int16_t (*WebRtcNetEQ_FuncUpdBWEst)(void* state, const uint16_t *encoded, - int32_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts); -typedef int (*WebRtcNetEQ_FuncDurationEst)(void* state, const uint8_t* payload, - int payload_length_bytes); -typedef int16_t (*WebRtcNetEQ_FuncGetErrorCode)(void* state); - -/********************************************************** - * Structures - */ - -typedef struct -{ - enum WebRtcNetEQDecoder codec; - int16_t payloadType; - WebRtcNetEQ_FuncDecode funcDecode; - WebRtcNetEQ_FuncDecode funcDecodeRCU; - WebRtcNetEQ_FuncDecodePLC funcDecodePLC; - WebRtcNetEQ_FuncDecodeInit funcDecodeInit; - WebRtcNetEQ_FuncAddLatePkt funcAddLatePkt; - WebRtcNetEQ_FuncGetMDinfo funcGetMDinfo; - WebRtcNetEQ_FuncGetPitchInfo funcGetPitch; - WebRtcNetEQ_FuncUpdBWEst funcUpdBWEst; - WebRtcNetEQ_FuncDurationEst funcDurationEst; - WebRtcNetEQ_FuncGetErrorCode funcGetErrorCode; - void* codec_state; - uint16_t codec_fs; -} WebRtcNetEQ_CodecDef; - -typedef struct -{ - uint16_t fraction_lost; - uint32_t cum_lost; - uint32_t ext_max; - uint32_t jitter; -} WebRtcNetEQ_RTCPStat; - -/********************************************************** - * NETEQ Functions - */ - -/* Info functions */ - -#define WEBRTC_NETEQ_MAX_ERROR_NAME 40 -int WebRtcNetEQ_GetErrorCode(void *inst); -int WebRtcNetEQ_GetErrorName(int errorCode, char *errorName, int maxStrLen); - -/* Instance memory assign functions */ - -int WebRtcNetEQ_AssignSize(int *sizeinbytes); -int WebRtcNetEQ_Assign(void **inst, void *NETEQ_inst_Addr); -int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecoder *codec, - int noOfCodecs, enum WebRtcNetEQNetworkType nwType, - int *MaxNoOfPackets, int *sizeinbytes, - int* per_packet_overhead_bytes); -int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr, - int sizeinbytes); - -/* Init functions */ - -int WebRtcNetEQ_Init(void *inst, uint16_t fs); -int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon); -int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs); -int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode); -int WebRtcNetEQ_SetBGNMode(void *inst, enum WebRtcNetEQBGNMode bgnMode); -int WebRtcNetEQ_GetBGNMode(const void *inst, enum WebRtcNetEQBGNMode *bgnMode); - -/* Codec Database functions */ - -int WebRtcNetEQ_CodecDbReset(void *inst); -int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst); -int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec); -int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, int16_t *UsedEntries, - int16_t *MaxEntries); -int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, int16_t Entry, - enum WebRtcNetEQDecoder *codec); - -/* Real-time functions */ - -int WebRtcNetEQ_RecIn(void *inst, int16_t *p_w16datagramstart, int16_t w16_RTPlen, - uint32_t uw32_timeRec); -int WebRtcNetEQ_RecOut(void *inst, int16_t *pw16_outData, int16_t *pw16_len); -int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst); -int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst); -int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, uint32_t *timestamp); -int WebRtcNetEQ_DecodedRtpInfo(const void* inst, - int* sequence_number, - uint32_t* timestamp); -int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType); - -/* VQmon related functions */ -int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, uint16_t *validVoiceDurationMs, - uint16_t *concealedVoiceDurationMs, - uint8_t *concealedVoiceFlags); -int WebRtcNetEQ_VQmonGetConfiguration(void *inst, uint16_t *absMaxDelayMs, - uint8_t *adaptationRate); -int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, uint16_t *avgDelayMs, - uint16_t *maxDelayMs); - -#ifdef __cplusplus -} -#endif - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h deleted file mode 100644 index bd93328108e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h +++ /dev/null @@ -1,454 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains some helper macros that can be used when loading the - * NetEQ codec database. - */ - -#ifndef WEBRTC_NETEQ_HELP_MACROS_H -#define WEBRTC_NETEQ_HELP_MACROS_H - -#ifndef NULL -#define NULL 0 -#endif - -/********************************************************** - * Help macros for NetEQ initialization - */ - -#define SET_CODEC_PAR(inst,decoder,pt,state,fs) \ - inst.codec=decoder; \ - inst.payloadType=pt; \ - inst.codec_state=state; \ - inst.codec_fs=fs; - -#define SET_PCMU_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG711_DecodeU; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=WebRtcG711_DurationEst; \ - inst.funcGetErrorCode=NULL; - -#define SET_PCMA_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG711_DecodeA; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=WebRtcG711_DurationEst; \ - inst.funcGetErrorCode=NULL; - -#define SET_ILBC_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIlbcfix_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcIlbcfix_NetEqPlc; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIlbcfix_Decoderinit30Ms; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_ISAC_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \ - inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode; - -#define SET_ISACfix_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsacfix_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsacfix_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsacfix_UpdateBwEstimate; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsacfix_GetErrorCode; - -#define SET_ISACSWB_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \ - inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode; - -#define SET_ISACFB_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \ - inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode; - -#define SET_G729_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG729_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG729_DecodePlc; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG729_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G729_1_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7291_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7291_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcG7291_DecodeBwe; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_PCM16B_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_PCM16B_WB_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_PCM16B_SWB32_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_PCM16B_SWB48_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG722_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG722_DecoderInit;\ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_1_16_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode16; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc16; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit16; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_1_24_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode24; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc24; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit24; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_1_32_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode32; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc32; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit32; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_1C_24_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode24; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc24; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit24; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_1C_32_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode32; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc32; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit32; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G722_1C_48_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode48; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc48; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit48; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_AMR_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcAmr_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcAmr_DecodePlc; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcAmr_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_AMRWB_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcAmrWb_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcAmrWb_DecodePlc; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcAmrWb_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_GSMFR_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcGSMFR_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcGSMFR_DecodePlc; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcGSMFR_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G726_16_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode16; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit16; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G726_24_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode24; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit24; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G726_32_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode32; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit32; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_G726_40_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode40; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit40; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_OPUS_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcOpus_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcOpus_DecodePlcMaster; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcOpus_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=(WebRtcNetEQ_FuncDurationEst)WebRtcOpus_DurationEst; \ - inst.funcGetErrorCode=NULL; - -#define SET_OPUSSLAVE_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcOpus_DecodeSlave; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcOpus_DecodePlcSlave; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcOpus_DecoderInitSlave; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=(WebRtcNetEQ_FuncDurationEst)WebRtcOpus_DurationEst; \ - inst.funcGetErrorCode=NULL; - -#define SET_SPEEX_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcSpeex_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcSpeex_DecodePlc; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcSpeex_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_CELT_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_Decode; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_CELTSLAVE_FUNCTIONS(inst) \ - inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_DecodeSlave; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInitSlave; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_RED_FUNCTIONS(inst) \ - inst.funcDecode=NULL; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_AVT_FUNCTIONS(inst) \ - inst.funcDecode=NULL; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#define SET_CNG_FUNCTIONS(inst) \ - inst.funcDecode=NULL; \ - inst.funcDecodeRCU=NULL; \ - inst.funcDecodePLC=NULL; \ - inst.funcDecodeInit=NULL; \ - inst.funcAddLatePkt=NULL; \ - inst.funcGetMDinfo=NULL; \ - inst.funcGetPitch=NULL; \ - inst.funcUpdBWEst=NULL; \ - inst.funcDurationEst=NULL; \ - inst.funcGetErrorCode=NULL; - -#endif /* WEBRTC_NETEQ_HELP_MACROS_H */ - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h deleted file mode 100644 index c46a3f62705..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h +++ /dev/null @@ -1,336 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the internal API functions. - */ - -#include "typedefs.h" - -#ifndef WEBRTC_NETEQ_INTERNAL_H -#define WEBRTC_NETEQ_INTERNAL_H - -#ifdef __cplusplus -extern "C" -{ -#endif - -typedef struct -{ - uint8_t payloadType; - uint16_t sequenceNumber; - uint32_t timeStamp; - uint32_t SSRC; - uint8_t markerBit; -} WebRtcNetEQ_RTPInfo; - -/**************************************************************************** - * WebRtcNetEQ_RecInRTPStruct(...) - * - * Alternative RecIn function, used when the RTP data has already been - * parsed into an RTP info struct (WebRtcNetEQ_RTPInfo). - * - * Input: - * - inst : NetEQ instance - * - rtpInfo : Pointer to RTP info - * - payloadPtr : Pointer to the RTP payload (first byte after header) - * - payloadLenBytes : Length (in bytes) of the payload in payloadPtr - * - timeRec : Receive time (in timestamps of the used codec) - * - * Return value : 0 - Ok - * -1 - Error - */ -int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo, - const uint8_t *payloadPtr, int16_t payloadLenBytes, - uint32_t timeRec); - -/**************************************************************************** - * WebRtcNetEQ_GetMasterSlaveInfoSize(...) - * - * Get size in bytes for master/slave struct msInfo used in - * WebRtcNetEQ_RecOutMasterSlave. - * - * Return value : Struct size in bytes - * - */ - -int WebRtcNetEQ_GetMasterSlaveInfoSize(); - -/**************************************************************************** - * WebRtcNetEQ_RecOutMasterSlave(...) - * - * RecOut function for running several NetEQ instances in master/slave mode. - * One master can be used to control several slaves. - * The MasterSlaveInfo struct must be allocated outside NetEQ. - * Use function WebRtcNetEQ_GetMasterSlaveInfoSize to get the size needed. - * - * Input: - * - inst : NetEQ instance - * - isMaster : Non-zero indicates that this is the master channel - * - msInfo : (slave only) Information from master - * - * Output: - * - inst : Updated NetEQ instance - * - pw16_outData : Pointer to vector where output should be written - * - pw16_len : Pointer to variable where output length is returned - * - msInfo : (master only) Information to slave(s) - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RecOutMasterSlave(void *inst, int16_t *pw16_outData, - int16_t *pw16_len, void *msInfo, - int16_t isMaster); - -typedef struct -{ - uint16_t currentBufferSize; /* Current jitter buffer size in ms. */ - uint16_t preferredBufferSize; /* Preferred buffer size in ms. */ - uint16_t jitterPeaksFound; /* 1 if adding extra delay due to peaky - * jitter; 0 otherwise. */ - uint16_t currentPacketLossRate; /* Loss rate (network + late) (Q14). */ - uint16_t currentDiscardRate; /* Late loss rate (Q14). */ - uint16_t currentExpandRate; /* Fraction (of original stream) of - * synthesized speech inserted through - * expansion (in Q14). */ - uint16_t currentPreemptiveRate; /* Fraction of data inserted through - * pre-emptive expansion (in Q14). */ - uint16_t currentAccelerateRate; /* Fraction of data removed through - * acceleration (in Q14). */ - int32_t clockDriftPPM; /* Average clock-drift in parts-per- - * million (positive or negative). */ - int addedSamples; /* Number of zero samples added in off - * mode */ -} WebRtcNetEQ_NetworkStatistics; - -/* - * Get the "in-call" statistics from NetEQ. - * The statistics are reset after the query. - */ -int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *stats); - - -typedef struct { - /* Samples removed from background noise only segments. */ - int accelerate_bgn_samples; - - /* Samples removed from normal audio segments. */ - int accelerate_normal_samples; - - /* Number of samples synthesized during background noise only segments. */ - int expand_bgn_sampels; - - /* Number of samples synthesized during normal audio segments. */ - int expand_normal_samples; - - /* Number of samples synthesized during background noise only segments, - * in preemptive mode. */ - int preemptive_expand_bgn_samples; - - /* Number of samples synthesized during normal audio segments, in preemptive - * mode. */ - int preemptive_expand_normal_samples; - - /* Number of samples synthesized during background noise only segments, - * while merging. */ - int merge_expand_bgn_samples; - - /* Number of samples synthesized during normal audio segments, while - * merging. */ - int merge_expand_normal_samples; -} WebRtcNetEQ_ProcessingActivity; - -/* - * Get the processing activities from NetEQ. - * The statistics are reset after the query. - * This API is meant to obtain processing activities in high granularity, - * e.g. per RecOut() call. - */ -void WebRtcNetEQ_GetProcessingActivity(void* inst, - WebRtcNetEQ_ProcessingActivity* stat); - -/* - * Get the raw waiting times for decoded frames. The function writes the last - * recorded waiting times (from frame arrival to frame decoding) to the memory - * pointed to by waitingTimeMs. The number of elements written is in the return - * value. No more than maxLength elements are written. Statistics are reset on - * each query. - */ -int WebRtcNetEQ_GetRawFrameWaitingTimes(void *inst, - int max_length, - int* waiting_times_ms); - -/***********************************************/ -/* Functions for post-decode VAD functionality */ -/***********************************************/ - -/* NetEQ must be compiled with the flag NETEQ_VAD enabled for these functions to work. */ - -/* - * VAD function pointer types - * - * These function pointers match the definitions of webrtc VAD functions WebRtcVad_Init, - * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in webrtc_vad.h. - */ -typedef int (*WebRtcNetEQ_VADInitFunction)(void *VAD_inst); -typedef int (*WebRtcNetEQ_VADSetmodeFunction)(void *VAD_inst, int mode); -typedef int (*WebRtcNetEQ_VADFunction)(void *VAD_inst, int fs, - int16_t *frame, int frameLen); - -/**************************************************************************** - * WebRtcNetEQ_SetVADInstance(...) - * - * Provide a pointer to an allocated VAD instance. If function is never - * called or it is called with NULL pointer as VAD_inst, the post-decode - * VAD functionality is disabled. Also provide pointers to init, setmode - * and VAD functions. These are typically pointers to WebRtcVad_Init, - * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in the - * interface file webrtc_vad.h. - * - * Input: - * - NetEQ_inst : NetEQ instance - * - VADinst : VAD instance - * - initFunction : Pointer to VAD init function - * - setmodeFunction : Pointer to VAD setmode function - * - VADfunction : Pointer to VAD function - * - * Output: - * - NetEQ_inst : Updated NetEQ instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_SetVADInstance(void *NetEQ_inst, void *VAD_inst, - WebRtcNetEQ_VADInitFunction initFunction, - WebRtcNetEQ_VADSetmodeFunction setmodeFunction, - WebRtcNetEQ_VADFunction VADFunction); - -/**************************************************************************** - * WebRtcNetEQ_SetVADMode(...) - * - * Pass an aggressiveness mode parameter to the post-decode VAD instance. - * If this function is never called, mode 0 (quality mode) is used as default. - * - * Input: - * - inst : NetEQ instance - * - mode : mode parameter (same range as WebRtc VAD mode) - * - * Output: - * - inst : Updated NetEQ instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_SetVADMode(void *NetEQ_inst, int mode); - -/**************************************************************************** - * WebRtcNetEQ_RecOutNoDecode(...) - * - * Special RecOut that does not do any decoding. - * - * Input: - * - inst : NetEQ instance - * - * Output: - * - inst : Updated NetEQ instance - * - pw16_outData : Pointer to vector where output should be written - * - pw16_len : Pointer to variable where output length is returned - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RecOutNoDecode(void *inst, int16_t *pw16_outData, - int16_t *pw16_len); - -/**************************************************************************** - * WebRtcNetEQ_FlushBuffers(...) - * - * Flush packet and speech buffers. Does not reset codec database or - * jitter statistics. - * - * Input: - * - inst : NetEQ instance - * - * Output: - * - inst : Updated NetEQ instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_FlushBuffers(void *inst); - -/***************************************************************************** - * void WebRtcNetEq_EnableAVSync(...) - * - * Enable AV-sync. If Enabled, NetEq will screen for sync payloads. For - * each sync payload a silence frame is generated. - * - * Input: - * - inst : NetEQ instance - * - enable : non-zero to enable, otherwise disabled. - * - * Output: - * - inst : Updated NetEQ instance - * - */ - -void WebRtcNetEQ_EnableAVSync(void* inst, int enable); - -/**************************************************************************** - * WebRtcNetEQ_RecInSyncRTP(...) - * - * Insert a sync packet with the given RTP specification. - * - * Input: - * - inst : NetEQ instance - * - rtpInfo : Pointer to RTP info - * - receive_timestamp : Receive time (in timestamps of the used codec) - * - * Output: - * - inst : Updated NetEQ instance - * - * Return value : if succeeded it returns the number of bytes pushed - * in, otherwise returns -1. - */ - -int WebRtcNetEQ_RecInSyncRTP(void* inst, - WebRtcNetEQ_RTPInfo* rtp_info, - uint32_t receive_timestamp); - -/* - * Set a minimum latency for the jitter buffer. The overall delay is the max of - * |minimum_delay_ms| and the latency that is internally computed based on the - * inter-arrival times. - */ -int WebRtcNetEQ_SetMinimumDelay(void *inst, int minimum_delay_ms); - -/* - * Set a maximum latency for the jitter buffer. The overall delay is the min of - * |maximum_delay_ms| and the latency that is internally computed based on the - * inter-arrival times. - */ -int WebRtcNetEQ_SetMaximumDelay(void *inst, int maximum_delay_ms); - -/* - * Get the least required delay in milliseconds given inter-arrival times - * and playout mode. - */ -int WebRtcNetEQ_GetRequiredDelayMs(const void* inst); - -#ifdef __cplusplus -} -#endif - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu.h deleted file mode 100644 index 931e6dcf561..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu.h +++ /dev/null @@ -1,300 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * MCU struct and functions related to the MCU side operations. - */ - -#ifndef MCU_H -#define MCU_H - -#include "typedefs.h" - -#include "codec_db.h" -#include "rtcp.h" -#include "packet_buffer.h" -#include "buffer_stats.h" -#include "neteq_statistics.h" - -#ifdef NETEQ_ATEVENT_DECODE -#include "dtmf_buffer.h" -#endif - -#define MAX_ONE_DESC 5 /* cannot do more than this many consecutive one-descriptor decodings */ -#define MAX_LOSS_REPORT_PERIOD 60 /* number of seconds between auto-reset */ - -enum TsScaling -{ - kTSnoScaling = 0, - kTSscalingTwo, - kTSscalingTwoThirds, - kTSscalingFourThirds -}; - -enum { kLenWaitingTimes = 100 }; - -typedef struct -{ - - int16_t current_Codec; - int16_t current_Payload; - uint32_t timeStamp; /* Next timestamp that should be played */ - int16_t millisecondsPerCall; - uint16_t timestampsPerCall; /* Output chunk size */ - uint16_t fs; - uint32_t ssrc; /* Current ssrc */ - int16_t new_codec; - int16_t first_packet; - - /* MCU/DSP Communication layer */ - int16_t *pw16_readAddress; - int16_t *pw16_writeAddress; - void *main_inst; - - CodecDbInst_t codec_DB_inst; /* Information about all the codecs, i.e. which - functions to use and which codpoints that - have been assigned */ - SplitInfo_t PayloadSplit_inst; /* Information about how the current codec - payload should be splitted */ - WebRtcNetEQ_RTCP_t RTCP_inst; /* RTCP statistics */ - PacketBuf_t PacketBuffer_inst; /* The packet buffer */ - BufstatsInst_t BufferStat_inst; /* Statistics that are used to make decision - for what the DSP should perform */ -#ifdef NETEQ_ATEVENT_DECODE - dtmf_inst_t DTMF_inst; -#endif - int NoOfExpandCalls; - int16_t AVT_PlayoutOn; - enum WebRtcNetEQPlayoutMode NetEqPlayoutMode; - - int16_t one_desc; /* Number of times running on one desc */ - - uint32_t lostTS; /* Number of timestamps lost */ - uint32_t lastReportTS; /* Timestamp elapsed since last report was given */ - - int waiting_times[kLenWaitingTimes]; /* Waiting time statistics storage. */ - int len_waiting_times; - int next_waiting_time_index; - - uint32_t externalTS; - uint32_t internalTS; - int16_t TSscalingInitialized; - enum TsScaling scalingFactor; - - /* AV-sync enabled. In AV-sync NetEq screens packets for specific sync - * packets. Sync packets are not decoded by a decoder but generate all-zero - * signal with the same number of samples as previously decoded payload. - * Also in AV-sync mode the sample-size of a sync payload is reported as - * previous frame-size. */ - int av_sync; - -#ifdef NETEQ_STEREO - int usingStereo; -#endif - - /* The sequence number of the latest decoded RTP payload. */ - int decoded_packet_sequence_number; - uint32_t decoded_packet_timestamp; -} MCUInst_t; - -/**************************************************************************** - * WebRtcNetEQ_McuReset(...) - * - * Reset the MCU instance. - * - * Input: - * - inst : MCU instance - * - * Return value : 0 - Ok - * <0 - Error - */ -int WebRtcNetEQ_McuReset(MCUInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_ResetMcuInCallStats(...) - * - * Reset MCU-side statistics variables for the in-call statistics. - * - * Input: - * - inst : MCU instance - * - * Return value : 0 - Ok - * <0 - Error - */ -int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_ResetWaitingTimeStats(...) - * - * Reset waiting-time statistics. - * - * Input: - * - inst : MCU instance. - * - * Return value : n/a - */ -void WebRtcNetEQ_ResetWaitingTimeStats(MCUInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_LogWaitingTime(...) - * - * Log waiting-time to the statistics. - * - * Input: - * - inst : MCU instance. - * - waiting_time : Waiting time in "RecOut calls" (i.e., 1 call = 10 ms). - * - * Return value : n/a - */ -void WebRtcNetEQ_StoreWaitingTime(MCUInst_t *inst, int waiting_time); - -/**************************************************************************** - * WebRtcNetEQ_ResetMcuJitterStat(...) - * - * Reset MCU-side statistics variables for the post-call statistics. - * - * Input: - * - inst : MCU instance - * - * Return value : 0 - Ok - * <0 - Error - */ -int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_McuAddressInit(...) - * - * Initializes MCU with read address and write address. - * - * Input: - * - inst : MCU instance - * - Data2McuAddress : Pointer to MCU address - * - Data2DspAddress : Pointer to DSP address - * - main_inst : Pointer to NetEQ main instance - * - * Return value : 0 - Ok - * <0 - Error - */ -int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress, - void * Data2DspAddress, void *main_inst); - -/**************************************************************************** - * WebRtcNetEQ_McuSetFs(...) - * - * Initializes MCU with read address and write address. - * - * Input: - * - inst : MCU instance - * - fs_hz : Sample rate in Hz -- 8000, 16000, 32000, (48000) - * - * Return value : 0 - Ok - * <0 - Error - */ -int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, uint16_t fs_hz); - -/**************************************************************************** - * WebRtcNetEQ_SignalMcu(...) - * - * Signal the MCU that data is available and ask for a RecOut decision. - * - * Input: - * - inst : MCU instance - * - av_sync : 1 if NetEQ is in AV-sync mode, otherwise 0. - * - * Return value : 0 - Ok - * <0 - Error - */ -int WebRtcNetEQ_SignalMcu(MCUInst_t *inst); - -/**************************************************************************** - * WebRtcNetEQ_RecInInternal(...) - * - * This function inserts a packet into the jitter buffer. - * - * Input: - * - MCU_inst : MCU instance - * - RTPpacket : The RTP packet, parsed into NetEQ's internal RTP struct - * - uw32_timeRec : Time stamp for the arrival of the packet (not RTP timestamp) - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacket, - uint32_t uw32_timeRec); - -/**************************************************************************** - * WebRtcNetEQ_RecInInternal(...) - * - * Split the packet according to split_inst and inserts the parts into - * Buffer_inst. - * - * Input: - * - MCU_inst : MCU instance - * - RTPpacket : The RTP packet, parsed into NetEQ's internal RTP struct - * - uw32_timeRec : Time stamp for the arrival of the packet (not RTP timestamp) - * - av_sync : indicates if AV-sync is enabled, 1 enabled, - * 0 disabled. - * - * Return value : 0 - Ok - * -1 - Error - */ -int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t* packet, - PacketBuf_t* Buffer_inst, - SplitInfo_t* split_inst, - int16_t* flushed, - int av_sync); - -/**************************************************************************** - * WebRtcNetEQ_GetTimestampScaling(...) - * - * Update information about timestamp scaling for a payload type - * in MCU_inst->scalingFactor. - * - * Input: - * - MCU_inst : MCU instance - * - rtpPayloadType : RTP payload number - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType); - -/**************************************************************************** - * WebRtcNetEQ_ScaleTimestampExternalToInternal(...) - * - * Convert from external to internal timestamp using current scaling info. - * - * Input: - * - MCU_inst : MCU instance - * - externalTS : External timestamp - * - * Return value : Internal timestamp - */ - -uint32_t WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst, - uint32_t externalTS); - -/**************************************************************************** - * WebRtcNetEQ_ScaleTimestampInternalToExternal(...) - * - * Convert from external to internal timestamp using current scaling info. - * - * Input: - * - MCU_inst : MCU instance - * - externalTS : Internal timestamp - * - * Return value : External timestamp - */ - -uint32_t WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst, - uint32_t internalTS); -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c deleted file mode 100644 index 666ecc85612..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "mcu.h" - -#include <string.h> /* to define NULL */ - -/* - * Initializes MCU with read address and write address - */ -int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress, - void * Data2DspAddress, void *main_inst) -{ - - inst->pw16_readAddress = (int16_t*) Data2McuAddress; - inst->pw16_writeAddress = (int16_t*) Data2DspAddress; - inst->main_inst = main_inst; - - inst->millisecondsPerCall = 10; - - /* Do expansions in the beginning */ - if (inst->pw16_writeAddress != NULL) inst->pw16_writeAddress[0] = DSP_INSTR_EXPAND; - - return (0); -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c deleted file mode 100644 index 2c48ec7dde2..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c +++ /dev/null @@ -1,45 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Communication between MCU and DSP sides. - */ - -#include "mcu_dsp_common.h" - -#include <string.h> - -/* Initialize instances with read and write address */ -int WebRtcNetEQ_DSPinit(MainInst_t *inst) -{ - int res = 0; - - res |= WebRtcNetEQ_AddressInit(&inst->DSPinst, NULL, NULL, inst); - res |= WebRtcNetEQ_McuAddressInit(&inst->MCUinst, NULL, NULL, inst); - - return res; - -} - -/* The DSP side will call this function to interrupt the MCU side */ -int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, int16_t *pw16_shared_mem) -{ - inst->MCUinst.pw16_readAddress = pw16_shared_mem; - inst->MCUinst.pw16_writeAddress = pw16_shared_mem; - return WebRtcNetEQ_SignalMcu(&inst->MCUinst); -} - -int WebRtcNetEQ_IsSyncPayload(const void* payload, int payload_len_bytes) { - if (payload_len_bytes != SYNC_PAYLOAD_LEN_BYTES || - memcmp(payload, kSyncPayload, SYNC_PAYLOAD_LEN_BYTES) != 0) { - return 0; - } - return 1; -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.h deleted file mode 100644 index b4ab514bc95..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.h +++ /dev/null @@ -1,69 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * The main NetEQ instance, which is where the DSP and MCU sides join. - */ - -#ifndef MCU_DSP_COMMON_H -#define MCU_DSP_COMMON_H - -#include "typedefs.h" - -#include "dsp.h" -#include "mcu.h" - -/* Define size of shared memory area. */ -#if defined(NETEQ_48KHZ_WIDEBAND) - #define SHARED_MEM_SIZE (6*640) -#elif defined(NETEQ_32KHZ_WIDEBAND) - #define SHARED_MEM_SIZE (4*640) -#elif defined(NETEQ_WIDEBAND) - #define SHARED_MEM_SIZE (2*640) -#else - #define SHARED_MEM_SIZE 640 -#endif - -#define SYNC_PAYLOAD_LEN_BYTES 7 -static const uint8_t kSyncPayload[SYNC_PAYLOAD_LEN_BYTES] = { - 'a', 'v', 's', 'y', 'n', 'c', '\0' }; - -/* Struct to hold the NetEQ instance */ -typedef struct -{ - DSPInst_t DSPinst; /* DSP part of the NetEQ instance */ - MCUInst_t MCUinst; /* MCU part of the NetEQ instance */ - int16_t ErrorCode; /* Store last error code */ -#ifdef NETEQ_STEREO - int16_t masterSlave; /* 0 = not set, 1 = master, 2 = slave */ -#endif /* NETEQ_STEREO */ -} MainInst_t; - -/* Struct used for communication between DSP and MCU sides of NetEQ */ -typedef struct -{ - uint32_t playedOutTS; /* Timestamp position at end of DSP data */ - uint16_t samplesLeft; /* Number of samples stored */ - int16_t MD; /* Multiple description codec information */ - int16_t lastMode; /* Latest mode of NetEQ playout */ - int16_t frameLen; /* Frame length of previously decoded packet */ -} DSP2MCU_info_t; - -/* Initialize instances with read and write address */ -int WebRtcNetEQ_DSPinit(MainInst_t *inst); - -/* The DSP side will call this function to interrupt the MCU side */ -int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, int16_t *pw16_shared_mem); - -/* Returns 1 if the given payload matches |kSyncPayload| payload, otherwise - * 0 is returned. */ -int WebRtcNetEQ_IsSyncPayload(const void* payload, int payload_len_bytes); - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c deleted file mode 100644 index ddbb798af87..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c +++ /dev/null @@ -1,131 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Reset MCU side data. - */ - -#include "mcu.h" - -#include <assert.h> -#include <string.h> - -#include "automode.h" - -int WebRtcNetEQ_McuReset(MCUInst_t *inst) -{ - -#ifdef NETEQ_ATEVENT_DECODE - int ok; -#endif - - /* MCU/DSP Communication layer */ - inst->pw16_readAddress = NULL; - inst->pw16_writeAddress = NULL; - inst->main_inst = NULL; - inst->one_desc = 0; - inst->BufferStat_inst.Automode_inst.extraDelayMs = 0; - inst->BufferStat_inst.Automode_inst.minimum_delay_ms = 0; - inst->BufferStat_inst.Automode_inst.maximum_delay_ms = 10000; - inst->NetEqPlayoutMode = kPlayoutOn; - inst->av_sync = 0; - - WebRtcNetEQ_DbReset(&inst->codec_DB_inst); - memset(&inst->PayloadSplit_inst, 0, sizeof(SplitInfo_t)); - - /* Clear the Packet buffer and the pointer to memory storage */ - WebRtcNetEQ_PacketBufferFlush(&inst->PacketBuffer_inst); - inst->PacketBuffer_inst.memorySizeW16 = 0; - inst->PacketBuffer_inst.maxInsertPositions = 0; - - /* Clear the decision and delay history */ - memset(&inst->BufferStat_inst, 0, sizeof(BufstatsInst_t)); -#ifdef NETEQ_ATEVENT_DECODE - ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 8000, 560); - if (ok != 0) - { - return ok; - } -#endif - inst->NoOfExpandCalls = 0; - inst->current_Codec = -1; - inst->current_Payload = -1; - - inst->millisecondsPerCall = 10; - inst->timestampsPerCall = inst->millisecondsPerCall * 8; - inst->fs = 8000; - inst->first_packet = 1; - - WebRtcNetEQ_ResetMcuInCallStats(inst); - - WebRtcNetEQ_ResetWaitingTimeStats(inst); - - WebRtcNetEQ_ResetMcuJitterStat(inst); - - WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst), - inst->PacketBuffer_inst.maxInsertPositions); - - return 0; -} - -/* - * Reset MCU-side statistics variables for the in-call statistics. - */ - -int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst) -{ - inst->lostTS = 0; - inst->lastReportTS = 0; - inst->PacketBuffer_inst.discardedPackets = 0; - - return 0; -} - -/* - * Reset waiting-time statistics. - */ - -void WebRtcNetEQ_ResetWaitingTimeStats(MCUInst_t *inst) { - memset(inst->waiting_times, 0, - kLenWaitingTimes * sizeof(inst->waiting_times[0])); - inst->len_waiting_times = 0; - inst->next_waiting_time_index = 0; -} - -/* - * Store waiting-time in the statistics. - */ - -void WebRtcNetEQ_StoreWaitingTime(MCUInst_t *inst, int waiting_time) { - assert(inst->next_waiting_time_index < kLenWaitingTimes); - inst->waiting_times[inst->next_waiting_time_index] = waiting_time; - inst->next_waiting_time_index++; - if (inst->next_waiting_time_index >= kLenWaitingTimes) { - inst->next_waiting_time_index = 0; - } - if (inst->len_waiting_times < kLenWaitingTimes) { - inst->len_waiting_times++; - } -} - -/* - * Reset all MCU-side statistics variables for the post-call statistics. - */ - -int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst) -{ - inst->BufferStat_inst.Automode_inst.countIAT500ms = 0; - inst->BufferStat_inst.Automode_inst.countIAT1000ms = 0; - inst->BufferStat_inst.Automode_inst.countIAT2000ms = 0; - inst->BufferStat_inst.Automode_inst.longestIATms = 0; - - return 0; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.c deleted file mode 100644 index 78da2c7c7db..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.c +++ /dev/null @@ -1,570 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This is the function to merge a new packet with expanded data after a packet loss. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" -#include "neteq_error_codes.h" - -/**************************************************************************** - * WebRtcNetEQ_Merge(...) - * - * This function... - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector. - * - decoded : Pointer to new decoded speech. - * - len : Number of samples in pw16_decoded. - * - * - * Output: - * - inst : Updated user information - * - outData : Pointer to a memory space where the output data - * should be stored - * - pw16_len : Number of samples written to pw16_outData - * - * Return value : 0 - Ok - * <0 - Error - */ - -/* Scratch usage: - - Type Name size startpos endpos - int16_t pw16_expanded 210*fs/8000 0 209*fs/8000 - int16_t pw16_expandedLB 100 210*fs/8000 99+210*fs/8000 - int16_t pw16_decodedLB 40 100+210*fs/8000 139+210*fs/8000 - int32_t pw32_corr 2*60 140+210*fs/8000 260+210*fs/8000 - int16_t pw16_corrVec 68 210*fs/8000 67+210*fs/8000 - - [gap in scratch vector] - - func WebRtcNetEQ_Expand 40+370*fs/8000 126*fs/8000 39+496*fs/8000 - - Total: 40+496*fs/8000 - */ - -#define SCRATCH_pw16_expanded 0 -#if (defined(NETEQ_48KHZ_WIDEBAND)) -#define SCRATCH_pw16_expandedLB 1260 -#define SCRATCH_pw16_decodedLB 1360 -#define SCRATCH_pw32_corr 1400 -#define SCRATCH_pw16_corrVec 1260 -#define SCRATCH_NETEQ_EXPAND 756 -#elif (defined(NETEQ_32KHZ_WIDEBAND)) -#define SCRATCH_pw16_expandedLB 840 -#define SCRATCH_pw16_decodedLB 940 -#define SCRATCH_pw32_corr 980 -#define SCRATCH_pw16_corrVec 840 -#define SCRATCH_NETEQ_EXPAND 504 -#elif (defined(NETEQ_WIDEBAND)) -#define SCRATCH_pw16_expandedLB 420 -#define SCRATCH_pw16_decodedLB 520 -#define SCRATCH_pw32_corr 560 -#define SCRATCH_pw16_corrVec 420 -#define SCRATCH_NETEQ_EXPAND 252 -#else /* NB */ -#define SCRATCH_pw16_expandedLB 210 -#define SCRATCH_pw16_decodedLB 310 -#define SCRATCH_pw32_corr 350 -#define SCRATCH_pw16_corrVec 210 -#define SCRATCH_NETEQ_EXPAND 126 -#endif - -int WebRtcNetEQ_Merge(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_decoded, int len, int16_t *pw16_outData, - int16_t *pw16_len) -{ - - int16_t fs_mult; - int16_t fs_shift; - int32_t w32_En_new_frame, w32_En_old_frame; - int16_t w16_expmax, w16_newmax; - int16_t w16_tmp, w16_tmp2; - int32_t w32_tmp; -#ifdef SCRATCH - int16_t *pw16_expanded = pw16_scratchPtr + SCRATCH_pw16_expanded; - int16_t *pw16_expandedLB = pw16_scratchPtr + SCRATCH_pw16_expandedLB; - int16_t *pw16_decodedLB = pw16_scratchPtr + SCRATCH_pw16_decodedLB; - int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_pw32_corr); - int16_t *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec; -#else - int16_t pw16_expanded[(125+80+5)*FSMULT]; - int16_t pw16_expandedLB[100]; - int16_t pw16_decodedLB[40]; - int32_t pw32_corr[60]; - int16_t pw16_corrVec[4+60+4]; -#endif - int16_t *pw16_corr = &pw16_corrVec[4]; - int16_t w16_stopPos = 0, w16_bestIndex, w16_interpLen; - int16_t w16_bestVal; /* bestVal is dummy */ - int16_t w16_startfact, w16_inc; - int16_t w16_expandedLen; - int16_t w16_startPos; - int16_t w16_expLen, w16_newLen = 0; - int16_t *pw16_decodedOut; - int16_t w16_muted; - - int w16_decodedLen = len; - -#ifdef NETEQ_STEREO - MasterSlaveInfo *msInfo = inst->msInfo; -#endif - - fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000); - fs_shift = 30 - WebRtcSpl_NormW32(fs_mult); /* Note that this is not "exact" for 48kHz */ - - /************************************* - * Generate data to merge with - *************************************/ - /* - * Check how much data that is left since earlier - * (at least there should be the overlap)... - */ - w16_startPos = inst->endPosition - inst->curPosition; - /* Get one extra expansion to merge and overlap with */ - inst->ExpandInst.w16_stopMuting = 1; - inst->ExpandInst.w16_lagsDirection = 1; /* make sure we get the "optimal" lag */ - inst->ExpandInst.w16_lagsPosition = -1; /* out of the 3 possible ones */ - w16_expandedLen = 0; /* Does not fill any function currently */ - - if (w16_startPos >= 210 * FSMULT) - { - /* - * The number of samples available in the sync buffer is more than what fits in - * pw16_expanded.Keep the first 210*FSMULT samples, but shift them towards the end of - * the buffer. This is ok, since all of the buffer will be expand data anyway, so as - * long as the beginning is left untouched, we're fine. - */ - - w16_tmp = w16_startPos - 210 * FSMULT; /* length difference */ - - WEBRTC_SPL_MEMMOVE_W16(&inst->speechBuffer[inst->curPosition+w16_tmp] , - &inst->speechBuffer[inst->curPosition], 210*FSMULT); - - inst->curPosition += w16_tmp; /* move start position of sync buffer accordingly */ - w16_startPos = 210 * FSMULT; /* this is the truncated length */ - } - - WebRtcNetEQ_Expand(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_expanded, /* let Expand write to beginning of pw16_expanded to avoid overflow */ - &w16_newLen, 0); - - /* - * Now shift the data in pw16_expanded to where it belongs. - * Truncate all that ends up outside the vector. - */ - - WEBRTC_SPL_MEMMOVE_W16(&pw16_expanded[w16_startPos], pw16_expanded, - WEBRTC_SPL_MIN(w16_newLen, - WEBRTC_SPL_MAX(210*FSMULT - w16_startPos, 0) ) ); - - inst->ExpandInst.w16_stopMuting = 0; - - /* Copy what is left since earlier into the expanded vector */ - - WEBRTC_SPL_MEMCPY_W16(pw16_expanded, &inst->speechBuffer[inst->curPosition], w16_startPos); - - /* - * Do "ugly" copy and paste from the expanded in order to generate more data - * to correlate (but not interpolate) with. - */ - w16_expandedLen = (120 + 80 + 2) * fs_mult; - w16_expLen = w16_startPos + w16_newLen; - - if (w16_expLen < w16_expandedLen) - { - while ((w16_expLen + w16_newLen) < w16_expandedLen) - { - WEBRTC_SPL_MEMCPY_W16(&pw16_expanded[w16_expLen], &pw16_expanded[w16_startPos], - w16_newLen); - w16_expLen += w16_newLen; - } - - /* Copy last part (fraction of a whole expansion) */ - - WEBRTC_SPL_MEMCPY_W16(&pw16_expanded[w16_expLen], &pw16_expanded[w16_startPos], - (w16_expandedLen-w16_expLen)); - } - w16_expLen = w16_expandedLen; - - /* Adjust muting factor (main muting factor times expand muting factor) */ - inst->w16_muteFactor - = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor, - inst->ExpandInst.w16_expandMuteFactor, 14); - - /* Adjust muting factor if new vector is more or less of the BGN energy */ - len = WEBRTC_SPL_MIN(64*fs_mult, w16_decodedLen); - w16_expmax = WebRtcSpl_MaxAbsValueW16(pw16_expanded, (int16_t) len); - w16_newmax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len); - - /* Calculate energy of old data */ - w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_expmax, w16_expmax)); - w16_tmp = WEBRTC_SPL_MAX(w16_tmp,0); - w32_En_old_frame = WebRtcNetEQ_DotW16W16(pw16_expanded, pw16_expanded, len, w16_tmp); - - /* Calculate energy of new data */ - w16_tmp2 = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_newmax, w16_newmax)); - w16_tmp2 = WEBRTC_SPL_MAX(w16_tmp2,0); - w32_En_new_frame = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, len, w16_tmp2); - - /* Align to same Q-domain */ - if (w16_tmp2 > w16_tmp) - { - w32_En_old_frame = WEBRTC_SPL_RSHIFT_W32(w32_En_old_frame, (w16_tmp2-w16_tmp)); - } - else - { - w32_En_new_frame = WEBRTC_SPL_RSHIFT_W32(w32_En_new_frame, (w16_tmp-w16_tmp2)); - } - - /* Calculate muting factor to use for new frame */ - if (w32_En_new_frame > w32_En_old_frame) - { - /* Normalize w32_En_new_frame to 14 bits */ - w16_tmp = WebRtcSpl_NormW32(w32_En_new_frame) - 17; - w32_En_new_frame = WEBRTC_SPL_SHIFT_W32(w32_En_new_frame, w16_tmp); - - /* - * Put w32_En_old_frame in a domain 14 higher, so that - * w32_En_old_frame/w32_En_new_frame is in Q14 - */ - w16_tmp = w16_tmp + 14; - w32_En_old_frame = WEBRTC_SPL_SHIFT_W32(w32_En_old_frame, w16_tmp); - w16_tmp - = WebRtcSpl_DivW32W16ResW16(w32_En_old_frame, (int16_t) w32_En_new_frame); - /* Calculate sqrt(w32_En_old_frame/w32_En_new_frame) in Q14 */ - w16_muted = (int16_t) WebRtcSpl_SqrtFloor( - WEBRTC_SPL_LSHIFT_W32((int32_t)w16_tmp,14)); - } - else - { - w16_muted = 16384; /* Set = 1.0 when old frame has higher energy than new */ - } - - /* Set the raise the continued muting factor w16_muted if w16_muteFactor is lower */ - if (w16_muted > inst->w16_muteFactor) - { - inst->w16_muteFactor = WEBRTC_SPL_MIN(w16_muted, 16384); - } - -#ifdef NETEQ_STEREO - - /* Sanity for msInfo */ - if (msInfo == NULL) - { - /* this should not happen here */ - return MASTER_SLAVE_ERROR; - } - - /* do not downsample and calculate correlations for slave instance(s) */ - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)) - { -#endif - - /********************************************* - * Downsample to 4kHz and find best overlap - *********************************************/ - - /* Downsample to 4 kHz */ - if (inst->fs == 8000) - { - WebRtcSpl_DownsampleFast(&pw16_expanded[2], (int16_t) (w16_expandedLen - 2), - pw16_expandedLB, (int16_t) (100), - (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl, (int16_t) 3, - (int16_t) 2, (int16_t) 0); - if (w16_decodedLen <= 80) - { - /* Not quite long enough, so we have to cheat a bit... */ - int16_t temp_len = w16_decodedLen - 2; - w16_tmp = temp_len / 2; - WebRtcSpl_DownsampleFast(&pw16_decoded[2], temp_len, - pw16_decodedLB, w16_tmp, - (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl, - (int16_t) 3, (int16_t) 2, (int16_t) 0); - WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40 - w16_tmp)); - } - else - { - WebRtcSpl_DownsampleFast(&pw16_decoded[2], - (int16_t) (w16_decodedLen - 2), pw16_decodedLB, - (int16_t) (40), (int16_t*) WebRtcNetEQ_kDownsample8kHzTbl, - (int16_t) 3, (int16_t) 2, (int16_t) 0); - } -#ifdef NETEQ_WIDEBAND - } - else if (inst->fs==16000) - { - WebRtcSpl_DownsampleFast( - &pw16_expanded[4], (int16_t)(w16_expandedLen-4), - pw16_expandedLB, (int16_t)(100), - (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl, (int16_t)5, - (int16_t)4, (int16_t)0); - if (w16_decodedLen<=160) - { - /* Not quite long enough, so we have to cheat a bit... */ - int16_t temp_len = w16_decodedLen - 4; - w16_tmp = temp_len / 4; - WebRtcSpl_DownsampleFast( - &pw16_decoded[4], temp_len, - pw16_decodedLB, w16_tmp, - (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl, (int16_t)5, - (int16_t)4, (int16_t)0); - WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp)); - } - else - { - WebRtcSpl_DownsampleFast( - &pw16_decoded[4], (int16_t)(w16_decodedLen-4), - pw16_decodedLB, (int16_t)(40), - (int16_t*)WebRtcNetEQ_kDownsample16kHzTbl, (int16_t)5, - (int16_t)4, (int16_t)0); - } -#endif -#ifdef NETEQ_32KHZ_WIDEBAND - } - else if (inst->fs==32000) - { - /* - * TODO(hlundin) Why is the offset into pw16_expanded 6? - */ - WebRtcSpl_DownsampleFast( - &pw16_expanded[6], (int16_t)(w16_expandedLen-6), - pw16_expandedLB, (int16_t)(100), - (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl, (int16_t)7, - (int16_t)8, (int16_t)0); - if (w16_decodedLen<=320) - { - /* Not quite long enough, so we have to cheat a bit... */ - int16_t temp_len = w16_decodedLen - 6; - w16_tmp = temp_len / 8; - WebRtcSpl_DownsampleFast( - &pw16_decoded[6], temp_len, - pw16_decodedLB, w16_tmp, - (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl, (int16_t)7, - (int16_t)8, (int16_t)0); - WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp)); - } - else - { - WebRtcSpl_DownsampleFast( - &pw16_decoded[6], (int16_t)(w16_decodedLen-6), - pw16_decodedLB, (int16_t)(40), - (int16_t*)WebRtcNetEQ_kDownsample32kHzTbl, (int16_t)7, - (int16_t)8, (int16_t)0); - } -#endif -#ifdef NETEQ_48KHZ_WIDEBAND - } - else /* if (inst->fs==48000) */ - { - /* - * TODO(hlundin) Why is the offset into pw16_expanded 6? - */ - WebRtcSpl_DownsampleFast( - &pw16_expanded[6], (int16_t)(w16_expandedLen-6), - pw16_expandedLB, (int16_t)(100), - (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl, (int16_t)7, - (int16_t)12, (int16_t)0); - if (w16_decodedLen<=320) - { - /* Not quite long enough, so we have to cheat a bit... */ - /* - * TODO(hlundin): Is this correct? Downsampling is a factor 12 - * but w16_tmp = temp_len / 8. - * (Was w16_tmp = ((w16_decodedLen-6)>>3) before re-write.) - */ - int16_t temp_len = w16_decodedLen - 6; - w16_tmp = temp_len / 8; - WebRtcSpl_DownsampleFast( - &pw16_decoded[6], temp_len, - pw16_decodedLB, w16_tmp, - (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl, (int16_t)7, - (int16_t)12, (int16_t)0); - WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp)); - } - else - { - WebRtcSpl_DownsampleFast( - &pw16_decoded[6], (int16_t)(w16_decodedLen-6), - pw16_decodedLB, (int16_t)(40), - (int16_t*)WebRtcNetEQ_kDownsample48kHzTbl, (int16_t)7, - (int16_t)12, (int16_t)0); - } -#endif - } - - /* Calculate correlation without any normalization (40 samples) */ - w16_tmp = WebRtcSpl_DivW32W16ResW16((int32_t) inst->ExpandInst.w16_maxLag, - (int16_t) (fs_mult * 2)) + 1; - w16_stopPos = WEBRTC_SPL_MIN(60, w16_tmp); - w32_tmp = WEBRTC_SPL_MUL_16_16(w16_expmax, w16_newmax); - if (w32_tmp > 26843546) - { - w16_tmp = 3; - } - else - { - w16_tmp = 0; - } - - WebRtcNetEQ_CrossCorr(pw32_corr, pw16_decodedLB, pw16_expandedLB, 40, - (int16_t) w16_stopPos, w16_tmp, 1); - - /* Normalize correlation to 14 bits and put in a int16_t vector */ - WebRtcSpl_MemSetW16(pw16_corrVec, 0, (4 + 60 + 4)); - w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_stopPos); - w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp); - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_stopPos, pw32_corr, w16_tmp); - - /* Calculate allowed starting point for peak finding. - The peak location bestIndex must fulfill two criteria: - (1) w16_bestIndex+w16_decodedLen < inst->timestampsPerCall+inst->ExpandInst.w16_overlap - (2) w16_bestIndex+w16_decodedLen < w16_startPos */ - w16_tmp = WEBRTC_SPL_MAX(0, WEBRTC_SPL_MAX(w16_startPos, - inst->timestampsPerCall+inst->ExpandInst.w16_overlap) - w16_decodedLen); - /* Downscale starting index to 4kHz domain */ - w16_tmp2 = WebRtcSpl_DivW32W16ResW16((int32_t) w16_tmp, - (int16_t) (fs_mult << 1)); - -#ifdef NETEQ_STEREO - } /* end if (msInfo->msMode != NETEQ_SLAVE) */ - - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)) - { - /* This is master or mono instance; find peak */ - WebRtcNetEQ_PeakDetection(&pw16_corr[w16_tmp2], w16_stopPos, 1, fs_mult, &w16_bestIndex, - &w16_bestVal); - w16_bestIndex += w16_tmp; /* compensate for modified starting index */ - msInfo->bestIndex = w16_bestIndex; - } - else if (msInfo->msMode == NETEQ_SLAVE) - { - /* Get peak location from master instance */ - w16_bestIndex = msInfo->bestIndex; - } - else - { - /* Invalid mode */ - return MASTER_SLAVE_ERROR; - } - -#else /* NETEQ_STEREO */ - - /* Find peak */ - WebRtcNetEQ_PeakDetection(&pw16_corr[w16_tmp2], w16_stopPos, 1, fs_mult, &w16_bestIndex, - &w16_bestVal); - w16_bestIndex += w16_tmp; /* compensate for modified starting index */ - -#endif /* NETEQ_STEREO */ - - /* - * Ensure that underrun does not occur for 10ms case => we have to get at least - * 10ms + overlap . (This should never happen thanks to the above modification of - * peak-finding starting point.) - * */ - while ((w16_bestIndex + w16_decodedLen) < (inst->timestampsPerCall - + inst->ExpandInst.w16_overlap) || w16_bestIndex + w16_decodedLen < w16_startPos) - { - w16_bestIndex += w16_newLen; /* Jump one lag ahead */ - } - pw16_decodedOut = pw16_outData + w16_bestIndex; - - /* Mute the new decoded data if needed (and unmute it linearly) */ - w16_interpLen = WEBRTC_SPL_MIN(60*fs_mult, - w16_expandedLen-w16_bestIndex); /* this is the overlapping part of pw16_expanded */ - w16_interpLen = WEBRTC_SPL_MIN(w16_interpLen, w16_decodedLen); - w16_inc = WebRtcSpl_DivW32W16ResW16(4194, - fs_mult); /* in Q20, 0.004 for NB and 0.002 for WB */ - if (inst->w16_muteFactor < 16384) - { - WebRtcNetEQ_UnmuteSignal(pw16_decoded, &inst->w16_muteFactor, pw16_decoded, w16_inc, - (int16_t) w16_interpLen); - WebRtcNetEQ_UnmuteSignal(&pw16_decoded[w16_interpLen], &inst->w16_muteFactor, - &pw16_decodedOut[w16_interpLen], w16_inc, - (int16_t) (w16_decodedLen - w16_interpLen)); - } - else - { - /* No muting needed */ - - WEBRTC_SPL_MEMMOVE_W16(&pw16_decodedOut[w16_interpLen], &pw16_decoded[w16_interpLen], - (w16_decodedLen-w16_interpLen)); - } - - /* Do overlap and interpolate linearly */ - w16_inc = WebRtcSpl_DivW32W16ResW16(16384, (int16_t) (w16_interpLen + 1)); /* Q14 */ - w16_startfact = (16384 - w16_inc); - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_expanded, w16_bestIndex); - WebRtcNetEQ_MixVoiceUnvoice(pw16_decodedOut, &pw16_expanded[w16_bestIndex], pw16_decoded, - &w16_startfact, w16_inc, w16_interpLen); - - inst->w16_mode = MODE_MERGE; - inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */ - - /* New added length (w16_startPos samples were borrowed) */ - *pw16_len = w16_bestIndex + w16_decodedLen - w16_startPos; - - /* Update VQmon parameter */ - inst->w16_concealedTS += (*pw16_len - w16_decodedLen); - inst->w16_concealedTS = WEBRTC_SPL_MAX(0, inst->w16_concealedTS); - - /* Update in-call and post-call statistics */ - if (inst->ExpandInst.w16_expandMuteFactor == 0) - { - /* expansion generates noise only */ - inst->statInst.expandedNoiseSamples += (*pw16_len - w16_decodedLen); - /* Short-term activity statistics. */ - inst->activity_stats.merge_expand_bgn_samples += - (*pw16_len - w16_decodedLen); - } - else - { - /* expansion generates more than only noise */ - inst->statInst.expandedVoiceSamples += (*pw16_len - w16_decodedLen); - /* Short-term activity statistics. */ - inst->activity_stats.merge_expand_normal_samples += - (*pw16_len - w16_decodedLen); - } - inst->statInst.expandLength += (*pw16_len - w16_decodedLen); - - - /* Copy back the first part of the data to the speechHistory */ - - WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->curPosition], pw16_outData, w16_startPos); - - - /* Move data to within outData */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, &pw16_outData[w16_startPos], (*pw16_len)); - - return 0; -} - -#undef SCRATCH_pw16_expanded -#undef SCRATCH_pw16_expandedLB -#undef SCRATCH_pw16_decodedLB -#undef SCRATCH_pw32_corr -#undef SCRATCH_pw16_corrVec -#undef SCRATCH_NETEQ_EXPAND diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/merge.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.cc index 463b2ca784c..d3d8077516b 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/merge.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/merge.h" +#include "webrtc/modules/audio_coding/neteq/merge.h" #include <assert.h> #include <string.h> // memmove, memcpy, memset, size_t @@ -16,10 +16,11 @@ #include <algorithm> // min, max #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { @@ -248,7 +249,7 @@ void Merge::Downsample(const int16_t* input, int input_length, int num_coefficients; int decimation_factor = fs_hz_ / 4000; static const int kCompensateDelay = 0; - int length_limit = fs_hz_ / 100; + int length_limit = fs_hz_ / 100; // 10 ms in samples. if (fs_hz_ == 8000) { filter_coefficients = DspHelper::kDownsample8kHzTbl; num_coefficients = 3; @@ -261,8 +262,6 @@ void Merge::Downsample(const int16_t* input, int input_length, } else { // fs_hz_ == 48000 filter_coefficients = DspHelper::kDownsample48kHzTbl; num_coefficients = 7; - // TODO(hlundin) Why is |length_limit| not 480 (legacy)? - length_limit = 320; } int signal_offset = num_coefficients - 1; WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset], @@ -309,9 +308,11 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max, stop_position_downsamp, correlation_shift, 1); // Normalize correlation to 14 bits and copy to a 16-bit array. - static const int kPadLength = 4; - int16_t correlation16[kPadLength + kMaxCorrelationLength + kPadLength] = {0}; - int16_t* correlation_ptr = &correlation16[kPadLength]; + const int pad_length = static_cast<int>(expand_->overlap_length() - 1); + const int correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength; + scoped_ptr<int16_t[]> correlation16(new int16_t[correlation_buffer_size]); + memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t)); + int16_t* correlation_ptr = &correlation16[pad_length]; int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation, stop_position_downsamp); int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation)); @@ -334,7 +335,7 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max, // start index |start_index_downsamp| and the effective array length. int modified_stop_pos = std::min(stop_position_downsamp, - kMaxCorrelationLength + kPadLength - start_index_downsamp); + kMaxCorrelationLength + pad_length - start_index_downsamp); int best_correlation_index; int16_t best_correlation; static const int kNumCorrelationCandidates = 1; @@ -357,4 +358,9 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max, return best_correlation_index; } +int Merge::RequiredFutureSamples() { + return static_cast<int>(fs_hz_ / 100 * num_channels_); // 10 ms. +} + + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/merge.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h index f1f64e6c538..1bf0483dfe1 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/merge.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MERGE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MERGE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_ #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -35,8 +35,8 @@ class Merge { public: Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer) : fs_hz_(fs_hz), - fs_mult_(fs_hz_ / 8000), num_channels_(num_channels), + fs_mult_(fs_hz_ / 8000), timestamps_per_call_(fs_hz_ / 100), expand_(expand), sync_buffer_(sync_buffer), @@ -44,6 +44,8 @@ class Merge { assert(num_channels_ > 0); } + virtual ~Merge() {} + // The main method to produce the audio data. The decoded data is supplied in // |input|, having |input_length| samples in total for all channels // (interleaved). The result is written to |output|. The number of channels @@ -51,9 +53,15 @@ class Merge { // de-interleaving |input|. The values in |external_mute_factor_array| (Q14) // will be used to scale the audio, and is updated in the process. The array // must have |num_channels_| elements. - int Process(int16_t* input, size_t input_length, - int16_t* external_mute_factor_array, - AudioMultiVector* output); + virtual int Process(int16_t* input, size_t input_length, + int16_t* external_mute_factor_array, + AudioMultiVector* output); + + virtual int RequiredFutureSamples(); + + protected: + const int fs_hz_; + const size_t num_channels_; private: static const int kMaxSampleRate = 48000; @@ -87,9 +95,7 @@ class Merge { int start_position, int input_length, int expand_period) const; - const int fs_hz_; const int fs_mult_; // fs_hz_ / 8000. - const size_t num_channels_; const int timestamps_per_call_; Expand* expand_; SyncBuffer* sync_buffer_; @@ -101,4 +107,4 @@ class Merge { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MERGE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/merge_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge_unittest.cc index 1d7b1f1fed1..fb5f789ff1e 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/merge_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge_unittest.cc @@ -10,15 +10,15 @@ // Unit tests for Merge class. -#include "webrtc/modules/audio_coding/neteq4/merge.h" +#include "webrtc/modules/audio_coding/neteq/merge.h" #include <vector> #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/min_distortion.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/min_distortion.c deleted file mode 100644 index 47e2b442cd6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/min_distortion.c +++ /dev/null @@ -1,55 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Calculate best overlap fit according to distortion measure. - */ - -#include "dsp_helpfunctions.h" - -#include "signal_processing_library.h" - -int16_t WebRtcNetEQ_MinDistortion(const int16_t *pw16_data, - int16_t w16_minLag, int16_t w16_maxLag, - int16_t len, int32_t *pw16_dist) -{ - int i, j; - const int16_t *pw16_data1; - const int16_t *pw16_data2; - int32_t w32_diff; - int32_t w32_sumdiff; - int16_t bestIndex = -1; - int32_t minDist = WEBRTC_SPL_WORD32_MAX; - - for (i = w16_minLag; i <= w16_maxLag; i++) - { - w32_sumdiff = 0; - pw16_data1 = pw16_data; - pw16_data2 = pw16_data - i; - - for (j = 0; j < len; j++) - { - w32_diff = pw16_data1[j] - pw16_data2[j]; - w32_sumdiff += WEBRTC_SPL_ABS_W32(w32_diff); - } - - /* Compare with previous minimum */ - if (w32_sumdiff < minDist) - { - minDist = w32_sumdiff; - bestIndex = i; - } - } - - *pw16_dist = minDist; - - return bestIndex; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c deleted file mode 100644 index 6c70d4916a3..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c +++ /dev/null @@ -1,41 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This function mixes a voiced signal with an unvoiced signal and - * updates the weight on a sample by sample basis. - */ - -#include "dsp_helpfunctions.h" - -#include "signal_processing_library.h" - -void WebRtcNetEQ_MixVoiceUnvoice(int16_t *pw16_outData, int16_t *pw16_voicedVec, - int16_t *pw16_unvoicedVec, - int16_t *w16_current_vfraction, - int16_t w16_vfraction_change, int16_t N) -{ - int i; - int16_t w16_tmp2; - int16_t vfraction = *w16_current_vfraction; - - w16_tmp2 = 16384 - vfraction; - for (i = 0; i < N; i++) - { - pw16_outData[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32( - WEBRTC_SPL_MUL_16_16(vfraction, pw16_voicedVec[i]) + - WEBRTC_SPL_MUL_16_16(w16_tmp2, pw16_unvoicedVec[i]) + 8192, - 14); - vfraction -= w16_vfraction_change; - w16_tmp2 += w16_vfraction_change; - } - *w16_current_vfraction = vfraction; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h index a6d587447d6..edf3b54e9ea 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_AUDIO_DECODER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_AUDIO_DECODER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_ -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "gmock/gmock.h" @@ -35,4 +35,4 @@ class MockAudioDecoder : public AudioDecoder { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_AUDIO_DECODER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_audio_vector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_audio_vector.h index 7a4747b0d31..a5a787c7aa4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_audio_vector.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_audio_vector.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_AUDIO_VECTOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_AUDIO_VECTOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_VECTOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_VECTOR_H_ -#include "webrtc/modules/audio_coding/neteq4/audio_vector.h" +#include "webrtc/modules/audio_coding/neteq/audio_vector.h" #include "gmock/gmock.h" @@ -48,4 +48,4 @@ class MockAudioVector : public AudioVector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_AUDIO_VECTOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_VECTOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_buffer_level_filter.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h index 87265517290..d9210668dde 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_buffer_level_filter.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" #include "gmock/gmock.h" @@ -34,4 +34,4 @@ class MockBufferLevelFilter : public BufferLevelFilter { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h index c4ca25a5274..583fa54ba02 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DECODER_DATABASE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DECODER_DATABASE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_ -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" #include "gmock/gmock.h" @@ -61,4 +61,4 @@ class MockDecoderDatabase : public DecoderDatabase { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DECODER_DATABASE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_delay_manager.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h index 1edfb873716..c21a1c28c73 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_delay_manager.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DELAY_MANAGER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DELAY_MANAGER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_ -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" #include "gmock/gmock.h" @@ -60,4 +60,4 @@ class MockDelayManager : public DelayManager { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DELAY_MANAGER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_delay_peak_detector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h index 211b2b91e12..26e09329e69 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_delay_peak_detector.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_ -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" #include "gmock/gmock.h" @@ -31,4 +31,4 @@ class MockDelayPeakDetector : public DelayPeakDetector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h index 5a89db46f68..0351d6b1e51 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DTMF_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DTMF_BUFFER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_ -#include "webrtc/modules/audio_coding/neteq4/dtmf_buffer.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h" #include "gmock/gmock.h" @@ -35,4 +35,4 @@ class MockDtmfBuffer : public DtmfBuffer { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DTMF_BUFFER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h index d34f7470ef2..3bed4d152b9 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ -#include "webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h" #include "gmock/gmock.h" @@ -32,4 +32,4 @@ class MockDtmfToneGenerator : public DtmfToneGenerator { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h index efc0c715837..9522b537e3e 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_ -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "gmock/gmock.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -96,4 +96,4 @@ class MockExternalPcm16B : public ExternalPcm16B { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_packet_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h index 37fa90de737..2882248c192 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_packet_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_PACKET_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_PACKET_BUFFER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_ -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" #include "gmock/gmock.h" @@ -19,8 +19,8 @@ namespace webrtc { class MockPacketBuffer : public PacketBuffer { public: - MockPacketBuffer(size_t max_number_of_packets, size_t max_payload_memory) - : PacketBuffer(max_number_of_packets, max_payload_memory) {} + MockPacketBuffer(size_t max_number_of_packets) + : PacketBuffer(max_number_of_packets) {} virtual ~MockPacketBuffer() { Die(); } MOCK_METHOD0(Die, void()); MOCK_METHOD0(Flush, @@ -55,4 +55,4 @@ class MockPacketBuffer : public PacketBuffer { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_PACKET_BUFFER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_payload_splitter.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h index f3d8c9b048d..f1665423afd 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/mock/mock_payload_splitter.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_PAYLOAD_SPLITTER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_PAYLOAD_SPLITTER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PAYLOAD_SPLITTER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PAYLOAD_SPLITTER_H_ -#include "webrtc/modules/audio_coding/neteq4/payload_splitter.h" +#include "webrtc/modules/audio_coding/neteq/payload_splitter.h" #include "gmock/gmock.h" @@ -21,6 +21,8 @@ class MockPayloadSplitter : public PayloadSplitter { public: MOCK_METHOD1(SplitRed, int(PacketList* packet_list)); + MOCK_METHOD2(SplitFec, + int(PacketList* packet_list, DecoderDatabase* decoder_database)); MOCK_METHOD2(CheckRedPayloads, int(PacketList* packet_list, const DecoderDatabase& decoder_database)); MOCK_METHOD2(SplitAudio, @@ -34,4 +36,4 @@ class MockPayloadSplitter : public PayloadSplitter { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MOCK_MOCK_PAYLOAD_SPLITTER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PAYLOAD_SPLITTER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mute_signal.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/mute_signal.c deleted file mode 100644 index 767a71dee1f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mute_signal.c +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This function mutes a signal linearly on a sample by sample basis. - */ - -#include "dsp_helpfunctions.h" - -#include "signal_processing_library.h" - -void WebRtcNetEQ_MuteSignal(int16_t *pw16_inout, int16_t muteSlope, - int16_t N) -{ - int i; - int32_t w32_tmp = 1048608; /* (16384<<6 + 32) */ - - for (i = 0; i < N; i++) - { - pw16_inout[i] - = (int16_t) ((WEBRTC_SPL_MUL_16_16((int16_t)(w32_tmp>>6), pw16_inout[i]) - + 8192) >> 14); - w32_tmp -= muteSlope; - } -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc new file mode 100644 index 00000000000..7edacde7633 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc @@ -0,0 +1,62 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" + +#include "webrtc/modules/audio_coding/neteq/accelerate.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/neteq_impl.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/payload_splitter.h" +#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" +#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" + +namespace webrtc { + +// Creates all classes needed and inject them into a new NetEqImpl object. +// Return the new object. +NetEq* NetEq::Create(const NetEq::Config& config) { + BufferLevelFilter* buffer_level_filter = new BufferLevelFilter; + DecoderDatabase* decoder_database = new DecoderDatabase; + DelayPeakDetector* delay_peak_detector = new DelayPeakDetector; + DelayManager* delay_manager = + new DelayManager(config.max_packets_in_buffer, delay_peak_detector); + delay_manager->SetMaximumDelay(config.max_delay_ms); + DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz); + DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator; + PacketBuffer* packet_buffer = new PacketBuffer(config.max_packets_in_buffer); + PayloadSplitter* payload_splitter = new PayloadSplitter; + TimestampScaler* timestamp_scaler = new TimestampScaler(*decoder_database); + AccelerateFactory* accelerate_factory = new AccelerateFactory; + ExpandFactory* expand_factory = new ExpandFactory; + PreemptiveExpandFactory* preemptive_expand_factory = + new PreemptiveExpandFactory; + return new NetEqImpl(config.sample_rate_hz, + buffer_level_filter, + decoder_database, + delay_manager, + delay_peak_detector, + dtmf_buffer, + dtmf_tone_generator, + packet_buffer, + payload_splitter, + timestamp_scaler, + accelerate_factory, + expand_factory, + preemptive_expand_factory); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi index 27e5c37aab7..21ccee41e1c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi @@ -7,242 +7,208 @@ # be found in the AUTHORS file in the root of the source tree. { + 'variables': { + 'codecs': [ + 'G711', + 'G722', + 'PCM16B', + 'iLBC', + 'iSAC', + 'iSACFix', + 'CNG', + ], + 'neteq_defines': [], + 'conditions': [ + ['include_opus==1', { + 'codecs': ['webrtc_opus',], + 'neteq_defines': ['WEBRTC_CODEC_OPUS',], + }], + ], + 'neteq_dependencies': [ + '<@(codecs)', + '<(DEPTH)/third_party/opus/opus.gyp:opus', + '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + ], + }, 'targets': [ { - 'target_name': 'NetEq', + 'target_name': 'neteq', 'type': 'static_library', 'dependencies': [ - 'CNG', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', + '<@(neteq_dependencies)', ], 'defines': [ - 'NETEQ_VOICEENGINE_CODECS', # TODO: Should create a Chrome define which - 'SCRATCH', # specifies a subset of codecs to support. + '<@(neteq_defines)', ], 'include_dirs': [ - 'interface', - '<(webrtc_root)', + # Need Opus header files for the audio classifier. + '<(DEPTH)/third_party/opus/src/celt', + '<(DEPTH)/third_party/opus/src/src', ], 'direct_dependent_settings': { 'include_dirs': [ - 'interface', - '<(webrtc_root)', + # Need Opus header files for the audio classifier. + '<(DEPTH)/third_party/opus/src/celt', + '<(DEPTH)/third_party/opus/src/src', ], }, + 'export_dependent_settings': [ + '<(DEPTH)/third_party/opus/opus.gyp:opus', + ], 'sources': [ - 'interface/webrtc_neteq.h', - 'interface/webrtc_neteq_help_macros.h', - 'interface/webrtc_neteq_internal.h', - 'accelerate.c', - 'automode.c', - 'automode.h', - 'bgn_update.c', - 'buffer_stats.h', - 'bufstats_decision.c', - 'cng_internal.c', - 'codec_db.c', - 'codec_db.h', - 'codec_db_defines.h', - 'correlator.c', - 'delay_logging.h', - 'dsp.c', - 'dsp.h', - 'dsp_helpfunctions.c', - 'dsp_helpfunctions.h', - 'dtmf_buffer.c', + 'interface/audio_decoder.h', + 'interface/neteq.h', + 'accelerate.cc', + 'accelerate.h', + 'audio_classifier.cc', + 'audio_classifier.h', + 'audio_decoder_impl.cc', + 'audio_decoder_impl.h', + 'audio_decoder.cc', + 'audio_multi_vector.cc', + 'audio_multi_vector.h', + 'audio_vector.cc', + 'audio_vector.h', + 'background_noise.cc', + 'background_noise.h', + 'buffer_level_filter.cc', + 'buffer_level_filter.h', + 'comfort_noise.cc', + 'comfort_noise.h', + 'decision_logic.cc', + 'decision_logic.h', + 'decision_logic_fax.cc', + 'decision_logic_fax.h', + 'decision_logic_normal.cc', + 'decision_logic_normal.h', + 'decoder_database.cc', + 'decoder_database.h', + 'defines.h', + 'delay_manager.cc', + 'delay_manager.h', + 'delay_peak_detector.cc', + 'delay_peak_detector.h', + 'dsp_helper.cc', + 'dsp_helper.h', + 'dtmf_buffer.cc', 'dtmf_buffer.h', - 'dtmf_tonegen.c', - 'dtmf_tonegen.h', - 'expand.c', - 'mcu.h', - 'mcu_address_init.c', - 'mcu_dsp_common.c', - 'mcu_dsp_common.h', - 'mcu_reset.c', - 'merge.c', - 'min_distortion.c', - 'mix_voice_unvoice.c', - 'mute_signal.c', - 'neteq_defines.h', - 'neteq_error_codes.h', - 'neteq_statistics.h', - 'normal.c', - 'packet_buffer.c', + 'dtmf_tone_generator.cc', + 'dtmf_tone_generator.h', + 'expand.cc', + 'expand.h', + 'merge.cc', + 'merge.h', + 'neteq_impl.cc', + 'neteq_impl.h', + 'neteq.cc', + 'statistics_calculator.cc', + 'statistics_calculator.h', + 'normal.cc', + 'normal.h', + 'packet_buffer.cc', 'packet_buffer.h', - 'peak_detection.c', - 'preemptive_expand.c', - 'random_vector.c', - 'recin.c', - 'recout.c', - 'rtcp.c', + 'payload_splitter.cc', + 'payload_splitter.h', + 'post_decode_vad.cc', + 'post_decode_vad.h', + 'preemptive_expand.cc', + 'preemptive_expand.h', + 'random_vector.cc', + 'random_vector.h', + 'rtcp.cc', 'rtcp.h', - 'rtp.c', - 'rtp.h', - 'set_fs.c', - 'signal_mcu.c', - 'split_and_insert.c', - 'unmute_signal.c', - 'webrtc_neteq.c', + 'sync_buffer.cc', + 'sync_buffer.h', + 'timestamp_scaler.cc', + 'timestamp_scaler.h', + 'time_stretch.cc', + 'time_stretch.h', ], }, ], # targets 'conditions': [ ['include_tests==1', { + 'includes': ['neteq_tests.gypi',], 'targets': [ { - 'target_name': 'neteq_unittests', + 'target_name': 'audio_decoder_unittests', 'type': '<(gtest_target_type)', 'dependencies': [ - 'NetEq', - 'NetEqTestTools', - 'neteq_unittest_tools', + '<@(codecs)', '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', '<(webrtc_root)/test/test.gyp:test_support_main', ], - 'sources': [ - 'webrtc_neteq_unittest.cc', + 'defines': [ + 'AUDIO_DECODER_UNITTEST', + 'WEBRTC_CODEC_G722', + 'WEBRTC_CODEC_ILBC', + 'WEBRTC_CODEC_ISACFX', + 'WEBRTC_CODEC_ISAC', + 'WEBRTC_CODEC_PCM16', + '<@(neteq_defines)', ], - # Disable warnings to enable Win64 build, issue 1323. - 'msvs_disabled_warnings': [ - 4267, # size_t to int truncation. + 'sources': [ + 'audio_decoder_impl.cc', + 'audio_decoder_impl.h', + 'audio_decoder_unittest.cc', + 'audio_decoder.cc', + 'interface/audio_decoder.h', ], 'conditions': [ # TODO(henrike): remove build_with_chromium==1 when the bots are # using Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { + ['build_with_chromium==1 and OS=="android"', { 'dependencies': [ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', ], }], ], - }, # neteq_unittests - { - 'target_name': 'NetEqRTPplay', - 'type': 'executable', - 'dependencies': [ - 'NetEq', # NetEQ library defined above - 'NetEqTestTools', # Test helpers - 'G711', - 'G722', - 'PCM16B', - 'iLBC', - 'iSAC', - 'CNG', - ], - 'defines': [ - # TODO: Make codec selection conditional on definitions in target NetEq - 'CODEC_ILBC', - 'CODEC_PCM16B', - 'CODEC_G711', - 'CODEC_G722', - 'CODEC_ISAC', - 'CODEC_PCM16B_WB', - 'CODEC_ISAC_SWB', - 'CODEC_ISAC_FB', - 'CODEC_PCM16B_32KHZ', - 'CODEC_CNGCODEC8', - 'CODEC_CNGCODEC16', - 'CODEC_CNGCODEC32', - 'CODEC_ATEVENT_DECODE', - 'CODEC_RED', - ], - 'include_dirs': [ - '.', - 'test', - ], - 'sources': [ - 'test/NetEqRTPplay.cc', - ], - # Disable warnings to enable Win64 build, issue 1323. - 'msvs_disabled_warnings': [ - 4267, # size_t to int truncation. - ], - }, - - { - 'target_name': 'neteq3_speed_test', - 'type': 'executable', - 'dependencies': [ - 'NetEq', - 'PCM16B', - 'neteq_unittest_tools', - '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', - ], - 'sources': [ - 'test/neteq_speed_test.cc', - ], - }, + }, # audio_decoder_unittests { - 'target_name': 'NetEqTestTools', - # Collection of useful functions used in other tests + 'target_name': 'neteq_unittest_tools', 'type': 'static_library', - 'variables': { - # Expects RTP packets without payloads when enabled. - 'neteq_dummy_rtp%': 0, - }, 'dependencies': [ - 'G711', - 'G722', - 'PCM16B', - 'iLBC', - 'iSAC', - 'CNG', - '<(DEPTH)/testing/gtest.gyp:gtest', + 'rtp_rtcp', ], 'direct_dependent_settings': { 'include_dirs': [ - 'interface', - 'test', + 'tools', ], }, - 'defines': [ - # TODO: Make codec selection conditional on definitions in target NetEq - 'CODEC_ILBC', - 'CODEC_PCM16B', - 'CODEC_G711', - 'CODEC_G722', - 'CODEC_ISAC', - 'CODEC_PCM16B_WB', - 'CODEC_ISAC_SWB', - 'CODEC_ISAC_FB', - 'CODEC_PCM16B_32KHZ', - 'CODEC_CNGCODEC8', - 'CODEC_CNGCODEC16', - 'CODEC_CNGCODEC32', - 'CODEC_ATEVENT_DECODE', - 'CODEC_RED', - ], 'include_dirs': [ - 'interface', - 'test', + 'tools', ], 'sources': [ - 'test/NETEQTEST_CodecClass.cc', - 'test/NETEQTEST_CodecClass.h', - 'test/NETEQTEST_DummyRTPpacket.cc', - 'test/NETEQTEST_DummyRTPpacket.h', - 'test/NETEQTEST_NetEQClass.cc', - 'test/NETEQTEST_NetEQClass.h', - 'test/NETEQTEST_RTPpacket.cc', - 'test/NETEQTEST_RTPpacket.h', - ], - # Disable warnings to enable Win64 build, issue 1323. - 'msvs_disabled_warnings': [ - 4267, # size_t to int truncation. - ], - }, + 'tools/audio_checksum.h', + 'tools/audio_loop.cc', + 'tools/audio_loop.h', + 'tools/audio_sink.h', + 'tools/input_audio_file.cc', + 'tools/input_audio_file.h', + 'tools/output_audio_file.h', + 'tools/packet.cc', + 'tools/packet.h', + 'tools/packet_source.h', + 'tools/rtp_file_source.cc', + 'tools/rtp_file_source.h', + 'tools/rtp_generator.cc', + 'tools/rtp_generator.h', + ], + }, # neteq_unittest_tools ], # targets 'conditions': [ # TODO(henrike): remove build_with_chromium==1 when the bots are using # Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { + ['build_with_chromium==1 and OS=="android"', { 'targets': [ { - 'target_name': 'neteq_unittests_apk_target', + 'target_name': 'audio_decoder_unittests_apk_target', 'type': 'none', 'dependencies': [ - '<(apk_tests_path):neteq_unittests_apk', + '<(apk_tests_path):audio_decoder_unittests_apk', ], }, ], @@ -250,17 +216,17 @@ ['test_isolation_mode != "noop"', { 'targets': [ { - 'target_name': 'neteq_unittests_run', + 'target_name': 'audio_decoder_unittests_run', 'type': 'none', 'dependencies': [ - 'neteq_unittests', + 'audio_decoder_unittests', ], 'includes': [ '../../../build/isolate.gypi', - 'neteq_unittests.isolate', + 'audio_decoder_unittests.isolate', ], 'sources': [ - 'neteq_unittests.isolate', + 'audio_decoder_unittests.isolate', ], }, ], diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_defines.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_defines.h deleted file mode 100644 index b3b3da5b70e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_defines.h +++ /dev/null @@ -1,374 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/***************************************************************************************** - * - * Compilation flags in NetEQ: - * - ***************************************************************************************** - * - ***** Platform flags ****** - * - * SCRATCH Run NetEQ with "Scratch memory" to save some stack memory. - * Definition can be used on all platforms - * - ***** Summary flags ****** - * - * NETEQ_ALL_SPECIAL_CODECS Add support for special codecs (CN/RED/DTMF) - * - * NETEQ_ALL_NB_CODECS Add support for all NB codecs (except CN/RED/DTMF) - * - * NETEQ_ALL_WB_CODECS Add support for all WB codecs (except CN/RED/DTMF) - * - * NETEQ_VOICEENGINE_CODECS Support for all NB, WB and SWB32 codecs and CN, RED and DTMF - * - * NETEQ_ALL_CODECS Support for all NB, WB, SWB 32kHz and SWB 48kHz as well as - * CN, RED and DTMF - * - ***** Sampling frequency ****** - * (Note: usually not needed when Summary flags are used) - * - * NETEQ_WIDEBAND Wideband enabled - * - * NETEQ_32KHZ_WIDEBAND Super wideband @ 32kHz enabled - * - * NETEQ_48KHZ_WIDEBAND Super wideband @ 48kHz enabled - * - ***** Special Codec ****** - * (Note: not needed if NETEQ_ALL_CODECS is used) - * - * NETEQ_RED_CODEC With this flag you enable NetEQ to understand redundancy in - * the RTP. NetEQ will use the redundancy if it's the same - * codec - * - * NETEQ_CNG_CODEC Enable DTX with the CN payload - * - * NETEQ_ATEVENT_DECODE Enable AVT event and play out the corresponding DTMF tone - * - ***** Speech Codecs ***** - * (Note: Not needed if Summary flags are used) - * - * NETEQ_G711_CODEC Enable G.711 u- and A-law - * - * NETEQ_PCM16B_CODEC Enable uncompressed 16-bit - * - * NETEQ_ILBC_CODEC Enable iLBC - * - * NETEQ_ISAC_CODEC Enable iSAC - * - * NETEQ_ISAC_SWB_CODEC Enable iSAC-SWB - * - * Note that the decoder of iSAC full-band operates at 32 kHz, that is the - * decoded signal is at 32 kHz. - * NETEQ_ISAC_FB_CODEC Enable iSAC-FB - * - * NETEQ_G722_CODEC Enable G.722 - * - * NETEQ_G729_CODEC Enable G.729 - * - * NETEQ_G729_1_CODEC Enable G.729.1 - * - * NETEQ_G726_CODEC Enable G.726 - * - * NETEQ_G722_1_CODEC Enable G722.1 - * - * NETEQ_G722_1C_CODEC Enable G722.1 Annex C - * - * NETEQ_OPUS_CODEC Enable Opus - * - * NETEQ_SPEEX_CODEC Enable Speex (at 8 and 16 kHz sample rate) - * - * NETEQ_CELT_CODEC Enable Celt (at 32 kHz sample rate) - * - * NETEQ_GSMFR_CODEC Enable GSM-FR - * - * NETEQ_AMR_CODEC Enable AMR (narrowband) - * - * NETEQ_AMRWB_CODEC Enable AMR-WB - * - * NETEQ_CNG_CODEC Enable DTX with the CNG payload - * - * NETEQ_ATEVENT_DECODE Enable AVT event and play out the corresponding DTMF tone - * - ***** Test flags ****** - * - * WEBRTC_NETEQ_40BITACC_TEST Run NetEQ with simulated 40-bit accumulator to run - * bit-exact to a DSP implementation where the main (splib - * and NetEQ) functions have been 40-bit optimized - * - ***************************************************************************************** - */ - -#if !defined NETEQ_DEFINES_H -#define NETEQ_DEFINES_H - -/* Data block structure for MCU to DSP communication: - * - * - * First 3 16-bit words are pre-header that contains instructions and timestamp update - * Fourth 16-bit word is length of data block 1 - * Rest is payload data - * - * 0 48 64 80 - * -------------...---------------------------------------------------------------------- - * | PreHeader ... | Length 1 | Payload data 1 ...... | Lenght 2| Data block 2.... | ... - * -------------...---------------------------------------------------------------------- - * - * - * Preheader: - * 4 MSB can be either of: - */ - -#define DSP_INSTR_NORMAL 0x1000 -/* Payload data will contain the encoded frames */ - -#define DSP_INSTR_MERGE 0x2000 -/* Payload data block 1 will contain the encoded frame */ -/* Info block will contain the number of missing samples */ - -#define DSP_INSTR_EXPAND 0x3000 -/* Payload data will be empty */ - -#define DSP_INSTR_ACCELERATE 0x4000 -/* Payload data will contain the encoded frame */ - -#define DSP_INSTR_DO_RFC3389CNG 0x5000 -/* Payload data will contain the SID frame if there is one*/ - -#define DSP_INSTR_DTMF_GENERATE 0x6000 -/* Payload data will be one int16_t with the current DTMF value and one - * int16_t with the current volume value - */ -#define DSP_INSTR_NORMAL_ONE_DESC 0x7000 -/* No encoded frames */ - -#define DSP_INSTR_DO_CODEC_INTERNAL_CNG 0x8000 -/* Codec has a built-in VAD/DTX scheme (use the above for "no transmission") */ - -#define DSP_INSTR_PREEMPTIVE_EXPAND 0x9000 -/* Payload data will contain the encoded frames, if any */ - -#define DSP_INSTR_DO_ALTERNATIVE_PLC 0xB000 -/* NetEQ switched off and packet missing... */ - -#define DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS 0xC000 -/* NetEQ switched off and packet missing... */ - -#define DSP_INSTR_DO_AUDIO_REPETITION 0xD000 -/* NetEQ switched off and packet missing... */ - -#define DSP_INSTR_DO_AUDIO_REPETITION_INC_TS 0xE000 -/* NetEQ switched off and packet missing... */ - -#define DSP_INSTR_FADE_TO_BGN 0xF000 -/* Exception handling: fade out to BGN (expand) */ - -/* - * Next 4 bits signal additional data that needs to be transmitted - */ - -#define DSP_CODEC_NO_CHANGE 0x0100 -#define DSP_CODEC_NEW_CODEC 0x0200 -#define DSP_CODEC_ADD_LATE_PKT 0x0300 -#define DSP_CODEC_RESET 0x0400 -#define DSP_DTMF_PAYLOAD 0x0010 - -/* - * The most significant bit of the payload-length - * is used to flag whether the associated payload - * is redundant payload. This currently useful only for - * iSAC, where redundant payloads have to be treated - * differently. Every time the length is read it must be - * masked by DSP_CODEC_MASK_RED_FLAG to ignore the flag. - * Use DSP_CODEC_RED_FLAG to set or retrieve the flag. - */ -#define DSP_CODEC_MASK_RED_FLAG 0x7FFF -#define DSP_CODEC_RED_FLAG 0x8000 - -/* - * The first block of payload data consist of decode function pointers, - * and then the speech blocks. - * - */ - - -/* - * The playout modes that NetEq produced (i.e. gives more info about if the - * Accelerate was successful or not) - */ - -#define MODE_NORMAL 0x0000 -#define MODE_EXPAND 0x0001 -#define MODE_MERGE 0x0002 -#define MODE_SUCCESS_ACCELERATE 0x0003 -#define MODE_UNSUCCESS_ACCELERATE 0x0004 -#define MODE_RFC3389CNG 0x0005 -#define MODE_LOWEN_ACCELERATE 0x0006 -#define MODE_DTMF 0x0007 -#define MODE_ONE_DESCRIPTOR 0x0008 -#define MODE_CODEC_INTERNAL_CNG 0x0009 -#define MODE_SUCCESS_PREEMPTIVE 0x000A -#define MODE_UNSUCCESS_PREEMPTIVE 0x000B -#define MODE_LOWEN_PREEMPTIVE 0x000C -#define MODE_FADE_TO_BGN 0x000D - -#define MODE_ERROR 0x0010 - -#define MODE_AWAITING_CODEC_PTR 0x0100 - -#define MODE_BGN_ONLY 0x0200 - -#define MODE_MASTER_DTMF_SIGNAL 0x0400 - -#define MODE_USING_STEREO 0x0800 - - - -/***********************/ -/* Group codec defines */ -/***********************/ - -#if (defined(NETEQ_ALL_SPECIAL_CODECS)) - #define NETEQ_CNG_CODEC - #define NETEQ_ATEVENT_DECODE - #define NETEQ_RED_CODEC - #define NETEQ_VAD - #define NETEQ_ARBITRARY_CODEC -#endif - -#if (defined(NETEQ_ALL_NB_CODECS)) /* Except RED, DTMF and CNG */ - #define NETEQ_PCM16B_CODEC - #define NETEQ_G711_CODEC - #define NETEQ_ILBC_CODEC - #define NETEQ_G729_CODEC - #define NETEQ_G726_CODEC - #define NETEQ_GSMFR_CODEC - #define NETEQ_OPUS_CODEC - #define NETEQ_AMR_CODEC -#endif - -#if (defined(NETEQ_ALL_WB_CODECS)) /* Except RED, DTMF and CNG */ - #define NETEQ_ISAC_CODEC - #define NETEQ_G722_CODEC - #define NETEQ_G722_1_CODEC - #define NETEQ_G729_1_CODEC - #define NETEQ_OPUS_CODEC - #define NETEQ_SPEEX_CODEC - #define NETEQ_AMRWB_CODEC - #define NETEQ_WIDEBAND -#endif - -#if (defined(NETEQ_ALL_WB32_CODECS)) /* AAC, RED, DTMF and CNG */ - #define NETEQ_ISAC_SWB_CODEC - #define NETEQ_32KHZ_WIDEBAND - #define NETEQ_G722_1C_CODEC - #define NETEQ_CELT_CODEC - #define NETEQ_OPUS_CODEC -#endif - -#if (defined(NETEQ_VOICEENGINE_CODECS)) - /* Special codecs */ - #define NETEQ_CNG_CODEC - #define NETEQ_ATEVENT_DECODE - #define NETEQ_RED_CODEC - #define NETEQ_VAD - #define NETEQ_ARBITRARY_CODEC - - /* Narrowband codecs */ - #define NETEQ_PCM16B_CODEC - #define NETEQ_G711_CODEC - #define NETEQ_ILBC_CODEC - #define NETEQ_AMR_CODEC - #define NETEQ_G729_CODEC - #define NETEQ_GSMFR_CODEC - - /* Wideband codecs */ - #define NETEQ_WIDEBAND - #define NETEQ_ISAC_CODEC - #define NETEQ_G722_CODEC - #define NETEQ_G722_1_CODEC - #define NETEQ_G729_1_CODEC - #define NETEQ_AMRWB_CODEC - #define NETEQ_SPEEX_CODEC - - /* Super wideband 32kHz codecs */ - #define NETEQ_ISAC_SWB_CODEC - #define NETEQ_32KHZ_WIDEBAND - #define NETEQ_G722_1C_CODEC - #define NETEQ_CELT_CODEC - - /* Fullband 48 kHz codecs */ - #define NETEQ_OPUS_CODEC - #define NETEQ_ISAC_FB_CODEC -#endif - -#if (defined(NETEQ_ALL_CODECS)) - /* Special codecs */ - #define NETEQ_CNG_CODEC - #define NETEQ_ATEVENT_DECODE - #define NETEQ_RED_CODEC - #define NETEQ_VAD - #define NETEQ_ARBITRARY_CODEC - - /* Narrowband codecs */ - #define NETEQ_PCM16B_CODEC - #define NETEQ_G711_CODEC - #define NETEQ_ILBC_CODEC - #define NETEQ_G729_CODEC - #define NETEQ_G726_CODEC - #define NETEQ_GSMFR_CODEC - #define NETEQ_AMR_CODEC - - /* Wideband codecs */ - #define NETEQ_WIDEBAND - #define NETEQ_ISAC_CODEC - #define NETEQ_G722_CODEC - #define NETEQ_G722_1_CODEC - #define NETEQ_G729_1_CODEC - #define NETEQ_SPEEX_CODEC - #define NETEQ_AMRWB_CODEC - - /* Super wideband 32kHz codecs */ - #define NETEQ_ISAC_SWB_CODEC - #define NETEQ_32KHZ_WIDEBAND - #define NETEQ_G722_1C_CODEC - #define NETEQ_CELT_CODEC - - /* Super wideband 48kHz codecs */ - #define NETEQ_48KHZ_WIDEBAND - #define NETEQ_OPUS_CODEC - #define NETEQ_ISAC_FB_CODEC -#endif - -/* Max output size from decoding one frame */ -#if defined(NETEQ_48KHZ_WIDEBAND) - #define NETEQ_MAX_FRAME_SIZE 5760 /* 120 ms super wideband */ - #define NETEQ_MAX_OUTPUT_SIZE 6480 /* 120+15 ms super wideband (120 ms - * decoded + 15 ms for merge overlap) */ -#elif defined(NETEQ_32KHZ_WIDEBAND) - #define NETEQ_MAX_FRAME_SIZE 3840 /* 120 ms super wideband */ - #define NETEQ_MAX_OUTPUT_SIZE 4320 /* 120+15 ms super wideband (120 ms - * decoded + 15 ms for merge overlap) */ -#elif defined(NETEQ_WIDEBAND) - #define NETEQ_MAX_FRAME_SIZE 1920 /* 120 ms wideband */ - #define NETEQ_MAX_OUTPUT_SIZE 2160 /* 120+15 ms wideband (120 ms decoded + - * 15 ms for merge overlap) */ -#else - #define NETEQ_MAX_FRAME_SIZE 960 /* 120 ms narrowband */ - #define NETEQ_MAX_OUTPUT_SIZE 1080 /* 120+15 ms narrowband (120 ms decoded - * + 15 ms for merge overlap) */ -#endif - - -/* Enable stereo */ -#define NETEQ_STEREO - -#endif /* #if !defined NETEQ_DEFINES_H */ - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_error_codes.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_error_codes.h deleted file mode 100644 index ab639d9c3e6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_error_codes.h +++ /dev/null @@ -1,81 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Definition of error codes. - * - * NOTE: When modifying the error codes, - * also modify the function WebRtcNetEQ_GetErrorCode! - */ - -#if !defined NETEQ_ERROR_CODES_H -#define NETEQ_ERROR_CODES_H - -/* Misc Error */ -#define NETEQ_OTHER_ERROR -1000 - -/* Misc Recout Errors */ -#define FAULTY_INSTRUCTION -1001 -#define FAULTY_NETWORK_TYPE -1002 -#define FAULTY_DELAYVALUE -1003 -#define FAULTY_PLAYOUTMODE -1004 -#define CORRUPT_INSTANCE -1005 -#define ILLEGAL_MASTER_SLAVE_SWITCH -1006 -#define MASTER_SLAVE_ERROR -1007 - -/* Misc Recout problems */ -#define UNKNOWN_BUFSTAT_DECISION -2001 -#define RECOUT_ERROR_DECODING -2002 -#define RECOUT_ERROR_SAMPLEUNDERRUN -2003 -#define RECOUT_ERROR_DECODED_TOO_MUCH -2004 - -/* Misc RecIn problems */ -#define RECIN_CNG_ERROR -3001 -#define RECIN_UNKNOWNPAYLOAD -3002 -#define RECIN_BUFFERINSERT_ERROR -3003 -#define RECIN_SYNC_RTP_CHANGED_CODEC -3004 -#define RECIN_SYNC_RTP_NOT_ACCEPTABLE -3005 - -/* PBUFFER/BUFSTAT ERRORS */ -#define PBUFFER_INIT_ERROR -4001 -#define PBUFFER_INSERT_ERROR1 -4002 -#define PBUFFER_INSERT_ERROR2 -4003 -#define PBUFFER_INSERT_ERROR3 -4004 -#define PBUFFER_INSERT_ERROR4 -4005 -#define PBUFFER_INSERT_ERROR5 -4006 -#define UNKNOWN_G723_HEADER -4007 -#define PBUFFER_NONEXISTING_PACKET -4008 -#define PBUFFER_NOT_INITIALIZED -4009 -#define AMBIGUOUS_ILBC_FRAME_SIZE -4010 - -/* CODEC DATABASE ERRORS */ -#define CODEC_DB_FULL -5001 -#define CODEC_DB_NOT_EXIST1 -5002 -#define CODEC_DB_NOT_EXIST2 -5003 -#define CODEC_DB_NOT_EXIST3 -5004 -#define CODEC_DB_NOT_EXIST4 -5005 -#define CODEC_DB_UNKNOWN_CODEC -5006 -#define CODEC_DB_PAYLOAD_TAKEN -5007 -#define CODEC_DB_UNSUPPORTED_CODEC -5008 -#define CODEC_DB_UNSUPPORTED_FS -5009 - -/* DTMF ERRORS */ -#define DTMF_DEC_PARAMETER_ERROR -6001 -#define DTMF_INSERT_ERROR -6002 -#define DTMF_GEN_UNKNOWN_SAMP_FREQ -6003 -#define DTMF_NOT_SUPPORTED -6004 - -/* RTP/PACKET ERRORS */ -#define RED_SPLIT_ERROR1 -7001 -#define RED_SPLIT_ERROR2 -7002 -#define RTP_TOO_SHORT_PACKET -7003 -#define RTP_CORRUPT_PACKET -7004 - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc index fec25e985e1..a40107651d0 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc @@ -15,10 +15,10 @@ #include "gmock/gmock.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h" -#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h" -#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -45,13 +45,15 @@ class NetEqExternalDecoderTest : public ::testing::Test { frame_size_ms_(10), frame_size_samples_(frame_size_ms_ * samples_per_ms_), output_size_samples_(frame_size_ms_ * samples_per_ms_), - neteq_external_(NetEq::Create(sample_rate_hz_)), - neteq_(NetEq::Create(sample_rate_hz_)), external_decoder_(new MockExternalPcm16B(kDecoderPCM16Bswb32kHz)), rtp_generator_(samples_per_ms_), payload_size_bytes_(0), last_send_time_(0), last_arrival_time_(0) { + NetEq::Config config; + config.sample_rate_hz = sample_rate_hz_; + neteq_external_ = NetEq::Create(config); + neteq_ = NetEq::Create(config); input_ = new int16_t[frame_size_samples_]; encoded_ = new uint8_t[2 * frame_size_samples_]; } @@ -78,7 +80,6 @@ class NetEqExternalDecoderTest : public ::testing::Test { ASSERT_EQ(NetEq::kOK, neteq_external_->RegisterExternalDecoder(external_decoder_, decoder, - sample_rate_hz_, kPayloadType)); ASSERT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(decoder, kPayloadType)); @@ -202,7 +203,7 @@ class NetEqExternalDecoderTest : public ::testing::Test { scoped_ptr<test::InputAudioFile> input_file_; }; -TEST_F(NetEqExternalDecoderTest, DISABLED_ON_ANDROID(RunTest)) { +TEST_F(NetEqExternalDecoderTest, RunTest) { RunTest(100); // Run 100 laps @ 10 ms each in the test loop. } diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.cc index fb27af2cff8..64a86603943 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/neteq_impl.h" +#include "webrtc/modules/audio_coding/neteq/neteq_impl.h" #include <assert.h> #include <memory.h> // memset @@ -16,28 +16,28 @@ #include <algorithm> #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/neteq4/accelerate.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" -#include "webrtc/modules/audio_coding/neteq4/comfort_noise.h" -#include "webrtc/modules/audio_coding/neteq4/decision_logic.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/defines.h" -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" -#include "webrtc/modules/audio_coding/neteq4/dtmf_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/modules/audio_coding/neteq4/merge.h" -#include "webrtc/modules/audio_coding/neteq4/normal.h" -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" -#include "webrtc/modules/audio_coding/neteq4/payload_splitter.h" -#include "webrtc/modules/audio_coding/neteq4/post_decode_vad.h" -#include "webrtc/modules/audio_coding/neteq4/preemptive_expand.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/timestamp_scaler.h" +#include "webrtc/modules/audio_coding/neteq/accelerate.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/comfort_noise.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/defines.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h" +#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/merge.h" +#include "webrtc/modules/audio_coding/neteq/normal.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" +#include "webrtc/modules/audio_coding/neteq/payload_splitter.h" +#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h" +#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" @@ -58,8 +58,13 @@ NetEqImpl::NetEqImpl(int fs, DtmfToneGenerator* dtmf_tone_generator, PacketBuffer* packet_buffer, PayloadSplitter* payload_splitter, - TimestampScaler* timestamp_scaler) - : buffer_level_filter_(buffer_level_filter), + TimestampScaler* timestamp_scaler, + AccelerateFactory* accelerate_factory, + ExpandFactory* expand_factory, + PreemptiveExpandFactory* preemptive_expand_factory, + bool create_components) + : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + buffer_level_filter_(buffer_level_filter), decoder_database_(decoder_database), delay_manager_(delay_manager), delay_peak_detector_(delay_peak_detector), @@ -69,8 +74,10 @@ NetEqImpl::NetEqImpl(int fs, payload_splitter_(payload_splitter), timestamp_scaler_(timestamp_scaler), vad_(new PostDecodeVad()), + expand_factory_(expand_factory), + accelerate_factory_(accelerate_factory), + preemptive_expand_factory_(preemptive_expand_factory), last_mode_(kModeNormal), - mute_factor_array_(NULL), decoded_buffer_length_(kMaxFrameSize), decoded_buffer_(new int16_t[decoded_buffer_length_]), playout_timestamp_(0), @@ -83,7 +90,6 @@ NetEqImpl::NetEqImpl(int fs, first_packet_(true), error_code_(0), decoder_error_code_(0), - crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), decoded_packet_sequence_number_(-1), decoded_packet_timestamp_(0) { if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) { @@ -91,19 +97,15 @@ NetEqImpl::NetEqImpl(int fs, "Changing to 8000 Hz."; fs = 8000; } - LOG(LS_INFO) << "Create NetEqImpl object with fs = " << fs << "."; + LOG(LS_VERBOSE) << "Create NetEqImpl object with fs = " << fs << "."; fs_hz_ = fs; fs_mult_ = fs / 8000; output_size_samples_ = kOutputSizeMs * 8 * fs_mult_; decoder_frame_length_ = 3 * output_size_samples_; WebRtcSpl_Init(); - decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_, - kPlayoutOn, - decoder_database_.get(), - *packet_buffer_.get(), - delay_manager_.get(), - buffer_level_filter_.get())); - SetSampleRateAndChannels(fs, 1); // Default is 1 channel. + if (create_components) { + SetSampleRateAndChannels(fs, 1); // Default is 1 channel. + } } NetEqImpl::~NetEqImpl() { @@ -143,12 +145,12 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, int error = InsertPacketInternal( rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true); - if (error != 0) { - LOG_FERR1(LS_WARNING, InsertPacketInternal, error); - error_code_ = error; - return kFail; - } - return kOK; + if (error != 0) { + LOG_FERR1(LS_WARNING, InsertPacketInternal, error); + error_code_ = error; + return kFail; + } + return kOK; } int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio, @@ -198,7 +200,6 @@ int NetEqImpl::RegisterPayloadType(enum NetEqDecoder codec, int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder, enum NetEqDecoder codec, - int sample_rate_hz, uint8_t rtp_payload_type) { CriticalSectionScoped lock(crit_sect_.get()); LOG_API2(static_cast<int>(rtp_payload_type), codec); @@ -207,6 +208,7 @@ int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder, assert(false); return kFail; } + const int sample_rate_hz = AudioDecoder::CodecSampleRateHz(codec); int ret = decoder_database_->InsertExternal(rtp_payload_type, codec, sample_rate_hz, decoder); if (ret != DecoderDatabase::kOK) { @@ -278,12 +280,7 @@ void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) { CriticalSectionScoped lock(crit_sect_.get()); if (!decision_logic_.get() || mode != decision_logic_->playout_mode()) { // The reset() method calls delete for the old object. - decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_, - mode, - decoder_database_.get(), - *packet_buffer_.get(), - delay_manager_.get(), - buffer_level_filter_.get())); + CreateDecisionLogic(mode); } } @@ -338,9 +335,15 @@ void NetEqImpl::DisableVad() { vad_->Disable(); } -uint32_t NetEqImpl::PlayoutTimestamp() { +bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) { CriticalSectionScoped lock(crit_sect_.get()); - return timestamp_scaler_->ToExternal(playout_timestamp_); + if (first_packet_) { + // We don't have a valid RTP timestamp until we have decoded our first + // RTP packet. + return false; + } + *timestamp = timestamp_scaler_->ToExternal(playout_timestamp_); + return true; } int NetEqImpl::LastError() { @@ -367,12 +370,9 @@ void NetEqImpl::FlushBuffers() { } void NetEqImpl::PacketBufferStatistics(int* current_num_packets, - int* max_num_packets, - int* current_memory_size_bytes, - int* max_memory_size_bytes) const { + int* max_num_packets) const { CriticalSectionScoped lock(crit_sect_.get()); - packet_buffer_->BufferStat(current_num_packets, max_num_packets, - current_memory_size_bytes, max_memory_size_bytes); + packet_buffer_->BufferStat(current_num_packets, max_num_packets); } int NetEqImpl::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const { @@ -396,6 +396,11 @@ NetEqBackgroundNoiseMode NetEqImpl::BackgroundNoiseMode() const { return background_noise_->mode(); } +const SyncBuffer* NetEqImpl::sync_buffer_for_test() const { + CriticalSectionScoped lock(crit_sect_.get()); + return sync_buffer_.get(); +} + // Methods below this line are private. int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, @@ -552,10 +557,23 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, } } + // Check for FEC in packets, and separate payloads into several packets. + int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get()); + if (ret != PayloadSplitter::kOK) { + LOG_FERR1(LS_WARNING, SplitFec, packet_list.size()); + PacketBuffer::DeleteAllPackets(&packet_list); + switch (ret) { + case PayloadSplitter::kUnknownPayloadType: + return kUnknownRtpPayloadType; + default: + return kOtherError; + } + } + // Split payloads into smaller chunks. This also verifies that all payloads // are of a known payload type. SplitAudio() method is protected against // sync-packets. - int ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_); + ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_); if (ret != PayloadSplitter::kOK) { LOG_FERR1(LS_WARNING, SplitAudio, packet_list.size()); PacketBuffer::DeleteAllPackets(&packet_list); @@ -595,9 +613,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, new_codec_ = true; update_sample_rate_and_channels = true; LOG_F(LS_WARNING) << "Packet buffer flushed"; - } else if (ret == PacketBuffer::kOversizePacket) { - LOG_F(LS_WARNING) << "Packet larger than packet buffer"; - return kOversizePacket; } else if (ret != PacketBuffer::kOK) { LOG_FERR1(LS_WARNING, InsertPacketList, packet_list.size()); PacketBuffer::DeleteAllPackets(&packet_list); @@ -827,10 +842,10 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output, sync_buffer_->set_dtmf_index(sync_buffer_->Size()); } - if ((last_mode_ != kModeExpand) && (last_mode_ != kModeRfc3389Cng)) { - // If last operation was neither expand, nor comfort noise, calculate the - // |playout_timestamp_| from the |sync_buffer_|. However, do not update the - // |playout_timestamp_| if it would be moved "backwards". + if (last_mode_ != kModeExpand) { + // If last operation was not expand, calculate the |playout_timestamp_| from + // the |sync_buffer_|. However, do not update the |playout_timestamp_| if it + // would be moved "backwards". uint32_t temp_timestamp = sync_buffer_->end_timestamp() - static_cast<uint32_t>(sync_buffer_->FutureLength()); if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) { @@ -864,15 +879,15 @@ int NetEqImpl::GetDecision(Operations* operation, } const RTPHeader* header = packet_buffer_->NextRtpHeader(); - if (decision_logic_->CngRfc3389On()) { + if (decision_logic_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) { // Because of timestamp peculiarities, we have to "manually" disallow using // a CNG packet with the same timestamp as the one that was last played. // This can happen when using redundancy and will cause the timing to shift. - while (header && - decoder_database_->IsComfortNoise(header->payloadType) && - end_timestamp >= header->timestamp) { + while (header && decoder_database_->IsComfortNoise(header->payloadType) && + (end_timestamp >= header->timestamp || + end_timestamp + decision_logic_->generated_noise_samples() > + header->timestamp)) { // Don't use this packet, discard it. - // TODO(hlundin): Write test for this case. if (packet_buffer_->DiscardNextPacket() != PacketBuffer::kOK) { assert(false); // Must be ok by design. } @@ -924,7 +939,7 @@ int NetEqImpl::GetDecision(Operations* operation, return 0; } - decision_logic_->ExpandDecision(*operation == kExpand); + decision_logic_->ExpandDecision(*operation); // Check conditions for reset. if (new_codec_ || *operation == kUndefined) { @@ -1043,6 +1058,11 @@ int NetEqImpl::GetDecision(Operations* operation, // Move on with the preemptive expand decision. break; } + case kMerge: { + required_samples = + std::max(merge_->RequiredFutureSamples(), required_samples); + break; + } default: { // Do nothing. } @@ -1126,12 +1146,11 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation, PacketBuffer::DeleteAllPackets(packet_list); return kDecoderNotFound; } - // We should have correct sampling rate and number of channels. They - // are set when packets are inserted. + // If sampling rate or number of channels has changed, we need to make + // a reset. if (decoder_info->fs_hz != fs_hz_ || decoder->channels() != algorithm_buffer_->Channels()) { - LOG_F(LS_ERROR) << "Sampling rate or number of channels mismatch."; - assert(false); + // TODO(tlegrand): Add unittest to cover this event. SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels()); } sync_buffer_->set_end_timestamp(timestamp_); @@ -1771,8 +1790,14 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) { AudioDecoder* decoder = decoder_database_->GetDecoder( packet->header.payloadType); if (decoder) { - packet_duration = packet->sync_packet ? decoder_frame_length_ : - decoder->PacketDuration(packet->payload, packet->payload_length); + if (packet->sync_packet) { + packet_duration = decoder_frame_length_; + } else { + packet_duration = packet->primary ? + decoder->PacketDuration(packet->payload, packet->payload_length) : + decoder->PacketDurationRedundant(packet->payload, + packet->payload_length); + } } else { LOG_FERR1(LS_WARNING, GetDecoder, packet->header.payloadType) << "Could not find a decoder for a packet about to be extracted."; @@ -1805,6 +1830,14 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) { return extracted_samples; } +void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { + // Delete objects and create new ones. + expand_.reset(expand_factory_->Create(background_noise_.get(), + sync_buffer_.get(), &random_vector_, + fs_hz, channels)); + merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); +} + void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { LOG_API2(fs_hz, channels); // TODO(hlundin): Change to an enumerator and skip assert. @@ -1852,19 +1885,20 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { // Reset random vector. random_vector_.Reset(); - // Delete Expand object and create a new one. - expand_.reset(new Expand(background_noise_.get(), sync_buffer_.get(), - &random_vector_, fs_hz, channels)); + UpdatePlcComponents(fs_hz, channels); + // Move index so that we create a small set of future samples (all 0). sync_buffer_->set_next_index(sync_buffer_->next_index() - - expand_->overlap_length()); + expand_->overlap_length()); normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_, expand_.get())); - merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); - accelerate_.reset(new Accelerate(fs_hz, channels, *background_noise_)); - preemptive_expand_.reset(new PreemptiveExpand(fs_hz, channels, - *background_noise_)); + accelerate_.reset( + accelerate_factory_->Create(fs_hz, channels, *background_noise_)); + preemptive_expand_.reset(preemptive_expand_factory_->Create( + fs_hz, channels, + *background_noise_, + static_cast<int>(expand_->overlap_length()))); // Delete ComfortNoise object and create a new one. comfort_noise_.reset(new ComfortNoise(fs_hz, decoder_database_.get(), @@ -1877,8 +1911,11 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { decoded_buffer_.reset(new int16_t[decoded_buffer_length_]); } - // Communicate new sample rate and output size to DecisionLogic object. - assert(decision_logic_.get()); + // Create DecisionLogic if it is not created yet, then communicate new sample + // rate and output size to DecisionLogic object. + if (!decision_logic_.get()) { + CreateDecisionLogic(kPlayoutOn); + } decision_logic_->SetSampleRate(fs_hz_, output_size_samples_); } @@ -1899,4 +1936,12 @@ NetEqOutputType NetEqImpl::LastOutputType() { } } +void NetEqImpl::CreateDecisionLogic(NetEqPlayoutMode mode) { + decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_, + mode, + decoder_database_.get(), + *packet_buffer_.get(), + delay_manager_.get(), + buffer_level_filter_.get())); +} } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.h index 83dd58b5acb..e92babd8e35 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.h @@ -8,20 +8,21 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NETEQ_IMPL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NETEQ_IMPL_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ #include <vector> -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/defines.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" // Declare PacketList. -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" -#include "webrtc/modules/audio_coding/neteq4/rtcp.h" -#include "webrtc/modules/audio_coding/neteq4/statistics_calculator.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/defines.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList. +#include "webrtc/modules/audio_coding/neteq/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/rtcp.h" +#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -48,7 +49,10 @@ class PreemptiveExpand; class RandomVector; class SyncBuffer; class TimestampScaler; +struct AccelerateFactory; struct DtmfEvent; +struct ExpandFactory; +struct PreemptiveExpandFactory; class NetEqImpl : public webrtc::NetEq { public: @@ -63,7 +67,11 @@ class NetEqImpl : public webrtc::NetEq { DtmfToneGenerator* dtmf_tone_generator, PacketBuffer* packet_buffer, PayloadSplitter* payload_splitter, - TimestampScaler* timestamp_scaler); + TimestampScaler* timestamp_scaler, + AccelerateFactory* accelerate_factory, + ExpandFactory* expand_factory, + PreemptiveExpandFactory* preemptive_expand_factory, + bool create_components = true); virtual ~NetEqImpl(); @@ -107,11 +115,10 @@ class NetEqImpl : public webrtc::NetEq { // Provides an externally created decoder object |decoder| to insert in the // decoder database. The decoder implements a decoder of type |codec| and - // associates it with |rtp_payload_type|. The decoder operates at the - // frequency |sample_rate_hz|. Returns kOK on success, kFail on failure. + // associates it with |rtp_payload_type|. Returns kOK on success, kFail on + // failure. virtual int RegisterExternalDecoder(AudioDecoder* decoder, enum NetEqDecoder codec, - int sample_rate_hz, uint8_t rtp_payload_type); // Removes |rtp_payload_type| from the codec database. Returns 0 on success, @@ -159,8 +166,7 @@ class NetEqImpl : public webrtc::NetEq { // Disables post-decode VAD. virtual void DisableVad(); - // Returns the RTP timestamp for the last sample delivered by GetAudio(). - virtual uint32_t PlayoutTimestamp(); + virtual bool GetPlayoutTimestamp(uint32_t* timestamp); virtual int SetTargetNumberOfChannels() { return kNotImplemented; } @@ -179,9 +185,7 @@ class NetEqImpl : public webrtc::NetEq { virtual void FlushBuffers(); virtual void PacketBufferStatistics(int* current_num_packets, - int* max_num_packets, - int* current_memory_size_bytes, - int* max_memory_size_bytes) const; + int* max_num_packets) const; // Get sequence number and timestamp of the latest RTP. // This method is to facilitate NACK. @@ -193,7 +197,10 @@ class NetEqImpl : public webrtc::NetEq { // Gets background noise mode. virtual NetEqBackgroundNoiseMode BackgroundNoiseMode() const; - private: + // This accessor method is only intended for testing purposes. + virtual const SyncBuffer* sync_buffer_for_test() const; + + protected: static const int kOutputSizeMs = 10; static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz. // TODO(hlundin): Provide a better value for kSyncBufferSize. @@ -206,8 +213,8 @@ class NetEqImpl : public webrtc::NetEq { const uint8_t* payload, int length_bytes, uint32_t receive_timestamp, - bool is_sync_packet); - + bool is_sync_packet) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Delivers 10 ms of audio data. The data is written to |output|, which can // hold (at least) |max_length| elements. The number of channels that were @@ -215,9 +222,10 @@ class NetEqImpl : public webrtc::NetEq { // and each channel contains |samples_per_channel| elements. If more than one // channel is written, the samples are interleaved. // Returns 0 on success, otherwise an error code. - int GetAudioInternal(size_t max_length, int16_t* output, - int* samples_per_channel, int* num_channels); - + int GetAudioInternal(size_t max_length, + int16_t* output, + int* samples_per_channel, + int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Provides a decision to the GetAudioInternal method. The decision what to // do is written to |operation|. Packets to decode are written to @@ -227,7 +235,7 @@ class NetEqImpl : public webrtc::NetEq { int GetDecision(Operations* operation, PacketList* packet_list, DtmfEvent* dtmf_event, - bool* play_dtmf); + bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Decodes the speech packets in |packet_list|, and writes the results to // |decoded_buffer|, which is allocated to hold |decoded_buffer_length| @@ -235,113 +243,150 @@ class NetEqImpl : public webrtc::NetEq { // The speech type -- speech or (codec-internal) comfort noise -- is written // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389 // comfort noise, those are not decoded. - int Decode(PacketList* packet_list, Operations* operation, - int* decoded_length, AudioDecoder::SpeechType* speech_type); + int Decode(PacketList* packet_list, + Operations* operation, + int* decoded_length, + AudioDecoder::SpeechType* speech_type) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method to Decode(). Performs the actual decoding. - int DecodeLoop(PacketList* packet_list, Operations* operation, - AudioDecoder* decoder, int* decoded_length, - AudioDecoder::SpeechType* speech_type); + int DecodeLoop(PacketList* packet_list, + Operations* operation, + AudioDecoder* decoder, + int* decoded_length, + AudioDecoder::SpeechType* speech_type) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method which calls the Normal class to perform the normal operation. - void DoNormal(const int16_t* decoded_buffer, size_t decoded_length, - AudioDecoder::SpeechType speech_type, bool play_dtmf); + void DoNormal(const int16_t* decoded_buffer, + size_t decoded_length, + AudioDecoder::SpeechType speech_type, + bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method which calls the Merge class to perform the merge operation. - void DoMerge(int16_t* decoded_buffer, size_t decoded_length, - AudioDecoder::SpeechType speech_type, bool play_dtmf); + void DoMerge(int16_t* decoded_buffer, + size_t decoded_length, + AudioDecoder::SpeechType speech_type, + bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method which calls the Expand class to perform the expand operation. - int DoExpand(bool play_dtmf); + int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method which calls the Accelerate class to perform the accelerate // operation. - int DoAccelerate(int16_t* decoded_buffer, size_t decoded_length, - AudioDecoder::SpeechType speech_type, bool play_dtmf); + int DoAccelerate(int16_t* decoded_buffer, + size_t decoded_length, + AudioDecoder::SpeechType speech_type, + bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method which calls the PreemptiveExpand class to perform the // preemtive expand operation. - int DoPreemptiveExpand(int16_t* decoded_buffer, size_t decoded_length, - AudioDecoder::SpeechType speech_type, bool play_dtmf); + int DoPreemptiveExpand(int16_t* decoded_buffer, + size_t decoded_length, + AudioDecoder::SpeechType speech_type, + bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort // noise. |packet_list| can either contain one SID frame to update the // noise parameters, or no payload at all, in which case the previously // received parameters are used. - int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf); + int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Calls the audio decoder to generate codec-internal comfort noise when // no packet was received. - void DoCodecInternalCng(); + void DoCodecInternalCng() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Calls the DtmfToneGenerator class to generate DTMF tones. - int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf); + int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Produces packet-loss concealment using alternative methods. If the codec // has an internal PLC, it is called to generate samples. Otherwise, the // method performs zero-stuffing. - void DoAlternativePlc(bool increase_timestamp); + void DoAlternativePlc(bool increase_timestamp) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Overdub DTMF on top of |output|. - int DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels, - int16_t* output) const; + int DtmfOverdub(const DtmfEvent& dtmf_event, + size_t num_channels, + int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Extracts packets from |packet_buffer_| to produce at least // |required_samples| samples. The packets are inserted into |packet_list|. // Returns the number of samples that the packets in the list will produce, or // -1 in case of an error. - int ExtractPackets(int required_samples, PacketList* packet_list); + int ExtractPackets(int required_samples, PacketList* packet_list) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Resets various variables and objects to new values based on the sample rate // |fs_hz| and |channels| number audio channels. - void SetSampleRateAndChannels(int fs_hz, size_t channels); + void SetSampleRateAndChannels(int fs_hz, size_t channels) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Returns the output type for the audio produced by the latest call to // GetAudio(). - NetEqOutputType LastOutputType(); - - scoped_ptr<BackgroundNoise> background_noise_; - scoped_ptr<BufferLevelFilter> buffer_level_filter_; - scoped_ptr<DecoderDatabase> decoder_database_; - scoped_ptr<DelayManager> delay_manager_; - scoped_ptr<DelayPeakDetector> delay_peak_detector_; - scoped_ptr<DtmfBuffer> dtmf_buffer_; - scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_; - scoped_ptr<PacketBuffer> packet_buffer_; - scoped_ptr<PayloadSplitter> payload_splitter_; - scoped_ptr<TimestampScaler> timestamp_scaler_; - scoped_ptr<DecisionLogic> decision_logic_; - scoped_ptr<PostDecodeVad> vad_; - scoped_ptr<AudioMultiVector> algorithm_buffer_; - scoped_ptr<SyncBuffer> sync_buffer_; - scoped_ptr<Expand> expand_; - scoped_ptr<Normal> normal_; - scoped_ptr<Merge> merge_; - scoped_ptr<Accelerate> accelerate_; - scoped_ptr<PreemptiveExpand> preemptive_expand_; - RandomVector random_vector_; - scoped_ptr<ComfortNoise> comfort_noise_; - Rtcp rtcp_; - StatisticsCalculator stats_; - int fs_hz_; - int fs_mult_; - int output_size_samples_; - int decoder_frame_length_; - Modes last_mode_; - scoped_array<int16_t> mute_factor_array_; - size_t decoded_buffer_length_; - scoped_array<int16_t> decoded_buffer_; - uint32_t playout_timestamp_; - bool new_codec_; - uint32_t timestamp_; - bool reset_decoder_; - uint8_t current_rtp_payload_type_; - uint8_t current_cng_rtp_payload_type_; - uint32_t ssrc_; - bool first_packet_; - int error_code_; // Store last error code. - int decoder_error_code_; - scoped_ptr<CriticalSectionWrapper> crit_sect_; + NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); + + // Updates Expand and Merge. + virtual void UpdatePlcComponents(int fs_hz, size_t channels) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); + + // Creates DecisionLogic object for the given mode. + virtual void CreateDecisionLogic(NetEqPlayoutMode mode) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); + + const scoped_ptr<CriticalSectionWrapper> crit_sect_; + const scoped_ptr<BufferLevelFilter> buffer_level_filter_ + GUARDED_BY(crit_sect_); + const scoped_ptr<DecoderDatabase> decoder_database_ GUARDED_BY(crit_sect_); + const scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_); + const scoped_ptr<DelayPeakDetector> delay_peak_detector_ + GUARDED_BY(crit_sect_); + const scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_); + const scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_ + GUARDED_BY(crit_sect_); + const scoped_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_); + const scoped_ptr<PayloadSplitter> payload_splitter_ GUARDED_BY(crit_sect_); + const scoped_ptr<TimestampScaler> timestamp_scaler_ GUARDED_BY(crit_sect_); + const scoped_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_); + const scoped_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_); + const scoped_ptr<AccelerateFactory> accelerate_factory_ + GUARDED_BY(crit_sect_); + const scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_ + GUARDED_BY(crit_sect_); + + scoped_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_); + scoped_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_); + scoped_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_); + scoped_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_); + scoped_ptr<Expand> expand_ GUARDED_BY(crit_sect_); + scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_); + scoped_ptr<Merge> merge_ GUARDED_BY(crit_sect_); + scoped_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_); + scoped_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_); + RandomVector random_vector_ GUARDED_BY(crit_sect_); + scoped_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_); + Rtcp rtcp_ GUARDED_BY(crit_sect_); + StatisticsCalculator stats_ GUARDED_BY(crit_sect_); + int fs_hz_ GUARDED_BY(crit_sect_); + int fs_mult_ GUARDED_BY(crit_sect_); + int output_size_samples_ GUARDED_BY(crit_sect_); + int decoder_frame_length_ GUARDED_BY(crit_sect_); + Modes last_mode_ GUARDED_BY(crit_sect_); + scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_); + size_t decoded_buffer_length_ GUARDED_BY(crit_sect_); + scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_); + uint32_t playout_timestamp_ GUARDED_BY(crit_sect_); + bool new_codec_ GUARDED_BY(crit_sect_); + uint32_t timestamp_ GUARDED_BY(crit_sect_); + bool reset_decoder_ GUARDED_BY(crit_sect_); + uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_); + uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_); + uint32_t ssrc_ GUARDED_BY(crit_sect_); + bool first_packet_ GUARDED_BY(crit_sect_); + int error_code_ GUARDED_BY(crit_sect_); // Store last error code. + int decoder_error_code_ GUARDED_BY(crit_sect_); // These values are used by NACK module to estimate time-to-play of // a missing packet. Occasionally, NetEq might decide to decode more @@ -350,11 +395,12 @@ class NetEqImpl : public webrtc::NetEq { // such cases, these values do not exactly represent the sequence number // or timestamp associated with a 10ms audio pulled from NetEq. NACK // module is designed to compensate for this. - int decoded_packet_sequence_number_; - uint32_t decoded_packet_timestamp_; + int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_); + uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_); + private: DISALLOW_COPY_AND_ASSIGN(NetEqImpl); }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NETEQ_IMPL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc new file mode 100644 index 00000000000..2e66487fae5 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -0,0 +1,498 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/neteq_impl.h" + +#include "gmock/gmock.h" +#include "gtest/gtest.h" +#include "webrtc/modules/audio_coding/neteq/accelerate.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h" +#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" + +using ::testing::Return; +using ::testing::ReturnNull; +using ::testing::_; +using ::testing::SetArgPointee; +using ::testing::InSequence; +using ::testing::Invoke; +using ::testing::WithArg; + +namespace webrtc { + +// This function is called when inserting a packet list into the mock packet +// buffer. The purpose is to delete all inserted packets properly, to avoid +// memory leaks in the test. +int DeletePacketsAndReturnOk(PacketList* packet_list) { + PacketBuffer::DeleteAllPackets(packet_list); + return PacketBuffer::kOK; +} + +class NetEqImplTest : public ::testing::Test { + protected: + NetEqImplTest() + : neteq_(NULL), + config_(), + mock_buffer_level_filter_(NULL), + buffer_level_filter_(NULL), + use_mock_buffer_level_filter_(true), + mock_decoder_database_(NULL), + decoder_database_(NULL), + use_mock_decoder_database_(true), + mock_delay_peak_detector_(NULL), + delay_peak_detector_(NULL), + use_mock_delay_peak_detector_(true), + mock_delay_manager_(NULL), + delay_manager_(NULL), + use_mock_delay_manager_(true), + mock_dtmf_buffer_(NULL), + dtmf_buffer_(NULL), + use_mock_dtmf_buffer_(true), + mock_dtmf_tone_generator_(NULL), + dtmf_tone_generator_(NULL), + use_mock_dtmf_tone_generator_(true), + mock_packet_buffer_(NULL), + packet_buffer_(NULL), + use_mock_packet_buffer_(true), + mock_payload_splitter_(NULL), + payload_splitter_(NULL), + use_mock_payload_splitter_(true), + timestamp_scaler_(NULL) { + config_.sample_rate_hz = 8000; + } + + void CreateInstance() { + if (use_mock_buffer_level_filter_) { + mock_buffer_level_filter_ = new MockBufferLevelFilter; + buffer_level_filter_ = mock_buffer_level_filter_; + } else { + buffer_level_filter_ = new BufferLevelFilter; + } + if (use_mock_decoder_database_) { + mock_decoder_database_ = new MockDecoderDatabase; + EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder()) + .WillOnce(ReturnNull()); + decoder_database_ = mock_decoder_database_; + } else { + decoder_database_ = new DecoderDatabase; + } + if (use_mock_delay_peak_detector_) { + mock_delay_peak_detector_ = new MockDelayPeakDetector; + EXPECT_CALL(*mock_delay_peak_detector_, Reset()).Times(1); + delay_peak_detector_ = mock_delay_peak_detector_; + } else { + delay_peak_detector_ = new DelayPeakDetector; + } + if (use_mock_delay_manager_) { + mock_delay_manager_ = new MockDelayManager(config_.max_packets_in_buffer, + delay_peak_detector_); + EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1); + delay_manager_ = mock_delay_manager_; + } else { + delay_manager_ = + new DelayManager(config_.max_packets_in_buffer, delay_peak_detector_); + } + if (use_mock_dtmf_buffer_) { + mock_dtmf_buffer_ = new MockDtmfBuffer(config_.sample_rate_hz); + dtmf_buffer_ = mock_dtmf_buffer_; + } else { + dtmf_buffer_ = new DtmfBuffer(config_.sample_rate_hz); + } + if (use_mock_dtmf_tone_generator_) { + mock_dtmf_tone_generator_ = new MockDtmfToneGenerator; + dtmf_tone_generator_ = mock_dtmf_tone_generator_; + } else { + dtmf_tone_generator_ = new DtmfToneGenerator; + } + if (use_mock_packet_buffer_) { + mock_packet_buffer_ = new MockPacketBuffer(config_.max_packets_in_buffer); + packet_buffer_ = mock_packet_buffer_; + } else { + packet_buffer_ = new PacketBuffer(config_.max_packets_in_buffer); + } + if (use_mock_payload_splitter_) { + mock_payload_splitter_ = new MockPayloadSplitter; + payload_splitter_ = mock_payload_splitter_; + } else { + payload_splitter_ = new PayloadSplitter; + } + timestamp_scaler_ = new TimestampScaler(*decoder_database_); + AccelerateFactory* accelerate_factory = new AccelerateFactory; + ExpandFactory* expand_factory = new ExpandFactory; + PreemptiveExpandFactory* preemptive_expand_factory = + new PreemptiveExpandFactory; + + neteq_ = new NetEqImpl(config_.sample_rate_hz, + buffer_level_filter_, + decoder_database_, + delay_manager_, + delay_peak_detector_, + dtmf_buffer_, + dtmf_tone_generator_, + packet_buffer_, + payload_splitter_, + timestamp_scaler_, + accelerate_factory, + expand_factory, + preemptive_expand_factory); + ASSERT_TRUE(neteq_ != NULL); + } + + void UseNoMocks() { + ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance"; + use_mock_buffer_level_filter_ = false; + use_mock_decoder_database_ = false; + use_mock_delay_peak_detector_ = false; + use_mock_delay_manager_ = false; + use_mock_dtmf_buffer_ = false; + use_mock_dtmf_tone_generator_ = false; + use_mock_packet_buffer_ = false; + use_mock_payload_splitter_ = false; + } + + virtual ~NetEqImplTest() { + if (use_mock_buffer_level_filter_) { + EXPECT_CALL(*mock_buffer_level_filter_, Die()).Times(1); + } + if (use_mock_decoder_database_) { + EXPECT_CALL(*mock_decoder_database_, Die()).Times(1); + } + if (use_mock_delay_manager_) { + EXPECT_CALL(*mock_delay_manager_, Die()).Times(1); + } + if (use_mock_delay_peak_detector_) { + EXPECT_CALL(*mock_delay_peak_detector_, Die()).Times(1); + } + if (use_mock_dtmf_buffer_) { + EXPECT_CALL(*mock_dtmf_buffer_, Die()).Times(1); + } + if (use_mock_dtmf_tone_generator_) { + EXPECT_CALL(*mock_dtmf_tone_generator_, Die()).Times(1); + } + if (use_mock_packet_buffer_) { + EXPECT_CALL(*mock_packet_buffer_, Die()).Times(1); + } + delete neteq_; + } + + NetEqImpl* neteq_; + NetEq::Config config_; + MockBufferLevelFilter* mock_buffer_level_filter_; + BufferLevelFilter* buffer_level_filter_; + bool use_mock_buffer_level_filter_; + MockDecoderDatabase* mock_decoder_database_; + DecoderDatabase* decoder_database_; + bool use_mock_decoder_database_; + MockDelayPeakDetector* mock_delay_peak_detector_; + DelayPeakDetector* delay_peak_detector_; + bool use_mock_delay_peak_detector_; + MockDelayManager* mock_delay_manager_; + DelayManager* delay_manager_; + bool use_mock_delay_manager_; + MockDtmfBuffer* mock_dtmf_buffer_; + DtmfBuffer* dtmf_buffer_; + bool use_mock_dtmf_buffer_; + MockDtmfToneGenerator* mock_dtmf_tone_generator_; + DtmfToneGenerator* dtmf_tone_generator_; + bool use_mock_dtmf_tone_generator_; + MockPacketBuffer* mock_packet_buffer_; + PacketBuffer* packet_buffer_; + bool use_mock_packet_buffer_; + MockPayloadSplitter* mock_payload_splitter_; + PayloadSplitter* payload_splitter_; + bool use_mock_payload_splitter_; + TimestampScaler* timestamp_scaler_; +}; + + +// This tests the interface class NetEq. +// TODO(hlundin): Move to separate file? +TEST(NetEq, CreateAndDestroy) { + NetEq::Config config; + NetEq* neteq = NetEq::Create(config); + delete neteq; +} + +TEST_F(NetEqImplTest, RegisterPayloadType) { + CreateInstance(); + uint8_t rtp_payload_type = 0; + NetEqDecoder codec_type = kDecoderPCMu; + EXPECT_CALL(*mock_decoder_database_, + RegisterPayload(rtp_payload_type, codec_type)); + neteq_->RegisterPayloadType(codec_type, rtp_payload_type); +} + +TEST_F(NetEqImplTest, RemovePayloadType) { + CreateInstance(); + uint8_t rtp_payload_type = 0; + EXPECT_CALL(*mock_decoder_database_, Remove(rtp_payload_type)) + .WillOnce(Return(DecoderDatabase::kDecoderNotFound)); + // Check that kFail is returned when database returns kDecoderNotFound. + EXPECT_EQ(NetEq::kFail, neteq_->RemovePayloadType(rtp_payload_type)); +} + +TEST_F(NetEqImplTest, InsertPacket) { + CreateInstance(); + const int kPayloadLength = 100; + const uint8_t kPayloadType = 0; + const uint16_t kFirstSequenceNumber = 0x1234; + const uint32_t kFirstTimestamp = 0x12345678; + const uint32_t kSsrc = 0x87654321; + const uint32_t kFirstReceiveTime = 17; + uint8_t payload[kPayloadLength] = {0}; + WebRtcRTPHeader rtp_header; + rtp_header.header.payloadType = kPayloadType; + rtp_header.header.sequenceNumber = kFirstSequenceNumber; + rtp_header.header.timestamp = kFirstTimestamp; + rtp_header.header.ssrc = kSsrc; + + // Create a mock decoder object. + MockAudioDecoder mock_decoder; + // BWE update function called with first packet. + EXPECT_CALL(mock_decoder, IncomingPacket(_, + kPayloadLength, + kFirstSequenceNumber, + kFirstTimestamp, + kFirstReceiveTime)); + // BWE update function called with second packet. + EXPECT_CALL(mock_decoder, IncomingPacket(_, + kPayloadLength, + kFirstSequenceNumber + 1, + kFirstTimestamp + 160, + kFirstReceiveTime + 155)); + EXPECT_CALL(mock_decoder, Die()).Times(1); // Called when deleted. + + // Expectations for decoder database. + EXPECT_CALL(*mock_decoder_database_, IsRed(kPayloadType)) + .WillRepeatedly(Return(false)); // This is not RED. + EXPECT_CALL(*mock_decoder_database_, CheckPayloadTypes(_)) + .Times(2) + .WillRepeatedly(Return(DecoderDatabase::kOK)); // Payload type is valid. + EXPECT_CALL(*mock_decoder_database_, IsDtmf(kPayloadType)) + .WillRepeatedly(Return(false)); // This is not DTMF. + EXPECT_CALL(*mock_decoder_database_, GetDecoder(kPayloadType)) + .Times(3) + .WillRepeatedly(Return(&mock_decoder)); + EXPECT_CALL(*mock_decoder_database_, IsComfortNoise(kPayloadType)) + .WillRepeatedly(Return(false)); // This is not CNG. + DecoderDatabase::DecoderInfo info; + info.codec_type = kDecoderPCMu; + EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType)) + .WillRepeatedly(Return(&info)); + + // Expectations for packet buffer. + EXPECT_CALL(*mock_packet_buffer_, NumPacketsInBuffer()) + .WillOnce(Return(0)) // First packet. + .WillOnce(Return(1)) // Second packet. + .WillOnce(Return(2)); // Second packet, checking after it was inserted. + EXPECT_CALL(*mock_packet_buffer_, Empty()) + .WillOnce(Return(false)); // Called once after first packet is inserted. + EXPECT_CALL(*mock_packet_buffer_, Flush()) + .Times(1); + EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _)) + .Times(2) + .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType), + WithArg<0>(Invoke(DeletePacketsAndReturnOk)))); + // SetArgPointee<2>(kPayloadType) means that the third argument (zero-based + // index) is a pointer, and the variable pointed to is set to kPayloadType. + // Also invoke the function DeletePacketsAndReturnOk to properly delete all + // packets in the list (to avoid memory leaks in the test). + EXPECT_CALL(*mock_packet_buffer_, NextRtpHeader()) + .Times(1) + .WillOnce(Return(&rtp_header.header)); + + // Expectations for DTMF buffer. + EXPECT_CALL(*mock_dtmf_buffer_, Flush()) + .Times(1); + + // Expectations for delay manager. + { + // All expectations within this block must be called in this specific order. + InSequence sequence; // Dummy variable. + // Expectations when the first packet is inserted. + EXPECT_CALL(*mock_delay_manager_, LastDecoderType(kDecoderPCMu)) + .Times(1); + EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf()) + .Times(2) + .WillRepeatedly(Return(-1)); + EXPECT_CALL(*mock_delay_manager_, set_last_pack_cng_or_dtmf(0)) + .Times(1); + EXPECT_CALL(*mock_delay_manager_, ResetPacketIatCount()).Times(1); + // Expectations when the second packet is inserted. Slightly different. + EXPECT_CALL(*mock_delay_manager_, LastDecoderType(kDecoderPCMu)) + .Times(1); + EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf()) + .WillOnce(Return(0)); + EXPECT_CALL(*mock_delay_manager_, SetPacketAudioLength(30)) + .WillOnce(Return(0)); + } + + // Expectations for payload splitter. + EXPECT_CALL(*mock_payload_splitter_, SplitAudio(_, _)) + .Times(2) + .WillRepeatedly(Return(PayloadSplitter::kOK)); + + // Insert first packet. + neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime); + + // Insert second packet. + rtp_header.header.timestamp += 160; + rtp_header.header.sequenceNumber += 1; + neteq_->InsertPacket(rtp_header, payload, kPayloadLength, + kFirstReceiveTime + 155); +} + +TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) { + UseNoMocks(); + CreateInstance(); + + const int kPayloadLengthSamples = 80; + const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit. + const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint32_t kReceiveTime = 17; // Value doesn't matter for this test. + uint8_t payload[kPayloadLengthBytes] = {0}; + WebRtcRTPHeader rtp_header; + rtp_header.header.payloadType = kPayloadType; + rtp_header.header.sequenceNumber = 0x1234; + rtp_header.header.timestamp = 0x12345678; + rtp_header.header.ssrc = 0x87654321; + + EXPECT_EQ(NetEq::kOK, + neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType)); + + // Insert packets. The buffer should not flush. + for (int i = 1; i <= config_.max_packets_in_buffer; ++i) { + EXPECT_EQ(NetEq::kOK, + neteq_->InsertPacket( + rtp_header, payload, kPayloadLengthBytes, kReceiveTime)); + rtp_header.header.timestamp += kPayloadLengthSamples; + rtp_header.header.sequenceNumber += 1; + EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer()); + } + + // Insert one more packet and make sure the buffer got flushed. That is, it + // should only hold one single packet. + EXPECT_EQ(NetEq::kOK, + neteq_->InsertPacket( + rtp_header, payload, kPayloadLengthBytes, kReceiveTime)); + EXPECT_EQ(1, packet_buffer_->NumPacketsInBuffer()); + const RTPHeader* test_header = packet_buffer_->NextRtpHeader(); + EXPECT_EQ(rtp_header.header.timestamp, test_header->timestamp); + EXPECT_EQ(rtp_header.header.sequenceNumber, test_header->sequenceNumber); +} + +// This test verifies that timestamps propagate from the incoming packets +// through to the sync buffer and to the playout timestamp. +TEST_F(NetEqImplTest, VerifyTimestampPropagation) { + UseNoMocks(); + CreateInstance(); + + const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint32_t kReceiveTime = 17; // Value doesn't matter for this test. + const int kSampleRateHz = 8000; + const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms. + const size_t kPayloadLengthBytes = kPayloadLengthSamples; + uint8_t payload[kPayloadLengthBytes] = {0}; + WebRtcRTPHeader rtp_header; + rtp_header.header.payloadType = kPayloadType; + rtp_header.header.sequenceNumber = 0x1234; + rtp_header.header.timestamp = 0x12345678; + rtp_header.header.ssrc = 0x87654321; + + // This is a dummy decoder that produces as many output samples as the input + // has bytes. The output is an increasing series, starting at 1 for the first + // sample, and then increasing by 1 for each sample. + class CountingSamplesDecoder : public AudioDecoder { + public: + explicit CountingSamplesDecoder(enum NetEqDecoder type) + : AudioDecoder(type), next_value_(1) {} + + // Produce as many samples as input bytes (|encoded_len|). + virtual int Decode(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) { + for (size_t i = 0; i < encoded_len; ++i) { + decoded[i] = next_value_++; + } + *speech_type = kSpeech; + return encoded_len; + } + + virtual int Init() { + next_value_ = 1; + return 0; + } + + uint16_t next_value() const { return next_value_; } + + private: + int16_t next_value_; + } decoder_(kDecoderPCM16B); + + EXPECT_EQ(NetEq::kOK, + neteq_->RegisterExternalDecoder( + &decoder_, kDecoderPCM16B, kPayloadType)); + + // Insert one packet. + EXPECT_EQ(NetEq::kOK, + neteq_->InsertPacket( + rtp_header, payload, kPayloadLengthBytes, kReceiveTime)); + + // Pull audio once. + const int kMaxOutputSize = 10 * kSampleRateHz / 1000; + int16_t output[kMaxOutputSize]; + int samples_per_channel; + int num_channels; + NetEqOutputType type; + EXPECT_EQ( + NetEq::kOK, + neteq_->GetAudio( + kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); + ASSERT_EQ(kMaxOutputSize, samples_per_channel); + EXPECT_EQ(1, num_channels); + EXPECT_EQ(kOutputNormal, type); + + // Start with a simple check that the fake decoder is behaving as expected. + EXPECT_EQ(kPayloadLengthSamples, decoder_.next_value() - 1); + + // The value of the last of the output samples is the same as the number of + // samples played from the decoded packet. Thus, this number + the RTP + // timestamp should match the playout timestamp. + uint32_t timestamp = 0; + EXPECT_TRUE(neteq_->GetPlayoutTimestamp(×tamp)); + EXPECT_EQ(rtp_header.header.timestamp + output[samples_per_channel - 1], + timestamp); + + // Check the timestamp for the last value in the sync buffer. This should + // be one full frame length ahead of the RTP timestamp. + const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test(); + ASSERT_TRUE(sync_buffer != NULL); + EXPECT_EQ(rtp_header.header.timestamp + kPayloadLengthSamples, + sync_buffer->end_timestamp()); + + // Check that the number of samples still to play from the sync buffer add + // up with what was already played out. + EXPECT_EQ(kPayloadLengthSamples - output[samples_per_channel - 1], + static_cast<int>(sync_buffer->FutureLength())); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_statistics.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_statistics.h deleted file mode 100644 index bba5b06b964..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_statistics.h +++ /dev/null @@ -1,56 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Definitions of statistics data structures for MCU and DSP sides. - */ - -#include "typedefs.h" - -#ifndef NETEQ_STATISTICS_H -#define NETEQ_STATISTICS_H - -/* - * Statistics struct on DSP side - */ -typedef struct -{ - - /* variables for in-call statistics; queried through WebRtcNetEQ_GetNetworkStatistics */ - uint32_t expandLength; /* number of samples produced through expand */ - uint32_t preemptiveLength; /* number of samples produced through pre-emptive - expand */ - uint32_t accelerateLength; /* number of samples removed through accelerate */ - int addedSamples; /* number of samples inserted in off mode */ - - /* variables for post-call statistics; queried through WebRtcNetEQ_GetJitterStatistics */ - uint32_t expandedVoiceSamples; /* number of voice samples produced through expand */ - uint32_t expandedNoiseSamples; /* number of noise (background) samples produced - through expand */ - -} DSPStats_t; - -typedef struct { - int preemptive_expand_bgn_samples; - int preemptive_expand_normal_samples; - - int expand_bgn_samples; - int expand_normal_samples; - - int merge_expand_bgn_samples; - int merge_expand_normal_samples; - - int accelerate_bgn_samples; - int accelarate_normal_samples; -} ActivityStats; - - -#endif - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc index d6c4150ec62..3c695c81d04 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc @@ -10,14 +10,15 @@ // Test to verify correct stereo and multi-channel operation. +#include <algorithm> #include <string> #include <list> #include "gtest/gtest.h" #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h" -#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -53,14 +54,16 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> { frame_size_ms_(GetParam().frame_size), frame_size_samples_(frame_size_ms_ * samples_per_ms_), output_size_samples_(10 * samples_per_ms_), - neteq_mono_(NetEq::Create(sample_rate_hz_)), - neteq_(NetEq::Create(sample_rate_hz_)), rtp_generator_mono_(samples_per_ms_), rtp_generator_(samples_per_ms_), payload_size_bytes_(0), multi_payload_size_bytes_(0), last_send_time_(0), last_arrival_time_(0) { + NetEq::Config config; + config.sample_rate_hz = sample_rate_hz_; + neteq_mono_ = NetEq::Create(config); + neteq_ = NetEq::Create(config); input_ = new int16_t[frame_size_samples_]; encoded_ = new uint8_t[2 * frame_size_samples_]; input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_]; diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_tests.gypi index a2b9265613f..4d2ce252bdf 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_tests.gypi @@ -12,8 +12,10 @@ 'target_name': 'neteq_rtpplay', 'type': 'executable', 'dependencies': [ - 'NetEq4', - 'NetEq4TestTools', + 'neteq', + 'neteq_test_tools', + 'neteq_unittest_tools', + 'PCM16B', '<(webrtc_root)/test/test.gyp:test_support_main', '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', ], @@ -29,7 +31,7 @@ 'type': 'executable', 'dependencies': [ # TODO(hlundin): Make RTPencode use ACM to encode files. - 'NetEq4TestTools',# Test helpers + 'neteq_test_tools',# Test helpers 'G711', 'G722', 'PCM16B', @@ -47,6 +49,7 @@ 'CODEC_PCM16B_WB', 'CODEC_ISAC_SWB', 'CODEC_PCM16B_32KHZ', + 'CODEC_PCM16B_48KHZ', 'CODEC_CNGCODEC8', 'CODEC_CNGCODEC16', 'CODEC_CNGCODEC32', @@ -79,14 +82,16 @@ }, { - 'target_name': 'RTPanalyze', + 'target_name': 'rtp_analyze', 'type': 'executable', 'dependencies': [ - 'NetEq4TestTools', + 'neteq_unittest_tools', '<(DEPTH)/testing/gtest.gyp:gtest', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', ], 'sources': [ - 'test/RTPanalyze.cc', + 'tools/rtp_analyze.cc', ], }, @@ -94,7 +99,7 @@ 'target_name': 'RTPchange', 'type': 'executable', 'dependencies': [ - 'NetEq4TestTools', + 'neteq_test_tools', '<(DEPTH)/testing/gtest.gyp:gtest', ], 'sources': [ @@ -106,7 +111,7 @@ 'target_name': 'RTPtimeshift', 'type': 'executable', 'dependencies': [ - 'NetEq4TestTools', + 'neteq_test_tools', '<(DEPTH)/testing/gtest.gyp:gtest', ], 'sources': [ @@ -118,7 +123,7 @@ 'target_name': 'RTPcat', 'type': 'executable', 'dependencies': [ - 'NetEq4TestTools', + 'neteq_test_tools', '<(DEPTH)/testing/gtest.gyp:gtest', ], 'sources': [ @@ -130,7 +135,7 @@ 'target_name': 'rtp_to_text', 'type': 'executable', 'dependencies': [ - 'NetEq4TestTools', + 'neteq_test_tools', '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', ], 'sources': [ @@ -139,21 +144,66 @@ }, { - 'target_name': 'neteq4_speed_test', + 'target_name': 'audio_classifier_test', 'type': 'executable', 'dependencies': [ - 'NetEq4', - 'neteq_unittest_tools', + 'neteq', + ], + 'sources': [ + 'test/audio_classifier_test.cc', + ], + }, + + { + 'target_name': 'neteq_test_support', + 'type': 'static_library', + 'dependencies': [ + 'neteq', 'PCM16B', + 'neteq_unittest_tools', + '<(DEPTH)/testing/gtest.gyp:gtest', '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', ], 'sources': [ + 'tools/neteq_performance_test.cc', + 'tools/neteq_performance_test.h', + 'tools/neteq_quality_test.cc', + 'tools/neteq_quality_test.h', + ], + }, # neteq_test_support + + { + 'target_name': 'neteq_speed_test', + 'type': 'executable', + 'dependencies': [ + 'neteq', + 'neteq_test_support', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + '<(webrtc_root)/test/test.gyp:test_support_main', + ], + 'sources': [ 'test/neteq_speed_test.cc', ], }, { - 'target_name': 'NetEq4TestTools', + 'target_name': 'neteq_opus_fec_quality_test', + 'type': 'executable', + 'dependencies': [ + 'neteq', + 'neteq_test_support', + 'webrtc_opus', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + '<(webrtc_root)/test/test.gyp:test_support_main', + ], + 'sources': [ + 'test/neteq_opus_fec_quality_test.cc', + ], + }, + + { + 'target_name': 'neteq_test_tools', # Collection of useful functions used in other tests. 'type': 'static_library', 'variables': { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittest.cc index 965f75f2cdd..0233e195003 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittest.cc @@ -12,19 +12,20 @@ * This file includes unit tests for NetEQ. */ -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include <math.h> #include <stdlib.h> #include <string.h> // memset -#include <cmath> +#include <algorithm> #include <set> #include <string> #include <vector> #include "gflags/gflags.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h" +#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h" #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -140,7 +141,7 @@ void RefFiles::ReadFromFileAndCompare( NetEqNetworkStatistics ref_stats; ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_)); // Compare - EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size)); + ASSERT_EQ(0, memcmp(&stats, &ref_stats, stat_size)); } } @@ -173,11 +174,11 @@ void RefFiles::ReadFromFileAndCompare( ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1, input_fp_)); // Compare - EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost); - EXPECT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost); - EXPECT_EQ(ref_stats.extended_max_sequence_number, + ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost); + ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost); + ASSERT_EQ(ref_stats.extended_max_sequence_number, stats.extended_max_sequence_number); - EXPECT_EQ(ref_stats.jitter, stats.jitter); + ASSERT_EQ(ref_stats.jitter, stats.jitter); } } @@ -219,11 +220,22 @@ class NetEqDecodingTest : public ::testing::Test { const std::set<uint16_t>& drop_seq_numbers, bool expect_seq_no_wrap, bool expect_timestamp_wrap); + void LongCngWithClockDrift(double drift_factor, + double network_freeze_ms, + bool pull_audio_during_freeze, + int delay_tolerance_ms, + int max_time_to_speech_ms); + + void DuplicateCng(); + + uint32_t PlayoutTimestamp(); + NetEq* neteq_; FILE* rtp_fp_; unsigned int sim_clock_; int16_t out_data_[kMaxBlockSize]; int output_sample_rate_; + int algorithmic_delay_ms_; }; // Allocating the static const so that it can be passed by reference. @@ -238,12 +250,18 @@ NetEqDecodingTest::NetEqDecodingTest() : neteq_(NULL), rtp_fp_(NULL), sim_clock_(0), - output_sample_rate_(kInitSampleRateHz) { + output_sample_rate_(kInitSampleRateHz), + algorithmic_delay_ms_(0) { memset(out_data_, 0, sizeof(out_data_)); } void NetEqDecodingTest::SetUp() { - neteq_ = NetEq::Create(kInitSampleRateHz); + NetEq::Config config; + config.sample_rate_hz = kInitSampleRateHz; + neteq_ = NetEq::Create(config); + NetEqNetworkStatistics stat; + ASSERT_EQ(0, neteq_->NetworkStatistics(&stat)); + algorithmic_delay_ms_ = stat.current_buffer_size_ms; ASSERT_TRUE(neteq_); LoadDecoders(); } @@ -365,19 +383,20 @@ void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file, ASSERT_GT(rtp.readFromFile(rtp_fp_), 0); while (rtp.dataLen() >= 0) { int out_len; - Process(&rtp, &out_len); + ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len)); // Query the network statistics API once per second if (sim_clock_ % 1000 == 0) { // Process NetworkStatistics. NetEqNetworkStatistics network_stats; ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats)); - network_stat_files.ProcessReference(network_stats); + ASSERT_NO_FATAL_FAILURE( + network_stat_files.ProcessReference(network_stats)); // Process RTCPstat. RtcpStatistics rtcp_stats; neteq_->GetRtcpStatistics(&rtcp_stats); - rtcp_stat_files.ProcessReference(rtcp_stats); + ASSERT_NO_FATAL_FAILURE(rtcp_stat_files.ProcessReference(rtcp_stats)); } } } @@ -475,8 +494,8 @@ void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz, ASSERT_EQ(expected_samples_per_channel, samples_per_channel); // To be able to test the fading of background noise we need at lease to pull - // 610 frames. - const int kFadingThreshold = 610; + // 611 frames. + const int kFadingThreshold = 611; // Test several CNG-to-PLC packet for the expected behavior. The number 20 is // arbitrary, but sufficiently large to test enough number of frames. @@ -507,56 +526,46 @@ void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz, EXPECT_TRUE(plc_to_cng); // Just to be sure that PLC-to-CNG has occurred. } -#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS) -// Disabled for Windows 64-bit until webrtc:1458 is fixed. -#define MAYBE_TestBitExactness DISABLED_TestBitExactness -#else -#define MAYBE_TestBitExactness TestBitExactness -#endif - -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(MAYBE_TestBitExactness)) { - const std::string kInputRtpFile = webrtc::test::ProjectRootPath() + +TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) { + const std::string input_rtp_file = webrtc::test::ProjectRootPath() + "resources/audio_coding/neteq_universal_new.rtp"; -#if defined(_MSC_VER) && (_MSC_VER >= 1700) - // For Visual Studio 2012 and later, we will have to use the generic reference - // file, rather than the windows-specific one. - const std::string kInputRefFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq4_universal_ref.pcm"; -#else - const std::string kInputRefFile = + // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm + // are identical. The latter could have been removed, but if clients still + // have a copy of the file, the test will fail. + const std::string input_ref_file = webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm"); -#endif if (FLAGS_gen_ref) { - DecodeAndCompare(kInputRtpFile, ""); + DecodeAndCompare(input_rtp_file, ""); } else { - DecodeAndCompare(kInputRtpFile, kInputRefFile); + DecodeAndCompare(input_rtp_file, input_ref_file); } } TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) { - const std::string kInputRtpFile = webrtc::test::ProjectRootPath() + + const std::string input_rtp_file = webrtc::test::ProjectRootPath() + "resources/audio_coding/neteq_universal_new.rtp"; #if defined(_MSC_VER) && (_MSC_VER >= 1700) // For Visual Studio 2012 and later, we will have to use the generic reference // file, rather than the windows-specific one. - const std::string kNetworkStatRefFile = webrtc::test::ProjectRootPath() + + const std::string network_stat_ref_file = webrtc::test::ProjectRootPath() + "resources/audio_coding/neteq4_network_stats.dat"; #else - const std::string kNetworkStatRefFile = + const std::string network_stat_ref_file = webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat"); #endif - const std::string kRtcpStatRefFile = + const std::string rtcp_stat_ref_file = webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat"); if (FLAGS_gen_ref) { - DecodeAndCheckStats(kInputRtpFile, "", ""); + DecodeAndCheckStats(input_rtp_file, "", ""); } else { - DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile); + DecodeAndCheckStats(input_rtp_file, network_stat_ref_file, + rtcp_stat_ref_file); } } // TODO(hlundin): Re-enable test once the statistics interface is up and again. -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) { +TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) { // Use fax mode to avoid time-scaling. This is to simplify the testing of // packet waiting times in the packet buffer. neteq_->SetPlayoutMode(kPlayoutFax); @@ -630,8 +639,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) { EXPECT_EQ(100u, waiting_times.size()); } -TEST_F(NetEqDecodingTest, - DISABLED_ON_ANDROID(TestAverageInterArrivalTimeNegative)) { +TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) { const int kNumFrames = 3000; // Needed for convergence. int frame_index = 0; const int kSamples = 10 * 16; @@ -662,8 +670,7 @@ TEST_F(NetEqDecodingTest, EXPECT_EQ(-103196, network_stats.clockdrift_ppm); } -TEST_F(NetEqDecodingTest, - DISABLED_ON_ANDROID(TestAverageInterArrivalTimePositive)) { +TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) { const int kNumFrames = 5000; // Needed for convergence. int frame_index = 0; const int kSamples = 10 * 16; @@ -694,16 +701,20 @@ TEST_F(NetEqDecodingTest, EXPECT_EQ(110946, network_stats.clockdrift_ppm); } -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) { +void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor, + double network_freeze_ms, + bool pull_audio_during_freeze, + int delay_tolerance_ms, + int max_time_to_speech_ms) { uint16_t seq_no = 0; uint32_t timestamp = 0; const int kFrameSizeMs = 30; const int kSamples = kFrameSizeMs * 16; const int kPayloadBytes = kSamples * 2; - // Apply a clock drift of -25 ms / s (sender faster than receiver). - const double kDriftFactor = 1000.0 / (1000.0 + 25.0); double next_input_time_ms = 0.0; double t_ms; + int out_len; + int num_channels; NetEqOutputType type; // Insert speech for 5 seconds. @@ -718,18 +729,16 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) { ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0)); ++seq_no; timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor; + next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor; } // Pull out data once. - int out_len; - int num_channels; ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); ASSERT_EQ(kBlockSize16kHz, out_len); } EXPECT_EQ(kOutputNormal, type); - int32_t delay_before = timestamp - neteq_->PlayoutTimestamp(); + int32_t delay_before = timestamp - PlayoutTimestamp(); // Insert CNG for 1 minute (= 60000 ms). const int kCngPeriodMs = 100; @@ -746,11 +755,9 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) { ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0)); ++seq_no; timestamp += kCngPeriodSamples; - next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor; + next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor; } // Pull out data once. - int out_len; - int num_channels; ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); ASSERT_EQ(kBlockSize16kHz, out_len); @@ -758,7 +765,49 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) { EXPECT_EQ(kOutputCNG, type); + if (network_freeze_ms > 0) { + // First keep pulling audio for |network_freeze_ms| without inserting + // any data, then insert CNG data corresponding to |network_freeze_ms| + // without pulling any output audio. + const double loop_end_time = t_ms + network_freeze_ms; + for (; t_ms < loop_end_time; t_ms += 10) { + // Pull out data once. + ASSERT_EQ(0, + neteq_->GetAudio( + kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); + ASSERT_EQ(kBlockSize16kHz, out_len); + EXPECT_EQ(kOutputCNG, type); + } + bool pull_once = pull_audio_during_freeze; + // If |pull_once| is true, GetAudio will be called once half-way through + // the network recovery period. + double pull_time_ms = (t_ms + next_input_time_ms) / 2; + while (next_input_time_ms <= t_ms) { + if (pull_once && next_input_time_ms >= pull_time_ms) { + pull_once = false; + // Pull out data once. + ASSERT_EQ( + 0, + neteq_->GetAudio( + kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); + ASSERT_EQ(kBlockSize16kHz, out_len); + EXPECT_EQ(kOutputCNG, type); + t_ms += 10; + } + // Insert one CNG frame each 100 ms. + uint8_t payload[kPayloadBytes]; + int payload_len; + WebRtcRTPHeader rtp_info; + PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len); + ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0)); + ++seq_no; + timestamp += kCngPeriodSamples; + next_input_time_ms += kCngPeriodMs * drift_factor; + } + } + // Insert speech again until output type is speech. + double speech_restart_time_ms = t_ms; while (type != kOutputNormal) { // Each turn in this for loop is 10 ms. while (next_input_time_ms <= t_ms) { @@ -769,11 +818,9 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) { ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0)); ++seq_no; timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor; + next_input_time_ms += kFrameSizeMs * drift_factor; } // Pull out data once. - int out_len; - int num_channels; ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); ASSERT_EQ(kBlockSize16kHz, out_len); @@ -781,13 +828,99 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockDrift)) { t_ms += 10; } - int32_t delay_after = timestamp - neteq_->PlayoutTimestamp(); + // Check that the speech starts again within reasonable time. + double time_until_speech_returns_ms = t_ms - speech_restart_time_ms; + EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms); + int32_t delay_after = timestamp - PlayoutTimestamp(); // Compare delay before and after, and make sure it differs less than 20 ms. - EXPECT_LE(delay_after, delay_before + 20 * 16); - EXPECT_GE(delay_after, delay_before - 20 * 16); + EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16); + EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16); +} + +TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) { + // Apply a clock drift of -25 ms / s (sender faster than receiver). + const double kDriftFactor = 1000.0 / (1000.0 + 25.0); + const double kNetworkFreezeTimeMs = 0.0; + const bool kGetAudioDuringFreezeRecovery = false; + const int kDelayToleranceMs = 20; + const int kMaxTimeToSpeechMs = 100; + LongCngWithClockDrift(kDriftFactor, + kNetworkFreezeTimeMs, + kGetAudioDuringFreezeRecovery, + kDelayToleranceMs, + kMaxTimeToSpeechMs); +} + +TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDrift) { + // Apply a clock drift of +25 ms / s (sender slower than receiver). + const double kDriftFactor = 1000.0 / (1000.0 - 25.0); + const double kNetworkFreezeTimeMs = 0.0; + const bool kGetAudioDuringFreezeRecovery = false; + const int kDelayToleranceMs = 20; + const int kMaxTimeToSpeechMs = 100; + LongCngWithClockDrift(kDriftFactor, + kNetworkFreezeTimeMs, + kGetAudioDuringFreezeRecovery, + kDelayToleranceMs, + kMaxTimeToSpeechMs); +} + +TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDriftNetworkFreeze) { + // Apply a clock drift of -25 ms / s (sender faster than receiver). + const double kDriftFactor = 1000.0 / (1000.0 + 25.0); + const double kNetworkFreezeTimeMs = 5000.0; + const bool kGetAudioDuringFreezeRecovery = false; + const int kDelayToleranceMs = 50; + const int kMaxTimeToSpeechMs = 200; + LongCngWithClockDrift(kDriftFactor, + kNetworkFreezeTimeMs, + kGetAudioDuringFreezeRecovery, + kDelayToleranceMs, + kMaxTimeToSpeechMs); } -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(UnknownPayloadType)) { +TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreeze) { + // Apply a clock drift of +25 ms / s (sender slower than receiver). + const double kDriftFactor = 1000.0 / (1000.0 - 25.0); + const double kNetworkFreezeTimeMs = 5000.0; + const bool kGetAudioDuringFreezeRecovery = false; + const int kDelayToleranceMs = 20; + const int kMaxTimeToSpeechMs = 100; + LongCngWithClockDrift(kDriftFactor, + kNetworkFreezeTimeMs, + kGetAudioDuringFreezeRecovery, + kDelayToleranceMs, + kMaxTimeToSpeechMs); +} + +TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreezeExtraPull) { + // Apply a clock drift of +25 ms / s (sender slower than receiver). + const double kDriftFactor = 1000.0 / (1000.0 - 25.0); + const double kNetworkFreezeTimeMs = 5000.0; + const bool kGetAudioDuringFreezeRecovery = true; + const int kDelayToleranceMs = 20; + const int kMaxTimeToSpeechMs = 100; + LongCngWithClockDrift(kDriftFactor, + kNetworkFreezeTimeMs, + kGetAudioDuringFreezeRecovery, + kDelayToleranceMs, + kMaxTimeToSpeechMs); +} + +TEST_F(NetEqDecodingTest, LongCngWithoutClockDrift) { + const double kDriftFactor = 1.0; // No drift. + const double kNetworkFreezeTimeMs = 0.0; + const bool kGetAudioDuringFreezeRecovery = false; + const int kDelayToleranceMs = 10; + const int kMaxTimeToSpeechMs = 50; + LongCngWithClockDrift(kDriftFactor, + kNetworkFreezeTimeMs, + kGetAudioDuringFreezeRecovery, + kDelayToleranceMs, + kMaxTimeToSpeechMs); +} + +TEST_F(NetEqDecodingTest, UnknownPayloadType) { const int kPayloadBytes = 100; uint8_t payload[kPayloadBytes] = {0}; WebRtcRTPHeader rtp_info; @@ -798,18 +931,6 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(UnknownPayloadType)) { EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError()); } -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(OversizePacket)) { - // Payload size is greater than packet buffer size - const int kPayloadBytes = NetEq::kMaxBytesInBuffer + 1; - uint8_t payload[kPayloadBytes] = {0}; - WebRtcRTPHeader rtp_info; - PopulateRtpInfo(0, 0, &rtp_info); - rtp_info.header.payloadType = 103; // iSAC, no packet splitting. - EXPECT_EQ(NetEq::kFail, - neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0)); - EXPECT_EQ(NetEq::kOversizePacket, neteq_->LastError()); -} - TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) { const int kPayloadBytes = 100; uint8_t payload[kPayloadBytes] = {0}; @@ -849,7 +970,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) { } } -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(GetAudioBeforeInsertPacket)) { +TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) { NetEqOutputType type; // Set all of |out_data_| to 1, and verify that it was set to 0 by the call // to GetAudio. @@ -872,7 +993,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(GetAudioBeforeInsertPacket)) { } } -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(BackgroundNoise)) { +TEST_F(NetEqDecodingTest, BackgroundNoise) { neteq_->SetBackgroundNoiseMode(kBgnOn); CheckBgnOff(8000, kBgnOn); CheckBgnOff(16000, kBgnOn); @@ -892,7 +1013,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(BackgroundNoise)) { EXPECT_EQ(kBgnFade, neteq_->BackgroundNoiseMode()); } -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketInsert)) { +TEST_F(NetEqDecodingTest, SyncPacketInsert) { WebRtcRTPHeader rtp_info; uint32_t receive_timestamp = 0; // For the readability use the following payloads instead of the defaults of @@ -971,12 +1092,16 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketInsert)) { // First insert several noise like packets, then sync-packets. Decoding all // packets should not produce error, statistics should not show any packet loss // and sync-packets should decode to zero. -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) { +// TODO(turajs) we will have a better test if we have a referece NetEq, and +// when Sync packets are inserted in "test" NetEq we insert all-zero payload +// in reference NetEq and compare the output of those two. +TEST_F(NetEqDecodingTest, SyncPacketDecode) { WebRtcRTPHeader rtp_info; PopulateRtpInfo(0, 0, &rtp_info); const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t); uint8_t payload[kPayloadBytes]; int16_t decoded[kBlockSize16kHz]; + int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1; for (int n = 0; n < kPayloadBytes; ++n) { payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence. } @@ -986,7 +1111,6 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) { int num_channels; int samples_per_channel; uint32_t receive_timestamp = 0; - int delay_samples = 0; for (int n = 0; n < 100; ++n) { ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, receive_timestamp)); @@ -996,16 +1120,15 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) { ASSERT_EQ(kBlockSize16kHz, samples_per_channel); ASSERT_EQ(1, num_channels); - // Even if there is RTP packet in NetEq's buffer, the first frame pulled - // from NetEq starts with few zero samples. Here we measure this delay. - if (n == 0) { - while(decoded[delay_samples] == 0) delay_samples++; - } rtp_info.header.sequenceNumber++; rtp_info.header.timestamp += kBlockSize16kHz; receive_timestamp += kBlockSize16kHz; } const int kNumSyncPackets = 10; + + // Make sure sufficient number of sync packets are inserted that we can + // conduct a test. + ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay); // Insert sync-packets, the decoded sequence should be all-zero. for (int n = 0; n < kNumSyncPackets; ++n) { ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); @@ -1014,38 +1137,44 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) { &output_type)); ASSERT_EQ(kBlockSize16kHz, samples_per_channel); ASSERT_EQ(1, num_channels); - EXPECT_TRUE(IsAllZero(&decoded[delay_samples], - samples_per_channel * num_channels - delay_samples)); - delay_samples = 0; // Delay only matters in the first frame. + if (n > algorithmic_frame_delay) { + EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels)); + } rtp_info.header.sequenceNumber++; rtp_info.header.timestamp += kBlockSize16kHz; receive_timestamp += kBlockSize16kHz; } - // We insert a regular packet, if sync packet are not correctly buffered then + + // We insert regular packets, if sync packet are not correctly buffered then // network statistics would show some packet loss. - ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, - receive_timestamp)); - ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded, - &samples_per_channel, &num_channels, - &output_type)); - // Make sure the last inserted packet is decoded and there are non-zero - // samples. - EXPECT_FALSE(IsAllZero(decoded, samples_per_channel * num_channels)); + for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) { + ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, + receive_timestamp)); + ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded, + &samples_per_channel, &num_channels, + &output_type)); + if (n >= algorithmic_frame_delay + 1) { + // Expect that this frame contain samples from regular RTP. + EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels)); + } + rtp_info.header.sequenceNumber++; + rtp_info.header.timestamp += kBlockSize16kHz; + receive_timestamp += kBlockSize16kHz; + } NetEqNetworkStatistics network_stats; ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats)); // Expecting a "clean" network. EXPECT_EQ(0, network_stats.packet_loss_rate); EXPECT_EQ(0, network_stats.expand_rate); EXPECT_EQ(0, network_stats.accelerate_rate); - EXPECT_EQ(0, network_stats.preemptive_rate); + EXPECT_LE(network_stats.preemptive_rate, 150); } // Test if the size of the packet buffer reported correctly when containing // sync packets. Also, test if network packets override sync packets. That is to // prefer decoding a network packet to a sync packet, if both have same sequence // number and timestamp. -TEST_F(NetEqDecodingTest, - DISABLED_ON_ANDROID(SyncPacketBufferSizeAndOverridenByNetworkPackets)) { +TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) { WebRtcRTPHeader rtp_info; PopulateRtpInfo(0, 0, &rtp_info); const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t); @@ -1060,7 +1189,8 @@ TEST_F(NetEqDecodingTest, int num_channels; int samples_per_channel; uint32_t receive_timestamp = 0; - for (int n = 0; n < 1; ++n) { + int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1; + for (int n = 0; n < algorithmic_frame_delay; ++n) { ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, receive_timestamp)); ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded, @@ -1086,7 +1216,8 @@ TEST_F(NetEqDecodingTest, } NetEqNetworkStatistics network_stats; ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats)); - EXPECT_EQ(kNumSyncPackets * 10, network_stats.current_buffer_size_ms); + EXPECT_EQ(kNumSyncPackets * 10 + algorithmic_delay_ms_, + network_stats.current_buffer_size_ms); // Rewind |rtp_info| to that of the first sync packet. memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info)); @@ -1129,7 +1260,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no, NetEqOutputType output_type; uint32_t receive_timestamp = 0; - // Insert speech for 1 second. + // Insert speech for 2 seconds. const int kSpeechDurationMs = 2000; int packets_inserted = 0; uint16_t last_seq_no; @@ -1159,7 +1290,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no, if (packets_inserted > 4) { // Expect preferred and actual buffer size to be no more than 2 frames. EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2); - EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2); + EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 + + algorithmic_delay_ms_); } last_seq_no = seq_no; last_timestamp = timestamp; @@ -1180,9 +1312,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no, ASSERT_EQ(1, num_channels); // Expect delay (in samples) to be less than 2 packets. - EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(), + EXPECT_LE(timestamp - PlayoutTimestamp(), static_cast<uint32_t>(kSamples * 2)); - } // Make sure we have actually tested wrap-around. ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped); @@ -1216,4 +1347,91 @@ TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) { WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true); } -} // namespace +void NetEqDecodingTest::DuplicateCng() { + uint16_t seq_no = 0; + uint32_t timestamp = 0; + const int kFrameSizeMs = 10; + const int kSampleRateKhz = 16; + const int kSamples = kFrameSizeMs * kSampleRateKhz; + const int kPayloadBytes = kSamples * 2; + + const int algorithmic_delay_samples = std::max( + algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8); + // Insert three speech packet. Three are needed to get the frame length + // correct. + int out_len; + int num_channels; + NetEqOutputType type; + uint8_t payload[kPayloadBytes] = {0}; + WebRtcRTPHeader rtp_info; + for (int i = 0; i < 3; ++i) { + PopulateRtpInfo(seq_no, timestamp, &rtp_info); + ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0)); + ++seq_no; + timestamp += kSamples; + + // Pull audio once. + ASSERT_EQ(0, + neteq_->GetAudio( + kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); + ASSERT_EQ(kBlockSize16kHz, out_len); + } + // Verify speech output. + EXPECT_EQ(kOutputNormal, type); + + // Insert same CNG packet twice. + const int kCngPeriodMs = 100; + const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz; + int payload_len; + PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len); + // This is the first time this CNG packet is inserted. + ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0)); + + // Pull audio once and make sure CNG is played. + ASSERT_EQ(0, + neteq_->GetAudio( + kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); + ASSERT_EQ(kBlockSize16kHz, out_len); + EXPECT_EQ(kOutputCNG, type); + EXPECT_EQ(timestamp - algorithmic_delay_samples, PlayoutTimestamp()); + + // Insert the same CNG packet again. Note that at this point it is old, since + // we have already decoded the first copy of it. + ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0)); + + // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since + // we have already pulled out CNG once. + for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) { + ASSERT_EQ(0, + neteq_->GetAudio( + kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); + ASSERT_EQ(kBlockSize16kHz, out_len); + EXPECT_EQ(kOutputCNG, type); + EXPECT_EQ(timestamp - algorithmic_delay_samples, + PlayoutTimestamp()); + } + + // Insert speech again. + ++seq_no; + timestamp += kCngPeriodSamples; + PopulateRtpInfo(seq_no, timestamp, &rtp_info); + ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0)); + + // Pull audio once and verify that the output is speech again. + ASSERT_EQ(0, + neteq_->GetAudio( + kMaxBlockSize, out_data_, &out_len, &num_channels, &type)); + ASSERT_EQ(kBlockSize16kHz, out_len); + EXPECT_EQ(kOutputNormal, type); + EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples, + PlayoutTimestamp()); +} + +uint32_t NetEqDecodingTest::PlayoutTimestamp() { + uint32_t playout_timestamp = 0; + EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&playout_timestamp)); + return playout_timestamp; +} + +TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); } +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittests.isolate b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittests.isolate deleted file mode 100644 index e8f4e482aaf..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittests.isolate +++ /dev/null @@ -1,44 +0,0 @@ -# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. -{ - 'conditions': [ - ['OS=="android"', { - # When doing Android builds, the WebRTC code is put in third_party/webrtc - # of a Chromium checkout, this is one level above the standalone build. - 'variables': { - 'isolate_dependency_untracked': [ - '../../../../../data/', - '../../../../../resources/', - ], - }, - }], - ['OS=="linux" or OS=="mac" or OS=="win"', { - 'variables': { - 'command': [ - '../../../../testing/test_env.py', - '<(PRODUCT_DIR)/neteq_unittests<(EXECUTABLE_SUFFIX)', - ], - 'isolate_dependency_touched': [ - '../../../../DEPS', - ], - 'isolate_dependency_tracked': [ - '../../../../resources/audio_coding/neteq_network_stats.dat', - '../../../../resources/audio_coding/neteq_rtcp_stats.dat', - '../../../../resources/audio_coding/neteq_universal.rtp', - '../../../../resources/audio_coding/neteq_universal_ref.pcm', - '../../../../resources/audio_coding/testfile32kHz.pcm', - '../../../../testing/test_env.py', - '<(PRODUCT_DIR)/neteq_unittests<(EXECUTABLE_SUFFIX)', - ], - 'isolate_dependency_untracked': [ - '../../../../tools/swarming_client/', - ], - }, - }], - ], -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.c deleted file mode 100644 index 8cbda521542..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.c +++ /dev/null @@ -1,279 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the function for handling "normal" speech operation. - */ -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" - -/* Scratch usage: - - Type Name size startpos endpos - int16_t pw16_expanded 125*fs/8000 0 125*fs/8000-1 - - func WebRtcNetEQ_Expand 40+370*fs/8000 125*fs/8000 39+495*fs/8000 - - Total: 40+495*fs/8000 - */ - -#define SCRATCH_PW16_EXPANDED 0 -#if (defined(NETEQ_48KHZ_WIDEBAND)) -#define SCRATCH_NETEQ_EXPAND 756 -#elif (defined(NETEQ_32KHZ_WIDEBAND)) -#define SCRATCH_NETEQ_EXPAND 504 -#elif (defined(NETEQ_WIDEBAND)) -#define SCRATCH_NETEQ_EXPAND 252 -#else /* NB */ -#define SCRATCH_NETEQ_EXPAND 126 -#endif - -/**************************************************************************** - * WebRtcNetEQ_Normal(...) - * - * This function has the possibility to modify data that is played out in Normal - * mode, for example adjust the gain of the signal. The length of the signal - * can not be changed. - * - * Input: - * - inst : NetEq instance, i.e. the user that requests more - * speech/audio data - * - scratchPtr : Pointer to scratch vector - * - decoded : Pointer to vector of new data from decoder - * (Vector contents may be altered by the function) - * - len : Number of input samples - * - * Output: - * - inst : Updated user information - * - outData : Pointer to a memory space where the output data - * should be stored - * - pw16_len : Pointer to variable where the number of samples - * produced will be written - * - * Return value : >=0 - Number of samples written to outData - * -1 - Error - */ - -int WebRtcNetEQ_Normal(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - int16_t *pw16_decoded, int16_t len, - int16_t *pw16_outData, int16_t *pw16_len) -{ - - int i; - int16_t fs_mult; - int16_t fs_shift; - int32_t w32_En_speech; - int16_t enLen; - int16_t w16_muted; - int16_t w16_inc, w16_frac; - int16_t w16_tmp; - int32_t w32_tmp; - - /* Sanity check */ - if (len < 0) - { - /* Cannot have negative length of input vector */ - return (-1); - } - - if (len == 0) - { - /* Still got some data to play => continue with the same mode */ - *pw16_len = len; - return (len); - } - - fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000); - fs_shift = 30 - WebRtcSpl_NormW32(fs_mult); /* Note that this is not "exact" for 48kHz */ - - /* - * Check if last RecOut call resulted in an Expand or a FadeToBGN. If so, we have to take - * care of some cross-fading and unmuting. - */ - if (inst->w16_mode == MODE_EXPAND || inst->w16_mode == MODE_FADE_TO_BGN) - { - - /* Define memory where temporary result from Expand algorithm can be stored. */ -#ifdef SCRATCH - int16_t *pw16_expanded = pw16_scratchPtr + SCRATCH_PW16_EXPANDED; -#else - int16_t pw16_expanded[FSMULT * 125]; -#endif - int16_t expandedLen = 0; - int16_t w16_decodedMax; - - /* Find largest value in new data */ - w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len); - - /* Generate interpolation data using Expand */ - /* First, set Expand parameters to appropriate values. */ - inst->ExpandInst.w16_lagsPosition = 0; - inst->ExpandInst.w16_lagsDirection = 0; - inst->ExpandInst.w16_stopMuting = 1; /* Do not mute signal any more */ - - /* Call Expand */ - WebRtcNetEQ_Expand(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_expanded, &expandedLen, (int16_t) (inst->w16_mode == MODE_FADE_TO_BGN)); - - inst->ExpandInst.w16_stopMuting = 0; /* Restore value */ - inst->ExpandInst.w16_consecExp = 0; /* Last was not Expand any more */ - - /* Adjust muting factor (main muting factor times expand muting factor) */ - if (inst->w16_mode == MODE_FADE_TO_BGN) - { - /* If last mode was FadeToBGN, the mute factor should be zero. */ - inst->w16_muteFactor = 0; - } - else - { - /* w16_muteFactor * w16_expandMuteFactor */ - inst->w16_muteFactor - = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor, - inst->ExpandInst.w16_expandMuteFactor, 14); - } - - /* Adjust muting factor if needed (to BGN level) */ - enLen = WEBRTC_SPL_MIN(fs_mult<<6, len); /* min( fs_mult * 64, len ) */ - w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32( - WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)); - w16_tmp = WEBRTC_SPL_MAX(w16_tmp, 0); - w32_En_speech = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, enLen, w16_tmp); - w32_En_speech = WebRtcSpl_DivW32W16(w32_En_speech, (int16_t) (enLen >> w16_tmp)); - - if ((w32_En_speech != 0) && (w32_En_speech > inst->BGNInst.w32_energy)) - { - /* Normalize new frame energy to 15 bits */ - w16_tmp = WebRtcSpl_NormW32(w32_En_speech) - 16; - /* we want inst->BGNInst.energy/En_speech in Q14 */ - w32_tmp = WEBRTC_SPL_SHIFT_W32(inst->BGNInst.w32_energy, (w16_tmp+14)); - w16_tmp = (int16_t) WEBRTC_SPL_SHIFT_W32(w32_En_speech, w16_tmp); - w16_tmp = (int16_t) WebRtcSpl_DivW32W16(w32_tmp, w16_tmp); - w16_muted = (int16_t) WebRtcSpl_SqrtFloor( - WEBRTC_SPL_LSHIFT_W32((int32_t) w16_tmp, - 14)); /* w16_muted in Q14 (sqrt(Q28)) */ - } - else - { - w16_muted = 16384; /* 1.0 in Q14 */ - } - if (w16_muted > inst->w16_muteFactor) - { - inst->w16_muteFactor = WEBRTC_SPL_MIN(w16_muted, 16384); - } - - /* If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14) */ - w16_inc = WebRtcSpl_DivW32W16ResW16(64, fs_mult); - for (i = 0; i < len; i++) - { - /* scale with mute factor */ - w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor); - /* shift 14 with proper rounding */ - pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14); - /* increase mute_factor towards 16384 */ - inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc)); - } - - /* - * Interpolate the expanded data into the new vector - * (NB/WB/SWB32/SWB40 8/16/32/32 samples) - */ - fs_shift = WEBRTC_SPL_MIN(3, fs_shift); /* Set to 3 for >32kHz */ - w16_inc = 4 >> fs_shift; - w16_frac = w16_inc; - for (i = 0; i < 8 * fs_mult; i++) - { - pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32( - (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) + - WEBRTC_SPL_MUL_16_16((32 - w16_frac), pw16_expanded[i]) + 8), - 5); - w16_frac += w16_inc; - } - -#ifdef NETEQ_CNG_CODEC - } - else if (inst->w16_mode==MODE_RFC3389CNG) - { /* previous was RFC 3389 CNG...*/ - int16_t pw16_CngInterp[32]; - /* Reset mute factor and start up fresh */ - inst->w16_muteFactor = 16384; - if (inst->CNG_Codec_inst != NULL) - { - /* Generate long enough for 32kHz */ - if(WebRtcCng_Generate(inst->CNG_Codec_inst,pw16_CngInterp, 32, 0)<0) - { - /* error returned; set return vector to all zeros */ - WebRtcSpl_MemSetW16(pw16_CngInterp, 0, 32); - } - } - else - { - /* - * If no CNG instance is defined, just copy from the decoded data. - * (This will result in interpolating the decoded with itself.) - */ - WEBRTC_SPL_MEMCPY_W16(pw16_CngInterp, pw16_decoded, fs_mult * 8); - } - /* - * Interpolate the CNG into the new vector - * (NB/WB/SWB32kHz/SWB48kHz 8/16/32/32 samples) - */ - fs_shift = WEBRTC_SPL_MIN(3, fs_shift); /* Set to 3 for >32kHz */ - w16_inc = 4>>fs_shift; - w16_frac = w16_inc; - for (i = 0; i < 8 * fs_mult; i++) - { - pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32( - (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) + - WEBRTC_SPL_MUL_16_16((32-w16_frac), pw16_CngInterp[i]) + 8), - 5); - w16_frac += w16_inc; - } -#endif - - } - else if (inst->w16_muteFactor < 16384) - { - /* - * Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are still - * ramping up from previous muting. - * If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14) - */ - w16_inc = WebRtcSpl_DivW32W16ResW16(64, fs_mult); - for (i = 0; i < len; i++) - { - /* scale with mute factor */ - w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor); - /* shift 14 with proper rounding */ - pw16_decoded[i] = (int16_t) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14); - /* increase mute_factor towards 16384 */ - inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc)); - } - } - - /* Copy data to other buffer */WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, len); - - inst->w16_mode = MODE_NORMAL; - *pw16_len = len; - return (len); - -} - -#undef SCRATCH_PW16_EXPANDED -#undef SCRATCH_NETEQ_EXPAND - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/normal.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.cc index 8d9c020f96d..bfde179bd17 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/normal.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/normal.h" +#include "webrtc/modules/audio_coding/neteq/normal.h" #include <string.h> // memset, memcpy @@ -16,11 +16,11 @@ #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/normal.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h index fa14685f9bb..aa24b528af4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/normal.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ #include <string.h> // Access to size_t. #include <vector> -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/defines.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/defines.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -65,4 +65,4 @@ class Normal { }; } // namespace webrtc -#endif // SRC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/normal_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc index 2bd7b894f42..c855865cfaa 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/normal_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc @@ -10,16 +10,16 @@ // Unit tests for Normal class. -#include "webrtc/modules/audio_coding/neteq4/normal.h" +#include "webrtc/modules/audio_coding/neteq/normal.h" #include <vector> #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/expand.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/expand.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.h index 4518f91381a..89ddda782cf 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PACKET_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PACKET_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_ #include <list> @@ -85,4 +85,4 @@ struct Packet { typedef std::list<Packet*> PacketList; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PACKET_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c deleted file mode 100644 index a542333cf87..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c +++ /dev/null @@ -1,851 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of the actual packet buffer data structure. - */ - -#include <assert.h> -#include "packet_buffer.h" - -#include <string.h> /* to define NULL */ - -#include "signal_processing_library.h" - -#include "mcu_dsp_common.h" - -#include "neteq_error_codes.h" - -#ifdef NETEQ_DELAY_LOGGING -/* special code for offline delay logging */ -#include "delay_logging.h" -#include <stdio.h> - -extern FILE *delay_fid2; /* file pointer to delay log file */ -extern uint32_t tot_received_packets; -#endif /* NETEQ_DELAY_LOGGING */ - - -int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets, - int16_t *pw16_memory, int memorySize) -{ - int i; - int pos = 0; - - /* Sanity check */ - if ((memorySize < PBUFFER_MIN_MEMORY_SIZE) || (pw16_memory == NULL) - || (maxNoOfPackets < 2) || (maxNoOfPackets > 600)) - { - /* Invalid parameters */ - return (PBUFFER_INIT_ERROR); - } - - /* Clear the buffer instance */ - WebRtcSpl_MemSetW16((int16_t*) bufferInst, 0, - sizeof(PacketBuf_t) / sizeof(int16_t)); - - /* Clear the buffer memory */ - WebRtcSpl_MemSetW16((int16_t*) pw16_memory, 0, memorySize); - - /* Set maximum number of packets */ - bufferInst->maxInsertPositions = maxNoOfPackets; - - /* Initialize array pointers */ - /* After each pointer has been set, the index pos is advanced to point immediately - * after the the recently allocated vector. Note that one step for the pos index - * corresponds to a int16_t. - */ - - bufferInst->timeStamp = (uint32_t*) &pw16_memory[pos]; - pos += maxNoOfPackets << 1; /* advance maxNoOfPackets * uint32_t */ - - bufferInst->payloadLocation = (int16_t**) &pw16_memory[pos]; - pos += maxNoOfPackets * (sizeof(int16_t*) / sizeof(int16_t)); /* advance */ - - bufferInst->seqNumber = (uint16_t*) &pw16_memory[pos]; - pos += maxNoOfPackets; /* advance maxNoOfPackets * uint16_t */ - - bufferInst->payloadType = &pw16_memory[pos]; - pos += maxNoOfPackets; /* advance maxNoOfPackets * int16_t */ - - bufferInst->payloadLengthBytes = &pw16_memory[pos]; - pos += maxNoOfPackets; /* advance maxNoOfPackets * int16_t */ - - bufferInst->rcuPlCntr = &pw16_memory[pos]; - pos += maxNoOfPackets; /* advance maxNoOfPackets * int16_t */ - - bufferInst->waitingTime = (int*) (&pw16_memory[pos]); - /* Advance maxNoOfPackets * sizeof(waitingTime element). */ - pos += maxNoOfPackets * - sizeof(*bufferInst->waitingTime) / sizeof(*pw16_memory); - - /* The payload memory starts after the slot arrays */ - bufferInst->startPayloadMemory = &pw16_memory[pos]; - bufferInst->currentMemoryPos = bufferInst->startPayloadMemory; - bufferInst->memorySizeW16 = (memorySize - pos); /* Remaining memory */ - - /* Initialize each payload slot as empty with infinite delay */ - for (i = 0; i < bufferInst->maxInsertPositions; i++) - { - bufferInst->payloadType[i] = -1; - } - - /* Reset buffer parameters */ - bufferInst->numPacketsInBuffer = 0; - bufferInst->packSizeSamples = 0; - bufferInst->insertPosition = 0; - - /* Reset buffer statistics */ - bufferInst->discardedPackets = 0; - - return (0); -} - - -int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst) -{ - int i; - - /* Sanity check */ - if (bufferInst->startPayloadMemory == NULL) - { - /* Packet buffer has not been initialized */ - /* Don't do the flushing operation, since we do not - know the state of the struct variables */ - return (0); - } - - /* Set all payload lengths to zero */ - WebRtcSpl_MemSetW16(bufferInst->payloadLengthBytes, 0, bufferInst->maxInsertPositions); - - /* Reset buffer variables */ - bufferInst->numPacketsInBuffer = 0; - bufferInst->currentMemoryPos = bufferInst->startPayloadMemory; - bufferInst->insertPosition = 0; - - /* Clear all slots, starting with the last one */ - for (i = (bufferInst->maxInsertPositions - 1); i >= 0; i--) - { - bufferInst->payloadType[i] = -1; - bufferInst->timeStamp[i] = 0; - bufferInst->seqNumber[i] = 0; - } - - return (0); -} - - -int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket, - int16_t *flushed, int av_sync) -{ - int nextPos; - int i; - -#ifdef NETEQ_DELAY_LOGGING - /* special code for offline delay logging */ - int temp_var; -#endif /* NETEQ_DELAY_LOGGING */ - - /* Initialize to "no flush" */ - *flushed = 0; - - /* Sanity check */ - if (bufferInst->startPayloadMemory == NULL) - { - /* packet buffer has not been initialized */ - return (-1); - } - - /* Sanity check for payload length - (payloadLen in bytes and memory size in int16_t) */ - if ((RTPpacket->payloadLen > (bufferInst->memorySizeW16 << 1)) || (RTPpacket->payloadLen - <= 0)) - { - /* faulty or too long payload length */ - return (-1); - } - - /* If we are in AV-sync mode, there is a risk that we have inserted a sync - * packet but now received the real version of it. Or because of some timing - * we might be overwriting a true payload with sync (I'm not sure why this - * should happen in regular case, but in some FEC enabled case happens). - * Go through packets and delete the sync version of the packet in hand. Or - * if this is sync packet and the regular version of it exists in the buffer - * refrain from inserting. - * - * TODO(turajs): Could we get this for free if we had set the RCU-counter of - * the sync packet to a number larger than 2? - */ - if (av_sync) { - for (i = 0; i < bufferInst->maxInsertPositions; ++i) { - /* Check if sequence numbers match and the payload actually exists. */ - if (bufferInst->seqNumber[i] == RTPpacket->seqNumber && - bufferInst->payloadLengthBytes[i] > 0) { - if (WebRtcNetEQ_IsSyncPayload(RTPpacket->payload, - RTPpacket->payloadLen)) { - return 0; - } - - if (WebRtcNetEQ_IsSyncPayload(bufferInst->payloadLocation[i], - bufferInst->payloadLengthBytes[i])) { - /* Clear the position in the buffer. */ - bufferInst->payloadType[i] = -1; - bufferInst->payloadLengthBytes[i] = 0; - - /* Reduce packet counter by one. */ - bufferInst->numPacketsInBuffer--; - /* TODO(turajs) if this is the latest packet better we rewind - * insertPosition and related variables. */ - break; /* There should be only one match. */ - } - } - } - } - - /* Find a position in the buffer for this packet */ - if (bufferInst->numPacketsInBuffer != 0) - { - /* Get the next slot */ - bufferInst->insertPosition++; - if (bufferInst->insertPosition >= bufferInst->maxInsertPositions) - { - /* "Wrap around" and start from the beginning */ - bufferInst->insertPosition = 0; - } - - /* Check if there is enough space for the new packet */ - if (bufferInst->currentMemoryPos + ((RTPpacket->payloadLen + 1) >> 1) - >= &bufferInst->startPayloadMemory[bufferInst->memorySizeW16]) - { - int16_t *tempMemAddress; - - /* - * Payload does not fit at the end of the memory, put it in the beginning - * instead - */ - bufferInst->currentMemoryPos = bufferInst->startPayloadMemory; - - /* - * Now, we must search for the next non-empty payload, - * finding the one with the lowest start address for the payload - */ - tempMemAddress = &bufferInst->startPayloadMemory[bufferInst->memorySizeW16]; - nextPos = -1; - - /* Loop through all slots again */ - for (i = 0; i < bufferInst->maxInsertPositions; i++) - { - /* Look for the non-empty slot with the lowest - payload location address */ - if (bufferInst->payloadLengthBytes[i] != 0 && bufferInst->payloadLocation[i] - < tempMemAddress) - { - tempMemAddress = bufferInst->payloadLocation[i]; - nextPos = i; - } - } - - /* Check that we did find a previous payload */ - if (nextPos == -1) - { - /* The buffer is corrupt => flush and return error */ - WebRtcNetEQ_PacketBufferFlush(bufferInst); - *flushed = 1; - return (-1); - } - } - else - { - /* Payload fits at the end of memory. */ - - /* Find the next non-empty slot. */ - nextPos = bufferInst->insertPosition + 1; - - /* Increase nextPos until a non-empty slot is found or end of array is encountered*/ - while ((bufferInst->payloadLengthBytes[nextPos] == 0) && (nextPos - < bufferInst->maxInsertPositions)) - { - nextPos++; - } - - if (nextPos == bufferInst->maxInsertPositions) - { - /* - * Reached the end of the array, so there must be a packet in the first - * position instead - */ - nextPos = 0; - - /* Increase nextPos until a non-empty slot is found */ - while (bufferInst->payloadLengthBytes[nextPos] == 0) - { - nextPos++; - } - } - } /* end if-else */ - - /* - * Check if the new payload will extend into a payload later in memory. - * If so, the buffer is full. - */ - if ((bufferInst->currentMemoryPos <= bufferInst->payloadLocation[nextPos]) - && ((&bufferInst->currentMemoryPos[(RTPpacket->payloadLen + 1) >> 1]) - > bufferInst->payloadLocation[nextPos])) - { - /* Buffer is full, so the buffer must be flushed */ - WebRtcNetEQ_PacketBufferFlush(bufferInst); - *flushed = 1; - } - - if (bufferInst->payloadLengthBytes[bufferInst->insertPosition] != 0) - { - /* All positions are already taken and entire buffer should be flushed */ - WebRtcNetEQ_PacketBufferFlush(bufferInst); - *flushed = 1; - } - - } - else - { - /* Buffer is empty, just insert the packet at the beginning */ - bufferInst->currentMemoryPos = bufferInst->startPayloadMemory; - bufferInst->insertPosition = 0; - } - - /* Insert packet in the found position */ - if (RTPpacket->starts_byte1 == 0) - { - /* Payload is 16-bit aligned => just copy it */ - - WEBRTC_SPL_MEMCPY_W8(bufferInst->currentMemoryPos, - RTPpacket->payload, RTPpacket->payloadLen); - } - else - { - /* Payload is not 16-bit aligned => align it during copy operation */ - for (i = 0; i < RTPpacket->payloadLen; i++) - { - /* copy the (i+1)-th byte to the i-th byte */ - - WEBRTC_SPL_SET_BYTE(bufferInst->currentMemoryPos, - (WEBRTC_SPL_GET_BYTE(RTPpacket->payload, (i + 1))), i); - } - } - - /* Copy the packet information */ - bufferInst->payloadLocation[bufferInst->insertPosition] = bufferInst->currentMemoryPos; - bufferInst->payloadLengthBytes[bufferInst->insertPosition] = RTPpacket->payloadLen; - bufferInst->payloadType[bufferInst->insertPosition] = RTPpacket->payloadType; - bufferInst->seqNumber[bufferInst->insertPosition] = RTPpacket->seqNumber; - bufferInst->timeStamp[bufferInst->insertPosition] = RTPpacket->timeStamp; - bufferInst->rcuPlCntr[bufferInst->insertPosition] = RTPpacket->rcuPlCntr; - bufferInst->waitingTime[bufferInst->insertPosition] = 0; - /* Update buffer parameters */ - bufferInst->numPacketsInBuffer++; - bufferInst->currentMemoryPos += (RTPpacket->payloadLen + 1) >> 1; - -#ifdef NETEQ_DELAY_LOGGING - /* special code for offline delay logging */ - if (*flushed) - { - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_FLUSH; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - } - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_RECIN; - if ((fwrite(&temp_var, sizeof(int), - 1, delay_fid2) != 1) || - (fwrite(&RTPpacket->timeStamp, sizeof(uint32_t), - 1, delay_fid2) != 1) || - (fwrite(&RTPpacket->seqNumber, sizeof(uint16_t), - 1, delay_fid2) != 1) || - (fwrite(&RTPpacket->payloadType, sizeof(int), - 1, delay_fid2) != 1) || - (fwrite(&RTPpacket->payloadLen, sizeof(int16_t), - 1, delay_fid2) != 1)) { - return -1; - } - tot_received_packets++; -#endif /* NETEQ_DELAY_LOGGING */ - - return (0); -} - - -int WebRtcNetEQ_PacketBufferExtract(PacketBuf_t *bufferInst, RTPPacket_t *RTPpacket, - int bufferPosition, int *waitingTime) -{ - - /* Sanity check */ - if (bufferInst->startPayloadMemory == NULL) - { - /* packet buffer has not been initialized */ - return (PBUFFER_NOT_INITIALIZED); - } - - if (bufferPosition < 0 || bufferPosition >= bufferInst->maxInsertPositions) - { - /* buffer position is outside valid range */ - return (NETEQ_OTHER_ERROR); - } - - /* Check that there is a valid payload in the specified position */ - if (bufferInst->payloadLengthBytes[bufferPosition] <= 0) - { - /* The position does not contain a valid payload */ - RTPpacket->payloadLen = 0; /* Set zero length */ - return (PBUFFER_NONEXISTING_PACKET); /* Return error */ - } - - /* Payload exists => extract payload data */ - - /* Copy the actual data payload to RTP packet struct */ - - WEBRTC_SPL_MEMCPY_W16((int16_t*) RTPpacket->payload, - bufferInst->payloadLocation[bufferPosition], - (bufferInst->payloadLengthBytes[bufferPosition] + 1) >> 1); /*length in int16_t*/ - - /* Copy payload parameters */ - RTPpacket->payloadLen = bufferInst->payloadLengthBytes[bufferPosition]; - RTPpacket->payloadType = bufferInst->payloadType[bufferPosition]; - RTPpacket->seqNumber = bufferInst->seqNumber[bufferPosition]; - RTPpacket->timeStamp = bufferInst->timeStamp[bufferPosition]; - RTPpacket->rcuPlCntr = bufferInst->rcuPlCntr[bufferPosition]; - *waitingTime = bufferInst->waitingTime[bufferPosition]; - RTPpacket->starts_byte1 = 0; /* payload is 16-bit aligned */ - - /* Clear the position in the packet buffer */ - bufferInst->payloadType[bufferPosition] = -1; - bufferInst->payloadLengthBytes[bufferPosition] = 0; - bufferInst->seqNumber[bufferPosition] = 0; - bufferInst->timeStamp[bufferPosition] = 0; - bufferInst->waitingTime[bufferPosition] = 0; - bufferInst->payloadLocation[bufferPosition] = bufferInst->startPayloadMemory; - - /* Reduce packet counter with one */ - bufferInst->numPacketsInBuffer--; - - return (0); -} - -int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t* buffer_inst, - uint32_t current_time_stamp, - uint32_t* time_stamp, - int* buffer_position, - int erase_old_packets, - int16_t* payload_type) { - int32_t time_stamp_diff = WEBRTC_SPL_WORD32_MAX; /* Smallest diff found. */ - int32_t new_diff; - int i; - int16_t rcu_payload_cntr; - if (buffer_inst->startPayloadMemory == NULL) { - /* Packet buffer has not been initialized. */ - return PBUFFER_NOT_INITIALIZED; - } - - /* Initialize all return values. */ - *time_stamp = 0; - *payload_type = -1; /* Indicates that no packet was found. */ - *buffer_position = -1; /* Indicates that no packet was found. */ - rcu_payload_cntr = WEBRTC_SPL_WORD16_MAX; /* Indicates no packet found. */ - - /* Check if buffer is empty. */ - if (buffer_inst->numPacketsInBuffer <= 0) { - return 0; - } - - /* Loop through all slots in buffer. */ - if (erase_old_packets) { /* If old payloads should be discarded. */ - for (i = 0; i < buffer_inst->maxInsertPositions; ++i) { - /* Calculate difference between this slot and current_time_stamp. */ - new_diff = (int32_t)(buffer_inst->timeStamp[i] - current_time_stamp); - - /* Check if payload should be discarded. */ - if ((new_diff < 0) /* Payload is too old */ - && (new_diff > -30000) /* Account for TS wrap-around. */ - && (buffer_inst->payloadLengthBytes[i] > 0)) { /* Payload exists. */ - /* Throw away old packet. */ - - /* Clear the position in the buffer. */ - buffer_inst->payloadType[i] = -1; - buffer_inst->payloadLengthBytes[i] = 0; - - /* Reduce packet counter by one. */ - buffer_inst->numPacketsInBuffer--; - /* Increase discard counter for in-call statistics. */ - buffer_inst->discardedPackets++; - } else if (((new_diff < time_stamp_diff) - || ((new_diff == time_stamp_diff) - && (buffer_inst->rcuPlCntr[i] < rcu_payload_cntr))) - && (buffer_inst->payloadLengthBytes[i] > 0)) { - /* New diff is smaller than previous diffs or we have a candidate with a - * time stamp as previous candidate but better RCU-counter; - * and the payload exists. - */ - /* Save this position as the best candidate. */ - *buffer_position = i; - time_stamp_diff = new_diff; - *payload_type = buffer_inst->payloadType[i]; - rcu_payload_cntr = buffer_inst->rcuPlCntr[i]; - } - } - } else { - for (i = 0; i < buffer_inst->maxInsertPositions; ++i) { - /* Calculate difference between this slot and current_time_stamp. */ - new_diff = (int32_t)(buffer_inst->timeStamp[i] - current_time_stamp); - - /* Check if this is the oldest packet. */ - if (((new_diff < time_stamp_diff) - || ((new_diff == time_stamp_diff) - && (buffer_inst->rcuPlCntr[i] < rcu_payload_cntr))) - && (buffer_inst->payloadLengthBytes[i] > 0)) { - /* New diff is smaller than previous diffs or we have a candidate with a - * time_stamp as previous candidate but better RCU-counter; - * and the payload exists. - */ - /* Save this position as the best candidate. */ - *buffer_position = i; - time_stamp_diff = new_diff; - *payload_type = buffer_inst->payloadType[i]; - rcu_payload_cntr = buffer_inst->rcuPlCntr[i]; - } - } - } - - /* Check that we did find a real position. */ - if (*buffer_position >= 0) { - /* Get the time_stamp for the best position. */ - *time_stamp = buffer_inst->timeStamp[*buffer_position]; - } - - return 0; -} - -int WebRtcNetEQ_PacketBufferGetPacketSize(const PacketBuf_t* buffer_inst, - int buffer_pos, - const CodecDbInst_t* codec_database, - int codec_pos, int last_duration, - int av_sync) { - if (codec_database->funcDurationEst[codec_pos] == NULL) { - return last_duration; - } - - if (av_sync != 0 && - WebRtcNetEQ_IsSyncPayload(buffer_inst->payloadLocation[buffer_pos], - buffer_inst->payloadLengthBytes[buffer_pos])) { - // In AV-sync and sync payload, report |last_duration| as current duration. - return last_duration; - } - - return (*codec_database->funcDurationEst[codec_pos])( - codec_database->codec_state[codec_pos], - (const uint8_t *)buffer_inst->payloadLocation[buffer_pos], - buffer_inst->payloadLengthBytes[buffer_pos]); -} - -int32_t WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t* buffer_inst, - const CodecDbInst_t* codec_database, - int av_sync) { - int i, count; - int last_duration; - int last_codec_pos; - int last_payload_type; - int32_t size_samples; - - count = 0; - last_duration = buffer_inst->packSizeSamples; - last_codec_pos = -1; - last_payload_type = -1; - size_samples = 0; - - /* Loop through all slots in the buffer */ - for (i = 0; i < buffer_inst->maxInsertPositions; i++) { - /* Only count the packets with non-zero size */ - if (buffer_inst->payloadLengthBytes[i] != 0) { - int payload_type; - int codec_pos; - /* Figure out the codec database entry for this payload_type. */ - payload_type = buffer_inst->payloadType[i]; - /* Remember the last one, to avoid the database search. */ - if(payload_type == last_payload_type) { - codec_pos = last_codec_pos; - } - else { - codec_pos = WebRtcNetEQ_DbGetCodec(codec_database, - payload_type); - if (codec_pos >= 0) { - codec_pos = codec_database->position[codec_pos]; - } - } - last_codec_pos = codec_pos; - last_payload_type = payload_type; - if (codec_pos >= 0) { - /* - * Right now WebRtcNetEQ_PacketBufferGetPacketSize either always - * returns last_duration or always computes the real duration without - * looking at last_duration. If an implementation really wanted to use - * last_duration to compute a changing duration, we would have to - * iterate through the packets in chronological order by timestamp. - */ - /* Check for error before setting. */ - int temp_last_duration = WebRtcNetEQ_PacketBufferGetPacketSize( - buffer_inst, i, codec_database, codec_pos, - last_duration, av_sync); - if (temp_last_duration >= 0) - last_duration = temp_last_duration; - } - /* Add in the size of this packet. */ - size_samples += last_duration; - count++; - } - } - - /* Sanity check; size cannot be negative */ - if (size_samples < 0) { - size_samples = 0; - } - return size_samples; -} - -void WebRtcNetEQ_IncrementWaitingTimes(PacketBuf_t *buffer_inst) { - int i; - /* Loop through all slots in the buffer. */ - for (i = 0; i < buffer_inst->maxInsertPositions; ++i) { - /* Only increment waiting time for the packets with non-zero size. */ - if (buffer_inst->payloadLengthBytes[i] != 0) { - buffer_inst->waitingTime[i]++; - } - } -} - -int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID, - int noOfCodecs, int *maxBytes, - int *maxSlots, - int* per_slot_overhead_bytes) -{ - int i; - int ok = 0; - int16_t w16_tmp; - int16_t codecBytes; - int16_t codecBuffers; - - /* Initialize return variables to zero */ - *maxBytes = 0; - *maxSlots = 0; - - /* Loop through all codecs supplied to function */ - for (i = 0; i < noOfCodecs; i++) - { - /* Find current codec and set parameters accordingly */ - - if ((codecID[i] == kDecoderPCMu) || (codecID[i] == kDecoderPCMu_2ch)) - { - codecBytes = 1680; /* Up to 210ms @ 64kbps */ - codecBuffers = 30; /* Down to 5ms frames */ - } - else if ((codecID[i] == kDecoderPCMa) || - (codecID[i] == kDecoderPCMa_2ch)) - { - codecBytes = 1680; /* Up to 210ms @ 64kbps */ - codecBuffers = 30; /* Down to 5ms frames */ - } - else if (codecID[i] == kDecoderILBC) - { - codecBytes = 380; /* 200ms @ 15.2kbps (20ms frames) */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderISAC) - { - codecBytes = 960; /* 240ms @ 32kbps (60ms frames) */ - codecBuffers = 8; - } - else if ((codecID[i] == kDecoderISACswb) || - (codecID[i] == kDecoderISACfb)) - { - codecBytes = 1560; /* 240ms @ 52kbps (30ms frames) */ - codecBuffers = 8; - } - else if (codecID[i] == kDecoderOpus) - { - codecBytes = 15300; /* 240ms @ 510kbps (60ms frames) */ - codecBuffers = 30; /* Replicating the value for PCMu/a */ - } - else if ((codecID[i] == kDecoderPCM16B) || - (codecID[i] == kDecoderPCM16B_2ch)) - { - codecBytes = 3360; /* 210ms */ - codecBuffers = 15; - } - else if ((codecID[i] == kDecoderPCM16Bwb) || - (codecID[i] == kDecoderPCM16Bwb_2ch)) - { - codecBytes = 6720; /* 210ms */ - codecBuffers = 15; - } - else if ((codecID[i] == kDecoderPCM16Bswb32kHz) || - (codecID[i] == kDecoderPCM16Bswb32kHz_2ch)) - { - codecBytes = 13440; /* 210ms */ - codecBuffers = 15; - } - else if (codecID[i] == kDecoderPCM16Bswb48kHz) - { - codecBytes = 20160; /* 210ms */ - codecBuffers = 15; - } - else if ((codecID[i] == kDecoderG722) || - (codecID[i] == kDecoderG722_2ch)) - { - codecBytes = 1680; /* 210ms @ 64kbps */ - codecBuffers = 15; - } - else if (codecID[i] == kDecoderRED) - { - codecBytes = 0; /* Should not be max... */ - codecBuffers = 0; - } - else if (codecID[i] == kDecoderAVT) - { - codecBytes = 0; /* Should not be max... */ - codecBuffers = 0; - } - else if (codecID[i] == kDecoderCNG) - { - codecBytes = 0; /* Should not be max... */ - codecBuffers = 0; - } - else if (codecID[i] == kDecoderG729) - { - codecBytes = 210; /* 210ms @ 8kbps */ - codecBuffers = 20; /* max 200ms supported for 10ms frames */ - } - else if (codecID[i] == kDecoderG729_1) - { - codecBytes = 840; /* 210ms @ 32kbps */ - codecBuffers = 10; /* max 200ms supported for 20ms frames */ - } - else if (codecID[i] == kDecoderG726_16) - { - codecBytes = 400; /* 200ms @ 16kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG726_24) - { - codecBytes = 600; /* 200ms @ 24kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG726_32) - { - codecBytes = 800; /* 200ms @ 32kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG726_40) - { - codecBytes = 1000; /* 200ms @ 40kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG722_1_16) - { - codecBytes = 420; /* 210ms @ 16kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG722_1_24) - { - codecBytes = 630; /* 210ms @ 24kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG722_1_32) - { - codecBytes = 840; /* 210ms @ 32kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG722_1C_24) - { - codecBytes = 630; /* 210ms @ 24kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG722_1C_32) - { - codecBytes = 840; /* 210ms @ 32kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderG722_1C_48) - { - codecBytes = 1260; /* 210ms @ 48kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderSPEEX_8) - { - codecBytes = 1250; /* 210ms @ 50kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderSPEEX_16) - { - codecBytes = 1250; /* 210ms @ 50kbps */ - codecBuffers = 10; - } - else if ((codecID[i] == kDecoderCELT_32) || - (codecID[i] == kDecoderCELT_32_2ch)) - { - codecBytes = 1250; /* 210ms @ 50kbps */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderGSMFR) - { - codecBytes = 340; /* 200ms */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderAMR) - { - codecBytes = 384; /* 240ms @ 12.2kbps+headers (60ms frames) */ - codecBuffers = 10; - } - else if (codecID[i] == kDecoderAMRWB) - { - codecBytes = 744; - codecBuffers = 10; - } - else if (codecID[i] == kDecoderArbitrary) - { - codecBytes = 6720; /* Assume worst case uncompressed WB 210ms */ - codecBuffers = 15; - } - else - { - /*Unknow codec */ - codecBytes = 0; - codecBuffers = 0; - ok = CODEC_DB_UNKNOWN_CODEC; - } - - /* Update max variables */ - *maxBytes = WEBRTC_SPL_MAX((*maxBytes), codecBytes); - *maxSlots = WEBRTC_SPL_MAX((*maxSlots), codecBuffers); - - } /* end of for loop */ - - /* - * Add size needed by the additional pointers for each slot inside struct, - * as indicated on each line below. - */ - w16_tmp = (sizeof(uint32_t) /* timeStamp */ - + sizeof(int16_t*) /* payloadLocation */ - + sizeof(uint16_t) /* seqNumber */ - + sizeof(int16_t) /* payloadType */ - + sizeof(int16_t) /* payloadLengthBytes */ - + sizeof(int16_t) /* rcuPlCntr */ - + sizeof(int)); /* waitingTime */ - /* Add the extra size per slot to the memory count */ - *maxBytes += w16_tmp * (*maxSlots); - - *per_slot_overhead_bytes = w16_tmp; - return ok; -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.cc index d19abbaa84d..8a81c2598bc 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.cc @@ -12,12 +12,12 @@ // an STL list. The list is kept sorted at all times so that the next packet to // decode is at the beginning of the list. -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" #include <algorithm> // find_if() -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" namespace webrtc { @@ -36,14 +36,8 @@ class NewTimestampIsLarger { const Packet* new_packet_; }; -// Constructor. The arguments define the maximum number of slots and maximum -// payload memory (excluding RTP headers) that the buffer will accept. -PacketBuffer::PacketBuffer(size_t max_number_of_packets, - size_t max_memory_bytes) - : max_number_of_packets_(max_number_of_packets), - max_memory_bytes_(max_memory_bytes), - current_memory_bytes_(0) { -} +PacketBuffer::PacketBuffer(size_t max_number_of_packets) + : max_number_of_packets_(max_number_of_packets) {} // Destructor. All packets in the buffer will be destroyed. PacketBuffer::~PacketBuffer() { @@ -53,7 +47,6 @@ PacketBuffer::~PacketBuffer() { // Flush the buffer. All packets in the buffer will be destroyed. void PacketBuffer::Flush() { DeleteAllPackets(&buffer_); - current_memory_bytes_ = 0; } int PacketBuffer::InsertPacket(Packet* packet) { @@ -66,22 +59,10 @@ int PacketBuffer::InsertPacket(Packet* packet) { int return_val = kOK; - if ((buffer_.size() >= max_number_of_packets_) || - (current_memory_bytes_ + packet->payload_length - > static_cast<int>(max_memory_bytes_))) { + if (buffer_.size() >= max_number_of_packets_) { // Buffer is full. Flush it. Flush(); return_val = kFlushed; - if ((buffer_.size() >= max_number_of_packets_) || - (current_memory_bytes_ + packet->payload_length - > static_cast<int>(max_memory_bytes_))) { - // Buffer is still too small for the packet. Either the buffer limits are - // really small, or the packet is really large. Delete the packet and - // return an error. - delete [] packet->payload; - delete packet; - return kOversizePacket; - } } // Get an iterator pointing to the place in the buffer where the new packet @@ -91,7 +72,6 @@ int PacketBuffer::InsertPacket(Packet* packet) { buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet)); buffer_.insert(rit.base(), packet); // Insert the packet at that position. - current_memory_bytes_ += packet->payload_length; return return_val; } @@ -183,8 +163,6 @@ Packet* PacketBuffer::GetNextPacket(int* discard_count) { // Assert that the packet sanity checks in InsertPacket method works. assert(packet && packet->payload); buffer_.pop_front(); - current_memory_bytes_ -= packet->payload_length; - assert(current_memory_bytes_ >= 0); // Assert bookkeeping is correct. // Discard other packets with the same timestamp. These are duplicates or // redundant payloads that should not be used. if (discard_count) { @@ -206,17 +184,14 @@ int PacketBuffer::DiscardNextPacket() { if (Empty()) { return kBufferEmpty; } - Packet* temp_packet = buffer_.front(); // Assert that the packet sanity checks in InsertPacket method works. - assert(temp_packet && temp_packet->payload); - current_memory_bytes_ -= temp_packet->payload_length; - assert(current_memory_bytes_ >= 0); // Assert bookkeeping is correct. + assert(buffer_.front()); + assert(buffer_.front()->payload); DeleteFirstPacket(&buffer_); return kOK; } int PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit) { - int discard_count = 0; while (!Empty() && timestamp_limit != buffer_.front()->header.timestamp && static_cast<uint32_t>(timestamp_limit @@ -225,7 +200,6 @@ int PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit) { if (DiscardNextPacket() != kOK) { assert(false); // Must be ok by design. } - ++discard_count; } return 0; } @@ -240,8 +214,15 @@ int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database, AudioDecoder* decoder = decoder_database->GetDecoder(packet->header.payloadType); if (decoder) { - int duration = packet->sync_packet ? last_duration : - decoder->PacketDuration(packet->payload, packet->payload_length); + int duration; + if (packet->sync_packet) { + duration = last_duration; + } else if (packet->primary) { + duration = + decoder->PacketDuration(packet->payload, packet->payload_length); + } else { + continue; + } if (duration >= 0) { last_duration = duration; // Save the most up-to-date (valid) duration. } @@ -275,14 +256,9 @@ void PacketBuffer::DeleteAllPackets(PacketList* packet_list) { } } -void PacketBuffer::BufferStat(int* num_packets, - int* max_num_packets, - int* current_memory_bytes, - int* max_memory_bytes) const { +void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const { *num_packets = static_cast<int>(buffer_.size()); *max_num_packets = static_cast<int>(max_number_of_packets_); - *current_memory_bytes = current_memory_bytes_; - *max_memory_bytes = static_cast<int>(max_memory_bytes_); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h index 61ff2b970fd..76c4ddd161d 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -8,247 +8,128 @@ * be found in the AUTHORS file in the root of the source tree. */ -/* - * Interface for the actual packet buffer data structure. - */ - -#ifndef PACKET_BUFFER_H -#define PACKET_BUFFER_H - -#include "typedefs.h" - -#include "webrtc_neteq.h" -#include "rtp.h" - -/* Define minimum allowed buffer memory, in 16-bit words */ -#define PBUFFER_MIN_MEMORY_SIZE 150 - -/****************************/ -/* The packet buffer struct */ -/****************************/ - -typedef struct -{ - - /* Variables common to the entire buffer */ - uint16_t packSizeSamples; /* packet size in samples of last decoded packet */ - int16_t *startPayloadMemory; /* pointer to the payload memory */ - int memorySizeW16; /* the size (in int16_t) of the payload memory */ - int16_t *currentMemoryPos; /* The memory position to insert next payload */ - int numPacketsInBuffer; /* The number of packets in the buffer */ - int insertPosition; /* The position to insert next packet */ - int maxInsertPositions; /* Maximum number of packets allowed */ - - /* Arrays with one entry per packet slot */ - /* NOTE: If these are changed, the changes must be accounted for at the end of - the function WebRtcNetEQ_GetDefaultCodecSettings(). */ - uint32_t *timeStamp; /* Timestamp in slot n */ - int16_t **payloadLocation; /* Memory location of payload in slot n */ - uint16_t *seqNumber; /* Sequence number in slot n */ - int16_t *payloadType; /* Payload type of packet in slot n */ - int16_t *payloadLengthBytes; /* Payload length of packet in slot n */ - int16_t *rcuPlCntr; /* zero for non-RCU payload, 1 for main payload - 2 for redundant payload */ - int *waitingTime; - - /* Statistics counter */ - uint16_t discardedPackets; /* Number of discarded packets */ - -} PacketBuf_t; - -/*************************/ -/* Function declarations */ -/*************************/ - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferInit(...) - * - * This function initializes the packet buffer. - * - * Input: - * - bufferInst : Buffer instance to be initialized - * - noOfPackets : Maximum number of packets that buffer should hold - * - memory : Pointer to the storage memory for the payloads - * - memorySize : The size of the payload memory (in int16_t) - * - * Output: - * - bufferInst : Updated buffer instance - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets, - int16_t *pw16_memory, int memorySize); - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferFlush(...) - * - * This function flushes all the packets in the buffer. - * - * Input: - * - bufferInst : Buffer instance to be flushed - * - * Output: - * - bufferInst : Flushed buffer instance - * - * Return value : 0 - Ok - */ - -int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst); - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferInsert(...) - * - * This function inserts an RTP packet into the packet buffer. - * - * Input: - * - bufferInst : Buffer instance - * - RTPpacket : An RTP packet struct (with payload, sequence - * number, etc.) - * - av_sync : 1 indicates AV-sync enabled, 0 disabled. - * - * Output: - * - bufferInst : Updated buffer instance - * - flushed : 1 if buffer was flushed, 0 otherwise - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket, - int16_t *flushed, int av_sync); - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferExtract(...) - * - * This function extracts a payload from the buffer. - * - * Input: - * - bufferInst : Buffer instance - * - bufferPosition: Position of the packet that should be extracted - * - * Output: - * - RTPpacket : An RTP packet struct (with payload, sequence - * number, etc) - * - bufferInst : Updated buffer instance - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_PacketBufferExtract(PacketBuf_t *bufferInst, RTPPacket_t *RTPpacket, - int bufferPosition, int *waitingTime); - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferFindLowestTimestamp(...) - * - * This function finds the next packet with the lowest timestamp. - * - * Input: - * - buffer_inst : Buffer instance. - * - current_time_stamp : The timestamp to compare packet timestamps with. - * - erase_old_packets : If non-zero, erase packets older than currentTS. - * - * Output: - * - time_stamp : Lowest timestamp that was found. - * - buffer_position : Position of this packet (-1 if there are no - * packets in the buffer). - * - payload_type : Payload type of the found payload. - * - * Return value : 0 - Ok; - * < 0 - Error. - */ - -int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t* buffer_inst, - uint32_t current_time_stamp, - uint32_t* time_stamp, - int* buffer_position, - int erase_old_packets, - int16_t* payload_type); - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferGetPacketSize(...) - * - * Calculate and return an estimate of the data length (in samples) of the - * given packet. If no estimate is available (because we do not know how to - * compute packet durations for the associated payload type), last_duration - * will be returned instead. - * - * Input: - * - buffer_inst : Buffer instance - * - buffer_pos : The index of the buffer of which to estimate the - * duration - * - codec_database : Codec database instance - * - codec_pos : The codec database entry associated with the payload - * type of the specified buffer. - * - last_duration : The duration of the previous frame. - * - av_sync : 1 indicates AV-sync enabled, 0 disabled. - * - * Return value : The buffer size in samples - */ - -int WebRtcNetEQ_PacketBufferGetPacketSize(const PacketBuf_t* buffer_inst, - int buffer_pos, - const CodecDbInst_t* codec_database, - int codec_pos, int last_duration, - int av_sync); - -/**************************************************************************** - * WebRtcNetEQ_PacketBufferGetSize(...) - * - * Calculate and return an estimate of the total data length (in samples) - * currently in the buffer. The estimate is calculated as the number of - * packets currently in the buffer (which does not have any remaining waiting - * time), multiplied with the number of samples obtained from the last - * decoded packet. - * - * Input: - * - buffer_inst : Buffer instance - * - codec_database : Codec database instance - * - av_sync : 1 indicates AV-sync enabled, 0 disabled. - * - * Return value : The buffer size in samples - */ - -int32_t WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t* buffer_inst, - const CodecDbInst_t* codec_database, - int av_sync); - -/**************************************************************************** - * WebRtcNetEQ_IncrementWaitingTimes(...) - * - * Increment the waiting time for all packets in the buffer by one. - * - * Input: - * - bufferInst : Buffer instance - * - * Return value : n/a - */ - -void WebRtcNetEQ_IncrementWaitingTimes(PacketBuf_t *buffer_inst); - -/**************************************************************************** - * WebRtcNetEQ_GetDefaultCodecSettings(...) - * - * Calculates a recommended buffer size for a specific set of codecs. - * - * Input: - * - codecID : An array of codec types that will be used - * - noOfCodecs : Number of codecs in array codecID - * - * Output: - * - maxBytes : Recommended buffer memory size in bytes - * - maxSlots : Recommended number of slots in buffer - * - per_slot_overhead_bytes : overhead in bytes for each slot in buffer. - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID, - int noOfCodecs, int *maxBytes, - int *maxSlots, - int* per_slot_overhead_bytes); - -#endif /* PACKET_BUFFER_H */ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_ + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +// Forward declaration. +class DecoderDatabase; + +// This is the actual buffer holding the packets before decoding. +class PacketBuffer { + public: + enum BufferReturnCodes { + kOK = 0, + kFlushed, + kNotFound, + kBufferEmpty, + kInvalidPacket, + kInvalidPointer + }; + + // Constructor creates a buffer which can hold a maximum of + // |max_number_of_packets| packets. + PacketBuffer(size_t max_number_of_packets); + + // Deletes all packets in the buffer before destroying the buffer. + virtual ~PacketBuffer(); + + // Flushes the buffer and deletes all packets in it. + virtual void Flush(); + + // Returns true for an empty buffer. + virtual bool Empty() const { return buffer_.empty(); } + + // Inserts |packet| into the buffer. The buffer will take over ownership of + // the packet object. + // Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer + // was flushed due to overfilling. + virtual int InsertPacket(Packet* packet); + + // Inserts a list of packets into the buffer. The buffer will take over + // ownership of the packet objects. + // Returns PacketBuffer::kOK if all packets were inserted successfully. + // If the buffer was flushed due to overfilling, only a subset of the list is + // inserted, and PacketBuffer::kFlushed is returned. + // The last three parameters are included for legacy compatibility. + // TODO(hlundin): Redesign to not use current_*_payload_type and + // decoder_database. + virtual int InsertPacketList(PacketList* packet_list, + const DecoderDatabase& decoder_database, + uint8_t* current_rtp_payload_type, + uint8_t* current_cng_rtp_payload_type); + + // Gets the timestamp for the first packet in the buffer and writes it to the + // output variable |next_timestamp|. + // Returns PacketBuffer::kBufferEmpty if the buffer is empty, + // PacketBuffer::kOK otherwise. + virtual int NextTimestamp(uint32_t* next_timestamp) const; + + // Gets the timestamp for the first packet in the buffer with a timestamp no + // lower than the input limit |timestamp|. The result is written to the output + // variable |next_timestamp|. + // Returns PacketBuffer::kBufferEmpty if the buffer is empty, + // PacketBuffer::kOK otherwise. + virtual int NextHigherTimestamp(uint32_t timestamp, + uint32_t* next_timestamp) const; + + // Returns a (constant) pointer the RTP header of the first packet in the + // buffer. Returns NULL if the buffer is empty. + virtual const RTPHeader* NextRtpHeader() const; + + // Extracts the first packet in the buffer and returns a pointer to it. + // Returns NULL if the buffer is empty. The caller is responsible for deleting + // the packet. + // Subsequent packets with the same timestamp as the one extracted will be + // discarded and properly deleted. The number of discarded packets will be + // written to the output variable |discard_count|. + virtual Packet* GetNextPacket(int* discard_count); + + // Discards the first packet in the buffer. The packet is deleted. + // Returns PacketBuffer::kBufferEmpty if the buffer is empty, + // PacketBuffer::kOK otherwise. + virtual int DiscardNextPacket(); + + // Discards all packets that are (strictly) older than |timestamp_limit|. + // Returns number of packets discarded. + virtual int DiscardOldPackets(uint32_t timestamp_limit); + + // Returns the number of packets in the buffer, including duplicates and + // redundant packets. + virtual int NumPacketsInBuffer() const { + return static_cast<int>(buffer_.size()); + } + + // Returns the number of samples in the buffer, including samples carried in + // duplicate and redundant packets. + virtual int NumSamplesInBuffer(DecoderDatabase* decoder_database, + int last_decoded_length) const; + + // Increase the waiting time counter for every packet in the buffer by |inc|. + // The default value for |inc| is 1. + virtual void IncrementWaitingTimes(int inc = 1); + + virtual void BufferStat(int* num_packets, int* max_num_packets) const; + + // Static method that properly deletes the first packet, and its payload + // array, in |packet_list|. Returns false if |packet_list| already was empty, + // otherwise true. + static bool DeleteFirstPacket(PacketList* packet_list); + + // Static method that properly deletes all packets, and their payload arrays, + // in |packet_list|. + static void DeleteAllPackets(PacketList* packet_list); + + private: + size_t max_number_of_packets_; + PacketList buffer_; + DISALLOW_COPY_AND_ASSIGN(PacketBuffer); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc index c8109dc6dff..5e6b89fdc4b 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc @@ -10,12 +10,12 @@ // Unit tests for PacketBuffer class. -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" +#include "webrtc/modules/audio_coding/neteq/packet_buffer.h" #include "gmock/gmock.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" using ::testing::Return; using ::testing::_; @@ -70,13 +70,13 @@ void PacketGenerator::SkipPacket() { // Start of test definitions. TEST(PacketBuffer, CreateAndDestroy) { - PacketBuffer* buffer = new PacketBuffer(10, 1000); // 10 packets, 1000 bytes. + PacketBuffer* buffer = new PacketBuffer(10); // 10 packets. EXPECT_TRUE(buffer->Empty()); delete buffer; } TEST(PacketBuffer, InsertPacket) { - PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes. + PacketBuffer buffer(10); // 10 packets. PacketGenerator gen(17u, 4711u, 0, 10); const int payload_len = 100; @@ -88,7 +88,6 @@ TEST(PacketBuffer, InsertPacket) { EXPECT_EQ(4711u, next_ts); EXPECT_FALSE(buffer.Empty()); EXPECT_EQ(1, buffer.NumPacketsInBuffer()); - EXPECT_EQ(payload_len, buffer.current_memory_bytes()); const RTPHeader* hdr = buffer.NextRtpHeader(); EXPECT_EQ(&(packet->header), hdr); // Compare pointer addresses. @@ -98,7 +97,7 @@ TEST(PacketBuffer, InsertPacket) { // Test to flush buffer. TEST(PacketBuffer, FlushBuffer) { - PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes. + PacketBuffer buffer(10); // 10 packets. PacketGenerator gen(0, 0, 0, 10); const int payload_len = 10; @@ -109,18 +108,16 @@ TEST(PacketBuffer, FlushBuffer) { } EXPECT_EQ(10, buffer.NumPacketsInBuffer()); EXPECT_FALSE(buffer.Empty()); - EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes()); buffer.Flush(); // Buffer should delete the payloads itself. EXPECT_EQ(0, buffer.NumPacketsInBuffer()); EXPECT_TRUE(buffer.Empty()); - EXPECT_EQ(0, buffer.current_memory_bytes()); } // Test to fill the buffer over the limits, and verify that it flushes. TEST(PacketBuffer, OverfillBuffer) { - PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes. + PacketBuffer buffer(10); // 10 packets. PacketGenerator gen(0, 0, 0, 10); // Insert 10 small packets; should be ok. @@ -131,7 +128,6 @@ TEST(PacketBuffer, OverfillBuffer) { EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(packet)); } EXPECT_EQ(10, buffer.NumPacketsInBuffer()); - EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes()); uint32_t next_ts; EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts)); EXPECT_EQ(0u, next_ts); // Expect first inserted packet to be first in line. @@ -140,30 +136,17 @@ TEST(PacketBuffer, OverfillBuffer) { Packet* packet = gen.NextPacket(payload_len); EXPECT_EQ(PacketBuffer::kFlushed, buffer.InsertPacket(packet)); EXPECT_EQ(1, buffer.NumPacketsInBuffer()); - EXPECT_EQ(payload_len, buffer.current_memory_bytes()); EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts)); // Expect last inserted packet to be first in line. EXPECT_EQ(packet->header.timestamp, next_ts); - // Insert 2 large packets; expect to flush when inserting the second one. - const int large_payload_len = 500; - packet = gen.NextPacket(large_payload_len); - EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(packet)); - EXPECT_EQ(2, buffer.NumPacketsInBuffer()); - EXPECT_EQ(payload_len + large_payload_len, buffer.current_memory_bytes()); - - packet = gen.NextPacket(large_payload_len); - EXPECT_EQ(PacketBuffer::kFlushed, buffer.InsertPacket(packet)); - EXPECT_EQ(1, buffer.NumPacketsInBuffer()); - EXPECT_EQ(large_payload_len, buffer.current_memory_bytes()); - - // Flush buffer to delete remaining packets. + // Flush buffer to delete all packets. buffer.Flush(); } // Test inserting a list of packets. TEST(PacketBuffer, InsertPacketList) { - PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes. + PacketBuffer buffer(10); // 10 packets. PacketGenerator gen(0, 0, 0, 10); PacketList list; const int payload_len = 10; @@ -187,7 +170,6 @@ TEST(PacketBuffer, InsertPacketList) { ¤t_cng_pt)); EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list. EXPECT_EQ(10, buffer.NumPacketsInBuffer()); - EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes()); EXPECT_EQ(0, current_pt); // Current payload type changed to 0. EXPECT_EQ(0xFF, current_cng_pt); // CNG payload type not changed. @@ -200,7 +182,7 @@ TEST(PacketBuffer, InsertPacketList) { // Expecting the buffer to flush. // TODO(hlundin): Remove this test when legacy operation is no longer needed. TEST(PacketBuffer, InsertPacketListChangePayloadType) { - PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes. + PacketBuffer buffer(10); // 10 packets. PacketGenerator gen(0, 0, 0, 10); PacketList list; const int payload_len = 10; @@ -229,7 +211,6 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) { ¤t_cng_pt)); EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list. EXPECT_EQ(1, buffer.NumPacketsInBuffer()); // Only the last packet. - EXPECT_EQ(1 * payload_len, buffer.current_memory_bytes()); EXPECT_EQ(1, current_pt); // Current payload type changed to 0. EXPECT_EQ(0xFF, current_cng_pt); // CNG payload type not changed. @@ -252,7 +233,7 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) { // 8 0x0005 0x00000028 0x0000001E // 9 0x0006 0x00000032 0x00000028 TEST(PacketBuffer, ExtractOrderRedundancy) { - PacketBuffer buffer(100, 1000); // 100 packets, 1000 bytes. + PacketBuffer buffer(100); // 100 packets. const uint32_t ts_increment = 10; // Samples per packet. const uint16_t start_seq_no = 0xFFFF - 2; // Wraps after 3 packets. const uint32_t start_ts = 0xFFFFFFFF - @@ -321,7 +302,7 @@ TEST(PacketBuffer, ExtractOrderRedundancy) { } TEST(PacketBuffer, DiscardPackets) { - PacketBuffer buffer(100, 1000); // 100 packets, 1000 bytes. + PacketBuffer buffer(100); // 100 packets. const uint16_t start_seq_no = 17; const uint32_t start_ts = 4711; const uint32_t ts_increment = 10; @@ -335,7 +316,6 @@ TEST(PacketBuffer, DiscardPackets) { buffer.InsertPacket(packet); } EXPECT_EQ(10, buffer.NumPacketsInBuffer()); - EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes()); // Discard them one by one and make sure that the right packets are at the // front of the buffer. @@ -351,7 +331,7 @@ TEST(PacketBuffer, DiscardPackets) { } TEST(PacketBuffer, Reordering) { - PacketBuffer buffer(100, 1000); // 100 packets, 1000 bytes. + PacketBuffer buffer(100); // 100 packets. const uint16_t start_seq_no = 17; const uint32_t start_ts = 4711; const uint32_t ts_increment = 10; @@ -384,7 +364,6 @@ TEST(PacketBuffer, Reordering) { ¤t_pt, ¤t_cng_pt)); EXPECT_EQ(10, buffer.NumPacketsInBuffer()); - EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes()); // Extract them and make sure that come out in the right order. uint32_t current_ts = start_ts; @@ -408,18 +387,8 @@ TEST(PacketBuffer, Failures) { int payload_len = 100; PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment); - PacketBuffer* buffer = new PacketBuffer(0, 1000); // 0 packets, 1000 bytes. - Packet* packet = gen.NextPacket(payload_len); - EXPECT_EQ(PacketBuffer::kOversizePacket, buffer->InsertPacket(packet)); - delete buffer; - - buffer = new PacketBuffer(100, 10); // 100 packets, 10 bytes. - packet = gen.NextPacket(payload_len); - EXPECT_EQ(PacketBuffer::kOversizePacket, buffer->InsertPacket(packet)); - delete buffer; - - buffer = new PacketBuffer(100, 10000); // 100 packets, 10000 bytes. - packet = NULL; + PacketBuffer* buffer = new PacketBuffer(100); // 100 packets. + Packet* packet = NULL; EXPECT_EQ(PacketBuffer::kInvalidPacket, buffer->InsertPacket(packet)); packet = gen.NextPacket(payload_len); delete [] packet->payload; @@ -448,7 +417,7 @@ TEST(PacketBuffer, Failures) { // Insert packet list of three packets, where the second packet has an invalid // payload. Expect first packet to be inserted, and the remaining two to be // discarded. - buffer = new PacketBuffer(100, 1000); // 100 packets, 1000 bytes. + buffer = new PacketBuffer(100); // 100 packets. PacketList list; list.push_back(gen.NextPacket(payload_len)); // Valid packet. packet = gen.NextPacket(payload_len); diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/payload_splitter.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.cc index 56039a57ec6..1d61ef0cf40 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/payload_splitter.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.cc @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/payload_splitter.h" +#include "webrtc/modules/audio_coding/neteq/payload_splitter.h" #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" namespace webrtc { @@ -119,6 +119,64 @@ int PayloadSplitter::SplitRed(PacketList* packet_list) { return ret; } +int PayloadSplitter::SplitFec(PacketList* packet_list, + DecoderDatabase* decoder_database) { + PacketList::iterator it = packet_list->begin(); + // Iterate through all packets in |packet_list|. + while (it != packet_list->end()) { + Packet* packet = (*it); // Just to make the notation more intuitive. + // Get codec type for this payload. + uint8_t payload_type = packet->header.payloadType; + const DecoderDatabase::DecoderInfo* info = + decoder_database->GetDecoderInfo(payload_type); + if (!info) { + return kUnknownPayloadType; + } + // No splitting for a sync-packet. + if (packet->sync_packet) { + ++it; + continue; + } + + // Not an FEC packet. + AudioDecoder* decoder = decoder_database->GetDecoder(payload_type); + // decoder should not return NULL. + assert(decoder != NULL); + if (!decoder || + !decoder->PacketHasFec(packet->payload, packet->payload_length)) { + ++it; + continue; + } + + switch (info->codec_type) { + case kDecoderOpus: + case kDecoderOpus_2ch: { + Packet* new_packet = new Packet; + + new_packet->header = packet->header; + int duration = decoder-> + PacketDurationRedundant(packet->payload, packet->payload_length); + new_packet->header.timestamp -= duration; + new_packet->payload = new uint8_t[packet->payload_length]; + memcpy(new_packet->payload, packet->payload, packet->payload_length); + new_packet->payload_length = packet->payload_length; + new_packet->primary = false; + new_packet->waiting_time = packet->waiting_time; + new_packet->sync_packet = packet->sync_packet; + + packet_list->insert(it, new_packet); + break; + } + default: { + return kFecSplitError; + } + } + + ++it; + } + return kOK; +} + int PayloadSplitter::CheckRedPayloads(PacketList* packet_list, const DecoderDatabase& decoder_database) { PacketList::iterator it = packet_list->begin(); @@ -283,7 +341,7 @@ int PayloadSplitter::SplitAudio(PacketList* packet_list, // increment it manually. it = packet_list->erase(it); } - return 0; + return kOK; } void PayloadSplitter::SplitBySamples(const Packet* packet, diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/payload_splitter.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.h index 3768c2f2b1c..a3dd77e5a53 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/payload_splitter.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PAYLOAD_SPLITTER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PAYLOAD_SPLITTER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PAYLOAD_SPLITTER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PAYLOAD_SPLITTER_H_ -#include "webrtc/modules/audio_coding/neteq4/packet.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" namespace webrtc { @@ -32,7 +32,8 @@ class PayloadSplitter { kTooLargePayload = -1, kFrameSplitError = -2, kUnknownPayloadType = -3, - kRedLengthMismatch = -4 + kRedLengthMismatch = -4, + kFecSplitError = -5, }; PayloadSplitter() {} @@ -47,6 +48,12 @@ class PayloadSplitter { // Returns kOK or an error. virtual int SplitRed(PacketList* packet_list); + // Iterates through |packet_list| and, duplicate each audio payload that has + // FEC as new packet for redundant decoding. The decoder database is needed to + // get information about which payload type each packet contains. + virtual int SplitFec(PacketList* packet_list, + DecoderDatabase* decoder_database); + // Checks all packets in |packet_list|. Packets that are DTMF events or // comfort noise payloads are kept. Except that, only one single payload type // is accepted. Any packet with another payload type is discarded. @@ -80,4 +87,4 @@ class PayloadSplitter { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PAYLOAD_SPLITTER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PAYLOAD_SPLITTER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/payload_splitter_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc index 5a7a6ca3e4e..5cde1bda5e5 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/payload_splitter_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc @@ -10,15 +10,15 @@ // Unit tests for PayloadSplitter class. -#include "webrtc/modules/audio_coding/neteq4/payload_splitter.h" +#include "webrtc/modules/audio_coding/neteq/payload_splitter.h" #include <assert.h> #include <utility> // pair #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" using ::testing::Return; @@ -91,6 +91,34 @@ Packet* CreateRedPayload(int num_payloads, return packet; } + +// A possible Opus packet that contains FEC is the following. +// The frame is 20 ms in duration. +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |0|0|0|0|1|0|0|0|x|1|x|x|x|x|x|x|x| | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | +// | Compressed frame 1 (N-2 bytes)... : +// : | +// | | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +Packet* CreateOpusFecPacket(uint8_t payload_type, int payload_length, + uint8_t payload_value) { + Packet* packet = new Packet; + packet->header.payloadType = payload_type; + packet->header.timestamp = kBaseTimestamp; + packet->header.sequenceNumber = kSequenceNumber; + packet->payload_length = payload_length; + uint8_t* payload = new uint8_t[packet->payload_length]; + payload[0] = 0x08; + payload[1] = 0x40; + memset(&payload[2], payload_value, payload_length - 2); + packet->payload = payload; + return packet; +} + // Create a packet with all payload bytes set to |payload_value|. Packet* CreatePacket(uint8_t payload_type, int payload_length, uint8_t payload_value) { @@ -691,4 +719,59 @@ TEST(IlbcPayloadSplitter, UnevenPayload) { EXPECT_CALL(decoder_database, Die()); } +TEST(FecPayloadSplitter, MixedPayload) { + PacketList packet_list; + DecoderDatabase decoder_database; + + decoder_database.RegisterPayload(0, kDecoderOpus); + decoder_database.RegisterPayload(1, kDecoderPCMu); + + Packet* packet = CreateOpusFecPacket(0, 10, 0xFF); + packet_list.push_back(packet); + + packet = CreatePacket(0, 10, 0); // Non-FEC Opus payload. + packet_list.push_back(packet); + + packet = CreatePacket(1, 10, 0); // Non-Opus payload. + packet_list.push_back(packet); + + PayloadSplitter splitter; + EXPECT_EQ(PayloadSplitter::kOK, + splitter.SplitFec(&packet_list, &decoder_database)); + EXPECT_EQ(4u, packet_list.size()); + + // Check first packet. + packet = packet_list.front(); + EXPECT_EQ(0, packet->header.payloadType); + EXPECT_EQ(kBaseTimestamp - 20 * 32, packet->header.timestamp); + EXPECT_EQ(10, packet->payload_length); + EXPECT_FALSE(packet->primary); + delete [] packet->payload; + delete packet; + packet_list.pop_front(); + + // Check second packet. + packet = packet_list.front(); + EXPECT_EQ(0, packet->header.payloadType); + EXPECT_EQ(kBaseTimestamp, packet->header.timestamp); + EXPECT_EQ(10, packet->payload_length); + EXPECT_TRUE(packet->primary); + delete [] packet->payload; + delete packet; + packet_list.pop_front(); + + // Check third packet. + packet = packet_list.front(); + VerifyPacket(packet, 10, 0, kSequenceNumber, kBaseTimestamp, 0, true); + delete [] packet->payload; + delete packet; + packet_list.pop_front(); + + // Check fourth packet. + packet = packet_list.front(); + VerifyPacket(packet, 10, 1, kSequenceNumber, kBaseTimestamp, 0, true); + delete [] packet->payload; + delete packet; +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/peak_detection.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/peak_detection.c deleted file mode 100644 index 8c85d2a837e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/peak_detection.c +++ /dev/null @@ -1,232 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of the peak detection used for finding correlation peaks. - */ - -#include "dsp_helpfunctions.h" - -#include "signal_processing_library.h" - -/* Table of constants used in parabolic fit function WebRtcNetEQ_PrblFit */ -const int16_t WebRtcNetEQ_kPrblCf[17][3] = { { 120, 32, 64 }, { 140, 44, 75 }, - { 150, 50, 80 }, { 160, 57, 85 }, - { 180, 72, 96 }, { 200, 89, 107 }, - { 210, 98, 112 }, { 220, 108, 117 }, - { 240, 128, 128 }, { 260, 150, 139 }, - { 270, 162, 144 }, { 280, 174, 149 }, - { 300, 200, 160 }, { 320, 228, 171 }, - { 330, 242, 176 }, { 340, 257, 181 }, - { 360, 288, 192 } }; - -int16_t WebRtcNetEQ_PeakDetection(int16_t *pw16_data, int16_t w16_dataLen, - int16_t w16_nmbPeaks, int16_t fs_mult, - int16_t *pw16_winIndex, - int16_t *pw16_winValue) -{ - /* Local variables */ - int i; - int16_t w16_tmp; - int16_t w16_tmp2; - int16_t indMin = 0; - int16_t indMax = 0; - - /* Peak detection */ - - for (i = 0; i <= (w16_nmbPeaks - 1); i++) - { - if (w16_nmbPeaks == 1) - { - /* - * Single peak - * The parabola fit assumes that an extra point is available; worst case it gets - * a zero on the high end of the signal. - */ - w16_dataLen++; - } - - pw16_winIndex[i] = WebRtcSpl_MaxIndexW16(pw16_data, (int16_t) (w16_dataLen - 1)); - - if (i != w16_nmbPeaks - 1) - { - w16_tmp = pw16_winIndex[i] - 2; /* *fs_mult; */ - indMin = WEBRTC_SPL_MAX(0, w16_tmp); - w16_tmp = pw16_winIndex[i] + 2; /* *fs_mult; */ - w16_tmp2 = w16_dataLen - 1; - indMax = WEBRTC_SPL_MIN(w16_tmp2, w16_tmp); - } - - if ((pw16_winIndex[i] != 0) && (pw16_winIndex[i] != (w16_dataLen - 2))) - { - /* Parabola fit*/ - WebRtcNetEQ_PrblFit(&(pw16_data[pw16_winIndex[i] - 1]), &(pw16_winIndex[i]), - &(pw16_winValue[i]), fs_mult); - } - else - { - if (pw16_winIndex[i] == (w16_dataLen - 2)) - { - if (pw16_data[pw16_winIndex[i]] > pw16_data[pw16_winIndex[i] + 1]) - { - WebRtcNetEQ_PrblFit(&(pw16_data[pw16_winIndex[i] - 1]), - &(pw16_winIndex[i]), &(pw16_winValue[i]), fs_mult); - } - else if (pw16_data[pw16_winIndex[i]] <= pw16_data[pw16_winIndex[i] + 1]) - { - pw16_winValue[i] = (pw16_data[pw16_winIndex[i]] - + pw16_data[pw16_winIndex[i] + 1]) >> 1; /* lin approx */ - pw16_winIndex[i] = (pw16_winIndex[i] * 2 + 1) * fs_mult; - } - } - else - { - pw16_winValue[i] = pw16_data[pw16_winIndex[i]]; - pw16_winIndex[i] = pw16_winIndex[i] * 2 * fs_mult; - } - } - - if (i != w16_nmbPeaks - 1) - { - WebRtcSpl_MemSetW16(&(pw16_data[indMin]), 0, (indMax - indMin + 1)); - /* for (j=indMin; j<=indMax; j++) pw16_data[j] = 0; */ - } - } - - return 0; -} - -int16_t WebRtcNetEQ_PrblFit(int16_t *pw16_3pts, int16_t *pw16_Ind, - int16_t *pw16_outVal, int16_t fs_mult) -{ - /* Variables */ - int32_t Num, Den; - int32_t temp; - int16_t flag, stp, strt, lmt; - uint16_t PFind[13]; - - if (fs_mult == 1) - { - PFind[0] = 0; - PFind[1] = 8; - PFind[2] = 16; - } - else if (fs_mult == 2) - { - PFind[0] = 0; - PFind[1] = 4; - PFind[2] = 8; - PFind[3] = 12; - PFind[4] = 16; - } - else if (fs_mult == 4) - { - PFind[0] = 0; - PFind[1] = 2; - PFind[2] = 4; - PFind[3] = 6; - PFind[4] = 8; - PFind[5] = 10; - PFind[6] = 12; - PFind[7] = 14; - PFind[8] = 16; - } - else - { - PFind[0] = 0; - PFind[1] = 1; - PFind[2] = 3; - PFind[3] = 4; - PFind[4] = 5; - PFind[5] = 7; - PFind[6] = 8; - PFind[7] = 9; - PFind[8] = 11; - PFind[9] = 12; - PFind[10] = 13; - PFind[11] = 15; - PFind[12] = 16; - } - - /* Num = -3*pw16_3pts[0] + 4*pw16_3pts[1] - pw16_3pts[2]; */ - /* Den = pw16_3pts[0] - 2*pw16_3pts[1] + pw16_3pts[2]; */ - Num = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],-3) + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],4) - - pw16_3pts[2]; - - Den = pw16_3pts[0] + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],-2) + pw16_3pts[2]; - - temp = (int32_t) WEBRTC_SPL_MUL(Num, (int32_t)120); /* need 32_16 really */ - flag = 1; - stp = WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0] - WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0]; - strt = (WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0] - + WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0]) >> 1; - - if (temp < (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)strt)) - { - lmt = strt - stp; - while (flag) - { - if ((flag == fs_mult) || (temp - > (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)lmt))) - { - *pw16_outVal - = (int16_t) - (((int32_t) ((int32_t) WEBRTC_SPL_MUL(Den,(int32_t)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][1]) - + (int32_t) WEBRTC_SPL_MUL(Num,(int32_t)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][2]) - + WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256))) >> 8); - *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) - flag; - flag = 0; - } - else - { - flag++; - lmt -= stp; - } - } - } - else if (temp > (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)(strt+stp))) - { - lmt = strt + (stp << 1); - while (flag) - { - if ((flag == fs_mult) || (temp - < (int32_t) WEBRTC_SPL_MUL(-Den,(int32_t)lmt))) - { - int32_t temp_term_1, temp_term_2, temp_term_3; - - temp_term_1 = WEBRTC_SPL_MUL(Den, - (int32_t) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][1]); - temp_term_2 = WEBRTC_SPL_MUL(Num, - (int32_t) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][2]); - temp_term_3 = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256); - - *pw16_outVal - = (int16_t) ((temp_term_1 + temp_term_2 + temp_term_3) >> 8); - - *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) + flag; - flag = 0; - } - else - { - flag++; - lmt += stp; - } - } - - } - else - { - *pw16_outVal = pw16_3pts[1]; - *pw16_Ind = (*pw16_Ind) * 2 * fs_mult; - } - - return 0; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/post_decode_vad.cc index c3d5c7edd71..7ae7f97abc9 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/post_decode_vad.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/post_decode_vad.h" +#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/post_decode_vad.h index eb197d9ef24..e713009c85f 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/post_decode_vad.h @@ -8,17 +8,17 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_POST_DECODE_VAD_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_POST_DECODE_VAD_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_ #include <string> // size_t +#include "webrtc/base/constructormagic.h" #include "webrtc/common_audio/vad/include/webrtc_vad.h" #include "webrtc/common_types.h" // NULL -#include "webrtc/modules/audio_coding/neteq4/defines.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/modules/audio_coding/neteq/defines.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -69,4 +69,4 @@ class PostDecodeVad { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_POST_DECODE_VAD_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc index a4d9da8e166..ed48db858d4 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc @@ -10,7 +10,7 @@ // Unit tests for PostDecodeVad class. -#include "webrtc/modules/audio_coding/neteq4/post_decode_vad.h" +#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h" #include "gtest/gtest.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c deleted file mode 100644 index e56c0628415..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c +++ /dev/null @@ -1,527 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains the Pre-emptive Expand algorithm that is used to increase - * the delay by repeating a part of the audio stream. - */ - -#include "dsp.h" - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" -#include "neteq_error_codes.h" - -#define PREEMPTIVE_CORR_LEN 50 -#define PREEMPTIVE_MIN_LAG 10 -#define PREEMPTIVE_MAX_LAG 60 -#define PREEMPTIVE_DOWNSAMPLED_LEN (PREEMPTIVE_CORR_LEN + PREEMPTIVE_MAX_LAG) - -/* Scratch usage: - - Type Name size startpos endpos - int16_t pw16_downSampSpeech 110 0 109 - int32_t pw32_corr 2*50 110 209 - int16_t pw16_corr 50 0 49 - - Total: 110+2*50 - */ - -#define SCRATCH_PW16_DS_SPEECH 0 -#define SCRATCH_PW32_CORR PREEMPTIVE_DOWNSAMPLED_LEN -#define SCRATCH_PW16_CORR 0 - -/**************************************************************************** - * WebRtcNetEQ_PreEmptiveExpand(...) - * - * This function tries to extend the audio data by repeating one or several - * pitch periods. The operation is only carried out if the correlation is - * strong or if the signal energy is very low. The algorithm is the - * reciprocal of the Accelerate algorithm. - * - * Input: - * - inst : NetEQ DSP instance - * - scratchPtr : Pointer to scratch vector. - * - decoded : Pointer to newly decoded speech. - * - len : Length of decoded speech. - * - oldDataLen : Length of the part of decoded that has already been played out. - * - BGNonly : If non-zero, Pre-emptive Expand will only copy - * the first DEFAULT_TIME_ADJUST seconds of the - * input and append to the end. No signal matching is - * done. - * - * Output: - * - inst : Updated instance - * - outData : Pointer to a memory space where the output data - * should be stored. The vector must be at least - * min(len + 120*fs/8000, NETEQ_MAX_OUTPUT_SIZE) - * elements long. - * - pw16_len : Number of samples written to outData. - * - * Return value : 0 - Ok - * <0 - Error - */ - -int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst, -#ifdef SCRATCH - int16_t *pw16_scratchPtr, -#endif - const int16_t *pw16_decoded, int len, int oldDataLen, - int16_t *pw16_outData, int16_t *pw16_len, - int16_t BGNonly) -{ - -#ifdef SCRATCH - /* Use scratch memory for internal temporary vectors */ - int16_t *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH; - int32_t *pw32_corr = (int32_t*) (pw16_scratchPtr + SCRATCH_PW32_CORR); - int16_t *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR; -#else - /* Allocate memory for temporary vectors */ - int16_t pw16_downSampSpeech[PREEMPTIVE_DOWNSAMPLED_LEN]; - int32_t pw32_corr[PREEMPTIVE_CORR_LEN]; - int16_t pw16_corr[PREEMPTIVE_CORR_LEN]; -#endif - int16_t w16_decodedMax = 0; - int16_t w16_tmp = 0; - int16_t w16_tmp2; - int32_t w32_tmp; - int32_t w32_tmp2; - - const int16_t w16_startLag = PREEMPTIVE_MIN_LAG; - const int16_t w16_endLag = PREEMPTIVE_MAX_LAG; - const int16_t w16_corrLen = PREEMPTIVE_CORR_LEN; - const int16_t *pw16_vec1, *pw16_vec2; - int16_t *pw16_vectmp; - int16_t w16_inc, w16_startfact; - int16_t w16_bestIndex, w16_bestVal; - int16_t w16_VAD = 1; - int16_t fsMult; - int16_t fsMult120; - int32_t w32_en1, w32_en2, w32_cc; - int16_t w16_en1, w16_en2; - int16_t w16_en1Scale, w16_en2Scale; - int16_t w16_sqrtEn1En2; - int16_t w16_bestCorr = 0; - int ok; - -#ifdef NETEQ_STEREO - MasterSlaveInfo *msInfo = inst->msInfo; -#endif - - fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */ - - /* Pre-calculate common multiplication with fsMult */ - fsMult120 = (int16_t) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */ - - inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */ - - /* - * Sanity check for len variable; must be (almost) 30 ms (120*fsMult + max(bestIndex)). - * Also, the new part must be at least .625 ms (w16_overlap). - */ - if (len < (int16_t) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult) || oldDataLen >= len - - inst->ExpandInst.w16_overlap) - { - /* Length of decoded data too short */ - inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE; - *pw16_len = len; - - - /* simply move all data from decoded to outData */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return NETEQ_OTHER_ERROR; - } - - /***********************************/ - /* Special operations for BGN only */ - /***********************************/ - - /* Check if "background noise only" flag is set */ - if (BGNonly) - { - /* special operation for BGN only; simply insert a chunk of data */ - w16_bestIndex = DEFAULT_TIME_ADJUST * (fsMult << 3); /* X*fs/1000 */ - - /* Sanity check for bestIndex */ - if (w16_bestIndex > len) - { /* not good, do nothing instead */ - inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE; - *pw16_len = len; - - - /* simply move all data from decoded to outData */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return NETEQ_OTHER_ERROR; - } - - /* set length parameter */ - *pw16_len = len + w16_bestIndex; - - - /* copy to output */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, len); - WEBRTC_SPL_MEMCPY_W16(&pw16_outData[len], pw16_decoded, w16_bestIndex); - - /* set mode */ - inst->w16_mode = MODE_LOWEN_PREEMPTIVE; - - /* update statistics */ - inst->statInst.preemptiveLength += w16_bestIndex; - /* Short-term activity statistics. */ - inst->activity_stats.preemptive_expand_bgn_samples += w16_bestIndex; - - return 0; - } /* end of special code for BGN mode */ - -#ifdef NETEQ_STEREO - - /* Sanity for msInfo */ - if (msInfo == NULL) - { - /* this should not happen here */ - return MASTER_SLAVE_ERROR; - } - - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)) - { - /* Find correlation lag only for non-slave instances */ - -#endif - - /****************************************************************/ - /* Find the strongest correlation lag by downsampling to 4 kHz, */ - /* calculating correlation for downsampled signal and finding */ - /* the strongest correlation peak. */ - /****************************************************************/ - - /* find maximum absolute value */ - w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (int16_t) len); - - /* downsample the decoded speech to 4 kHz */ - ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech, - PREEMPTIVE_DOWNSAMPLED_LEN, 1 /* compensate delay*/); - if (ok != 0) - { - /* error */ - inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE; - *pw16_len = len; - - - /* simply move all data from decoded to outData */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return NETEQ_OTHER_ERROR; - } - - /* - * Set scaling factor for cross correlation to protect against - * overflow (log2(50) => 6) - */ - w16_tmp = 6 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)); - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */WebRtcNetEQ_CrossCorr( - pw32_corr, &pw16_downSampSpeech[w16_endLag], - &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen, - (int16_t) (w16_endLag - w16_startLag), w16_tmp, -1); - - /* Normalize correlation to 14 bits and put in a int16_t vector */ - w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen); - w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp); - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_corrLen, pw32_corr, w16_tmp); - - /* Find limits for peak finding, in order to avoid overful NetEQ algorithm buffer. */ - /* Calculate difference between MAX_OUTPUT_SIZE and len in 4 kHz domain. */ - w16_tmp = WebRtcSpl_DivW32W16ResW16((int32_t) (NETEQ_MAX_OUTPUT_SIZE - len), - (int16_t) (fsMult << 1)) - w16_startLag; - w16_tmp = WEBRTC_SPL_MIN(w16_corrLen, w16_tmp); /* no more than corrLen = 50 */ - -#ifdef NETEQ_STEREO - } /* end if (msInfo->msMode != NETEQ_SLAVE) */ - - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)) - { - /* Find the strongest correlation peak by using the parabolic fit method */ - WebRtcNetEQ_PeakDetection(pw16_corr, w16_tmp, 1, fsMult, &w16_bestIndex, &w16_bestVal); - /* 0 <= bestIndex <= (2*w16_tmp - 1)*fsMult <= (2*corrLen - 1)*fsMult = 99*fsMult */ - - /* Compensate bestIndex for displaced starting position */ - w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1); - /* 20*fsMult <= bestIndex <= 119*fsMult */ - - msInfo->bestIndex = w16_bestIndex; - } - else if (msInfo->msMode == NETEQ_SLAVE) - { - if (msInfo->extraInfo == PE_EXP_FAIL) - { - /* Master has signaled an unsuccessful preemptive expand */ - w16_bestIndex = 0; - } - else - { - /* Get best index from master */ - w16_bestIndex = msInfo->bestIndex; - } - } - else - { - /* Invalid mode */ - return (MASTER_SLAVE_ERROR); - } - -#else /* NETEQ_STEREO */ - - /* Find the strongest correlation peak by using the parabolic fit method */ - WebRtcNetEQ_PeakDetection(pw16_corr, w16_tmp, 1, fsMult, &w16_bestIndex, &w16_bestVal); - /* 0 <= bestIndex <= (2*w16_tmp - 1)*fsMult <= (2*corrLen - 1)*fsMult = 99*fsMult */ - - /* Compensate bestIndex for displaced starting position */ - w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1); - /* 20*fsMult <= bestIndex <= 119*fsMult */ - -#endif /* NETEQ_STEREO */ - -#ifdef NETEQ_STEREO - - if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)) - { - /* Calculate correlation only for non-slave instances */ - -#endif /* NETEQ_STEREO */ - - /*****************************************************/ - /* Calculate correlation bestCorr for the found lag. */ - /* Also do a simple VAD decision. */ - /*****************************************************/ - - /* - * Calculate scaling to ensure that bestIndex samples can be square-summed - * without overflowing - */ - w16_tmp = (31 - - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax))); - w16_tmp += (31 - WebRtcSpl_NormW32(w16_bestIndex)); - w16_tmp -= 31; - w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp); - - /* vec1 starts at 15 ms minus one pitch period */ - pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex]; - /* vec2 start at 15 ms */ - pw16_vec2 = &pw16_decoded[fsMult120]; - - /* Calculate energies for vec1 and vec2 */ - w32_en1 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1, - (int16_t*) pw16_vec1, w16_bestIndex, w16_tmp); - w32_en2 = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec2, - (int16_t*) pw16_vec2, w16_bestIndex, w16_tmp); - - /* Calculate cross-correlation at the found lag */ - w32_cc = WebRtcNetEQ_DotW16W16((int16_t*) pw16_vec1, (int16_t*) pw16_vec2, - w16_bestIndex, w16_tmp); - - /* Check VAD constraint - ((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */ - w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_en1 + w32_en2, 4); /* (en1+en2)/(2*8) */ - if (inst->BGNInst.w16_initialized == 1) - { - w32_tmp2 = inst->BGNInst.w32_energy; - } - else - { - /* if BGN parameters have not been estimated, use a fixed threshold */ - w32_tmp2 = 75000; - } - w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2); - w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2); - w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2); - w16_tmp2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2); - w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2); - - /* Scale w32_tmp properly before comparing with w32_tmp2 */ - /* (w16_tmp is scaling before energy calculation, thus 2*w16_tmp) */ - if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1)) - { - /* Cannot scale only w32_tmp, must scale w32_temp2 too */ - int16_t tempshift = WebRtcSpl_NormW32(w32_tmp); - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift); - w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2, - WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift); - } - else - { - w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, - WEBRTC_SPL_LSHIFT_W32(w16_tmp,1)); - } - - if (w32_tmp <= w32_tmp2) /*((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */ - { - /* The signal seems to be passive speech */ - w16_VAD = 0; - w16_bestCorr = 0; /* Correlation does not matter */ - - /* For low energy expansion, the new data can be less than 15 ms, - but we must ensure that bestIndex is not larger than the new data. */ - w16_bestIndex = WEBRTC_SPL_MIN( w16_bestIndex, len - oldDataLen ); - } - else - { - /* The signal is active speech */ - w16_VAD = 1; - - /* Calculate correlation (cc/sqrt(en1*en2)) */ - - /* Start with calculating scale values */ - w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1); - w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale); - w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2); - w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale); - - /* Make sure total scaling is even (to simplify scale factor after sqrt) */ - if ((w16_en1Scale + w16_en2Scale) & 1) - { - w16_en1Scale += 1; - } - - /* Convert energies to int16_t */ - w16_en1 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale); - w16_en2 = (int16_t) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale); - - /* Calculate energy product */ - w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2); - - /* Calculate square-root of energy product */ - w16_sqrtEn1En2 = (int16_t) WebRtcSpl_SqrtFloor(w32_tmp); - - /* Calculate cc/sqrt(en1*en2) in Q14 */ - w16_tmp = 14 - ((w16_en1Scale + w16_en2Scale) >> 1); - w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp); - w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */ - w16_bestCorr = (int16_t) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2); - w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */ - } - -#ifdef NETEQ_STEREO - - } /* end if (msInfo->msMode != NETEQ_SLAVE) */ - -#endif /* NETEQ_STEREO */ - - /*******************************************************/ - /* Check preemptive expand criteria and insert samples */ - /*******************************************************/ - - /* Check for strong correlation (>0.9) and at least 15 ms new data, - or passive speech */ -#ifdef NETEQ_STEREO - if (((((w16_bestCorr > 14746) && (oldDataLen <= fsMult120)) || (w16_VAD == 0)) - && (msInfo->msMode != NETEQ_SLAVE)) || ((msInfo->msMode == NETEQ_SLAVE) - && (msInfo->extraInfo != PE_EXP_FAIL))) -#else - if (((w16_bestCorr > 14746) && (oldDataLen <= fsMult120)) - || (w16_VAD == 0)) -#endif - { - /* Do expand operation by overlap add */ - - /* Set length of the first part, not to be modified */ - int16_t w16_startIndex = WEBRTC_SPL_MAX(oldDataLen, fsMult120); - - /* - * Calculate cross-fading slope so that the fading factor goes from - * 1 (16384 in Q14) to 0 in one pitch period (bestIndex). - */ - w16_inc = (int16_t) WebRtcSpl_DivW32W16((int32_t) 16384, - (int16_t) (w16_bestIndex + 1)); /* in Q14 */ - - /* Initiate fading factor */ - w16_startfact = 16384 - w16_inc; - - /* vec1 starts at 15 ms minus one pitch period */ - pw16_vec1 = &pw16_decoded[w16_startIndex - w16_bestIndex]; - /* vec2 start at 15 ms */ - pw16_vec2 = &pw16_decoded[w16_startIndex]; - - - /* Copy unmodified part [0 to 15 ms] */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, w16_startIndex); - - /* Generate interpolated part of length bestIndex (1 pitch period) */ - pw16_vectmp = pw16_outData + w16_startIndex; - /* Reuse mixing function from Expand */ - WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (int16_t*) pw16_vec2, - (int16_t*) pw16_vec1, &w16_startfact, w16_inc, w16_bestIndex); - - /* Move the last part (also unmodified) */ - /* Take from decoded at 15 ms */ - pw16_vec2 = &pw16_decoded[w16_startIndex]; - WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[w16_startIndex + w16_bestIndex], pw16_vec2, - (int16_t) (len - w16_startIndex)); - - /* Set the mode flag */ - if (w16_VAD) - { - inst->w16_mode = MODE_SUCCESS_PREEMPTIVE; - } - else - { - inst->w16_mode = MODE_LOWEN_PREEMPTIVE; - } - - /* Calculate resulting length = original length + pitch period */ - *pw16_len = len + w16_bestIndex; - - /* Update in-call statistics */ - inst->statInst.preemptiveLength += w16_bestIndex; - /* Short-term activity statistics. */ - inst->activity_stats.preemptive_expand_normal_samples += w16_bestIndex; - return 0; - } - else - { - /* Preemptive Expand not allowed */ - -#ifdef NETEQ_STEREO - /* Signal to slave(s) that this was unsuccessful */ - if (msInfo->msMode == NETEQ_MASTER) - { - msInfo->extraInfo = PE_EXP_FAIL; - } -#endif - - /* Set mode flag to unsuccessful preemptive expand */ - inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE; - - /* Length is unmodified */ - *pw16_len = len; - - - /* Simply move all data from decoded to outData */ - - WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (int16_t) len); - - return 0; - } -} - -#undef SCRATCH_PW16_DS_SPEECH -#undef SCRATCH_PW32_CORR -#undef SCRATCH_PW16_CORR diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.cc index ac787eb1d66..b2dc3e60cba 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/preemptive_expand.h" +#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" #include <algorithm> // min, max @@ -98,4 +98,13 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch( } } +PreemptiveExpand* PreemptiveExpandFactory::Create( + int sample_rate_hz, + size_t num_channels, + const BackgroundNoise& background_noise, + int overlap_samples) const { + return new PreemptiveExpand( + sample_rate_hz, num_channels, background_noise, overlap_samples); +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/preemptive_expand.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.h index 4cd92cc0bb6..1aa61330145 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/preemptive_expand.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PREEMPTIVE_EXPAND_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PREEMPTIVE_EXPAND_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_ #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/time_stretch.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" +#include "webrtc/modules/audio_coding/neteq/time_stretch.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -29,11 +29,13 @@ class BackgroundNoise; // PreemptiveExpand are implemented. class PreemptiveExpand : public TimeStretch { public: - PreemptiveExpand(int sample_rate_hz, size_t num_channels, - const BackgroundNoise& background_noise) + PreemptiveExpand(int sample_rate_hz, + size_t num_channels, + const BackgroundNoise& background_noise, + int overlap_samples) : TimeStretch(sample_rate_hz, num_channels, background_noise), old_data_length_per_channel_(-1), - overlap_samples_(5 * sample_rate_hz / 8000) { + overlap_samples_(overlap_samples) { } virtual ~PreemptiveExpand() {} @@ -70,5 +72,16 @@ class PreemptiveExpand : public TimeStretch { DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand); }; +struct PreemptiveExpandFactory { + PreemptiveExpandFactory() {} + virtual ~PreemptiveExpandFactory() {} + + virtual PreemptiveExpand* Create( + int sample_rate_hz, + size_t num_channels, + const BackgroundNoise& background_noise, + int overlap_samples) const; +}; + } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PREEMPTIVE_EXPAND_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.c deleted file mode 100644 index c168ab5437f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.c +++ /dev/null @@ -1,54 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This function generates a pseudo-random vector. - */ - -#include "dsp_helpfunctions.h" - -/* - * Values are normalized so that - * sqrt(dot(pw16_NETEQFIX_RANDN_TBL,pw16_NETEQFIX_RANDN_TBL)/256)=2^13 - */ -const int16_t WebRtcNetEQ_kRandnTbl[RANDVEC_NO_OF_SAMPLES] = -{ - 2680, 5532, 441, 5520, 16170, -5146, -1024, -8733, 3115, 9598, -10380, -4959, -1280, -21716, 7133, -1522, - 13458, -3902, 2789, -675, 3441, 5016, -13599, -4003, -2739, 3922, -7209, 13352, -11617, -7241, 12905, -2314, - 5426, 10121, -9702, 11207, -13542, 1373, 816, -5934, -12504, 4798, 1811, 4112, -613, 201, -10367, -2960, - -2419, 3442, 4299, -6116, -6092, 1552, -1650, -480, -1237, 18720, -11858, -8303, -8212, 865, -2890, -16968, - 12052, -5845, -5912, 9777, -5665, -6294, 5426, -4737, -6335, 1652, 761, 3832, 641, -8552, -9084, -5753, - 8146, 12156, -4915, 15086, -1231, -1869, 11749, -9319, -6403, 11407, 6232, -1683, 24340, -11166, 4017, -10448, - 3153, -2936, 6212, 2891, -866, -404, -4807, -2324, -1917, -2388, -6470, -3895, -10300, 5323, -5403, 2205, - 4640, 7022, -21186, -6244, -882, -10031, -3395, -12885, 7155, -5339, 5079, -2645, -9515, 6622, 14651, 15852, - 359, 122, 8246, -3502, -6696, -3679, -13535, -1409, -704, -7403, -4007, 1798, 279, -420, -12796, -14219, - 1141, 3359, 11434, 7049, -6684, -7473, 14283, -4115, -9123, -8969, 4152, 4117, 13792, 5742, 16168, 8661, - -1609, -6095, 1881, 14380, -5588, 6758, -6425, -22969, -7269, 7031, 1119, -1611, -5850, -11281, 3559, -8952, - -10146, -4667, -16251, -1538, 2062, -1012, -13073, 227, -3142, -5265, 20, 5770, -7559, 4740, -4819, 992, - -8208, -7130, -4652, 6725, 7369, -1036, 13144, -1588, -5304, -2344, -449, -5705, -8894, 5205, -17904, -11188, - -1022, 4852, 10101, -5255, -4200, -752, 7941, -1543, 5959, 14719, 13346, 17045, -15605, -1678, -1600, -9230, - 68, 23348, 1172, 7750, 11212, -18227, 9956, 4161, 883, 3947, 4341, 1014, -4889, -2603, 1246, -5630, - -3596, -870, -1298, 2784, -3317, -6612, -20541, 4166, 4181, -8625, 3562, 12890, 4761, 3205, -12259, -8579 -}; - - -void WebRtcNetEQ_RandomVec(uint32_t *w32_seed, int16_t *pw16_randVec, - int16_t w16_len, int16_t w16_incval) -{ - int i; - int16_t w16_pos; - for (i = 0; i < w16_len; i++) - { - *w32_seed = (*w32_seed) + w16_incval; - w16_pos = (int16_t) ((*w32_seed) & (RANDVEC_NO_OF_SAMPLES - 1)); - pw16_randVec[i] = WebRtcNetEQ_kRandnTbl[w16_pos]; - } -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/random_vector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.cc index 823909f1352..b12f2171553 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/random_vector.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/random_vector.h" namespace webrtc { @@ -54,4 +54,4 @@ void RandomVector::IncreaseSeedIncrement(int16_t increase_by) { seed_increment_+= increase_by; seed_increment_ &= kRandomTableSize - 1; } -} +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/random_vector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.h index 64cfe0d9dd0..767dc48eee3 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/random_vector.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_RANDOM_VECTOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_RANDOM_VECTOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_ #include <string.h> // size_t -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -47,4 +47,4 @@ class RandomVector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_RANDOM_VECTOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/random_vector_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector_unittest.cc index 83193e2a7bb..cbdcdf7c829 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/random_vector_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/random_vector_unittest.cc @@ -10,7 +10,7 @@ // Unit tests for RandomVector class. -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" +#include "webrtc/modules/audio_coding/neteq/random_vector.h" #include "gtest/gtest.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/recin.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/recin.c deleted file mode 100644 index 17bea5f5bbf..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/recin.c +++ /dev/null @@ -1,531 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of the RecIn function, which is the main function for inserting RTP - * packets into NetEQ. - */ - -#include "mcu.h" - -#include <string.h> - -#include "automode.h" -#include "dtmf_buffer.h" -#include "mcu_dsp_common.h" -#include "neteq_defines.h" -#include "neteq_error_codes.h" -#include "signal_processing_library.h" - -int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput, - uint32_t uw32_timeRec) -{ - RTPPacket_t RTPpacket[2]; - int i_k; - int i_ok = 0, i_No_Of_Payloads = 1; - int16_t flushed = 0; - int16_t codecPos; - int curr_Codec; - int16_t isREDPayload = 0; - int32_t temp_bufsize; - int is_sync_rtp = MCU_inst->av_sync && WebRtcNetEQ_IsSyncPayload( - RTPpacketInput->payload, RTPpacketInput->payloadLen); -#ifdef NETEQ_RED_CODEC - RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */ - RTPpacketPtr[0] = &RTPpacket[0]; - RTPpacketPtr[1] = &RTPpacket[1]; -#endif - - temp_bufsize = WebRtcNetEQ_PacketBufferGetSize(&MCU_inst->PacketBuffer_inst, - &MCU_inst->codec_DB_inst, - MCU_inst->av_sync); - /* - * Copy from input RTP packet to local copy - * (mainly to enable multiple payloads using RED) - */ - - WEBRTC_SPL_MEMCPY_W8(&RTPpacket[0], RTPpacketInput, sizeof(RTPPacket_t)); - - /* Reinitialize NetEq if it's needed (changed SSRC or first call) */ - - if ((RTPpacket[0].ssrc != MCU_inst->ssrc) || (MCU_inst->first_packet == 1)) - { - WebRtcNetEQ_RTCPInit(&MCU_inst->RTCP_inst, RTPpacket[0].seqNumber); - MCU_inst->first_packet = 0; - - /* Flush the buffer */ - WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); - - /* Store new SSRC */ - MCU_inst->ssrc = RTPpacket[0].ssrc; - - /* Update codecs */ - MCU_inst->timeStamp = RTPpacket[0].timeStamp; - MCU_inst->current_Payload = RTPpacket[0].payloadType; - - /*Set MCU to update codec on next SignalMCU call */ - MCU_inst->new_codec = 1; - - /* Reset timestamp scaling */ - MCU_inst->TSscalingInitialized = 0; - - } - - if (!is_sync_rtp) { /* Update only if it not sync packet. */ - /* Call RTCP statistics if it is not sync packet. */ - i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), - RTPpacket[0].seqNumber, - RTPpacket[0].timeStamp, uw32_timeRec); - } - - /* If Redundancy is supported and this is the redundancy payload, separate the payloads */ -#ifdef NETEQ_RED_CODEC - if (RTPpacket[0].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, - kDecoderRED)) - { - if (is_sync_rtp) - { - /* Sync packet should not have RED payload type. */ - return RECIN_SYNC_RTP_NOT_ACCEPTABLE; - } - - /* Split the payload into a main and a redundancy payloads */ - i_ok = WebRtcNetEQ_RedundancySplit(RTPpacketPtr, 2, &i_No_Of_Payloads); - if (i_ok < 0) - { - /* error returned */ - return i_ok; - } - - /* - * Only accept a few redundancies of the same type as the main data, - * AVT events and CNG. - */ - if ((i_No_Of_Payloads > 1) && (RTPpacket[0].payloadType != RTPpacket[1].payloadType) - && (RTPpacket[0].payloadType != WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, - kDecoderAVT)) && (RTPpacket[1].payloadType != WebRtcNetEQ_DbGetPayload( - &MCU_inst->codec_DB_inst, kDecoderAVT)) && (!WebRtcNetEQ_DbIsCNGPayload( - &MCU_inst->codec_DB_inst, RTPpacket[0].payloadType)) - && (!WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[1].payloadType))) - { - i_No_Of_Payloads = 1; - } - isREDPayload = 1; - } -#endif - - /* loop over the number of payloads */ - for (i_k = 0; i_k < i_No_Of_Payloads; i_k++) - { - - if (isREDPayload == 1) - { - RTPpacket[i_k].rcuPlCntr = i_k; - } - else - { - RTPpacket[i_k].rcuPlCntr = 0; - } - - /* Force update of SplitInfo if it's iLBC because of potential change between 20/30ms */ - if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, - kDecoderILBC) && !is_sync_rtp) /* Don't update if sync RTP. */ - { - i_ok = WebRtcNetEQ_DbGetSplitInfo( - &MCU_inst->PayloadSplit_inst, - (enum WebRtcNetEQDecoder) WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, - RTPpacket[i_k].payloadType), RTPpacket[i_k].payloadLen); - if (i_ok < 0) - { - /* error returned */ - return i_ok; - } - } - - /* Get information about timestamp scaling for this payload type */ - i_ok = WebRtcNetEQ_GetTimestampScaling(MCU_inst, RTPpacket[i_k].payloadType); - if (i_ok < 0) - { - /* error returned */ - return i_ok; - } - - if (MCU_inst->TSscalingInitialized == 0 && MCU_inst->scalingFactor != kTSnoScaling) - { - /* Must initialize scaling with current timestamps */ - MCU_inst->externalTS = RTPpacket[i_k].timeStamp; - MCU_inst->internalTS = RTPpacket[i_k].timeStamp; - MCU_inst->TSscalingInitialized = 1; - } - - /* Adjust timestamp if timestamp scaling is needed (e.g. SILK or G.722) */ - if (MCU_inst->TSscalingInitialized == 1) - { - uint32_t newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst, - RTPpacket[i_k].timeStamp); - - /* save the incoming timestamp for next time */ - MCU_inst->externalTS = RTPpacket[i_k].timeStamp; - - /* add the scaled difference to last scaled timestamp and save ... */ - MCU_inst->internalTS = newTS; - - RTPpacket[i_k].timeStamp = newTS; - } - - /* Is this a DTMF packet?*/ - if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, - kDecoderAVT)) - { - if (is_sync_rtp) - { - /* Sync RTP should not have AVT payload type. */ - return RECIN_SYNC_RTP_NOT_ACCEPTABLE; - } - -#ifdef NETEQ_ATEVENT_DECODE - if (MCU_inst->AVT_PlayoutOn) - { - i_ok = WebRtcNetEQ_DtmfInsertEvent(&MCU_inst->DTMF_inst, - RTPpacket[i_k].payload, RTPpacket[i_k].payloadLen, - RTPpacket[i_k].timeStamp); - if (i_ok != 0) - { - return i_ok; - } - } -#endif -#ifdef NETEQ_STEREO - if (MCU_inst->usingStereo == 0) - { - /* do not set this for DTMF packets when using stereo mode */ - MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; - } -#else - MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; -#endif - } - else if (WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, - RTPpacket[i_k].payloadType)) - { - /* Is this a CNG packet? how should we handle this?*/ -#ifdef NETEQ_CNG_CODEC - /* Get CNG sample rate */ - uint16_t fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst, - RTPpacket[i_k].payloadType); - if (is_sync_rtp) - { - /* Sync RTP should not have CNG payload type. */ - return RECIN_SYNC_RTP_NOT_ACCEPTABLE; - } - - /* Force sampling frequency to 32000 Hz CNG 48000 Hz. */ - /* TODO(tlegrand): remove limitation once ACM has full 48 kHz - * support. */ - if (fsCng > 32000) { - fsCng = 32000; - } - if ((fsCng != MCU_inst->fs) && (fsCng > 8000)) - { - /* - * We have received CNG with a different sample rate from what we are using - * now (must be > 8000, since we may use only one CNG type (default) for all - * frequencies). Flush buffer and signal new codec. - */ - WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); - MCU_inst->new_codec = 1; - MCU_inst->current_Codec = -1; - } - i_ok = WebRtcNetEQ_PacketBufferInsert(&MCU_inst->PacketBuffer_inst, - &RTPpacket[i_k], &flushed, MCU_inst->av_sync); - if (i_ok < 0) - { - return RECIN_CNG_ERROR; - } - MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; -#else /* NETEQ_CNG_CODEC not defined */ - return RECIN_UNKNOWNPAYLOAD; -#endif /* NETEQ_CNG_CODEC */ - } - else - { - /* Reinitialize the splitting if the payload and/or the payload length has changed */ - curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, - RTPpacket[i_k].payloadType); - if (curr_Codec != MCU_inst->current_Codec) - { - if (curr_Codec < 0) - { - return RECIN_UNKNOWNPAYLOAD; - } - if (is_sync_rtp) - { - /* Sync RTP should not cause codec change. */ - return RECIN_SYNC_RTP_CHANGED_CODEC; - } - MCU_inst->current_Codec = curr_Codec; - MCU_inst->current_Payload = RTPpacket[i_k].payloadType; - i_ok = WebRtcNetEQ_DbGetSplitInfo(&MCU_inst->PayloadSplit_inst, - (enum WebRtcNetEQDecoder) MCU_inst->current_Codec, - RTPpacket[i_k].payloadLen); - if (i_ok < 0) - { /* error returned */ - return i_ok; - } - WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); - MCU_inst->new_codec = 1; - } - - /* Parse the payload and insert it into the buffer */ - i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k], - &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, - &flushed, MCU_inst->av_sync); - if (i_ok < 0) - { - return i_ok; - } - if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF != 0) - { - /* first normal packet after CNG or DTMF */ - MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = -1; - } - } - /* Reset DSP timestamp etc. if packet buffer flushed */ - if (flushed) - { - MCU_inst->new_codec = 1; - } - } - - /* - * If not sync RTP, update Bandwidth Estimate. - * Only send the main payload to BWE. - */ - if (!is_sync_rtp && - (curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, - RTPpacket[0].payloadType)) >= 0) - { - codecPos = MCU_inst->codec_DB_inst.position[curr_Codec]; - if (MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos] != NULL) /* codec has BWE function */ - { - if (RTPpacket[0].starts_byte1) /* check for shifted byte alignment */ - { - /* re-align to 16-bit alignment */ - for (i_k = 0; i_k < RTPpacket[0].payloadLen; i_k++) - { - WEBRTC_SPL_SET_BYTE(RTPpacket[0].payload, - WEBRTC_SPL_GET_BYTE(RTPpacket[0].payload, i_k+1), - i_k); - } - RTPpacket[0].starts_byte1 = 0; - } - - MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos]( - MCU_inst->codec_DB_inst.codec_state[codecPos], - (const uint16_t *) RTPpacket[0].payload, - (int32_t) RTPpacket[0].payloadLen, RTPpacket[0].seqNumber, - (uint32_t) RTPpacket[0].timeStamp, (uint32_t) uw32_timeRec); - } - } - - if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == 0) - { - /* Calculate the total speech length carried in each packet */ - temp_bufsize = WebRtcNetEQ_PacketBufferGetSize( - &MCU_inst->PacketBuffer_inst, &MCU_inst->codec_DB_inst, - MCU_inst->av_sync) - temp_bufsize; - - if ((temp_bufsize > 0) && (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF - == 0) && (temp_bufsize - != MCU_inst->BufferStat_inst.Automode_inst.packetSpeechLenSamp)) - { - /* Change the auto-mode parameters if packet length has changed */ - WebRtcNetEQ_SetPacketSpeechLen(&(MCU_inst->BufferStat_inst.Automode_inst), - (int16_t) temp_bufsize, MCU_inst->fs); - } - - /* update statistics */ - if ((int32_t) (RTPpacket[0].timeStamp - MCU_inst->timeStamp) >= 0 - && !MCU_inst->new_codec) - { - /* - * Only update statistics if incoming packet is not older than last played out - * packet, and if new codec flag is not set. - */ - WebRtcNetEQ_UpdateIatStatistics(&MCU_inst->BufferStat_inst.Automode_inst, - MCU_inst->PacketBuffer_inst.maxInsertPositions, RTPpacket[0].seqNumber, - RTPpacket[0].timeStamp, MCU_inst->fs, - WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) MCU_inst->current_Codec), - (MCU_inst->NetEqPlayoutMode == kPlayoutStreaming)); - } - } - else if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == -1) - { - /* - * This is first "normal" packet after CNG or DTMF. - * Reset packet time counter and measure time until next packet, - * but don't update statistics. - */ - MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 0; - MCU_inst->BufferStat_inst.Automode_inst.packetIatCountSamp = 0; - } - return 0; - -} - -int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType) -{ - enum WebRtcNetEQDecoder codec; - int codecNumber; - - codecNumber = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, rtpPayloadType); - if (codecNumber < 0) - { - /* error */ - return codecNumber; - } - - /* cast to enumerator */ - codec = (enum WebRtcNetEQDecoder) codecNumber; - - /* - * The factor obtained below is the number with which the RTP timestamp must be - * multiplied to get the true sample count. - */ - switch (codec) - { - case kDecoderG722: - case kDecoderG722_2ch: - { - /* Use timestamp scaling with factor 2 (two output samples per RTP timestamp) */ - MCU_inst->scalingFactor = kTSscalingTwo; - break; - } - case kDecoderISACfb: - case kDecoderOpus: - { - /* We resample Opus internally to 32 kHz, and isac-fb decodes at - * 32 kHz, but timestamps are counted at 48 kHz. So there are two - * output samples per three RTP timestamp ticks. */ - MCU_inst->scalingFactor = kTSscalingTwoThirds; - break; - } - - case kDecoderAVT: - case kDecoderCNG: - { - /* TODO(tlegrand): remove scaling once ACM has full 48 kHz - * support. */ - uint16_t sample_freq = - WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst, - rtpPayloadType); - if (sample_freq == 48000) { - MCU_inst->scalingFactor = kTSscalingTwoThirds; - } - - /* For sample_freq <= 32 kHz, do not change the timestamp scaling - * settings. */ - break; - } - default: - { - /* do not use timestamp scaling */ - MCU_inst->scalingFactor = kTSnoScaling; - break; - } - } - return 0; -} - -uint32_t WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst, - uint32_t externalTS) -{ - int32_t timestampDiff; - uint32_t internalTS; - - /* difference between this and last incoming timestamp */ - timestampDiff = externalTS - MCU_inst->externalTS; - - switch (MCU_inst->scalingFactor) - { - case kTSscalingTwo: - { - /* multiply with 2 */ - timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 1); - break; - } - case kTSscalingTwoThirds: - { - /* multiply with 2/3 */ - timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 1); - timestampDiff = WebRtcSpl_DivW32W16(timestampDiff, 3); - break; - } - case kTSscalingFourThirds: - { - /* multiply with 4/3 */ - timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 2); - timestampDiff = WebRtcSpl_DivW32W16(timestampDiff, 3); - break; - } - default: - { - /* no scaling */ - } - } - - /* add the scaled difference to last scaled timestamp and save ... */ - internalTS = MCU_inst->internalTS + timestampDiff; - - return internalTS; -} - -uint32_t WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst, - uint32_t internalTS) -{ - int32_t timestampDiff; - uint32_t externalTS; - - /* difference between this and last incoming timestamp */ - timestampDiff = (int32_t) internalTS - MCU_inst->internalTS; - - switch (MCU_inst->scalingFactor) - { - case kTSscalingTwo: - { - /* divide by 2 */ - timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 1); - break; - } - case kTSscalingTwoThirds: - { - /* multiply with 3/2 */ - timestampDiff = WEBRTC_SPL_MUL_32_16(timestampDiff, 3); - timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 1); - break; - } - case kTSscalingFourThirds: - { - /* multiply with 3/4 */ - timestampDiff = WEBRTC_SPL_MUL_32_16(timestampDiff, 3); - timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 2); - break; - } - default: - { - /* no scaling */ - } - } - - /* add the scaled difference to last scaled timestamp and save ... */ - externalTS = MCU_inst->externalTS + timestampDiff; - - return externalTS; -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/recout.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/recout.c deleted file mode 100644 index 8f62007310c..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/recout.c +++ /dev/null @@ -1,1502 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of RecOut function, which is the main function for the audio output - * process. This function must be called (through the NetEQ API) once every 10 ms. - */ - -#include "dsp.h" - -#include <assert.h> -#include <string.h> /* to define NULL */ - -#include "signal_processing_library.h" - -#include "dsp_helpfunctions.h" -#include "neteq_error_codes.h" -#include "neteq_defines.h" -#include "mcu_dsp_common.h" - -/* Audio types */ -#define TYPE_SPEECH 1 -#define TYPE_CNG 2 - -#ifdef NETEQ_DELAY_LOGGING -#include "delay_logging.h" -#include <stdio.h> -#pragma message("*******************************************************************") -#pragma message("You have specified to use NETEQ_DELAY_LOGGING in the NetEQ library.") -#pragma message("Make sure that your test application supports this.") -#pragma message("*******************************************************************") -#endif - -/* Scratch usage: - - Type Name size startpos endpos - int16_t pw16_NetEqAlgorithm_buffer 1080*fs/8000 0 1080*fs/8000-1 - struct dspInfo 6 1080*fs/8000 1085*fs/8000 - - func WebRtcNetEQ_Normal 40+495*fs/8000 0 39+495*fs/8000 - func WebRtcNetEQ_Merge 40+496*fs/8000 0 39+496*fs/8000 - func WebRtcNetEQ_Expand 40+370*fs/8000 126*fs/800 39+496*fs/8000 - func WebRtcNetEQ_Accelerate 210 240*fs/8000 209+240*fs/8000 - func WebRtcNetEQ_BGNUpdate 69 480*fs/8000 68+480*fs/8000 - - Total: 1086*fs/8000 - */ - -#define SCRATCH_ALGORITHM_BUFFER 0 -#define SCRATCH_NETEQ_NORMAL 0 -#define SCRATCH_NETEQ_MERGE 0 - -#if (defined(NETEQ_48KHZ_WIDEBAND)) -#define SCRATCH_DSP_INFO 6480 -#define SCRATCH_NETEQ_ACCELERATE 1440 -#define SCRATCH_NETEQ_BGN_UPDATE 2880 -#define SCRATCH_NETEQ_EXPAND 756 -#elif (defined(NETEQ_32KHZ_WIDEBAND)) -#define SCRATCH_DSP_INFO 4320 -#define SCRATCH_NETEQ_ACCELERATE 960 -#define SCRATCH_NETEQ_BGN_UPDATE 1920 -#define SCRATCH_NETEQ_EXPAND 504 -#elif (defined(NETEQ_WIDEBAND)) -#define SCRATCH_DSP_INFO 2160 -#define SCRATCH_NETEQ_ACCELERATE 480 -#define SCRATCH_NETEQ_BGN_UPDATE 960 -#define SCRATCH_NETEQ_EXPAND 252 -#else /* NB */ -#define SCRATCH_DSP_INFO 1080 -#define SCRATCH_NETEQ_ACCELERATE 240 -#define SCRATCH_NETEQ_BGN_UPDATE 480 -#define SCRATCH_NETEQ_EXPAND 126 -#endif - -#if (defined(NETEQ_48KHZ_WIDEBAND)) -#define SIZE_SCRATCH_BUFFER 6516 -#elif (defined(NETEQ_32KHZ_WIDEBAND)) -#define SIZE_SCRATCH_BUFFER 4344 -#elif (defined(NETEQ_WIDEBAND)) -#define SIZE_SCRATCH_BUFFER 2172 -#else /* NB */ -#define SIZE_SCRATCH_BUFFER 1086 -#endif - -#ifdef NETEQ_DELAY_LOGGING -extern FILE *delay_fid2; /* file pointer to delay log file */ -extern uint32_t tot_received_packets; -#endif - - -int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, int16_t *pw16_outData, - int16_t *pw16_len, int16_t BGNonly, - int av_sync) -{ - - int16_t blockLen, payloadLen, len = 0, pos; - int16_t w16_tmp1, w16_tmp2, w16_tmp3, DataEnough; - int16_t *blockPtr; - int16_t MD = 0; - - int16_t speechType = TYPE_SPEECH; - uint16_t instr; - uint16_t uw16_tmp; -#ifdef SCRATCH - char pw8_ScratchBuffer[((SIZE_SCRATCH_BUFFER + 1) * 2)]; - int16_t *pw16_scratchPtr = (int16_t*) pw8_ScratchBuffer; - /* pad with 240*fs_mult to match the overflow guard below */ - int16_t pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6]; - int16_t *pw16_NetEqAlgorithm_buffer = pw16_scratchPtr - + SCRATCH_ALGORITHM_BUFFER; - DSP2MCU_info_t *dspInfo = (DSP2MCU_info_t*) (pw16_scratchPtr + SCRATCH_DSP_INFO); -#else - /* pad with 240*fs_mult to match the overflow guard below */ - int16_t pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6]; - int16_t pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE+240*6]; - DSP2MCU_info_t dspInfoStruct; - DSP2MCU_info_t *dspInfo = &dspInfoStruct; -#endif - int16_t fs_mult; - int borrowedSamples; - int oldBorrowedSamples; - int return_value = 0; - int16_t lastModeBGNonly = (inst->w16_mode & MODE_BGN_ONLY) != 0; /* check BGN flag */ - void *mainInstBackup = inst->main_inst; - -#ifdef NETEQ_DELAY_LOGGING - int temp_var; -#endif - int16_t dtmfValue = -1; - int16_t dtmfVolume = -1; - int playDtmf = 0; -#ifdef NETEQ_ATEVENT_DECODE - int dtmfSwitch = 0; -#endif -#ifdef NETEQ_STEREO - MasterSlaveInfo *msInfo = inst->msInfo; -#endif - int16_t *sharedMem = pw16_NetEqAlgorithm_buffer; /* Reuse memory SHARED_MEM_SIZE size */ - inst->pw16_readAddress = sharedMem; - inst->pw16_writeAddress = sharedMem; - - /* Get information about if there is one descriptor left */ - if (inst->codec_ptr_inst.funcGetMDinfo != NULL) - { - MD = inst->codec_ptr_inst.funcGetMDinfo(inst->codec_ptr_inst.codec_state); - if (MD > 0) - MD = 1; - else - MD = 0; - } - -#ifdef NETEQ_STEREO - if ((msInfo->msMode == NETEQ_SLAVE) && (inst->codec_ptr_inst.funcDecode != NULL)) - { - /* - * Valid function pointers indicate that we have decoded something, - * and that the timestamp information is correct. - */ - - /* Get the information from master to correct synchronization */ - uint32_t currentMasterTimestamp; - uint32_t currentSlaveTimestamp; - - currentMasterTimestamp = msInfo->endTimestamp - msInfo->samplesLeftWithOverlap; - currentSlaveTimestamp = inst->endTimestamp - (inst->endPosition - inst->curPosition); - - /* Partition the uint32_t space in three: [0 0.25) [0.25 0.75] (0.75 1] - * We consider a wrap to have occurred if the timestamps are in - * different edge partitions. - */ - if (currentSlaveTimestamp < 0x40000000 && - currentMasterTimestamp > 0xc0000000) { - // Slave has wrapped. - currentSlaveTimestamp += (0xffffffff - currentMasterTimestamp) + 1; - currentMasterTimestamp = 0; - } else if (currentMasterTimestamp < 0x40000000 && - currentSlaveTimestamp > 0xc0000000) { - // Master has wrapped. - currentMasterTimestamp += (0xffffffff - currentSlaveTimestamp) + 1; - currentSlaveTimestamp = 0; - } - - if (currentSlaveTimestamp < currentMasterTimestamp) - { - /* brute-force discard a number of samples to catch up */ - inst->curPosition += currentMasterTimestamp - currentSlaveTimestamp; - - } - else if (currentSlaveTimestamp > currentMasterTimestamp) - { - /* back off current position to slow down */ - inst->curPosition -= currentSlaveTimestamp - currentMasterTimestamp; - } - - /* make sure we have at least "overlap" samples left */ - inst->curPosition = WEBRTC_SPL_MIN(inst->curPosition, - inst->endPosition - inst->ExpandInst.w16_overlap); - - /* make sure we do not end up outside the speech history */ - inst->curPosition = WEBRTC_SPL_MAX(inst->curPosition, 0); - } -#endif - - /* Write status data to shared memory */ - dspInfo->playedOutTS = inst->endTimestamp; - dspInfo->samplesLeft = inst->endPosition - inst->curPosition - - inst->ExpandInst.w16_overlap; - dspInfo->MD = MD; - dspInfo->lastMode = inst->w16_mode; - dspInfo->frameLen = inst->w16_frameLen; - - /* Force update of codec if codec function is NULL */ - if (inst->codec_ptr_inst.funcDecode == NULL) - { - dspInfo->lastMode |= MODE_AWAITING_CODEC_PTR; - } - -#ifdef NETEQ_STEREO - if (msInfo->msMode == NETEQ_SLAVE && (msInfo->extraInfo == DTMF_OVERDUB - || msInfo->extraInfo == DTMF_ONLY)) - { - /* Signal that the master instance generated DTMF tones */ - dspInfo->lastMode |= MODE_MASTER_DTMF_SIGNAL; - } - - if (msInfo->msMode != NETEQ_MONO) - { - /* We are using stereo mode; signal this to MCU side */ - dspInfo->lastMode |= MODE_USING_STEREO; - } -#endif - - WEBRTC_SPL_MEMCPY_W8(inst->pw16_writeAddress,dspInfo,sizeof(DSP2MCU_info_t)); - - /* Signal MCU with "interrupt" call to main inst*/ -#ifdef NETEQ_STEREO - assert(msInfo != NULL); - if (msInfo->msMode == NETEQ_MASTER) - { - /* clear info to slave */ - WebRtcSpl_MemSetW16((int16_t *) msInfo, 0, - sizeof(MasterSlaveInfo) / sizeof(int16_t)); - /* re-set mode */ - msInfo->msMode = NETEQ_MASTER; - - /* Store some information to slave */ - msInfo->endTimestamp = inst->endTimestamp; - msInfo->samplesLeftWithOverlap = inst->endPosition - inst->curPosition; - } -#endif - - /* - * This call will trigger the MCU side to make a decision based on buffer contents and - * decision history. Instructions, encoded data and function pointers will be written - * to the shared memory. - */ - return_value = WebRtcNetEQ_DSP2MCUinterrupt((MainInst_t *) inst->main_inst, sharedMem); - - /* Read MCU data and instructions */ - instr = (uint16_t) (inst->pw16_readAddress[0] & 0xf000); - -#ifdef NETEQ_STEREO - if (msInfo->msMode == NETEQ_MASTER) - { - msInfo->instruction = instr; - } - else if (msInfo->msMode == NETEQ_SLAVE) - { - /* Nothing to do */ - } -#endif - - /* check for error returned from MCU side, if so, return error */ - if (return_value < 0) - { - inst->w16_mode = MODE_ERROR; - dspInfo->lastMode = MODE_ERROR; - return return_value; - } - - blockPtr = &((inst->pw16_readAddress)[3]); - - /* Check for DTMF payload flag */ - if ((inst->pw16_readAddress[0] & DSP_DTMF_PAYLOAD) != 0) - { - playDtmf = 1; - dtmfValue = blockPtr[1]; - dtmfVolume = blockPtr[2]; - blockPtr += 3; - -#ifdef NETEQ_STEREO - if (msInfo->msMode == NETEQ_MASTER) - { - /* signal to slave that master is using DTMF */ - msInfo->extraInfo = DTMF_OVERDUB; - } -#endif - } - - blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of int16_t */ - payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG); - blockPtr++; - - /* Do we have to change our decoder? */ - if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_NEW_CODEC) - { - WEBRTC_SPL_MEMCPY_W16(&inst->codec_ptr_inst,blockPtr,(payloadLen+1)>>1); - if (inst->codec_ptr_inst.codec_fs != 0) - { - return_value = WebRtcNetEQ_DSPInit(inst, inst->codec_ptr_inst.codec_fs); - if (return_value != 0) - { /* error returned */ - instr = DSP_INSTR_FADE_TO_BGN; /* emergency instruction */ - } -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS; - if ((fwrite(&temp_var, sizeof(int), - 1, delay_fid2) != 1) || - (fwrite(&inst->fs, sizeof(uint16_t), - 1, delay_fid2) != 1)) { - return -1; - } -#endif - } - - /* Copy it again since the init destroys this part */ - - WEBRTC_SPL_MEMCPY_W16(&inst->codec_ptr_inst,blockPtr,(payloadLen+1)>>1); - inst->endTimestamp = inst->codec_ptr_inst.timeStamp; - inst->videoSyncTimestamp = inst->codec_ptr_inst.timeStamp; - blockPtr += blockLen; - blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; - payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG); - blockPtr++; - if (inst->codec_ptr_inst.funcDecodeInit != NULL) - { - inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state); - } - -#ifdef NETEQ_CNG_CODEC - - /* Also update the CNG state as this might be uninitialized */ - - WEBRTC_SPL_MEMCPY_W16(&inst->CNG_Codec_inst,blockPtr,(payloadLen+1)>>1); - blockPtr += blockLen; - blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; - payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG); - blockPtr++; - if (inst->CNG_Codec_inst != NULL) - { - WebRtcCng_InitDec(inst->CNG_Codec_inst); - } -#endif - } - else if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_RESET) - { - /* Reset the current codec (but not DSP struct) */ - if (inst->codec_ptr_inst.funcDecodeInit != NULL) - { - inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state); - } - -#ifdef NETEQ_CNG_CODEC - /* And reset CNG */ - if (inst->CNG_Codec_inst != NULL) - { - WebRtcCng_InitDec(inst->CNG_Codec_inst); - } -#endif /*NETEQ_CNG_CODEC*/ - } - - fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs); - - /* Add late packet? */ - if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_ADD_LATE_PKT) - { - if (inst->codec_ptr_inst.funcAddLatePkt != NULL) - { - /* Only do this if the codec has support for Add Late Pkt */ - inst->codec_ptr_inst.funcAddLatePkt(inst->codec_ptr_inst.codec_state, blockPtr, - payloadLen); - } - blockPtr += blockLen; - blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of Word16 */ - payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG); - blockPtr++; - } - - /* Do we have to decode data? */ - if ((instr == DSP_INSTR_NORMAL) || (instr == DSP_INSTR_ACCELERATE) || (instr - == DSP_INSTR_MERGE) || (instr == DSP_INSTR_PREEMPTIVE_EXPAND)) - { - /* Do we need to update codec-internal PLC state? */ - if ((instr == DSP_INSTR_MERGE) && (inst->codec_ptr_inst.funcDecodePLC != NULL)) - { - len = 0; - len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state, - &pw16_decoded_buffer[len], 1); - } - len = 0; - - /* Do decoding */ - while ((blockLen > 0) && (len < (240 * fs_mult))) /* Guard somewhat against overflow */ - { - if (inst->codec_ptr_inst.funcDecode != NULL) - { - int16_t dec_Len; - if (!BGNonly) - { - /* Check if this is a sync payload. */ - if (av_sync && WebRtcNetEQ_IsSyncPayload(blockPtr, - payloadLen)) { - /* Zero-stuffing with same size as the last frame. */ - dec_Len = inst->w16_frameLen; - memset(&pw16_decoded_buffer[len], 0, dec_Len * - sizeof(pw16_decoded_buffer[len])); - } else { - /* Do decoding as normal - * - * blockPtr is pointing to payload, at this point, - * the most significant bit of *(blockPtr - 1) is a flag if - * set to 1 indicates that the following payload is the - * redundant payload. - */ - if (((*(blockPtr - 1) & DSP_CODEC_RED_FLAG) != 0) - && (inst->codec_ptr_inst.funcDecodeRCU != NULL)) - { - dec_Len = inst->codec_ptr_inst.funcDecodeRCU( - inst->codec_ptr_inst.codec_state, blockPtr, - payloadLen, &pw16_decoded_buffer[len], &speechType); - } - else - { - /* Regular decoding. */ - dec_Len = inst->codec_ptr_inst.funcDecode( - inst->codec_ptr_inst.codec_state, blockPtr, - payloadLen, &pw16_decoded_buffer[len], &speechType); - } - } - } - else - { - /* - * Background noise mode: don't decode, just produce the same length BGN. - * Don't call Expand for BGN here, since Expand uses the memory where the - * bitstreams are stored (sharemem). - */ - dec_Len = inst->w16_frameLen; - } - - if (dec_Len > 0) - { - len += dec_Len; - /* Update frameLen */ - inst->w16_frameLen = dec_Len; - } - else if (dec_Len < 0) - { - /* Error */ - len = -1; - break; - } - /* - * Sanity check (although we might still write outside memory when this - * happens...) - */ - if (len > NETEQ_MAX_FRAME_SIZE) - { - WebRtcSpl_MemSetW16(pw16_outData, 0, inst->timestampsPerCall); - *pw16_len = inst->timestampsPerCall; - inst->w16_mode = MODE_ERROR; - dspInfo->lastMode = MODE_ERROR; - return RECOUT_ERROR_DECODED_TOO_MUCH; - } - - /* Verify that instance was not corrupted by decoder */ - if (mainInstBackup != inst->main_inst) - { - /* Instance is corrupt */ - return CORRUPT_INSTANCE; - } - - } - blockPtr += blockLen; - blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of Word16 */ - payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG); - blockPtr++; - } - - if (len < 0) - { - len = 0; - inst->endTimestamp += inst->w16_frameLen; /* advance one frame */ - if (inst->codec_ptr_inst.funcGetErrorCode != NULL) - { - return_value = -inst->codec_ptr_inst.funcGetErrorCode( - inst->codec_ptr_inst.codec_state); - } - else - { - return_value = RECOUT_ERROR_DECODING; - } - instr = DSP_INSTR_FADE_TO_BGN; - } - if (speechType != TYPE_CNG) - { - /* - * Don't increment timestamp if codec returned CNG speech type - * since in this case, the MCU side will increment the CNGplayedTS counter. - */ - inst->endTimestamp += len; - } - } - else if (instr == DSP_INSTR_NORMAL_ONE_DESC) - { - if (inst->codec_ptr_inst.funcDecode != NULL) - { - len = inst->codec_ptr_inst.funcDecode(inst->codec_ptr_inst.codec_state, NULL, 0, - pw16_decoded_buffer, &speechType); -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - if (fwrite(&inst->endTimestamp, sizeof(uint32_t), - 1, delay_fid2) != 1) { - return -1; - } - if (fwrite(&dspInfo->samplesLeft, sizeof(uint16_t), - 1, delay_fid2) != 1) { - return -1; - } - tot_received_packets++; -#endif - } - if (speechType != TYPE_CNG) - { - /* - * Don't increment timestamp if codec returned CNG speech type - * since in this case, the MCU side will increment the CNGplayedTS counter. - */ - inst->endTimestamp += len; - } - - /* Verify that instance was not corrupted by decoder */ - if (mainInstBackup != inst->main_inst) - { - /* Instance is corrupt */ - return CORRUPT_INSTANCE; - } - - if (len <= 0) - { - len = 0; - if (inst->codec_ptr_inst.funcGetErrorCode != NULL) - { - return_value = -inst->codec_ptr_inst.funcGetErrorCode( - inst->codec_ptr_inst.codec_state); - } - else - { - return_value = RECOUT_ERROR_DECODING; - } - if ((inst->codec_ptr_inst.funcDecodeInit != NULL) - && (inst->codec_ptr_inst.codec_state != NULL)) - { - /* Reinitialize codec state as something is obviously wrong */ - inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state); - } - inst->endTimestamp += inst->w16_frameLen; /* advance one frame */ - instr = DSP_INSTR_FADE_TO_BGN; - } - } - - if (len == 0 && lastModeBGNonly) /* no new data */ - { - BGNonly = 1; /* force BGN this time too */ - } - -#ifdef NETEQ_VAD - if ((speechType == TYPE_CNG) /* decoder responded with codec-internal CNG */ - || ((instr == DSP_INSTR_DO_RFC3389CNG) && (blockLen > 0)) /* ... or, SID frame */ - || (inst->fs > 16000)) /* ... or, if not NB or WB */ - { - /* disable post-decode VAD upon first sign of send-side DTX/VAD active, or if SWB */ - inst->VADInst.VADEnabled = 0; - inst->VADInst.VADDecision = 1; /* set to always active, just to be on the safe side */ - inst->VADInst.SIDintervalCounter = 0; /* reset SID interval counter */ - } - else if (!inst->VADInst.VADEnabled) /* VAD disabled and no SID/CNG data observed this time */ - { - inst->VADInst.SIDintervalCounter++; /* increase counter */ - } - - /* check for re-enabling the VAD */ - if (inst->VADInst.SIDintervalCounter >= POST_DECODE_VAD_AUTO_ENABLE) - { - /* - * It's been a while since the last CNG/SID frame was observed => re-enable VAD. - * (Do not care to look for a VAD instance, since this is done inside the init - * function) - */ - WebRtcNetEQ_InitVAD(&inst->VADInst, inst->fs); - } - - if (len > 0 /* if we decoded any data */ - && inst->VADInst.VADEnabled /* and VAD enabled */ - && inst->fs <= 16000) /* can only do VAD for NB and WB */ - { - int VADframeSize; /* VAD frame size in ms */ - int VADSamplePtr = 0; - - inst->VADInst.VADDecision = 0; - - if (inst->VADInst.VADFunction != NULL) /* make sure that VAD function is provided */ - { - /* divide the data into groups, as large as possible */ - for (VADframeSize = 30; VADframeSize >= 10; VADframeSize -= 10) - { - /* loop through 30, 20, 10 */ - - while (inst->VADInst.VADDecision == 0 - && len - VADSamplePtr >= VADframeSize * fs_mult * 8) - { - /* - * Only continue until first active speech found, and as long as there is - * one VADframeSize left. - */ - - /* call VAD with new decoded data */ - inst->VADInst.VADDecision |= inst->VADInst.VADFunction( - inst->VADInst.VADState, (int) inst->fs, - (int16_t *) &pw16_decoded_buffer[VADSamplePtr], - (VADframeSize * fs_mult * 8)); - - VADSamplePtr += VADframeSize * fs_mult * 8; /* increment sample counter */ - } - } - } - else - { /* VAD function is NULL */ - inst->VADInst.VADDecision = 1; /* set decision to active */ - inst->VADInst.VADEnabled = 0; /* disable VAD since we have no VAD function */ - } - - } -#endif /* NETEQ_VAD */ - - /* Adjust timestamp if needed */ - uw16_tmp = (uint16_t) inst->pw16_readAddress[1]; - inst->endTimestamp += (((uint32_t) uw16_tmp) << 16); - uw16_tmp = (uint16_t) inst->pw16_readAddress[2]; - inst->endTimestamp += uw16_tmp; - - if (BGNonly && len > 0) - { - /* - * If BGN mode, we did not produce any data at decoding. - * Do it now instead. - */ - - WebRtcNetEQ_GenerateBGN(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_decoded_buffer, len); - } - - /* Switch on the instruction received from the MCU side. */ - switch (instr) - { - case DSP_INSTR_NORMAL: - - /* Allow for signal processing to apply gain-back etc */ - WebRtcNetEQ_Normal(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_NORMAL, -#endif - pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len); - - /* If last packet was decoded as a inband CNG set mode to CNG instead */ - if ((speechType == TYPE_CNG) || ((inst->w16_mode == MODE_CODEC_INTERNAL_CNG) - && (len == 0))) - { - inst->w16_mode = MODE_CODEC_INTERNAL_CNG; - } - -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - break; - case DSP_INSTR_NORMAL_ONE_DESC: - - /* Allow for signal processing to apply gain-back etc */ - WebRtcNetEQ_Normal(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_NORMAL, -#endif - pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len); -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - inst->w16_mode = MODE_ONE_DESCRIPTOR; - break; - case DSP_INSTR_MERGE: -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = -len; -#endif - /* Call Merge with history*/ - return_value = WebRtcNetEQ_Merge(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_MERGE, -#endif - pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len); - - if (return_value < 0) - { - /* error */ - return return_value; - } - -#ifdef NETEQ_DELAY_LOGGING - temp_var += len; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - /* If last packet was decoded as a inband CNG set mode to CNG instead */ - if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG; -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - break; - - case DSP_INSTR_EXPAND: - len = 0; - pos = 0; - while ((inst->endPosition - inst->curPosition - inst->ExpandInst.w16_overlap + pos) - < (inst->timestampsPerCall)) - { - return_value = WebRtcNetEQ_Expand(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_NetEqAlgorithm_buffer, &len, BGNonly); - if (return_value < 0) - { - /* error */ - return return_value; - } - - /* - * Update buffer, but only end part (otherwise expand state is destroyed - * since it reuses speechBuffer[] memory - */ - - WEBRTC_SPL_MEMMOVE_W16(inst->pw16_speechHistory, - inst->pw16_speechHistory + len, - (inst->w16_speechHistoryLen-len)); - WEBRTC_SPL_MEMCPY_W16(&inst->pw16_speechHistory[inst->w16_speechHistoryLen-len], - pw16_NetEqAlgorithm_buffer, len); - - inst->curPosition -= len; - - /* Update variables for VQmon */ - inst->w16_concealedTS += len; -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = len; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - len = 0; /* already written the data, so do not write it again further down. */ - } -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - break; - - case DSP_INSTR_ACCELERATE: - if (len < 3 * 80 * fs_mult) - { - /* We need to move data from the speechBuffer[] in order to get 30 ms */ - borrowedSamples = 3 * 80 * fs_mult - len; - - WEBRTC_SPL_MEMMOVE_W16(&pw16_decoded_buffer[borrowedSamples], - pw16_decoded_buffer, len); - WEBRTC_SPL_MEMCPY_W16(pw16_decoded_buffer, - &(inst->speechBuffer[inst->endPosition-borrowedSamples]), - borrowedSamples); - - return_value = WebRtcNetEQ_Accelerate(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE, -#endif - pw16_decoded_buffer, 3 * inst->timestampsPerCall, - pw16_NetEqAlgorithm_buffer, &len, BGNonly); - - if (return_value < 0) - { - /* error */ - return return_value; - } - - /* Copy back samples to the buffer */ - if (len < borrowedSamples) - { - /* - * This destroys the beginning of the buffer, but will not cause any - * problems - */ - - WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-borrowedSamples], - pw16_NetEqAlgorithm_buffer, len); - WEBRTC_SPL_MEMMOVE_W16(&inst->speechBuffer[borrowedSamples-len], - inst->speechBuffer, - (inst->endPosition-(borrowedSamples-len))); - - inst->curPosition += (borrowedSamples - len); -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = 3 * inst->timestampsPerCall - len; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - len = 0; - } - else - { - WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-borrowedSamples], - pw16_NetEqAlgorithm_buffer, borrowedSamples); - WEBRTC_SPL_MEMMOVE_W16(pw16_NetEqAlgorithm_buffer, - &pw16_NetEqAlgorithm_buffer[borrowedSamples], - (len-borrowedSamples)); -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = 3 * inst->timestampsPerCall - len; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - len = len - borrowedSamples; - } - - } - else - { -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = len; -#endif - return_value = WebRtcNetEQ_Accelerate(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE, -#endif - pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len, BGNonly); - - if (return_value < 0) - { - /* error */ - return return_value; - } - -#ifdef NETEQ_DELAY_LOGGING - temp_var -= len; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - } - /* If last packet was decoded as a inband CNG set mode to CNG instead */ - if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG; -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - break; - - case DSP_INSTR_DO_RFC3389CNG: -#ifdef NETEQ_CNG_CODEC - if (blockLen > 0) - { - if (WebRtcCng_UpdateSid(inst->CNG_Codec_inst, (uint8_t*) blockPtr, - payloadLen) < 0) - { - /* error returned from CNG function */ - return_value = -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst); - len = inst->timestampsPerCall; - WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len); - break; - } - } - - if (BGNonly) - { - /* Get data from BGN function instead of CNG */ - len = WebRtcNetEQ_GenerateBGN(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_NetEqAlgorithm_buffer, inst->timestampsPerCall); - if (len != inst->timestampsPerCall) - { - /* this is not good, treat this as an error */ - return_value = -1; - } - } - else - { - return_value = WebRtcNetEQ_Cng(inst, pw16_NetEqAlgorithm_buffer, - inst->timestampsPerCall); - } - len = inst->timestampsPerCall; - inst->ExpandInst.w16_consecExp = 0; - inst->w16_mode = MODE_RFC3389CNG; -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - - if (return_value < 0) - { - /* error returned */ - WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len); - } - - break; -#else - return FAULTY_INSTRUCTION; -#endif - case DSP_INSTR_DO_CODEC_INTERNAL_CNG: - /* - * This represents the case when there is no transmission and the decoder should - * do internal CNG. - */ - len = 0; - if (inst->codec_ptr_inst.funcDecode != NULL && !BGNonly) - { - len = inst->codec_ptr_inst.funcDecode(inst->codec_ptr_inst.codec_state, - blockPtr, 0, pw16_decoded_buffer, &speechType); - } - else - { - /* get BGN data */ - len = WebRtcNetEQ_GenerateBGN(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_decoded_buffer, inst->timestampsPerCall); - } - WebRtcNetEQ_Normal(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_NORMAL, -#endif - pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len); - inst->w16_mode = MODE_CODEC_INTERNAL_CNG; - inst->ExpandInst.w16_consecExp = 0; - break; - - case DSP_INSTR_DTMF_GENERATE: -#ifdef NETEQ_ATEVENT_DECODE - dtmfSwitch = 0; - if ((inst->w16_mode != MODE_DTMF) && (inst->DTMFInst.reinit == 0)) - { - /* Special case; see below. - * We must catch this before calling DTMFGenerate, - * since reinit is set to 0 in that call. - */ - dtmfSwitch = 1; - } - - len = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume, - pw16_NetEqAlgorithm_buffer, inst->fs, -1); - if (len < 0) - { - /* error occurred */ - return_value = len; - len = inst->timestampsPerCall; - WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len); - } - - if (dtmfSwitch == 1) - { - /* - * This is the special case where the previous operation was DTMF overdub. - * but the current instruction is "regular" DTMF. We must make sure that the - * DTMF does not have any discontinuities. The first DTMF sample that we - * generate now must be played out immediately, wherefore it must be copied to - * the speech buffer. - */ - - /* - * Generate extra DTMF data to fill the space between - * curPosition and endPosition - */ - int16_t tempLen; - - tempLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume, - &pw16_NetEqAlgorithm_buffer[len], inst->fs, - inst->endPosition - inst->curPosition); - if (tempLen < 0) - { - /* error occurred */ - return_value = tempLen; - len = inst->endPosition - inst->curPosition; - WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, - inst->endPosition - inst->curPosition); - } - - /* Add to total length */ - len += tempLen; - - /* Overwrite the "future" part of the speech buffer with the new DTMF data */ - - WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->curPosition], - pw16_NetEqAlgorithm_buffer, - inst->endPosition - inst->curPosition); - - /* Shuffle the remaining data to the beginning of algorithm buffer */ - len -= (inst->endPosition - inst->curPosition); - WEBRTC_SPL_MEMMOVE_W16(pw16_NetEqAlgorithm_buffer, - &pw16_NetEqAlgorithm_buffer[inst->endPosition - inst->curPosition], - len); - } - - inst->endTimestamp += inst->timestampsPerCall; - inst->DTMFInst.reinit = 0; - inst->ExpandInst.w16_consecExp = 0; - inst->w16_mode = MODE_DTMF; - BGNonly = 0; /* override BGN only and let DTMF through */ - - playDtmf = 0; /* set to zero because the DTMF is already in the Algorithm buffer */ - /* - * If playDtmf is 1, an extra DTMF vector will be generated and overdubbed - * on the output. - */ - -#ifdef NETEQ_STEREO - if (msInfo->msMode == NETEQ_MASTER) - { - /* signal to slave that master is using DTMF only */ - msInfo->extraInfo = DTMF_ONLY; - } -#endif - - break; -#else - inst->w16_mode = MODE_ERROR; - dspInfo->lastMode = MODE_ERROR; - return FAULTY_INSTRUCTION; -#endif - - case DSP_INSTR_DO_ALTERNATIVE_PLC: - if (inst->codec_ptr_inst.funcDecodePLC != 0) - { - len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state, - pw16_NetEqAlgorithm_buffer, 1); - } - else - { - len = inst->timestampsPerCall; - /* ZeroStuffing... */ - WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len); - /* By not advancing the timestamp, NetEq inserts samples. */ - inst->statInst.addedSamples += len; - } - inst->ExpandInst.w16_consecExp = 0; - break; - case DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS: - if (inst->codec_ptr_inst.funcDecodePLC != 0) - { - len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state, - pw16_NetEqAlgorithm_buffer, 1); - } - else - { - len = inst->timestampsPerCall; - /* ZeroStuffing... */ - WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len); - } - inst->ExpandInst.w16_consecExp = 0; - inst->endTimestamp += len; - break; - case DSP_INSTR_DO_AUDIO_REPETITION: - len = inst->timestampsPerCall; - /* copy->paste... */ - WEBRTC_SPL_MEMCPY_W16(pw16_NetEqAlgorithm_buffer, - &inst->speechBuffer[inst->endPosition-len], len); - inst->ExpandInst.w16_consecExp = 0; - break; - case DSP_INSTR_DO_AUDIO_REPETITION_INC_TS: - len = inst->timestampsPerCall; - /* copy->paste... */ - WEBRTC_SPL_MEMCPY_W16(pw16_NetEqAlgorithm_buffer, - &inst->speechBuffer[inst->endPosition-len], len); - inst->ExpandInst.w16_consecExp = 0; - inst->endTimestamp += len; - break; - - case DSP_INSTR_PREEMPTIVE_EXPAND: - if (len < 3 * inst->timestampsPerCall) - { - /* borrow samples from sync buffer if necessary */ - borrowedSamples = 3 * inst->timestampsPerCall - len; /* borrow this many samples */ - /* calculate how many of these are already played out */ - oldBorrowedSamples = WEBRTC_SPL_MAX(0, - borrowedSamples - (inst->endPosition - inst->curPosition)); - WEBRTC_SPL_MEMMOVE_W16(&pw16_decoded_buffer[borrowedSamples], - pw16_decoded_buffer, len); - WEBRTC_SPL_MEMCPY_W16(pw16_decoded_buffer, - &(inst->speechBuffer[inst->endPosition-borrowedSamples]), - borrowedSamples); - } - else - { - borrowedSamples = 0; - oldBorrowedSamples = 0; - } - -#ifdef NETEQ_DELAY_LOGGING - w16_tmp1 = len; -#endif - /* do the expand */ - return_value = WebRtcNetEQ_PreEmptiveExpand(inst, -#ifdef SCRATCH - /* use same scratch memory as Accelerate */ - pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE, -#endif - pw16_decoded_buffer, len + borrowedSamples, oldBorrowedSamples, - pw16_NetEqAlgorithm_buffer, &len, BGNonly); - - if (return_value < 0) - { - /* error */ - return return_value; - } - - if (borrowedSamples > 0) - { - /* return borrowed samples */ - - /* Copy back to last part of speechBuffer from beginning of output buffer */ - WEBRTC_SPL_MEMCPY_W16( &(inst->speechBuffer[inst->endPosition-borrowedSamples]), - pw16_NetEqAlgorithm_buffer, - borrowedSamples); - - len -= borrowedSamples; /* remove the borrowed samples from new total length */ - - /* Move to beginning of output buffer from end of output buffer */ - WEBRTC_SPL_MEMMOVE_W16( pw16_NetEqAlgorithm_buffer, - &pw16_NetEqAlgorithm_buffer[borrowedSamples], - len); - } - -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = len - w16_tmp1; /* number of samples added */ - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - /* If last packet was decoded as inband CNG, set mode to CNG instead */ - if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG; -#ifdef NETEQ_ATEVENT_DECODE - if (playDtmf == 0) - { - inst->DTMFInst.reinit = 1; - } -#endif - break; - - case DSP_INSTR_FADE_TO_BGN: - { - int tempReturnValue; - /* do not overwrite return_value, since it likely contains an error code */ - - /* calculate interpolation length */ - w16_tmp3 = WEBRTC_SPL_MIN(inst->endPosition - inst->curPosition, - inst->timestampsPerCall); - /* check that it will fit in pw16_NetEqAlgorithm_buffer */ - if (w16_tmp3 + inst->w16_frameLen > NETEQ_MAX_OUTPUT_SIZE) - { - w16_tmp3 = NETEQ_MAX_OUTPUT_SIZE - inst->w16_frameLen; - } - - /* call Expand */ - len = inst->timestampsPerCall + inst->ExpandInst.w16_overlap; - pos = 0; - - tempReturnValue = WebRtcNetEQ_Expand(inst, -#ifdef SCRATCH - pw16_scratchPtr + SCRATCH_NETEQ_EXPAND, -#endif - pw16_NetEqAlgorithm_buffer, &len, 1); - - if (tempReturnValue < 0) - { - /* error */ - /* this error value will override return_value */ - return tempReturnValue; - } - - pos += len; /* got len samples from expand */ - - /* copy to fill the demand */ - while (pos + len <= inst->w16_frameLen + w16_tmp3) - { - WEBRTC_SPL_MEMCPY_W16(&pw16_NetEqAlgorithm_buffer[pos], - pw16_NetEqAlgorithm_buffer, len); - pos += len; - } - - /* fill with fraction of the expand vector if needed */ - if (pos < inst->w16_frameLen + w16_tmp3) - { - WEBRTC_SPL_MEMCPY_W16(&pw16_NetEqAlgorithm_buffer[pos], pw16_NetEqAlgorithm_buffer, - inst->w16_frameLen + w16_tmp3 - pos); - } - - len = inst->w16_frameLen + w16_tmp3; /* truncate any surplus samples since we don't want these */ - - /* - * Mix with contents in sync buffer. Find largest power of two that is less than - * interpolate length divide 16384 with this number; result is in w16_tmp2. - */ - w16_tmp1 = 2; - w16_tmp2 = 16384; - while (w16_tmp1 <= w16_tmp3) - { - w16_tmp2 >>= 1; /* divide with 2 */ - w16_tmp1 <<= 1; /* increase with a factor of 2 */ - } - - w16_tmp1 = 0; - pos = 0; - while (w16_tmp1 < 16384) - { - inst->speechBuffer[inst->curPosition + pos] - = - (int16_t) WEBRTC_SPL_RSHIFT_W32( - WEBRTC_SPL_MUL_16_16( inst->speechBuffer[inst->endPosition - w16_tmp3 + pos], - 16384-w16_tmp1 ) + - WEBRTC_SPL_MUL_16_16( pw16_NetEqAlgorithm_buffer[pos], w16_tmp1 ), - 14 ); - w16_tmp1 += w16_tmp2; - pos++; - } - - /* overwrite remainder of speech buffer */ - - WEBRTC_SPL_MEMCPY_W16( &inst->speechBuffer[inst->endPosition - w16_tmp3 + pos], - &pw16_NetEqAlgorithm_buffer[pos], w16_tmp3 - pos); - - len -= w16_tmp3; - /* shift algorithm buffer */ - - WEBRTC_SPL_MEMMOVE_W16( pw16_NetEqAlgorithm_buffer, - &pw16_NetEqAlgorithm_buffer[w16_tmp3], - len ); - - /* Update variables for VQmon */ - inst->w16_concealedTS += len; - - inst->w16_mode = MODE_FADE_TO_BGN; -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } - temp_var = len; - if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) { - return -1; - } -#endif - - break; - } - - default: - inst->w16_mode = MODE_ERROR; - dspInfo->lastMode = MODE_ERROR; - return FAULTY_INSTRUCTION; - } /* end of grand switch */ - - /* Copy data directly to output buffer */ - - w16_tmp2 = 0; - if ((inst->endPosition + len - inst->curPosition - inst->ExpandInst.w16_overlap) - >= inst->timestampsPerCall) - { - w16_tmp2 = inst->endPosition - inst->curPosition; - w16_tmp2 = WEBRTC_SPL_MAX(w16_tmp2, 0); /* Additional error protection, just in case */ - w16_tmp1 = WEBRTC_SPL_MIN(w16_tmp2, inst->timestampsPerCall); - w16_tmp2 = inst->timestampsPerCall - w16_tmp1; - WEBRTC_SPL_MEMCPY_W16(pw16_outData, &inst->speechBuffer[inst->curPosition], w16_tmp1); - WEBRTC_SPL_MEMCPY_W16(&pw16_outData[w16_tmp1], pw16_NetEqAlgorithm_buffer, w16_tmp2); - DataEnough = 1; - } - else - { - DataEnough = 0; - } - - if (playDtmf != 0) - { -#ifdef NETEQ_ATEVENT_DECODE - int16_t outDataIndex = 0; - int16_t overdubLen = -1; /* default len */ - int16_t dtmfLen; - - /* - * Overdub the output with DTMF. Note that this is not executed if the - * DSP_INSTR_DTMF_GENERATE operation is performed above. - */ - if (inst->DTMFInst.lastDtmfSample - inst->curPosition > 0) - { - /* special operation for transition from "DTMF only" to "DTMF overdub" */ - outDataIndex - = WEBRTC_SPL_MIN(inst->DTMFInst.lastDtmfSample - inst->curPosition, - inst->timestampsPerCall); - overdubLen = inst->timestampsPerCall - outDataIndex; - } - - dtmfLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume, - &pw16_outData[outDataIndex], inst->fs, overdubLen); - if (dtmfLen < 0) - { - /* error occurred */ - return_value = dtmfLen; - } - inst->DTMFInst.reinit = 0; -#else - inst->w16_mode = MODE_ERROR; - dspInfo->lastMode = MODE_ERROR; - return FAULTY_INSTRUCTION; -#endif - } - - /* - * Shuffle speech buffer to allow more data. Move data from pw16_NetEqAlgorithm_buffer - * to speechBuffer. - */ - if (instr != DSP_INSTR_EXPAND) - { - w16_tmp1 = WEBRTC_SPL_MIN(inst->endPosition, len); - WEBRTC_SPL_MEMMOVE_W16(inst->speechBuffer, inst->speechBuffer + w16_tmp1, - (inst->endPosition-w16_tmp1)); - WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-w16_tmp1], - &pw16_NetEqAlgorithm_buffer[len-w16_tmp1], w16_tmp1); -#ifdef NETEQ_ATEVENT_DECODE - /* Update index to end of DTMF data in speech buffer */ - if (instr == DSP_INSTR_DTMF_GENERATE) - { - /* We have written DTMF data to the end of speech buffer */ - inst->DTMFInst.lastDtmfSample = inst->endPosition; - } - else if (inst->DTMFInst.lastDtmfSample > 0) - { - /* The end of DTMF data in speech buffer has been shuffled */ - inst->DTMFInst.lastDtmfSample -= w16_tmp1; - } -#endif - /* - * Update the BGN history if last operation was not expand (nor Merge, Accelerate - * or Pre-emptive expand, to save complexity). - */ - if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_MERGE) - && (inst->w16_mode != MODE_SUCCESS_ACCELERATE) && (inst->w16_mode - != MODE_LOWEN_ACCELERATE) && (inst->w16_mode != MODE_SUCCESS_PREEMPTIVE) - && (inst->w16_mode != MODE_LOWEN_PREEMPTIVE) && (inst->w16_mode - != MODE_FADE_TO_BGN) && (inst->w16_mode != MODE_DTMF) && (!BGNonly)) - { - WebRtcNetEQ_BGNUpdate(inst -#ifdef SCRATCH - , pw16_scratchPtr + SCRATCH_NETEQ_BGN_UPDATE -#endif - ); - } - } - else /* instr == DSP_INSTR_EXPAND */ - { - /* Nothing should be done since data is already copied to output. */ - } - - inst->curPosition -= len; - - /* - * Extra protection in case something should go totally wrong in terms of sizes... - * If everything is ok this should NEVER happen. - */ - if (inst->curPosition < -inst->timestampsPerCall) - { - inst->curPosition = -inst->timestampsPerCall; - } - - if ((instr != DSP_INSTR_EXPAND) && (instr != DSP_INSTR_MERGE) && (instr - != DSP_INSTR_FADE_TO_BGN)) - { - /* Reset concealed TS parameter if it does not seem to have been flushed */ - if (inst->w16_concealedTS > inst->timestampsPerCall) - { - inst->w16_concealedTS = 0; - } - } - - /* - * Double-check that we actually have 10 ms to play. If we haven't, there has been a - * serious error.The decoder might have returned way too few samples - */ - if (!DataEnough) - { - /* This should not happen. Set outdata to zeros, and return error. */ - WebRtcSpl_MemSetW16(pw16_outData, 0, inst->timestampsPerCall); - *pw16_len = inst->timestampsPerCall; - inst->w16_mode = MODE_ERROR; - dspInfo->lastMode = MODE_ERROR; - return RECOUT_ERROR_SAMPLEUNDERRUN; - } - - /* - * Update Videosync timestamp (this special timestamp is needed since the endTimestamp - * stops during CNG and Expand periods. - */ - if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_RFC3389CNG)) - { - uint32_t uw32_tmpTS; - uw32_tmpTS = inst->endTimestamp - (inst->endPosition - inst->curPosition); - if ((int32_t) (uw32_tmpTS - inst->videoSyncTimestamp) > 0) - { - inst->videoSyncTimestamp = uw32_tmpTS; - } - } - else - { - inst->videoSyncTimestamp += inst->timestampsPerCall; - } - - /* After this, regardless of what has happened, deliver 10 ms of future data */ - inst->curPosition += inst->timestampsPerCall; - *pw16_len = inst->timestampsPerCall; - - /* Remember if BGNonly was used */ - if (BGNonly) - { - inst->w16_mode |= MODE_BGN_ONLY; - } - - return return_value; -} - -#undef SCRATCH_ALGORITHM_BUFFER -#undef SCRATCH_NETEQ_NORMAL -#undef SCRATCH_NETEQ_MERGE -#undef SCRATCH_NETEQ_BGN_UPDATE -#undef SCRATCH_NETEQ_EXPAND -#undef SCRATCH_DSP_INFO -#undef SCRATCH_NETEQ_ACCELERATE -#undef SIZE_SCRATCH_BUFFER diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.c deleted file mode 100644 index d1ce934bcfb..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.c +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of RTCP statistics reporting. - */ - -#include "rtcp.h" - -#include <string.h> - -#include "signal_processing_library.h" - -int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo) -{ - /* - * Initialize everything to zero and then set the start values for the RTP packet stream. - */ - WebRtcSpl_MemSetW16((int16_t*) RTCP_inst, 0, - sizeof(WebRtcNetEQ_RTCP_t) / sizeof(int16_t)); - RTCP_inst->base_seq = uw16_seqNo; - RTCP_inst->max_seq = uw16_seqNo; - return 0; -} - -int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo, - uint32_t uw32_timeStamp, uint32_t uw32_recTime) -{ - int16_t w16_SeqDiff; - int32_t w32_TimeDiff; - int32_t w32_JitterDiff; - - /* - * Update number of received packets, and largest packet number received. - */ - RTCP_inst->received++; - w16_SeqDiff = uw16_seqNo - RTCP_inst->max_seq; - if (w16_SeqDiff >= 0) - { - if (uw16_seqNo < RTCP_inst->max_seq) - { - /* Wrap around detected */ - RTCP_inst->cycles++; - } - RTCP_inst->max_seq = uw16_seqNo; - } - - /* Calculate Jitter, and update previous timestamps */ - /* Note that the value in RTCP_inst->jitter is in Q4. */ - if (RTCP_inst->received > 1) - { - w32_TimeDiff = (uw32_recTime - (uw32_timeStamp - RTCP_inst->transit)); - w32_TimeDiff = WEBRTC_SPL_ABS_W32(w32_TimeDiff); - w32_JitterDiff = WEBRTC_SPL_LSHIFT_W16(w32_TimeDiff, 4) - RTCP_inst->jitter; - RTCP_inst->jitter = RTCP_inst->jitter + WEBRTC_SPL_RSHIFT_W32((w32_JitterDiff + 8), 4); - } - RTCP_inst->transit = (uw32_timeStamp - uw32_recTime); - return 0; -} - -int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst, - uint16_t *puw16_fraction_lost, - uint32_t *puw32_cum_lost, uint32_t *puw32_ext_max, - uint32_t *puw32_jitter, int16_t doNotReset) -{ - uint32_t uw32_exp_nr, uw32_exp_interval, uw32_rec_interval; - int32_t w32_lost; - - /* Extended highest sequence number received */ - *puw32_ext_max - = (uint32_t) WEBRTC_SPL_LSHIFT_W32((uint32_t)RTCP_inst->cycles, 16) - + RTCP_inst->max_seq; - - /* - * Calculate expected number of packets and compare it to the number of packets that - * were actually received => the cumulative number of packets lost can be extracted. - */ - uw32_exp_nr = *puw32_ext_max - RTCP_inst->base_seq + 1; - if (RTCP_inst->received == 0) - { - /* no packets received, assume none lost */ - *puw32_cum_lost = 0; - } - else if (uw32_exp_nr > RTCP_inst->received) - { - *puw32_cum_lost = uw32_exp_nr - RTCP_inst->received; - if (*puw32_cum_lost > (uint32_t) 0xFFFFFF) - { - *puw32_cum_lost = 0xFFFFFF; - } - } - else - { - *puw32_cum_lost = 0; - } - - /* Fraction lost (Since last report) */ - uw32_exp_interval = uw32_exp_nr - RTCP_inst->exp_prior; - if (!doNotReset) - { - RTCP_inst->exp_prior = uw32_exp_nr; - } - uw32_rec_interval = RTCP_inst->received - RTCP_inst->rec_prior; - if (!doNotReset) - { - RTCP_inst->rec_prior = RTCP_inst->received; - } - w32_lost = (int32_t) (uw32_exp_interval - uw32_rec_interval); - if (uw32_exp_interval == 0 || w32_lost <= 0 || RTCP_inst->received == 0) - { - *puw16_fraction_lost = 0; - } - else - { - *puw16_fraction_lost = (uint16_t) (WEBRTC_SPL_LSHIFT_W32(w32_lost, 8) - / uw32_exp_interval); - } - if (*puw16_fraction_lost > 0xFF) - { - *puw16_fraction_lost = 0xFF; - } - - /* Inter-arrival jitter */ - *puw32_jitter = (RTCP_inst->jitter) >> 4; /* scaling from Q4 */ - return 0; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/rtcp.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.cc index f9dcf449198..cf8e0280bb2 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/rtcp.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.cc @@ -8,11 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/rtcp.h" +#include "webrtc/modules/audio_coding/neteq/rtcp.h" -#include <algorithm> #include <string.h> +#include <algorithm> + #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/interface/module_common_types.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h index 5e066eb38f2..2a765efa588 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtcp.h @@ -8,95 +8,51 @@ * be found in the AUTHORS file in the root of the source tree. */ -/* - * RTCP statistics reporting. - */ - -#ifndef RTCP_H -#define RTCP_H - -#include "typedefs.h" - -typedef struct -{ - uint16_t cycles; /* The number of wrap-arounds for the sequence number */ - uint16_t max_seq; /* The maximum sequence number received - (starts from 0 again after wrap around) */ - uint16_t base_seq; /* The sequence number of the first packet that arrived */ - uint32_t received; /* The number of packets that has been received */ - uint32_t rec_prior; /* Number of packets received when last report was generated */ - uint32_t exp_prior; /* Number of packets that should have been received if no - packets were lost. Stored value from last report. */ - uint32_t jitter; /* Jitter statistics at this instance (calculated according to RFC) */ - int32_t transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */ -} WebRtcNetEQ_RTCP_t; - -/**************************************************************************** - * WebRtcNetEQ_RTCPInit(...) - * - * This function calculates the parameters that are needed for the RTCP - * report. - * - * Input: - * - RTCP_inst : RTCP instance, that contains information about the - * packets that have been received etc. - * - seqNo : Packet number of the first received frame. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo); - -/**************************************************************************** - * WebRtcNetEQ_RTCPUpdate(...) - * - * This function calculates the parameters that are needed for the RTCP - * report. - * - * Input: - * - RTCP_inst : RTCP instance, that contains information about the - * packets that have been received etc. - * - seqNo : Packet number of the first received frame. - * - timeStamp : Time stamp from the RTP header. - * - recTime : Time (in RTP timestamps) when this packet was received. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo, - uint32_t uw32_timeStamp, uint32_t uw32_recTime); - -/**************************************************************************** - * WebRtcNetEQ_RTCPGetStats(...) - * - * This function calculates the parameters that are needed for the RTCP - * report. - * - * Input: - * - RTCP_inst : RTCP instance, that contains information about the - * packets that have been received etc. - * - doNotReset : If non-zero, the fraction lost statistics will not - * be reset. - * - * Output: - * - RTCP_inst : Updated RTCP information (some statistics are - * reset when generating this report) - * - fraction_lost : Number of lost RTP packets divided by the number of - * expected packets, since the last RTCP Report. - * - cum_lost : Cumulative number of lost packets during this - * session. - * - ext_max : Extended highest sequence number received. - * - jitter : Inter-arrival jitter. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst, - uint16_t *puw16_fraction_lost, - uint32_t *puw32_cum_lost, uint32_t *puw32_ext_max, - uint32_t *puw32_jitter, int16_t doNotReset); - -#endif +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +// Forward declaration. +struct RTPHeader; + +class Rtcp { + public: + Rtcp() { + Init(0); + } + + ~Rtcp() {} + + // Resets the RTCP statistics, and sets the first received sequence number. + void Init(uint16_t start_sequence_number); + + // Updates the RTCP statistics with a new received packet. + void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp); + + // Returns the current RTCP statistics. If |no_reset| is true, the statistics + // are not reset, otherwise they are. + void GetStatistics(bool no_reset, RtcpStatistics* stats); + + private: + uint16_t cycles_; // The number of wrap-arounds for the sequence number. + uint16_t max_seq_no_; // The maximum sequence number received. Starts over + // from 0 after wrap-around. + uint16_t base_seq_no_; // The sequence number of the first received packet. + uint32_t received_packets_; // The number of packets that have been received. + uint32_t received_packets_prior_; // Number of packets received when last + // report was generated. + uint32_t expected_prior_; // Expected number of packets, at the time of the + // last report. + uint32_t jitter_; // Current jitter value. + int32_t transit_; // Clock difference for previous packet. + + DISALLOW_COPY_AND_ASSIGN(Rtcp); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtp.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtp.c deleted file mode 100644 index 6ab5944b5aa..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtp.c +++ /dev/null @@ -1,240 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * RTP related functions. - */ - -#include "rtp.h" - -#include "typedefs.h" /* to define endianness */ - -#include "neteq_error_codes.h" - -int WebRtcNetEQ_RTPPayloadInfo(int16_t* pw16_Datagram, int i_DatagramLen, - RTPPacket_t* RTPheader) -{ - int i_P, i_X, i_CC, i_startPosition; - int i_IPver; - int i_extlength = -1; /* Default value is there is no extension */ - int i_padlength = 0; /* Default value if there is no padding */ - - if (i_DatagramLen < 12) - { - return RTP_TOO_SHORT_PACKET; - } - -#ifdef WEBRTC_ARCH_BIG_ENDIAN - i_IPver = (((uint16_t) (pw16_Datagram[0] & 0xC000)) >> 14); /* Extract the version */ - i_P = (((uint16_t) (pw16_Datagram[0] & 0x2000)) >> 13); /* Extract the P bit */ - i_X = (((uint16_t) (pw16_Datagram[0] & 0x1000)) >> 12); /* Extract the X bit */ - i_CC = ((uint16_t) (pw16_Datagram[0] >> 8) & 0xF); /* Get the CC number */ - RTPheader->payloadType = pw16_Datagram[0] & 0x7F; /* Get the coder type */ - RTPheader->seqNumber = pw16_Datagram[1]; /* Get the sequence number */ - RTPheader->timeStamp = ((((uint32_t) ((uint16_t) pw16_Datagram[2])) << 16) - | (uint16_t) (pw16_Datagram[3])); /* Get timestamp */ - RTPheader->ssrc = (((uint32_t) pw16_Datagram[4]) << 16) - + (((uint32_t) pw16_Datagram[5])); /* Get the SSRC */ - - if (i_X == 1) - { - /* Extension header exists. Find out how many int32_t it consists of. */ - i_extlength = pw16_Datagram[7 + 2 * i_CC]; - } - if (i_P == 1) - { - /* Padding exists. Find out how many bytes the padding consists of. */ - if (i_DatagramLen & 0x1) - { - /* odd number of bytes => last byte in higher byte */ - i_padlength = (((uint16_t) pw16_Datagram[i_DatagramLen >> 1]) >> 8); - } - else - { - /* even number of bytes => last byte in lower byte */ - i_padlength = ((pw16_Datagram[(i_DatagramLen >> 1) - 1]) & 0xFF); - } - } -#else /* WEBRTC_ARCH_LITTLE_ENDIAN */ - i_IPver = (((uint16_t) (pw16_Datagram[0] & 0xC0)) >> 6); /* Extract the IP version */ - i_P = (((uint16_t) (pw16_Datagram[0] & 0x20)) >> 5); /* Extract the P bit */ - i_X = (((uint16_t) (pw16_Datagram[0] & 0x10)) >> 4); /* Extract the X bit */ - i_CC = (uint16_t) (pw16_Datagram[0] & 0xF); /* Get the CC number */ - RTPheader->payloadType = (pw16_Datagram[0] >> 8) & 0x7F; /* Get the coder type */ - RTPheader->seqNumber = (((((uint16_t) pw16_Datagram[1]) >> 8) & 0xFF) - | (((uint16_t) (pw16_Datagram[1] & 0xFF)) << 8)); /* Get the packet number */ - RTPheader->timeStamp = ((((uint16_t) pw16_Datagram[2]) & 0xFF) << 24) - | ((((uint16_t) pw16_Datagram[2]) & 0xFF00) << 8) - | ((((uint16_t) pw16_Datagram[3]) >> 8) & 0xFF) - | ((((uint16_t) pw16_Datagram[3]) & 0xFF) << 8); /* Get timestamp */ - RTPheader->ssrc = ((((uint16_t) pw16_Datagram[4]) & 0xFF) << 24) - | ((((uint16_t) pw16_Datagram[4]) & 0xFF00) << 8) - | ((((uint16_t) pw16_Datagram[5]) >> 8) & 0xFF) - | ((((uint16_t) pw16_Datagram[5]) & 0xFF) << 8); /* Get the SSRC */ - - if (i_X == 1) - { - /* Extension header exists. Find out how many int32_t it consists of. */ - i_extlength = (((((uint16_t) pw16_Datagram[7 + 2 * i_CC]) >> 8) & 0xFF) - | (((uint16_t) (pw16_Datagram[7 + 2 * i_CC] & 0xFF)) << 8)); - } - if (i_P == 1) - { - /* Padding exists. Find out how many bytes the padding consists of. */ - if (i_DatagramLen & 0x1) - { - /* odd number of bytes => last byte in higher byte */ - i_padlength = (pw16_Datagram[i_DatagramLen >> 1] & 0xFF); - } - else - { - /* even number of bytes => last byte in lower byte */ - i_padlength = (((uint16_t) pw16_Datagram[(i_DatagramLen >> 1) - 1]) >> 8); - } - } -#endif - - i_startPosition = 12 + 4 * (i_extlength + 1) + 4 * i_CC; - RTPheader->payload = &pw16_Datagram[i_startPosition >> 1]; - RTPheader->payloadLen = i_DatagramLen - i_startPosition - i_padlength; - RTPheader->starts_byte1 = 0; - - if ((i_IPver != 2) || (RTPheader->payloadLen <= 0) || (RTPheader->payloadLen >= 16000) - || (i_startPosition < 12) || (i_startPosition > i_DatagramLen)) - { - return RTP_CORRUPT_PACKET; - } - - return 0; -} - -#ifdef NETEQ_RED_CODEC - -int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads, - int *i_No_Of_Payloads) -{ - const int16_t *pw16_data = RTPheader[0]->payload; /* Pointer to the data */ - uint16_t uw16_offsetTimeStamp = 65535, uw16_secondPayload = 65535; - int i_blockLength, i_k; - int i_discardedBlockLength = 0; - int singlePayload = 0; - -#ifdef WEBRTC_ARCH_BIG_ENDIAN - if ((pw16_data[0] & 0x8000) == 0) - { - /* Only one payload in this packet*/ - singlePayload = 1; - /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */ - i_blockLength = -4; - RTPheader[0]->payloadType = ((((uint16_t)pw16_data[0]) & 0x7F00) >> 8); - } - else - { - /* Discard all but the two last payloads. */ - while (((pw16_data[2] & 0x8000) != 0) && - (pw16_data<((RTPheader[0]->payload)+((RTPheader[0]->payloadLen+1)>>1)))) - { - i_discardedBlockLength += (4+(((uint16_t)pw16_data[1]) & 0x3FF)); - pw16_data+=2; - } - if (pw16_data>=(RTPheader[0]->payload+((RTPheader[0]->payloadLen+1)>>1))) - { - return RED_SPLIT_ERROR2; /* Error, we are outside the packet */ - } - singlePayload = 0; /* the packet contains more than one payload */ - uw16_secondPayload = ((((uint16_t)pw16_data[0]) & 0x7F00) >> 8); - RTPheader[0]->payloadType = ((((uint16_t)pw16_data[2]) & 0x7F00) >> 8); - uw16_offsetTimeStamp = ((((uint16_t)pw16_data[0]) & 0xFF) << 6) + - ((((uint16_t)pw16_data[1]) & 0xFC00) >> 10); - i_blockLength = (((uint16_t)pw16_data[1]) & 0x3FF); - } -#else /* WEBRTC_ARCH_LITTLE_ENDIAN */ - if ((pw16_data[0] & 0x80) == 0) - { - /* Only one payload in this packet */ - singlePayload = 1; - /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */ - i_blockLength = -4; - RTPheader[0]->payloadType = (((uint16_t) pw16_data[0]) & 0x7F); - } - else - { - /* Discard all but the two last payloads. */ - while (((pw16_data[2] & 0x80) != 0) && (pw16_data < ((RTPheader[0]->payload) - + ((RTPheader[0]->payloadLen + 1) >> 1)))) - { - i_discardedBlockLength += (4 + ((((uint16_t) pw16_data[1]) & 0x3) << 8) - + ((((uint16_t) pw16_data[1]) & 0xFF00) >> 8)); - pw16_data += 2; - } - if (pw16_data >= (RTPheader[0]->payload + ((RTPheader[0]->payloadLen + 1) >> 1))) - { - return RED_SPLIT_ERROR2; /* Error, we are outside the packet */; - } - singlePayload = 0; /* the packet contains more than one payload */ - uw16_secondPayload = (((uint16_t) pw16_data[0]) & 0x7F); - RTPheader[0]->payloadType = (((uint16_t) pw16_data[2]) & 0x7F); - uw16_offsetTimeStamp = ((((uint16_t) pw16_data[0]) & 0xFF00) >> 2) - + ((((uint16_t) pw16_data[1]) & 0xFC) >> 2); - i_blockLength = ((((uint16_t) pw16_data[1]) & 0x3) << 8) - + ((((uint16_t) pw16_data[1]) & 0xFF00) >> 8); - } -#endif - - if (i_MaximumPayloads < 2 || singlePayload == 1) - { - /* Reject the redundancy; or no redundant payload present. */ - for (i_k = 1; i_k < i_MaximumPayloads; i_k++) - { - RTPheader[i_k]->payloadType = -1; - RTPheader[i_k]->payloadLen = 0; - } - - /* update the pointer for the main data */ - pw16_data = &pw16_data[(5 + i_blockLength) >> 1]; - RTPheader[0]->starts_byte1 = (5 + i_blockLength) & 0x1; - RTPheader[0]->payloadLen = RTPheader[0]->payloadLen - (i_blockLength + 5) - - i_discardedBlockLength; - RTPheader[0]->payload = pw16_data; - - *i_No_Of_Payloads = 1; - - } - else - { - /* Redundancy accepted, put the redundancy in second RTPheader. */ - RTPheader[1]->payloadType = uw16_secondPayload; - RTPheader[1]->payload = &pw16_data[5 >> 1]; - RTPheader[1]->starts_byte1 = 5 & 0x1; - RTPheader[1]->seqNumber = RTPheader[0]->seqNumber; - RTPheader[1]->timeStamp = RTPheader[0]->timeStamp - uw16_offsetTimeStamp; - RTPheader[1]->ssrc = RTPheader[0]->ssrc; - RTPheader[1]->payloadLen = i_blockLength; - - /* Modify first RTP packet, so that it contains the main data. */ - RTPheader[0]->payload = &pw16_data[(5 + i_blockLength) >> 1]; - RTPheader[0]->starts_byte1 = (5 + i_blockLength) & 0x1; - RTPheader[0]->payloadLen = RTPheader[0]->payloadLen - (i_blockLength + 5) - - i_discardedBlockLength; - - /* Clear the following payloads. */ - for (i_k = 2; i_k < i_MaximumPayloads; i_k++) - { - RTPheader[i_k]->payloadType = -1; - RTPheader[i_k]->payloadLen = 0; - } - - *i_No_Of_Payloads = 2; - } - return 0; -} - -#endif - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtp.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/rtp.h deleted file mode 100644 index 4642eaef770..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/rtp.h +++ /dev/null @@ -1,78 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * RTP data struct and related functions. - */ - -#ifndef RTP_H -#define RTP_H - -#include "typedefs.h" - -#include "codec_db.h" - -typedef struct -{ - uint16_t seqNumber; - uint32_t timeStamp; - uint32_t ssrc; - int payloadType; - const int16_t *payload; - int16_t payloadLen; - int16_t starts_byte1; - int16_t rcuPlCntr; -} RTPPacket_t; - -/**************************************************************************** - * WebRtcNetEQ_RTPPayloadInfo(...) - * - * Converts a datagram into an RTP header struct. - * - * Input: - * - Datagram : UDP datagram from the network - * - DatagramLen : Length in bytes of the datagram - * - * Output: - * - RTPheader : Structure with the datagram info - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RTPPayloadInfo(int16_t* pw16_Datagram, int i_DatagramLen, - RTPPacket_t* RTPheader); - -/**************************************************************************** - * WebRtcNetEQ_RedundancySplit(...) - * - * Splits a Redundancy RTP struct into two RTP structs. User has to check - * that it's really the redundancy payload. No such check is done inside this - * function. - * - * Input: - * - RTPheader : First header holds the whole RTP packet (with the redundancy payload) - * - MaximumPayloads: - * The maximum number of RTP payloads that should be - * extracted (1+maximum_no_of_Redundancies). - * - * Output: - * - RTPheader : First header holds the main RTP data, while 2..N - * holds the redundancy data. - * - No_Of - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads, - int *i_No_Of_Payloads); - -#endif diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/set_fs.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/set_fs.c deleted file mode 100644 index ac974548e61..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/set_fs.c +++ /dev/null @@ -1,78 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Function were the sample rate is set. - */ - -#include "mcu.h" - -#include "dtmf_buffer.h" -#include "neteq_error_codes.h" - -int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, uint16_t fs) -{ - int16_t ok = 0; - - switch (fs) - { - case 8000: - { -#ifdef NETEQ_ATEVENT_DECODE - ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 8000, 560); -#endif - inst->timestampsPerCall = inst->millisecondsPerCall * 8; - break; - } - -#ifdef NETEQ_WIDEBAND - case 16000: - { -#ifdef NETEQ_ATEVENT_DECODE - ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 16000, 1120); -#endif - inst->timestampsPerCall = inst->millisecondsPerCall * 16; - break; - } -#endif - -#ifdef NETEQ_32KHZ_WIDEBAND - case 32000: - { -#ifdef NETEQ_ATEVENT_DECODE - ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 32000, 2240); -#endif - inst->timestampsPerCall = inst->millisecondsPerCall * 32; - break; - } -#endif - -#ifdef NETEQ_48KHZ_WIDEBAND - case 48000: - { -#ifdef NETEQ_ATEVENT_DECODE - ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 48000, 3360); -#endif - inst->timestampsPerCall = inst->millisecondsPerCall * 48; - break; - } -#endif - - default: - { - /* Not supported yet */ - return CODEC_DB_UNSUPPORTED_FS; - } - } /* end switch */ - - inst->fs = fs; - - return ok; -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c deleted file mode 100644 index b795ec30e38..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c +++ /dev/null @@ -1,820 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Signal the MCU that data is available and ask for a RecOut decision. - */ - -#include "mcu.h" - -#include <string.h> - -#include "signal_processing_library.h" - -#include "automode.h" -#include "dtmf_buffer.h" -#include "mcu_dsp_common.h" -#include "neteq_error_codes.h" - -#ifdef NETEQ_DELAY_LOGGING -#include "delay_logging.h" -#include <stdio.h> - -extern FILE *delay_fid2; /* file pointer to delay log file */ -#endif - - -/* - * Update the frame size, if we can. - */ -static int WebRtcNetEQ_UpdatePackSizeSamples(MCUInst_t* inst, int buffer_pos, - int payload_type, - int pack_size_samples) { - if (buffer_pos >= 0) { - int codec_pos; - codec_pos = WebRtcNetEQ_DbGetCodec(&inst->codec_DB_inst, payload_type); - if (codec_pos >= 0) { - codec_pos = inst->codec_DB_inst.position[codec_pos]; - if (codec_pos >= 0) { - int temp_packet_size_samples = WebRtcNetEQ_PacketBufferGetPacketSize( - &inst->PacketBuffer_inst, buffer_pos, &inst->codec_DB_inst, - codec_pos, pack_size_samples, inst->av_sync); - if (temp_packet_size_samples > 0) - return temp_packet_size_samples; - return pack_size_samples; - } - } - } - return pack_size_samples; -} - -/* - * Signals the MCU that DSP status data is available. - */ -int WebRtcNetEQ_SignalMcu(MCUInst_t *inst) -{ - - int i_bufferpos, i_res; - uint16_t uw16_instr; - DSP2MCU_info_t dspInfo; - int16_t *blockPtr, blockLen; - uint32_t uw32_availableTS; - RTPPacket_t temp_pkt; - int32_t w32_bufsize, w32_tmp; - int16_t payloadType = -1; - int16_t wantedNoOfTimeStamps; - int32_t totalTS; - int16_t oldPT, latePacketExist = 0; - uint32_t oldTS, prevTS, uw32_tmp; - uint16_t prevSeqNo; - int16_t nextSeqNoAvail; - int16_t fs_mult, w16_tmp; - int16_t lastModeBGNonly = 0; -#ifdef NETEQ_DELAY_LOGGING - int temp_var; -#endif - int playDtmf = 0; - - fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000); - - /* Increment counter since last statistics report */ - inst->lastReportTS += inst->timestampsPerCall; - - /* Increment waiting time for all packets. */ - WebRtcNetEQ_IncrementWaitingTimes(&inst->PacketBuffer_inst); - - /* Read info from DSP so we now current status */ - - WEBRTC_SPL_MEMCPY_W8(&dspInfo,inst->pw16_readAddress,sizeof(DSP2MCU_info_t)); - - /* Set blockPtr to first payload block */ - blockPtr = &inst->pw16_writeAddress[3]; - - /* Clear instruction word and number of lost samples (2*int16_t) */ - inst->pw16_writeAddress[0] = 0; - inst->pw16_writeAddress[1] = 0; - inst->pw16_writeAddress[2] = 0; - - if ((dspInfo.lastMode & MODE_AWAITING_CODEC_PTR) != 0) - { - /* - * Make sure state is adjusted so that a codec update is - * performed when first packet arrives. - */ - if (inst->new_codec != 1) - { - inst->current_Codec = -1; - } - dspInfo.lastMode = (dspInfo.lastMode ^ MODE_AWAITING_CODEC_PTR); - } - -#ifdef NETEQ_STEREO - if ((dspInfo.lastMode & MODE_MASTER_DTMF_SIGNAL) != 0) - { - playDtmf = 1; /* force DTMF decision */ - dspInfo.lastMode = (dspInfo.lastMode ^ MODE_MASTER_DTMF_SIGNAL); - } - - if ((dspInfo.lastMode & MODE_USING_STEREO) != 0) - { - if (inst->usingStereo == 0) - { - /* stereo mode changed; reset automode instance to re-synchronize statistics */ - WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst), - inst->PacketBuffer_inst.maxInsertPositions); - } - inst->usingStereo = 1; - dspInfo.lastMode = (dspInfo.lastMode ^ MODE_USING_STEREO); - } - else - { - inst->usingStereo = 0; - } -#endif - - /* detect if BGN_ONLY flag is set in lastMode */ - if ((dspInfo.lastMode & MODE_BGN_ONLY) != 0) - { - lastModeBGNonly = 1; /* remember flag */ - dspInfo.lastMode ^= MODE_BGN_ONLY; /* clear the flag */ - } - - if ((dspInfo.lastMode == MODE_RFC3389CNG) || (dspInfo.lastMode == MODE_CODEC_INTERNAL_CNG) - || (dspInfo.lastMode == MODE_EXPAND)) - { - /* - * If last mode was CNG (or Expand, since this could be covering up for a lost CNG - * packet), increase the CNGplayedTS counter. - */ - inst->BufferStat_inst.uw32_CNGplayedTS += inst->timestampsPerCall; - - if (dspInfo.lastMode == MODE_RFC3389CNG) - { - /* remember that RFC3389CNG is on (needed if CNG is interrupted by DTMF) */ - inst->BufferStat_inst.w16_cngOn = CNG_RFC3389_ON; - } - else if (dspInfo.lastMode == MODE_CODEC_INTERNAL_CNG) - { - /* remember that internal CNG is on (needed if CNG is interrupted by DTMF) */ - inst->BufferStat_inst.w16_cngOn = CNG_INTERNAL_ON; - } - - } - - /* Update packet size from previously decoded packet */ - if (dspInfo.frameLen > 0) - { - inst->PacketBuffer_inst.packSizeSamples = dspInfo.frameLen; - } - - /* Look for late packet (unless codec has changed) */ - if (inst->new_codec != 1) - { - if (WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) inst->current_Codec)) - { - WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst, - inst->timeStamp, &uw32_availableTS, &i_bufferpos, 1, &payloadType); - if ((inst->new_codec != 1) && (inst->timeStamp == uw32_availableTS) - && (inst->timeStamp < dspInfo.playedOutTS) && (i_bufferpos != -1) - && (WebRtcNetEQ_DbGetPayload(&(inst->codec_DB_inst), - (enum WebRtcNetEQDecoder) inst->current_Codec) == payloadType)) - { - int waitingTime; - temp_pkt.payload = blockPtr + 1; - i_res = WebRtcNetEQ_PacketBufferExtract(&inst->PacketBuffer_inst, &temp_pkt, - i_bufferpos, &waitingTime); - if (i_res < 0) - { /* error returned */ - return i_res; - } - WebRtcNetEQ_StoreWaitingTime(inst, waitingTime); - *blockPtr = temp_pkt.payloadLen; - /* set the flag if this is a redundant payload */ - if (temp_pkt.rcuPlCntr > 0) - { - *blockPtr = (*blockPtr) | (DSP_CODEC_RED_FLAG); - } - blockPtr += ((temp_pkt.payloadLen + 1) >> 1) + 1; - - /* - * Close the data with a zero size block, in case we will not write any - * more data. - */ - *blockPtr = 0; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff) - | DSP_CODEC_ADD_LATE_PKT; - latePacketExist = 1; - } - } - } - - i_res = WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst, - dspInfo.playedOutTS, &uw32_availableTS, &i_bufferpos, (inst->new_codec == 0), - &payloadType); - if (i_res < 0) - { /* error returned */ - return i_res; - } - - if (inst->BufferStat_inst.w16_cngOn == CNG_RFC3389_ON) - { - /* - * Because of timestamp peculiarities, we have to "manually" disallow using a CNG - * packet with the same timestamp as the one that was last played. This can happen - * when using redundancy and will cause the timing to shift. - */ - while (i_bufferpos != -1 && WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, - payloadType) && dspInfo.playedOutTS >= uw32_availableTS) - { - - /* Don't use this packet, discard it */ - inst->PacketBuffer_inst.payloadType[i_bufferpos] = -1; - inst->PacketBuffer_inst.payloadLengthBytes[i_bufferpos] = 0; - inst->PacketBuffer_inst.numPacketsInBuffer--; - - /* Check buffer again */ - WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst, - dspInfo.playedOutTS, &uw32_availableTS, &i_bufferpos, (inst->new_codec == 0), - &payloadType); - } - } - - /* Check packet buffer */ - w32_bufsize = WebRtcNetEQ_PacketBufferGetSize(&inst->PacketBuffer_inst, - &inst->codec_DB_inst, inst->av_sync); - - if (dspInfo.lastMode == MODE_SUCCESS_ACCELERATE || dspInfo.lastMode - == MODE_LOWEN_ACCELERATE || dspInfo.lastMode == MODE_SUCCESS_PREEMPTIVE - || dspInfo.lastMode == MODE_LOWEN_PREEMPTIVE) - { - /* Subtract (dspInfo.samplesLeft + inst->timestampsPerCall) from sampleMemory */ - inst->BufferStat_inst.Automode_inst.sampleMemory -= dspInfo.samplesLeft - + inst->timestampsPerCall; - } - - /* calculate total current buffer size (in ms*8), including sync buffer */ - w32_bufsize = WebRtcSpl_DivW32W16((w32_bufsize + dspInfo.samplesLeft), fs_mult); - -#ifdef NETEQ_ATEVENT_DECODE - /* DTMF data will affect the decision */ - if (WebRtcNetEQ_DtmfDecode(&inst->DTMF_inst, blockPtr + 1, blockPtr + 2, - dspInfo.playedOutTS + inst->BufferStat_inst.uw32_CNGplayedTS) > 0) - { - playDtmf = 1; - - /* Flag DTMF payload */ - inst->pw16_writeAddress[0] = inst->pw16_writeAddress[0] | DSP_DTMF_PAYLOAD; - - /* Block Length in bytes */ - blockPtr[0] = 4; - /* Advance to next payload position */ - blockPtr += 3; - } -#endif - - /* Update the frame size, if we can. */ - inst->PacketBuffer_inst.packSizeSamples = - WebRtcNetEQ_UpdatePackSizeSamples(inst, i_bufferpos, payloadType, - inst->PacketBuffer_inst.packSizeSamples); - /* Update statistics and make decision */ - uw16_instr = WebRtcNetEQ_BufstatsDecision(&inst->BufferStat_inst, - inst->PacketBuffer_inst.packSizeSamples, w32_bufsize, dspInfo.playedOutTS, - uw32_availableTS, i_bufferpos == -1, - WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType), dspInfo.lastMode, - inst->NetEqPlayoutMode, inst->timestampsPerCall, inst->NoOfExpandCalls, fs_mult, - lastModeBGNonly, playDtmf); - - /* Check if time to reset loss counter */ - if (inst->lastReportTS > WEBRTC_SPL_UMUL(inst->fs, MAX_LOSS_REPORT_PERIOD)) - { - /* reset loss counter */ - WebRtcNetEQ_ResetMcuInCallStats(inst); - } - - /* Check sync buffer size */ - if ((dspInfo.samplesLeft >= inst->timestampsPerCall) && (uw16_instr - != BUFSTATS_DO_ACCELERATE) && (uw16_instr != BUFSTATS_DO_MERGE) && (uw16_instr - != BUFSTATS_DO_PREEMPTIVE_EXPAND)) - { - *blockPtr = 0; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) | DSP_INSTR_NORMAL; - return 0; - } - - if (uw16_instr == BUFSTATS_DO_EXPAND) - { - inst->NoOfExpandCalls++; - } - else - { - /* reset counter */ - inst->NoOfExpandCalls = 0; - } - - /* New codec or big change in packet number? */ - if ((inst->new_codec) || (uw16_instr == BUFSTAT_REINIT)) - { - CodecFuncInst_t cinst; - - /* Clear other instructions */ - blockPtr = &inst->pw16_writeAddress[3]; - /* Clear instruction word */ - inst->pw16_writeAddress[0] = 0; - - inst->timeStamp = uw32_availableTS; - dspInfo.playedOutTS = uw32_availableTS; - if (inst->current_Codec != -1) - { - i_res = WebRtcNetEQ_DbGetPtrs(&inst->codec_DB_inst, - (enum WebRtcNetEQDecoder) inst->current_Codec, &cinst); - if (i_res < 0) - { /* error returned */ - return i_res; - } - } - else - { - /* The main codec has not been initialized yet (first packets are DTMF or CNG). */ - if (WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType)) - { - /* The currently extracted packet is CNG; get CNG fs */ - uint16_t tempFs; - - tempFs = WebRtcNetEQ_DbGetSampleRate(&inst->codec_DB_inst, payloadType); - /* TODO(tlegrand): Remove this limitation once ACM has full - * 48 kHz support. */ - if (tempFs > 32000) - { - inst->fs = 32000; - } - else if (tempFs > 0) - { - inst->fs = tempFs; - } - } - WebRtcSpl_MemSetW16((int16_t*) &cinst, 0, - sizeof(CodecFuncInst_t) / sizeof(int16_t)); - cinst.codec_fs = inst->fs; - } - cinst.timeStamp = inst->timeStamp; - blockLen = (sizeof(CodecFuncInst_t)) >> (sizeof(int16_t) - 1); /* in Word16 */ - *blockPtr = blockLen * 2; - blockPtr++; - WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst,sizeof(CodecFuncInst_t)); - blockPtr += blockLen; - inst->new_codec = 0; - - /* Reinitialize the MCU fs */ - i_res = WebRtcNetEQ_McuSetFs(inst, cinst.codec_fs); - if (i_res < 0) - { /* error returned */ - return i_res; - } - - /* Set the packet size by guessing */ - inst->PacketBuffer_inst.packSizeSamples = - WebRtcNetEQ_UpdatePackSizeSamples(inst, i_bufferpos, payloadType, - inst->timestampsPerCall * 3); - - WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst), - inst->PacketBuffer_inst.maxInsertPositions); - -#ifdef NETEQ_CNG_CODEC - /* Also insert CNG state as this might be needed by DSP */ - i_res = WebRtcNetEQ_DbGetPtrs(&inst->codec_DB_inst, kDecoderCNG, &cinst); - if ((i_res < 0) && (i_res != CODEC_DB_NOT_EXIST1)) - { - /* other error returned */ - /* (CODEC_DB_NOT_EXIST1 simply indicates that CNG is not used */ - return i_res; - } - else - { - /* CNG exists */ - blockLen = (sizeof(cinst.codec_state)) >> (sizeof(int16_t) - 1); - *blockPtr = blockLen * 2; - blockPtr++; - WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst.codec_state,sizeof(cinst.codec_state)); - blockPtr += blockLen; - } -#endif - - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff) - | DSP_CODEC_NEW_CODEC; - - if (uw16_instr == BUFSTATS_DO_RFC3389CNG_NOPACKET) - { - /* - * Change decision to CNG packet, since we do have a CNG packet, but it was - * considered too early to use. Now, use it anyway. - */ - uw16_instr = BUFSTATS_DO_RFC3389CNG_PACKET; - } - else if (uw16_instr != BUFSTATS_DO_RFC3389CNG_PACKET) - { - uw16_instr = BUFSTATS_DO_NORMAL; - } - - /* reset loss counter */ - WebRtcNetEQ_ResetMcuInCallStats(inst); - } - - /* Should we just reset the decoder? */ - if (uw16_instr == BUFSTAT_REINIT_DECODER) - { - /* Change decision to normal and flag decoder reset */ - uw16_instr = BUFSTATS_DO_NORMAL; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff) | DSP_CODEC_RESET; - } - - /* Expand requires no new packet */ - if (uw16_instr == BUFSTATS_DO_EXPAND) - { - - inst->timeStamp = dspInfo.playedOutTS; - - /* Have we got one descriptor left? */ - if (WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) inst->current_Codec) - && (dspInfo.MD || latePacketExist)) - { - - if (dspInfo.lastMode != MODE_ONE_DESCRIPTOR) - { - /* this is the first "consecutive" one-descriptor decoding; reset counter */ - inst->one_desc = 0; - } - if (inst->one_desc < MAX_ONE_DESC) - { - /* use that one descriptor */ - inst->one_desc++; /* increase counter */ - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_NORMAL_ONE_DESC; - - /* decrease counter since we did no Expand */ - inst->NoOfExpandCalls = WEBRTC_SPL_MAX(inst->NoOfExpandCalls - 1, 0); - return 0; - } - else - { - /* too many consecutive one-descriptor decodings; do expand instead */ - inst->one_desc = 0; /* reset counter */ - } - - } - - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) | DSP_INSTR_EXPAND; - return 0; - } - - /* Merge is not needed if we still have a descriptor */ - if ((uw16_instr == BUFSTATS_DO_MERGE) && (dspInfo.MD != 0)) - { - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_NORMAL_ONE_DESC; - *blockPtr = 0; - return 0; - } - - /* Do CNG without trying to extract any packets from buffer */ - if (uw16_instr == BUFSTATS_DO_RFC3389CNG_NOPACKET) - { - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_RFC3389CNG; - *blockPtr = 0; - return 0; - } - - /* Do built-in CNG without extracting any new packets from buffer */ - if (uw16_instr == BUFSTATS_DO_INTERNAL_CNG_NOPACKET) - { - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_CODEC_INTERNAL_CNG; - *blockPtr = 0; - return 0; - } - - /* Do DTMF without extracting any new packets from buffer */ - if (uw16_instr == BUFSTATS_DO_DTMF_ONLY) - { - uint32_t timeStampJump = 0; - - /* Update timestamp */ - if ((inst->BufferStat_inst.uw32_CNGplayedTS > 0) && (dspInfo.lastMode != MODE_DTMF)) - { - /* Jump in timestamps if needed */ - timeStampJump = inst->BufferStat_inst.uw32_CNGplayedTS; - inst->pw16_writeAddress[1] = (uint16_t) (timeStampJump >> 16); - inst->pw16_writeAddress[2] = (uint16_t) (timeStampJump & 0xFFFF); - } - - inst->timeStamp = dspInfo.playedOutTS + timeStampJump; - - inst->BufferStat_inst.uw32_CNGplayedTS = 0; - inst->NoOfExpandCalls = 0; - - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DTMF_GENERATE; - *blockPtr = 0; - return 0; - } - - if (uw16_instr == BUFSTATS_DO_ACCELERATE) - { - /* In order to do a Accelerate we need at least 30 ms of data */ - if (dspInfo.samplesLeft >= (3 * 80 * fs_mult)) - { - /* Already have enough data, so we do not need to extract any more */ - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_ACCELERATE; - *blockPtr = 0; - inst->BufferStat_inst.Automode_inst.sampleMemory - = (int32_t) dspInfo.samplesLeft; - inst->BufferStat_inst.Automode_inst.prevTimeScale = 1; - return 0; - } - else if ((dspInfo.samplesLeft >= (1 * 80 * fs_mult)) - && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult))) - { - /* Avoid decoding more data as it might overflow playout buffer */ - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_NORMAL; - *blockPtr = 0; - return 0; - } - else if ((dspInfo.samplesLeft < (1 * 80 * fs_mult)) - && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult))) - { - /* For >= 30ms allow Accelerate with a decoding to avoid overflow in playout buffer */ - wantedNoOfTimeStamps = inst->timestampsPerCall; - } - else if (dspInfo.samplesLeft >= (2 * 80 * fs_mult)) - { - /* We need to decode another 10 ms in order to do an Accelerate */ - wantedNoOfTimeStamps = inst->timestampsPerCall; - } - else - { - /* - * Build up decoded data by decoding at least 20 ms of data. - * Do not perform Accelerate yet, but wait until we only need to do one decoding. - */ - wantedNoOfTimeStamps = 2 * inst->timestampsPerCall; - uw16_instr = BUFSTATS_DO_NORMAL; - } - } - else if (uw16_instr == BUFSTATS_DO_PREEMPTIVE_EXPAND) - { - /* In order to do a Preemptive Expand we need at least 30 ms of data */ - if (dspInfo.samplesLeft >= (3 * 80 * fs_mult)) - { - /* Already have enough data, so we do not need to extract any more */ - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_PREEMPTIVE_EXPAND; - *blockPtr = 0; - inst->BufferStat_inst.Automode_inst.sampleMemory - = (int32_t) dspInfo.samplesLeft; - inst->BufferStat_inst.Automode_inst.prevTimeScale = 1; - return 0; - } - else if ((dspInfo.samplesLeft >= (1 * 80 * fs_mult)) - && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult))) - { - /* - * Avoid decoding more data as it might overflow playout buffer; - * still try Preemptive Expand though. - */ - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_PREEMPTIVE_EXPAND; - *blockPtr = 0; - inst->BufferStat_inst.Automode_inst.sampleMemory - = (int32_t) dspInfo.samplesLeft; - inst->BufferStat_inst.Automode_inst.prevTimeScale = 1; - return 0; - } - else if ((dspInfo.samplesLeft < (1 * 80 * fs_mult)) - && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult))) - { - /* - * For >= 30ms allow Preemptive Expand with a decoding to avoid overflow in - * playout buffer - */ - wantedNoOfTimeStamps = inst->timestampsPerCall; - } - else if (dspInfo.samplesLeft >= (2 * 80 * fs_mult)) - { - /* We need to decode another 10 ms in order to do an Preemptive Expand */ - wantedNoOfTimeStamps = inst->timestampsPerCall; - } - else - { - /* - * Build up decoded data by decoding at least 20 ms of data, - * Still try to perform Preemptive Expand. - */ - wantedNoOfTimeStamps = 2 * inst->timestampsPerCall; - } - } - else - { - wantedNoOfTimeStamps = inst->timestampsPerCall; - } - - /* Otherwise get data from buffer, try to get at least 10ms */ - totalTS = 0; - oldTS = uw32_availableTS; - if ((i_bufferpos > -1) && (uw16_instr != BUFSTATS_DO_ALTERNATIVE_PLC) && (uw16_instr - != BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS) && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION) - && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION_INC_TS)) - { - uw32_tmp = (uw32_availableTS - dspInfo.playedOutTS); - inst->pw16_writeAddress[1] = (uint16_t) (uw32_tmp >> 16); - inst->pw16_writeAddress[2] = (uint16_t) (uw32_tmp & 0xFFFF); - if (inst->BufferStat_inst.w16_cngOn == CNG_OFF) - { - /* - * Adjustment of TS only corresponds to an actual packet loss - * if comfort noise is not played. If comfort noise was just played, - * this adjustment of TS is only done to get back in sync with the - * stream TS; no loss to report. - */ - inst->lostTS += uw32_tmp; - } - - if (uw16_instr != BUFSTATS_DO_RFC3389CNG_PACKET) - { - /* We are about to decode and use a non-CNG packet => CNG period is ended */ - inst->BufferStat_inst.w16_cngOn = CNG_OFF; - } - - /* - * Reset CNG timestamp as a new packet will be delivered. - * (Also if CNG packet, since playedOutTS is updated.) - */ - inst->BufferStat_inst.uw32_CNGplayedTS = 0; - - prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos]; - prevTS = inst->PacketBuffer_inst.timeStamp[i_bufferpos]; - oldPT = inst->PacketBuffer_inst.payloadType[i_bufferpos]; - - /* These values are used by NACK module to estimate time-to-play of - * a missing packet. Occasionally, NetEq might decide to decode more - * than one packet. Therefore, these values store sequence number and - * timestamp of the first packet pulled from the packet buffer. In - * such cases, these values do not exactly represent the sequence number - * or timestamp associated with a 10ms audio pulled from NetEq. NACK - * module is designed to compensate for this. - */ - inst->decoded_packet_sequence_number = prevSeqNo; - inst->decoded_packet_timestamp = prevTS; - - /* clear flag bits */ - inst->pw16_writeAddress[0] = inst->pw16_writeAddress[0] & 0xFF3F; - do - { - int waitingTime; - inst->timeStamp = uw32_availableTS; - /* Write directly to shared memory */ - temp_pkt.payload = blockPtr + 1; - i_res = WebRtcNetEQ_PacketBufferExtract(&inst->PacketBuffer_inst, &temp_pkt, - i_bufferpos, &waitingTime); - - if (i_res < 0) - { - /* error returned */ - return i_res; - } - WebRtcNetEQ_StoreWaitingTime(inst, waitingTime); - -#ifdef NETEQ_DELAY_LOGGING - temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE; - if ((fwrite(&temp_var, sizeof(int), - 1, delay_fid2) != 1) || - (fwrite(&temp_pkt.timeStamp, sizeof(uint32_t), - 1, delay_fid2) != 1) || - (fwrite(&dspInfo.samplesLeft, sizeof(uint16_t), - 1, delay_fid2) != 1)) { - return -1; - } -#endif - - *blockPtr = temp_pkt.payloadLen; - /* set the flag if this is a redundant payload */ - if (temp_pkt.rcuPlCntr > 0) - { - *blockPtr = (*blockPtr) | (DSP_CODEC_RED_FLAG); - } - blockPtr += ((temp_pkt.payloadLen + 1) >> 1) + 1; - - if (i_bufferpos > -1) - { - /* - * Store number of TS extracted (last extracted is assumed to be of - * packSizeSamples). - */ - totalTS = uw32_availableTS - oldTS + inst->PacketBuffer_inst.packSizeSamples; - } - /* Check what next packet is available */ - WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst, - inst->timeStamp, &uw32_availableTS, &i_bufferpos, 0, &payloadType); - - nextSeqNoAvail = 0; - if ((i_bufferpos > -1) && (oldPT - == inst->PacketBuffer_inst.payloadType[i_bufferpos])) - { - w16_tmp = inst->PacketBuffer_inst.seqNumber[i_bufferpos] - prevSeqNo; - w32_tmp = inst->PacketBuffer_inst.timeStamp[i_bufferpos] - prevTS; - if ((w16_tmp == 1) || /* Next packet */ - ((w16_tmp == 0) && (w32_tmp == inst->PacketBuffer_inst.packSizeSamples))) - { /* or packet split into frames */ - nextSeqNoAvail = 1; - } - prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos]; - } - /* Update the frame size, if we can. */ - inst->PacketBuffer_inst.packSizeSamples = - WebRtcNetEQ_UpdatePackSizeSamples(inst, i_bufferpos, - payloadType, inst->PacketBuffer_inst.packSizeSamples); - } - while ((totalTS < wantedNoOfTimeStamps) && (nextSeqNoAvail == 1)); - } - - if ((uw16_instr == BUFSTATS_DO_ACCELERATE) - || (uw16_instr == BUFSTATS_DO_PREEMPTIVE_EXPAND)) - { - /* Check that we have enough data (30ms) to do the Accelearate */ - if ((totalTS + dspInfo.samplesLeft) < WEBRTC_SPL_MUL(3,inst->timestampsPerCall) - && (uw16_instr == BUFSTATS_DO_ACCELERATE)) - { - /* Not enough, do normal operation instead */ - uw16_instr = BUFSTATS_DO_NORMAL; - } - else - { - inst->BufferStat_inst.Automode_inst.sampleMemory - = (int32_t) dspInfo.samplesLeft + totalTS; - inst->BufferStat_inst.Automode_inst.prevTimeScale = 1; - } - } - - /* Close the data with a zero size block */ - *blockPtr = 0; - - /* Write data to DSP */ - switch (uw16_instr) - { - case BUFSTATS_DO_NORMAL: - /* Normal with decoding included */ - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_NORMAL; - break; - case BUFSTATS_DO_ACCELERATE: - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_ACCELERATE; - break; - case BUFSTATS_DO_MERGE: - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_MERGE; - break; - case BUFSTATS_DO_RFC3389CNG_PACKET: - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_RFC3389CNG; - break; - case BUFSTATS_DO_ALTERNATIVE_PLC: - inst->pw16_writeAddress[1] = 0; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_ALTERNATIVE_PLC; - break; - case BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS: - inst->pw16_writeAddress[1] = 0; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS; - break; - case BUFSTATS_DO_AUDIO_REPETITION: - inst->pw16_writeAddress[1] = 0; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_AUDIO_REPETITION; - break; - case BUFSTATS_DO_AUDIO_REPETITION_INC_TS: - inst->pw16_writeAddress[1] = 0; - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_DO_AUDIO_REPETITION_INC_TS; - break; - case BUFSTATS_DO_PREEMPTIVE_EXPAND: - inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) - | DSP_INSTR_PREEMPTIVE_EXPAND; - break; - default: - return UNKNOWN_BUFSTAT_DECISION; - } - - inst->timeStamp = dspInfo.playedOutTS; - return 0; - -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c deleted file mode 100644 index d7f17fdc8d4..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c +++ /dev/null @@ -1,152 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Split an RTP payload (if possible and suitable) and insert into packet buffer. - */ - -#include "mcu.h" - -#include <string.h> - -#include "mcu_dsp_common.h" -#include "neteq_error_codes.h" -#include "signal_processing_library.h" - -int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t* packet, - PacketBuf_t* Buffer_inst, - SplitInfo_t* split_inst, - int16_t* flushed, - int av_sync) -{ - - int i_ok; - int len; - int i; - RTPPacket_t temp_packet; - int16_t localFlushed = 0; - const int16_t *pw16_startPayload; - const int is_sync_rtp = av_sync && - WebRtcNetEQ_IsSyncPayload(packet->payload, packet->payloadLen); - *flushed = 0; - - len = packet->payloadLen; - - /* Copy to temp packet that can be modified. */ - - WEBRTC_SPL_MEMCPY_W8(&temp_packet,packet,sizeof(RTPPacket_t)); - - if (split_inst->deltaBytes == NO_SPLIT || - is_sync_rtp) /* Don't split sync RTPs just insert. */ - { - /* Not splittable codec */ - i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, packet, - &localFlushed, av_sync); - *flushed |= localFlushed; - if (i_ok < 0) - { - return PBUFFER_INSERT_ERROR5; - } - } - else if (split_inst->deltaBytes < -10) - { - /* G711, PCM16B or G722, use "soft splitting" */ - int split_size = packet->payloadLen; - int mult = WEBRTC_SPL_ABS_W32(split_inst->deltaBytes) - 10; - - /* Find "chunk size" >= 20 ms and < 40 ms - * split_inst->deltaTime in this case contains the number of bytes per - * timestamp unit times 2 - */ - while (split_size >= ((80 << split_inst->deltaTime) * mult)) - { - split_size >>= 1; - } - - /* Make the size an even value. */ - if (split_size > 1) - { - split_size >>= 1; - split_size *= 2; - } - - temp_packet.payloadLen = split_size; - pw16_startPayload = temp_packet.payload; - i = 0; - while (len >= (2 * split_size)) - { - /* insert every chunk */ - i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, - &localFlushed, av_sync); - *flushed |= localFlushed; - temp_packet.timeStamp += ((2 * split_size) >> split_inst->deltaTime); - i++; - temp_packet.payload = &(pw16_startPayload[(i * split_size) >> 1]); - temp_packet.starts_byte1 = temp_packet.starts_byte1 ^ (split_size & 0x1); - - len -= split_size; - if (i_ok < 0) - { - return PBUFFER_INSERT_ERROR1; - } - } - - /* Insert the rest */ - temp_packet.payloadLen = len; - i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, - &localFlushed, av_sync); - *flushed |= localFlushed; - if (i_ok < 0) - { - return PBUFFER_INSERT_ERROR2; - } - } - else - { - /* Frame based codec, use hard splitting. */ - i = 0; - pw16_startPayload = temp_packet.payload; - while (len >= split_inst->deltaBytes) - { - - temp_packet.payloadLen = split_inst->deltaBytes; - i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, - &localFlushed, av_sync); - *flushed |= localFlushed; - i++; - temp_packet.payload = &(pw16_startPayload[(i * split_inst->deltaBytes) >> 1]); - temp_packet.timeStamp += split_inst->deltaTime; - temp_packet.starts_byte1 = temp_packet.starts_byte1 ^ (split_inst->deltaBytes - & 0x1); - - if (i_ok < 0) - { - return PBUFFER_INSERT_ERROR3; - } - len -= split_inst->deltaBytes; - - } - if (len > 0) - { - /* Must be a either an error or a SID frame at the end of the packet. */ - temp_packet.payloadLen = len; - i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, - &localFlushed, av_sync); - *flushed |= localFlushed; - if (i_ok < 0) - { - return PBUFFER_INSERT_ERROR4; - } - } - } - - return 0; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/statistics_calculator.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/statistics_calculator.cc index b6e9222d475..383f7055549 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/statistics_calculator.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/statistics_calculator.cc @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/statistics_calculator.h" +#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" #include <assert.h> #include <string.h> // memset -#include "webrtc/modules/audio_coding/neteq4/decision_logic.h" -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/decision_logic.h" +#include "webrtc/modules/audio_coding/neteq/delay_manager.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/statistics_calculator.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/statistics_calculator.h index 25f8a14bb9a..07ef8536fa8 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/statistics_calculator.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/statistics_calculator.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_STATISTICS_CALCULATOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_STATISTICS_CALCULATOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_ #include <vector> -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -106,4 +106,4 @@ class StatisticsCalculator { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_STATISTICS_CALCULATOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/sync_buffer.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc index 75ee6ece082..d1802e174fc 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/sync_buffer.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc @@ -12,7 +12,7 @@ #include <algorithm> // Access to min. -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/sync_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.h index e1e5daf1b78..59bd4d87e26 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/sync_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_SYNC_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_SYNC_BUFFER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_ -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -78,7 +78,8 @@ class SyncBuffer : public AudioMultiVector { // created. void Flush(); - const AudioVector& Channel(size_t n) { return *channels_[n]; } + const AudioVector& Channel(size_t n) const { return *channels_[n]; } + AudioVector& Channel(size_t n) { return *channels_[n]; } // Accessors and mutators. size_t next_index() const { return next_index_; } @@ -97,4 +98,4 @@ class SyncBuffer : public AudioMultiVector { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_SYNC_BUFFER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc index 1aafa22ab88..1a3d0fe781c 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h" +#include "webrtc/modules/audio_coding/neteq/sync_buffer.h" #include "gtest/gtest.h" diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc index 5b6b3ba9666..a9228d49889 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/time_stretch.h" +#include "webrtc/modules/audio_coding/neteq/time_stretch.h" #include <algorithm> // min, max #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" -#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" +#include "webrtc/modules/audio_coding/neteq/dsp_helper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { @@ -29,7 +29,7 @@ TimeStretch::ReturnCodes TimeStretch::Process( int fs_mult_120 = fs_mult_ * 120; // Corresponds to 15 ms. const int16_t* signal; - scoped_array<int16_t> signal_array; + scoped_ptr<int16_t[]> signal_array; size_t signal_len; if (num_channels_ == 1) { signal = input; diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.h index f0f58b83ad9..9396d8ff519 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIME_STRETCH_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIME_STRETCH_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_ #include <assert.h> #include <string.h> // memset, size_t -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -108,4 +108,4 @@ class TimeStretch { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIME_STRETCH_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc new file mode 100644 index 00000000000..64789b4d436 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc @@ -0,0 +1,52 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Unit tests for Accelerate and PreemptiveExpand classes. + +#include "webrtc/modules/audio_coding/neteq/accelerate.h" +#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" + +#include "gtest/gtest.h" +#include "webrtc/modules/audio_coding/neteq/background_noise.h" + +namespace webrtc { + +TEST(TimeStretch, CreateAndDestroy) { + const int kSampleRate = 8000; + const size_t kNumChannels = 1; + const int kOverlapSamples = 5 * kSampleRate / 8000; + BackgroundNoise bgn(kNumChannels); + Accelerate accelerate(kSampleRate, kNumChannels, bgn); + PreemptiveExpand preemptive_expand( + kSampleRate, kNumChannels, bgn, kOverlapSamples); +} + +TEST(TimeStretch, CreateUsingFactory) { + const int kSampleRate = 8000; + const size_t kNumChannels = 1; + const int kOverlapSamples = 5 * kSampleRate / 8000; + BackgroundNoise bgn(kNumChannels); + + AccelerateFactory accelerate_factory; + Accelerate* accelerate = + accelerate_factory.Create(kSampleRate, kNumChannels, bgn); + EXPECT_TRUE(accelerate != NULL); + delete accelerate; + + PreemptiveExpandFactory preemptive_expand_factory; + PreemptiveExpand* preemptive_expand = preemptive_expand_factory.Create( + kSampleRate, kNumChannels, bgn, kOverlapSamples); + EXPECT_TRUE(preemptive_expand != NULL); + delete preemptive_expand; +} + +// TODO(hlundin): Write more tests. + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc index b2b5b40a3a6..01890136a94 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/timestamp_scaler.h" +#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/defines.h" +#include "webrtc/modules/audio_coding/neteq/decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/defines.h" #include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler.h index e165076a5e5..59b8cc7d1dc 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIMESTAMP_SCALER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIMESTAMP_SCALER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_ -#include "webrtc/modules/audio_coding/neteq4/packet.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -65,4 +65,4 @@ class TimestampScaler { }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TIMESTAMP_SCALER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc index c676094672f..8cbbfa393ac 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/timestamp_scaler.h" +#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" #include "gmock/gmock.h" #include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" +#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" +#include "webrtc/modules/audio_coding/neteq/packet.h" using ::testing::Return; using ::testing::ReturnNull; diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h new file mode 100644 index 00000000000..ac5682651b5 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h @@ -0,0 +1,60 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_ + +#include <string> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/base/md5digest.h" +#include "webrtc/base/stringencode.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +namespace test { + +class AudioChecksum : public AudioSink { + public: + AudioChecksum() : finished_(false) {} + + virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE { + if (finished_) + return false; + +#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +#error "Big-endian gives a different checksum" +#endif + checksum_.Update(audio, num_samples * sizeof(*audio)); + return true; + } + + // Finalizes the computations, and returns the checksum. + std::string Finish() { + if (!finished_) { + finished_ = true; + checksum_.Finish(checksum_result_, rtc::Md5Digest::kSize); + } + return rtc::hex_encode(checksum_result_, rtc::Md5Digest::kSize); + } + + private: + rtc::Md5Digest checksum_; + char checksum_result_[rtc::Md5Digest::kSize]; + bool finished_; + + DISALLOW_COPY_AND_ASSIGN(AudioChecksum); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc index 94ea5bef015..2d2a7e3dd4a 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/tools/audio_loop.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" #include <assert.h> #include <stdio.h> diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.h index 038ca370e72..9647d827ac9 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_AUDIO_LOOP_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_AUDIO_LOOP_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_ #include <string> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" @@ -27,8 +27,7 @@ class AudioLoop { AudioLoop() : next_index_(0), loop_length_samples_(0), - block_length_samples_(0), - audio_array_(NULL) { + block_length_samples_(0) { } virtual ~AudioLoop() {} @@ -50,11 +49,11 @@ class AudioLoop { size_t next_index_; size_t loop_length_samples_; size_t block_length_samples_; - scoped_array<int16_t> audio_array_; + scoped_ptr<int16_t[]> audio_array_; DISALLOW_COPY_AND_ASSIGN(AudioLoop); }; } // namespace test } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_AUDIO_LOOP_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_sink.h new file mode 100644 index 00000000000..5743c3641de --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_sink.h @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +namespace test { + +// Interface class for an object receiving raw output audio from test +// applications. +class AudioSink { + public: + AudioSink() {} + virtual ~AudioSink() {} + + // Writes |num_samples| from |audio| to the AudioSink. Returns true if + // successful, otherwise false. + virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0; + + // Writes |audio_frame| to the AudioSink. Returns true if successful, + // otherwise false. + bool WriteAudioFrame(const AudioFrame& audio_frame) { + return WriteArray( + audio_frame.data_, + audio_frame.samples_per_channel_ * audio_frame.num_channels_); + } + + private: + DISALLOW_COPY_AND_ASSIGN(AudioSink); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/input_audio_file.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc index 62692e27dc4..806317320f8 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/input_audio_file.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" namespace webrtc { namespace test { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h index de51ff88b8c..274f8ea07e5 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_INPUT_AUDIO_FILE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_INPUT_AUDIO_FILE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_ #include <stdio.h> #include <string> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -48,4 +48,4 @@ class InputAudioFile { } // namespace test } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_INPUT_AUDIO_FILE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc new file mode 100644 index 00000000000..433546fbcbd --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc @@ -0,0 +1,132 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" + +#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/typedefs.h" + +using webrtc::NetEq; +using webrtc::test::AudioLoop; +using webrtc::test::RtpGenerator; +using webrtc::WebRtcRTPHeader; + +namespace webrtc { +namespace test { + +int64_t NetEqPerformanceTest::Run(int runtime_ms, + int lossrate, + double drift_factor) { + const std::string kInputFileName = + webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + const int kSampRateHz = 32000; + const webrtc::NetEqDecoder kDecoderType = webrtc::kDecoderPCM16Bswb32kHz; + const int kPayloadType = 95; + + // Initialize NetEq instance. + NetEq::Config config; + config.sample_rate_hz = kSampRateHz; + NetEq* neteq = NetEq::Create(config); + // Register decoder in |neteq|. + if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0) + return -1; + + // Set up AudioLoop object. + AudioLoop audio_loop; + const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop. + const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms. + if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, + kInputBlockSizeSamples)) + return -1; + + int32_t time_now_ms = 0; + + // Get first input packet. + WebRtcRTPHeader rtp_header; + RtpGenerator rtp_gen(kSampRateHz / 1000); + // Start with positive drift first half of simulation. + rtp_gen.set_drift_factor(drift_factor); + bool drift_flipped = false; + int32_t packet_input_time_ms = + rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); + const int16_t* input_samples = audio_loop.GetNextBlock(); + if (!input_samples) exit(1); + uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)]; + int payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples), + kInputBlockSizeSamples, + input_payload); + assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); + + // Main loop. + webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock(); + int64_t start_time_ms = clock->TimeInMilliseconds(); + while (time_now_ms < runtime_ms) { + while (packet_input_time_ms <= time_now_ms) { + // Drop every N packets, where N = FLAGS_lossrate. + bool lost = false; + if (lossrate > 0) { + lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0; + } + if (!lost) { + // Insert packet. + int error = neteq->InsertPacket( + rtp_header, input_payload, payload_len, + packet_input_time_ms * kSampRateHz / 1000); + if (error != NetEq::kOK) + return -1; + } + + // Get next packet. + packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType, + kInputBlockSizeSamples, + &rtp_header); + input_samples = audio_loop.GetNextBlock(); + if (!input_samples) return -1; + payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples), + kInputBlockSizeSamples, + input_payload); + assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); + } + + // Get output audio, but don't do anything with it. + static const int kMaxChannels = 1; + static const int kMaxSamplesPerMs = 48000 / 1000; + static const int kOutputBlockSizeMs = 10; + static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs * + kMaxChannels; + int16_t out_data[kOutDataLen]; + int num_channels; + int samples_per_channel; + int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel, + &num_channels, NULL); + if (error != NetEq::kOK) + return -1; + + assert(samples_per_channel == kSampRateHz * 10 / 1000); + + time_now_ms += kOutputBlockSizeMs; + if (time_now_ms >= runtime_ms / 2 && !drift_flipped) { + // Apply negative drift second half of simulation. + rtp_gen.set_drift_factor(-drift_factor); + drift_flipped = true; + } + } + int64_t end_time_ms = clock->TimeInMilliseconds(); + delete neteq; + return end_time_ms - start_time_ms; +} + +} // namespace test +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h new file mode 100644 index 00000000000..d094db0f9b3 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h @@ -0,0 +1,32 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_ + +#include "webrtc/typedefs.h" + +namespace webrtc { +namespace test { + +class NetEqPerformanceTest { + public: + // Runs a performance test with parameters as follows: + // |runtime_ms|: the simulation time, i.e., the duration of the audio data. + // |lossrate|: drop one out of |lossrate| packets, e.g., one out of 10. + // |drift_factor|: clock drift in [0, 1]. + // Returns the runtime in ms. + static int64_t Run(int runtime_ms, int lossrate, double drift_factor); +}; + +} // namespace test +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc new file mode 100644 index 00000000000..fc5d8abaf82 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc @@ -0,0 +1,115 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <stdio.h> +#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h" + +namespace webrtc { +namespace test { + +const uint8_t kPayloadType = 95; +const int kOutputSizeMs = 10; + +NetEqQualityTest::NetEqQualityTest(int block_duration_ms, + int in_sampling_khz, + int out_sampling_khz, + enum NetEqDecoder decoder_type, + int channels, + double drift_factor, + std::string in_filename, + std::string out_filename) + : decoded_time_ms_(0), + decodable_time_ms_(0), + drift_factor_(drift_factor), + block_duration_ms_(block_duration_ms), + in_sampling_khz_(in_sampling_khz), + out_sampling_khz_(out_sampling_khz), + decoder_type_(decoder_type), + channels_(channels), + in_filename_(in_filename), + out_filename_(out_filename), + in_size_samples_(in_sampling_khz_ * block_duration_ms_), + out_size_samples_(out_sampling_khz_ * kOutputSizeMs), + payload_size_bytes_(0), + max_payload_bytes_(0), + in_file_(new InputAudioFile(in_filename_)), + out_file_(NULL), + rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0, + decodable_time_ms_)) { + NetEq::Config config; + config.sample_rate_hz = out_sampling_khz_ * 1000; + neteq_.reset(NetEq::Create(config)); + max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t); + in_data_.reset(new int16_t[in_size_samples_ * channels_]); + payload_.reset(new uint8_t[max_payload_bytes_]); + out_data_.reset(new int16_t[out_size_samples_ * channels_]); +} + +void NetEqQualityTest::SetUp() { + out_file_ = fopen(out_filename_.c_str(), "wb"); + ASSERT_TRUE(out_file_ != NULL); + ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType)); + rtp_generator_->set_drift_factor(drift_factor_); +} + +void NetEqQualityTest::TearDown() { + fclose(out_file_); +} + +int NetEqQualityTest::Transmit() { + int packet_input_time_ms = + rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_, + &rtp_header_); + if (!PacketLost(packet_input_time_ms) && payload_size_bytes_ > 0) { + int ret = neteq_->InsertPacket(rtp_header_, &payload_[0], + payload_size_bytes_, + packet_input_time_ms * in_sampling_khz_); + if (ret != NetEq::kOK) + return -1; + } + return packet_input_time_ms; +} + +int NetEqQualityTest::DecodeBlock() { + int channels; + int samples; + int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0], + &samples, &channels, NULL); + + if (ret != NetEq::kOK) { + return -1; + } else { + assert(channels == channels_); + assert(samples == kOutputSizeMs * out_sampling_khz_); + fwrite(&out_data_[0], sizeof(int16_t), samples * channels, out_file_); + return samples; + } +} + +void NetEqQualityTest::Simulate(int end_time_ms) { + int audio_size_samples; + + while (decoded_time_ms_ < end_time_ms) { + while (decodable_time_ms_ - kOutputSizeMs < decoded_time_ms_) { + ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0])); + payload_size_bytes_ = EncodeBlock(&in_data_[0], + in_size_samples_, &payload_[0], + max_payload_bytes_); + decodable_time_ms_ = Transmit() + block_duration_ms_; + } + audio_size_samples = DecodeBlock(); + if (audio_size_samples > 0) { + decoded_time_ms_ += audio_size_samples / out_sampling_khz_; + } + } +} + +} // namespace test +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h new file mode 100644 index 00000000000..87fc50794ca --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h @@ -0,0 +1,100 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_ + +#include <string> +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +namespace test { + +class NetEqQualityTest : public ::testing::Test { + protected: + NetEqQualityTest(int block_duration_ms, + int in_sampling_khz, + int out_sampling_khz, + enum NetEqDecoder decoder_type, + int channels, + double drift_factor, + std::string in_filename, + std::string out_filename); + virtual void SetUp() OVERRIDE; + virtual void TearDown() OVERRIDE; + + // EncodeBlock(...) does the following: + // 1. encodes a block of audio, saved in |in_data| and has a length of + // |block_size_samples| (samples per channel), + // 2. save the bit stream to |payload| of |max_bytes| bytes in size, + // 3. returns the length of the payload (in bytes), + virtual int EncodeBlock(int16_t* in_data, int block_size_samples, + uint8_t* payload, int max_bytes) = 0; + + // PacketLoss(...) determines weather a packet sent at an indicated time gets + // lost or not. + virtual bool PacketLost(int packet_input_time_ms) { return false; } + + // DecodeBlock() decodes a block of audio using the payload stored in + // |payload_| with the length of |payload_size_bytes_| (bytes). The decoded + // audio is to be stored in |out_data_|. + int DecodeBlock(); + + // Transmit() uses |rtp_generator_| to generate a packet and passes it to + // |neteq_|. + int Transmit(); + + // Simulate(...) runs encoding / transmitting / decoding up to |end_time_ms| + // (miliseconds), the resulted audio is stored in the file with the name of + // |out_filename_|. + void Simulate(int end_time_ms); + + private: + int decoded_time_ms_; + int decodable_time_ms_; + double drift_factor_; + const int block_duration_ms_; + const int in_sampling_khz_; + const int out_sampling_khz_; + const enum NetEqDecoder decoder_type_; + const int channels_; + const std::string in_filename_; + const std::string out_filename_; + + // Number of samples per channel in a frame. + const int in_size_samples_; + + // Expected output number of samples per channel in a frame. + const int out_size_samples_; + + int payload_size_bytes_; + int max_payload_bytes_; + + scoped_ptr<InputAudioFile> in_file_; + FILE* out_file_; + + scoped_ptr<RtpGenerator> rtp_generator_; + scoped_ptr<NetEq> neteq_; + + scoped_ptr<int16_t[]> in_data_; + scoped_ptr<uint8_t[]> payload_; + scoped_ptr<int16_t[]> out_data_; + WebRtcRTPHeader rtp_header_; +}; + +} // namespace test +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/neteq_rtpplay.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc index f0ca51f2f70..3c5f6b0bbaa 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/neteq_rtpplay.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc @@ -8,6 +8,9 @@ * be found in the AUTHORS file in the root of the source tree. */ +// TODO(hlundin): The functionality in this file should be moved into one or +// several classes. + #include <assert.h> #include <stdio.h> @@ -16,10 +19,13 @@ #include <string> #include "google/gflags.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h" -#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_DummyRTPpacket.h" +#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" +#include "webrtc/modules/audio_coding/neteq/interface/neteq.h" +#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h" +#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" #include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/typedefs.h" @@ -88,11 +94,23 @@ DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and " "codec"); DEFINE_bool(dummy_rtp, false, "The input file contains ""dummy"" RTP data, " "i.e., only headers"); +DEFINE_string(replacement_audio_file, "", + "A PCM file that will be used to populate ""dummy"" RTP packets"); // Declaring helper functions (defined further down in this file). std::string CodecName(webrtc::NetEqDecoder codec); void RegisterPayloadTypes(NetEq* neteq); void PrintCodecMapping(); +size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file, + webrtc::scoped_ptr<int16_t[]>* replacement_audio, + webrtc::scoped_ptr<uint8_t[]>* payload, + size_t* payload_mem_size_bytes, + size_t* frame_size_samples, + WebRtcRTPHeader* rtp_header, + NETEQTEST_RTPpacket* next_rtp); +int CodecSampleRate(uint8_t payload_type); +int CodecTimestampRate(uint8_t payload_type); +bool IsComfortNosie(uint8_t payload_type); int main(int argc, char* argv[]) { static const int kMaxChannels = 5; @@ -135,6 +153,15 @@ int main(int argc, char* argv[]) { } std::cout << "Output file: " << argv[2] << std::endl; + // Check if a replacement audio file was provided, and if so, open it. + bool replace_payload = false; + webrtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file; + if (!FLAGS_replacement_audio_file.empty()) { + replacement_audio_file.reset( + new webrtc::test::InputAudioFile(FLAGS_replacement_audio_file)); + replace_payload = true; + } + // Read RTP file header. if (NETEQTEST_RTPpacket::skipFileHeader(in_file) != 0) { std::cerr << "Wrong format in RTP file" << std::endl; @@ -149,21 +176,47 @@ int main(int argc, char* argv[]) { // Initialize NetEq instance. int sample_rate_hz = 16000; - NetEq* neteq = NetEq::Create(sample_rate_hz); + NetEq::Config config; + config.sample_rate_hz = sample_rate_hz; + NetEq* neteq = NetEq::Create(config); RegisterPayloadTypes(neteq); // Read first packet. - NETEQTEST_RTPpacket *rtp; + NETEQTEST_RTPpacket* rtp; + NETEQTEST_RTPpacket* next_rtp = NULL; if (!FLAGS_dummy_rtp) { rtp = new NETEQTEST_RTPpacket(); + if (replace_payload) { + next_rtp = new NETEQTEST_RTPpacket(); + } } else { rtp = new NETEQTEST_DummyRTPpacket(); + if (replace_payload) { + next_rtp = new NETEQTEST_DummyRTPpacket(); + } } rtp->readFromFile(in_file); - if (!rtp) { + if (rtp->dataLen() < 0) { std::cout << "Warning: RTP file is empty" << std::endl; } + // Set up variables for audio replacement if needed. + size_t input_frame_size_timestamps = 0; + webrtc::scoped_ptr<int16_t[]> replacement_audio; + webrtc::scoped_ptr<uint8_t[]> payload; + size_t payload_mem_size_bytes = 0; + if (replace_payload) { + // Initially assume that the frame size is 30 ms at the initial sample rate. + // This value will be replaced with the correct one as soon as two + // consecutive packets are found. + input_frame_size_timestamps = 30 * sample_rate_hz / 1000; + replacement_audio.reset(new int16_t[input_frame_size_timestamps]); + payload_mem_size_bytes = 2 * input_frame_size_timestamps; + payload.reset(new uint8_t[payload_mem_size_bytes]); + assert(next_rtp); + next_rtp->readFromFile(in_file); + } + // This is the main simulation loop. int time_now_ms = rtp->time(); // Start immediately with the first packet. int next_input_time_ms = rtp->time(); @@ -181,9 +234,21 @@ int main(int argc, char* argv[]) { // Parse RTP header. WebRtcRTPHeader rtp_header; rtp->parseHeader(&rtp_header); - int error = neteq->InsertPacket(rtp_header, rtp->payload(), - rtp->payloadLen(), - rtp->time() * sample_rate_hz / 1000); + uint8_t* payload_ptr = rtp->payload(); + size_t payload_len = rtp->payloadLen(); + if (replace_payload) { + payload_len = ReplacePayload(replacement_audio_file.get(), + &replacement_audio, + &payload, + &payload_mem_size_bytes, + &input_frame_size_timestamps, + &rtp_header, + next_rtp); + payload_ptr = payload.get(); + } + int error = neteq->InsertPacket(rtp_header, payload_ptr, + static_cast<int>(payload_len), + rtp->time() * sample_rate_hz / 1000); if (error != NetEq::kOK) { std::cerr << "InsertPacket returned error code " << neteq->LastError() << std::endl; @@ -191,6 +256,13 @@ int main(int argc, char* argv[]) { } // Get next packet from file. rtp->readFromFile(in_file); + if (replace_payload) { + // At this point |rtp| contains the packet *after* |next_rtp|. + // Swap RTP packet objects between |rtp| and |next_rtp|. + NETEQTEST_RTPpacket* temp_rtp = rtp; + rtp = next_rtp; + next_rtp = temp_rtp; + } next_input_time_ms = rtp->time(); } @@ -212,6 +284,7 @@ int main(int argc, char* argv[]) { } // Write to file. + // TODO(hlundin): Make writing to file optional. size_t write_len = samples_per_channel * num_channels; if (fwrite(out_data, sizeof(out_data[0]), write_len, out_file) != write_len) { @@ -229,6 +302,8 @@ int main(int argc, char* argv[]) { fclose(in_file); fclose(out_file); + delete rtp; + delete next_rtp; delete neteq; webrtc::Trace::ReturnTrace(); return 0; @@ -421,3 +496,133 @@ void PrintCodecMapping() { std::cout << CodecName(webrtc::kDecoderCNGswb48kHz).c_str() << ": " << FLAGS_cn_swb48 << std::endl; } + +size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file, + webrtc::scoped_ptr<int16_t[]>* replacement_audio, + webrtc::scoped_ptr<uint8_t[]>* payload, + size_t* payload_mem_size_bytes, + size_t* frame_size_samples, + WebRtcRTPHeader* rtp_header, + NETEQTEST_RTPpacket* next_rtp) { + size_t payload_len = 0; + // Check for CNG. + if (IsComfortNosie(rtp_header->header.payloadType)) { + // If CNG, simply insert a zero-energy one-byte payload. + if (*payload_mem_size_bytes < 1) { + (*payload).reset(new uint8_t[1]); + *payload_mem_size_bytes = 1; + } + (*payload)[0] = 127; // Max attenuation of CNG. + payload_len = 1; + } else { + if (next_rtp->payloadLen() > 0) { + // Check if payload length has changed. + if (next_rtp->sequenceNumber() == rtp_header->header.sequenceNumber + 1) { + if (*frame_size_samples != + next_rtp->timeStamp() - rtp_header->header.timestamp) { + *frame_size_samples = + next_rtp->timeStamp() - rtp_header->header.timestamp; + (*replacement_audio).reset( + new int16_t[*frame_size_samples]); + *payload_mem_size_bytes = 2 * *frame_size_samples; + (*payload).reset(new uint8_t[*payload_mem_size_bytes]); + } + } + } + // Get new speech. + assert((*replacement_audio).get()); + if (CodecTimestampRate(rtp_header->header.payloadType) != + CodecSampleRate(rtp_header->header.payloadType) || + rtp_header->header.payloadType == FLAGS_red || + rtp_header->header.payloadType == FLAGS_avt) { + // Some codecs have different sample and timestamp rates. And neither + // RED nor DTMF is supported for replacement. + std::cerr << "Codec not supported for audio replacement." << + std::endl; + webrtc::Trace::ReturnTrace(); + exit(1); + } + assert(*frame_size_samples > 0); + if (!replacement_audio_file->Read(*frame_size_samples, + (*replacement_audio).get())) { + std::cerr << "Could no read replacement audio file." << std::endl; + webrtc::Trace::ReturnTrace(); + exit(1); + } + // Encode it as PCM16. + assert((*payload).get()); + payload_len = WebRtcPcm16b_Encode((*replacement_audio).get(), + static_cast<int16_t>(*frame_size_samples), + (*payload).get()); + assert(payload_len == 2 * *frame_size_samples); + // Change payload type to PCM16. + switch (CodecSampleRate(rtp_header->header.payloadType)) { + case 8000: + rtp_header->header.payloadType = FLAGS_pcm16b; + break; + case 16000: + rtp_header->header.payloadType = FLAGS_pcm16b_wb; + break; + case 32000: + rtp_header->header.payloadType = FLAGS_pcm16b_swb32; + break; + case 48000: + rtp_header->header.payloadType = FLAGS_pcm16b_swb48; + break; + default: + std::cerr << "Payload type " << + static_cast<int>(rtp_header->header.payloadType) << + " not supported or unknown." << std::endl; + webrtc::Trace::ReturnTrace(); + exit(1); + assert(false); + } + } + return payload_len; +} + +int CodecSampleRate(uint8_t payload_type) { + if (payload_type == FLAGS_pcmu || + payload_type == FLAGS_pcma || + payload_type == FLAGS_ilbc || + payload_type == FLAGS_pcm16b || + payload_type == FLAGS_cn_nb) { + return 8000; + } else if (payload_type == FLAGS_isac || + payload_type == FLAGS_pcm16b_wb || + payload_type == FLAGS_g722 || + payload_type == FLAGS_cn_wb) { + return 16000; + } else if (payload_type == FLAGS_isac_swb || + payload_type == FLAGS_pcm16b_swb32 || + payload_type == FLAGS_cn_swb32) { + return 32000; + } else if (payload_type == FLAGS_pcm16b_swb48 || + payload_type == FLAGS_cn_swb48) { + return 48000; + } else if (payload_type == FLAGS_avt || + payload_type == FLAGS_red) { + return 0; + } else { + return -1; + } +} + +int CodecTimestampRate(uint8_t payload_type) { + if (payload_type == FLAGS_g722) { + return 8000; + } else { + return CodecSampleRate(payload_type); + } +} + +bool IsComfortNosie(uint8_t payload_type) { + if (payload_type == FLAGS_cn_nb || + payload_type == FLAGS_cn_wb || + payload_type == FLAGS_cn_swb32 || + payload_type == FLAGS_cn_swb48) { + return true; + } else { + return false; + } +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h new file mode 100644 index 00000000000..1d6128076e1 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h @@ -0,0 +1,50 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_ + +#include <assert.h> +#include <stdio.h> +#include <string> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" + +namespace webrtc { +namespace test { + +class OutputAudioFile : public AudioSink { + public: + // Creates an OutputAudioFile, opening a file named |file_name| for writing. + // The file format is 16-bit signed host-endian PCM. + explicit OutputAudioFile(const std::string& file_name) { + out_file_ = fopen(file_name.c_str(), "wb"); + } + + virtual ~OutputAudioFile() { + if (out_file_) + fclose(out_file_); + } + + virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE { + assert(out_file_); + return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples; + } + + private: + FILE* out_file_; + + DISALLOW_COPY_AND_ASSIGN(OutputAudioFile); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.cc new file mode 100644 index 00000000000..d8fb7134f10 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.cc @@ -0,0 +1,155 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" + +namespace webrtc { +namespace test { + +Packet::Packet(uint8_t* packet_memory, + size_t allocated_bytes, + double time_ms, + const RtpHeaderParser& parser) + : payload_memory_(packet_memory), + payload_(NULL), + packet_length_bytes_(allocated_bytes), + payload_length_bytes_(0), + virtual_packet_length_bytes_(allocated_bytes), + virtual_payload_length_bytes_(0), + time_ms_(time_ms) { + valid_header_ = ParseHeader(parser); +} + +Packet::Packet(uint8_t* packet_memory, + size_t allocated_bytes, + size_t virtual_packet_length_bytes, + double time_ms, + const RtpHeaderParser& parser) + : payload_memory_(packet_memory), + payload_(NULL), + packet_length_bytes_(allocated_bytes), + payload_length_bytes_(0), + virtual_packet_length_bytes_(virtual_packet_length_bytes), + virtual_payload_length_bytes_(0), + time_ms_(time_ms) { + valid_header_ = ParseHeader(parser); +} + +Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms) + : payload_memory_(packet_memory), + payload_(NULL), + packet_length_bytes_(allocated_bytes), + payload_length_bytes_(0), + virtual_packet_length_bytes_(allocated_bytes), + virtual_payload_length_bytes_(0), + time_ms_(time_ms) { + scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + valid_header_ = ParseHeader(*parser); +} + +Packet::Packet(uint8_t* packet_memory, + size_t allocated_bytes, + size_t virtual_packet_length_bytes, + double time_ms) + : payload_memory_(packet_memory), + payload_(NULL), + packet_length_bytes_(allocated_bytes), + payload_length_bytes_(0), + virtual_packet_length_bytes_(virtual_packet_length_bytes), + virtual_payload_length_bytes_(0), + time_ms_(time_ms) { + scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + valid_header_ = ParseHeader(*parser); +} + +bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const { + // + // 0 1 2 3 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |1| block PT | timestamp offset | block length | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |1| ... | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // |0| block PT | + // +-+-+-+-+-+-+-+-+ + // + + assert(payload_); + const uint8_t* payload_ptr = payload_; + const uint8_t* payload_end_ptr = payload_ptr + payload_length_bytes_; + + // Find all RED headers with the extension bit set to 1. That is, all headers + // but the last one. + while ((payload_ptr < payload_end_ptr) && (*payload_ptr & 0x80)) { + RTPHeader* header = new RTPHeader; + CopyToHeader(header); + header->payloadType = payload_ptr[0] & 0x7F; + uint32_t offset = (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2); + header->timestamp -= offset; + headers->push_front(header); + payload_ptr += 4; + } + // Last header. + assert(payload_ptr < payload_end_ptr); + if (payload_ptr >= payload_end_ptr) { + return false; // Payload too short. + } + RTPHeader* header = new RTPHeader; + CopyToHeader(header); + header->payloadType = payload_ptr[0] & 0x7F; + headers->push_front(header); + return true; +} + +void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) { + while (!headers->empty()) { + delete headers->front(); + headers->pop_front(); + } +} + +bool Packet::ParseHeader(const RtpHeaderParser& parser) { + bool valid_header = parser.Parse( + payload_memory_.get(), static_cast<int>(packet_length_bytes_), &header_); + assert(valid_header); + if (!valid_header) { + return false; + } + assert(header_.headerLength <= packet_length_bytes_); + payload_ = &payload_memory_[header_.headerLength]; + assert(packet_length_bytes_ >= header_.headerLength); + payload_length_bytes_ = packet_length_bytes_ - header_.headerLength; + assert(virtual_packet_length_bytes_ >= header_.headerLength); + virtual_payload_length_bytes_ = + virtual_packet_length_bytes_ - header_.headerLength; + return true; +} + +void Packet::CopyToHeader(RTPHeader* destination) const { + destination->markerBit = header_.markerBit; + destination->payloadType = header_.payloadType; + destination->sequenceNumber = header_.sequenceNumber; + destination->timestamp = header_.timestamp; + destination->ssrc = header_.ssrc; + destination->numCSRCs = header_.numCSRCs; + destination->paddingLength = header_.paddingLength; + destination->headerLength = header_.headerLength; + destination->payload_type_frequency = header_.payload_type_frequency; + memcpy(&destination->arrOfCSRCs, + &header_.arrOfCSRCs, + sizeof(header_.arrOfCSRCs)); + memcpy( + &destination->extension, &header_.extension, sizeof(header_.extension)); +} + +} // namespace test +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.h new file mode 100644 index 00000000000..eb8ce28a227 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.h @@ -0,0 +1,117 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ + +#include <list> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/common_types.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class RtpHeaderParser; + +namespace test { + +// Class for handling RTP packets in test applications. +class Packet { + public: + // Creates a packet, with the packet payload (including header bytes) in + // |packet_memory|. The length of |packet_memory| is |allocated_bytes|. + // The new object assumes ownership of |packet_memory| and will delete it + // when the Packet object is deleted. The |time_ms| is an extra time + // associated with this packet, typically used to denote arrival time. + // The first bytes in |packet_memory| will be parsed using |parser|. + Packet(uint8_t* packet_memory, + size_t allocated_bytes, + double time_ms, + const RtpHeaderParser& parser); + + // Same as above, but with the extra argument |virtual_packet_length_bytes|. + // This is typically used when reading RTP dump files that only contain the + // RTP headers, and no payload (a.k.a RTP dummy files or RTP light). The + // |virtual_packet_length_bytes| tells what size the packet had on wire, + // including the now discarded payload, whereas |allocated_bytes| is the + // length of the remaining payload (typically only the RTP header). + Packet(uint8_t* packet_memory, + size_t allocated_bytes, + size_t virtual_packet_length_bytes, + double time_ms, + const RtpHeaderParser& parser); + + // The following two constructors are the same as above, but without a + // parser. Note that when the object is constructed using any of these + // methods, the header will be parsed using a default RtpHeaderParser object. + // In particular, RTP header extensions won't be parsed. + Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms); + + Packet(uint8_t* packet_memory, + size_t allocated_bytes, + size_t virtual_packet_length_bytes, + double time_ms); + + virtual ~Packet() {} + + // Parses the first bytes of the RTP payload, interpreting them as RED headers + // according to RFC 2198. The headers will be inserted into |headers|. The + // caller of the method assumes ownership of the objects in the list, and + // must delete them properly. + bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const; + + // Deletes all RTPHeader objects in |headers|, but does not delete |headers| + // itself. + static void DeleteRedHeaders(std::list<RTPHeader*>* headers); + + const uint8_t* payload() const { return payload_; } + + size_t packet_length_bytes() const { return packet_length_bytes_; } + + size_t payload_length_bytes() const { return payload_length_bytes_; } + + size_t virtual_packet_length_bytes() const { + return virtual_packet_length_bytes_; + } + + size_t virtual_payload_length_bytes() const { + return virtual_payload_length_bytes_; + } + + const RTPHeader& header() const { return header_; } + + void set_time_ms(double time) { time_ms_ = time; } + double time_ms() const { return time_ms_; } + bool valid_header() const { return valid_header_; } + + private: + bool ParseHeader(const RtpHeaderParser& parser); + void CopyToHeader(RTPHeader* destination) const; + + RTPHeader header_; + scoped_ptr<uint8_t[]> payload_memory_; + const uint8_t* payload_; // First byte after header. + const size_t packet_length_bytes_; // Total length of packet. + size_t payload_length_bytes_; // Length of the payload, after RTP header. + // Zero for dummy RTP packets. + // Virtual lengths are used when parsing RTP header files (dummy RTP files). + const size_t virtual_packet_length_bytes_; + size_t virtual_payload_length_bytes_; + double time_ms_; // Used to denote a packet's arrival time. + bool valid_header_; // Set by the RtpHeaderParser. + + DISALLOW_COPY_AND_ASSIGN(Packet); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet_source.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet_source.h new file mode 100644 index 00000000000..669bc14e461 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet_source.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ + +#include "webrtc/base/constructormagic.h" + +namespace webrtc { +namespace test { + +class Packet; + +// Interface class for an object delivering RTP packets to test applications. +class PacketSource { + public: + PacketSource() {} + virtual ~PacketSource() {} + + // Returns a pointer to the next packet. Returns NULL if the source is + // depleted, or if an error occurred. + virtual Packet* NextPacket() = 0; + + private: + DISALLOW_COPY_AND_ASSIGN(PacketSource); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc new file mode 100644 index 00000000000..df844ee84f7 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc @@ -0,0 +1,202 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Unit tests for test Packet class. + +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" + +#include "gtest/gtest.h" + +namespace webrtc { +namespace test { + +namespace { +const int kHeaderLengthBytes = 12; + +void MakeRtpHeader(int payload_type, + int seq_number, + uint32_t timestamp, + uint32_t ssrc, + uint8_t* rtp_data) { + rtp_data[0] = 0x80; + rtp_data[1] = payload_type & 0xFF; + rtp_data[2] = (seq_number >> 8) & 0xFF; + rtp_data[3] = (seq_number) & 0xFF; + rtp_data[4] = (timestamp >> 24) & 0xFF; + rtp_data[5] = (timestamp >> 16) & 0xFF; + rtp_data[6] = (timestamp >> 8) & 0xFF; + rtp_data[7] = timestamp & 0xFF; + rtp_data[8] = (ssrc >> 24) & 0xFF; + rtp_data[9] = (ssrc >> 16) & 0xFF; + rtp_data[10] = (ssrc >> 8) & 0xFF; + rtp_data[11] = ssrc & 0xFF; +} +} // namespace + +TEST(TestPacket, RegularPacket) { + const size_t kPacketLengthBytes = 100; + uint8_t* packet_memory = new uint8_t[kPacketLengthBytes]; + const uint8_t kPayloadType = 17; + const uint16_t kSequenceNumber = 4711; + const uint32_t kTimestamp = 47114711; + const uint32_t kSsrc = 0x12345678; + MakeRtpHeader( + kPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory); + const double kPacketTime = 1.0; + // Hand over ownership of |packet_memory| to |packet|. + Packet packet(packet_memory, kPacketLengthBytes, kPacketTime); + ASSERT_TRUE(packet.valid_header()); + EXPECT_EQ(kPayloadType, packet.header().payloadType); + EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber); + EXPECT_EQ(kTimestamp, packet.header().timestamp); + EXPECT_EQ(kSsrc, packet.header().ssrc); + EXPECT_EQ(0, packet.header().numCSRCs); + EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes()); + EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes, + packet.payload_length_bytes()); + EXPECT_EQ(kPacketLengthBytes, packet.virtual_packet_length_bytes()); + EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes, + packet.virtual_payload_length_bytes()); + EXPECT_EQ(kPacketTime, packet.time_ms()); +} + +TEST(TestPacket, DummyPacket) { + const size_t kPacketLengthBytes = kHeaderLengthBytes; // Only RTP header. + const size_t kVirtualPacketLengthBytes = 100; + uint8_t* packet_memory = new uint8_t[kPacketLengthBytes]; + const uint8_t kPayloadType = 17; + const uint16_t kSequenceNumber = 4711; + const uint32_t kTimestamp = 47114711; + const uint32_t kSsrc = 0x12345678; + MakeRtpHeader( + kPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory); + const double kPacketTime = 1.0; + // Hand over ownership of |packet_memory| to |packet|. + Packet packet(packet_memory, + kPacketLengthBytes, + kVirtualPacketLengthBytes, + kPacketTime); + ASSERT_TRUE(packet.valid_header()); + EXPECT_EQ(kPayloadType, packet.header().payloadType); + EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber); + EXPECT_EQ(kTimestamp, packet.header().timestamp); + EXPECT_EQ(kSsrc, packet.header().ssrc); + EXPECT_EQ(0, packet.header().numCSRCs); + EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes()); + EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes, + packet.payload_length_bytes()); + EXPECT_EQ(kVirtualPacketLengthBytes, packet.virtual_packet_length_bytes()); + EXPECT_EQ(kVirtualPacketLengthBytes - kHeaderLengthBytes, + packet.virtual_payload_length_bytes()); + EXPECT_EQ(kPacketTime, packet.time_ms()); +} + +namespace { +// Writes one RED block header starting at |rtp_data|, according to RFC 2198. +// returns the number of bytes written (1 or 4). +// +// Format if |last_payoad| is false: +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |1| block PT | timestamp offset | block length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// Format if |last_payoad| is true: +// 0 1 2 3 4 5 6 7 +// +-+-+-+-+-+-+-+-+ +// |0| Block PT | +// +-+-+-+-+-+-+-+-+ + +int MakeRedHeader(int payload_type, + uint32_t timestamp_offset, + int block_length, + bool last_payload, + uint8_t* rtp_data) { + rtp_data[0] = 0x80 | (payload_type & 0x7F); // Set the first bit to 1. + if (last_payload) { + rtp_data[0] &= 0x7F; // Reset the first but to 0 to indicate last block. + return 1; + } + rtp_data[1] = timestamp_offset >> 6; + rtp_data[2] = (timestamp_offset & 0x3F) << 2; + rtp_data[2] |= block_length >> 8; + rtp_data[3] = block_length & 0xFF; + return 4; +} +} // namespace + +TEST(TestPacket, RED) { + const size_t kPacketLengthBytes = 100; + uint8_t* packet_memory = new uint8_t[kPacketLengthBytes]; + const uint8_t kRedPayloadType = 17; + const uint16_t kSequenceNumber = 4711; + const uint32_t kTimestamp = 47114711; + const uint32_t kSsrc = 0x12345678; + MakeRtpHeader( + kRedPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory); + // Create four RED headers. + // Payload types are just the same as the block index the offset is 100 times + // the block index. + const int kRedBlocks = 4; + uint8_t* payload_ptr = + &packet_memory[kHeaderLengthBytes]; // First byte after header. + for (int i = 0; i < kRedBlocks; ++i) { + int payload_type = i; + // Offset value is not used for the last block. + uint32_t timestamp_offset = 100 * i; + int block_length = 10 * i; + bool last_block = (i == kRedBlocks - 1) ? true : false; + payload_ptr += MakeRedHeader( + payload_type, timestamp_offset, block_length, last_block, payload_ptr); + } + const double kPacketTime = 1.0; + // Hand over ownership of |packet_memory| to |packet|. + Packet packet(packet_memory, kPacketLengthBytes, kPacketTime); + ASSERT_TRUE(packet.valid_header()); + EXPECT_EQ(kRedPayloadType, packet.header().payloadType); + EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber); + EXPECT_EQ(kTimestamp, packet.header().timestamp); + EXPECT_EQ(kSsrc, packet.header().ssrc); + EXPECT_EQ(0, packet.header().numCSRCs); + EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes()); + EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes, + packet.payload_length_bytes()); + EXPECT_EQ(kPacketLengthBytes, packet.virtual_packet_length_bytes()); + EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes, + packet.virtual_payload_length_bytes()); + EXPECT_EQ(kPacketTime, packet.time_ms()); + std::list<RTPHeader*> red_headers; + EXPECT_TRUE(packet.ExtractRedHeaders(&red_headers)); + EXPECT_EQ(kRedBlocks, static_cast<int>(red_headers.size())); + int block_index = 0; + for (std::list<RTPHeader*>::reverse_iterator it = red_headers.rbegin(); + it != red_headers.rend(); + ++it) { + // Reading list from the back, since the extraction puts the main payload + // (which is the last one on wire) first. + RTPHeader* red_block = *it; + EXPECT_EQ(block_index, red_block->payloadType); + EXPECT_EQ(kSequenceNumber, red_block->sequenceNumber); + if (block_index == kRedBlocks - 1) { + // Last block has zero offset per definition. + EXPECT_EQ(kTimestamp, red_block->timestamp); + } else { + EXPECT_EQ(kTimestamp - 100 * block_index, red_block->timestamp); + } + EXPECT_EQ(kSsrc, red_block->ssrc); + EXPECT_EQ(0, red_block->numCSRCs); + ++block_index; + } + Packet::DeleteRedHeaders(&red_headers); +} + +} // namespace test +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc new file mode 100644 index 00000000000..773cc2c8962 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc @@ -0,0 +1,147 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <assert.h> +#include <stdio.h> +#include <vector> + +#include "google/gflags.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +// Flag validator. +static bool ValidatePayloadType(const char* flagname, int32_t value) { + if (value >= 0 && value <= 127) // Value is ok. + return true; + printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value)); + return false; +} +static bool ValidateExtensionId(const char* flagname, int32_t value) { + if (value > 0 && value <= 255) // Value is ok. + return true; + printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value)); + return false; +} + +// Define command line flags. +DEFINE_int32(red, 117, "RTP payload type for RED"); +static const bool red_dummy = + google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType); +DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)"); +static const bool audio_level_dummy = + google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId); + +int main(int argc, char* argv[]) { + std::string program_name = argv[0]; + std::string usage = + "Tool for parsing an RTP dump file to text output.\n" + "Run " + + program_name + + " --helpshort for usage.\n" + "Example usage:\n" + + program_name + " input.rtp output.txt\n\n" + + "Output is sent to stdout if no output file is given." + + "Note that this tool can read files with our without payloads."; + google::SetUsageMessage(usage); + google::ParseCommandLineFlags(&argc, &argv, true); + + if (argc != 2 && argc != 3) { + // Print usage information. + printf("%s", google::ProgramUsage()); + return 0; + } + + FILE* in_file = fopen(argv[1], "rb"); + if (!in_file) { + printf("Cannot open input file %s\n", argv[1]); + return -1; + } + printf("Input file: %s\n", argv[1]); + webrtc::scoped_ptr<webrtc::test::RtpFileSource> file_source( + webrtc::test::RtpFileSource::Create(argv[1])); + assert(file_source.get()); + // Set RTP extension ID. + bool print_audio_level = false; + if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) { + print_audio_level = true; + file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, + FLAGS_audio_level); + } + + FILE* out_file; + if (argc == 3) { + out_file = fopen(argv[2], "wt"); + if (!out_file) { + printf("Cannot open output file %s\n", argv[2]); + return -1; + } + printf("Output file: %s\n\n", argv[2]); + } else { + out_file = stdout; + } + + // Print file header. + fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M SSRC"); + if (print_audio_level) { + fprintf(out_file, " AuLvl (V)"); + } + fprintf(out_file, "\n"); + + webrtc::scoped_ptr<webrtc::test::Packet> packet; + while (!file_source->EndOfFile()) { + packet.reset(file_source->NextPacket()); + if (!packet.get()) { + // This is probably an RTCP packet. Move on to the next one. + continue; + } + assert(packet.get()); + // Write packet data to file. + fprintf(out_file, + "%5u %10u %10u %5i %5i %2i %#08X", + packet->header().sequenceNumber, + packet->header().timestamp, + static_cast<unsigned int>(packet->time_ms()), + static_cast<int>(packet->packet_length_bytes()), + packet->header().payloadType, + packet->header().markerBit, + packet->header().ssrc); + if (print_audio_level && packet->header().extension.hasAudioLevel) { + // |audioLevel| consists of one bit for "V" and then 7 bits level. + fprintf(out_file, + " %5u (%1i)", + packet->header().extension.audioLevel & 0x7F, + (packet->header().extension.audioLevel & 0x80) == 0 ? 0 : 1); + } + fprintf(out_file, "\n"); + + if (packet->header().payloadType == FLAGS_red) { + std::list<webrtc::RTPHeader*> red_headers; + packet->ExtractRedHeaders(&red_headers); + while (!red_headers.empty()) { + webrtc::RTPHeader* red = red_headers.front(); + assert(red); + fprintf(out_file, + "* %5u %10u %10u %5i\n", + red->sequenceNumber, + red->timestamp, + static_cast<unsigned int>(packet->time_ms()), + red->payloadType); + red_headers.pop_front(); + delete red; + } + } + } + + fclose(in_file); + fclose(out_file); + + return 0; +} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc new file mode 100644 index 00000000000..6490d46857c --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc @@ -0,0 +1,147 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" + +#include <assert.h> +#include <string.h> +#ifdef WIN32 +#include <winsock2.h> +#else +#include <netinet/in.h> +#endif + +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" + +namespace webrtc { +namespace test { + +RtpFileSource* RtpFileSource::Create(const std::string& file_name) { + RtpFileSource* source = new RtpFileSource; + assert(source); + if (!source->OpenFile(file_name) || !source->SkipFileHeader()) { + assert(false); + delete source; + return NULL; + } + return source; +} + +RtpFileSource::~RtpFileSource() { + if (in_file_) + fclose(in_file_); +} + +bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type, + uint8_t id) { + assert(parser_.get()); + return parser_->RegisterRtpHeaderExtension(type, id); +} + +Packet* RtpFileSource::NextPacket() { + while (!EndOfFile()) { + uint16_t length; + if (fread(&length, sizeof(length), 1, in_file_) == 0) { + assert(false); + return NULL; + } + length = ntohs(length); + + uint16_t plen; + if (fread(&plen, sizeof(plen), 1, in_file_) == 0) { + assert(false); + return NULL; + } + plen = ntohs(plen); + + uint32_t offset; + if (fread(&offset, sizeof(offset), 1, in_file_) == 0) { + assert(false); + return NULL; + } + offset = ntohl(offset); + + // Use length here because a plen of 0 specifies RTCP. + assert(length >= kPacketHeaderSize); + size_t packet_size_bytes = length - kPacketHeaderSize; + if (packet_size_bytes == 0) { + // May be an RTCP packet. + // Read the next one. + continue; + } + scoped_ptr<uint8_t> packet_memory(new uint8_t[packet_size_bytes]); + if (fread(packet_memory.get(), 1, packet_size_bytes, in_file_) != + packet_size_bytes) { + assert(false); + return NULL; + } + scoped_ptr<Packet> packet(new Packet(packet_memory.release(), + packet_size_bytes, + plen, + offset, + *parser_.get())); + if (!packet->valid_header()) { + assert(false); + return NULL; + } + return packet.release(); + } + return NULL; +} + +bool RtpFileSource::EndOfFile() const { + assert(in_file_); + return ftell(in_file_) >= file_end_; +} + +RtpFileSource::RtpFileSource() + : PacketSource(), + in_file_(NULL), + file_end_(-1), + parser_(RtpHeaderParser::Create()) {} + +bool RtpFileSource::OpenFile(const std::string& file_name) { + in_file_ = fopen(file_name.c_str(), "rb"); + assert(in_file_); + if (in_file_ == NULL) { + return false; + } + + // Find out how long the file is. + fseek(in_file_, 0, SEEK_END); + file_end_ = ftell(in_file_); + rewind(in_file_); + return true; +} + +bool RtpFileSource::SkipFileHeader() { + char firstline[kFirstLineLength]; + assert(in_file_); + if (fgets(firstline, kFirstLineLength, in_file_) == NULL) { + assert(false); + return false; + } + // Check that the first line is ok. + if ((strncmp(firstline, "#!rtpplay1.0", 12) != 0) && + (strncmp(firstline, "#!RTPencode1.0", 14) != 0)) { + assert(false); + return false; + } + // Skip the file header. + if (fseek(in_file_, kRtpFileHeaderSize, SEEK_CUR) != 0) { + assert(false); + return false; + } + return true; +} + +} // namespace test +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h new file mode 100644 index 00000000000..6b92a88698f --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h @@ -0,0 +1,67 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ + +#include <stdio.h> +#include <string> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/common_types.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +class RtpHeaderParser; + +namespace test { + +class RtpFileSource : public PacketSource { + public: + // Creates an RtpFileSource reading from |file_name|. If the file cannot be + // opened, or has the wrong format, NULL will be returned. + static RtpFileSource* Create(const std::string& file_name); + + virtual ~RtpFileSource(); + + // Registers an RTP header extension and binds it to |id|. + virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); + + // Returns a pointer to the next packet. Returns NULL if end of file was + // reached, or if a the data was corrupt. + virtual Packet* NextPacket(); + + // Returns true if the end of file has been reached. + virtual bool EndOfFile() const; + + private: + static const int kFirstLineLength = 40; + static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; + static const size_t kPacketHeaderSize = 8; + + RtpFileSource(); + + bool OpenFile(const std::string& file_name); + + bool SkipFileHeader(); + + FILE* in_file_; + int64_t file_end_; + scoped_ptr<RtpHeaderParser> parser_; + + DISALLOW_COPY_AND_ASSIGN(RtpFileSource); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc index 8d9a89d5450..17ac209f1d9 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc @@ -10,7 +10,7 @@ #include <assert.h> -#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" namespace webrtc { namespace test { diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h index ece7ef29808..d3824c8d22d 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_RTP_GENERATOR_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_RTP_GENERATOR_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -54,4 +54,4 @@ class RtpGenerator { } // namespace test } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_RTP_GENERATOR_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c deleted file mode 100644 index 3128f21f4a0..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c +++ /dev/null @@ -1,41 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This function "unmutes" a vector on a sample by sample basis. - */ - -#include "dsp_helpfunctions.h" - -#include "signal_processing_library.h" - - -void WebRtcNetEQ_UnmuteSignal(int16_t *pw16_inVec, int16_t *startMuteFact, - int16_t *pw16_outVec, int16_t unmuteFact, - int16_t N) -{ - int i; - uint16_t w16_tmp; - int32_t w32_tmp; - - w16_tmp = (uint16_t) *startMuteFact; - w32_tmp = WEBRTC_SPL_LSHIFT_W32((int32_t)w16_tmp,6) + 32; - for (i = 0; i < N; i++) - { - pw16_outVec[i] - = (int16_t) ((WEBRTC_SPL_MUL_16_16(w16_tmp, pw16_inVec[i]) + 8192) >> 14); - w32_tmp += unmuteFact; - w32_tmp = WEBRTC_SPL_MAX(0, w32_tmp); - w16_tmp = (uint16_t) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 6); /* 20 - 14 = 6 */ - w16_tmp = WEBRTC_SPL_MIN(16384, w16_tmp); - } - *startMuteFact = (int16_t) w16_tmp; -} - diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c b/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c deleted file mode 100644 index fad690d0810..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c +++ /dev/null @@ -1,1769 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Implementation of main NetEQ API. - */ - -#include "webrtc_neteq.h" -#include "webrtc_neteq_internal.h" - -#include <assert.h> -#include <string.h> - -#include "typedefs.h" -#include "signal_processing_library.h" - -#include "neteq_error_codes.h" -#include "mcu_dsp_common.h" -#include "rtcp.h" - -#define RETURN_ON_ERROR( macroExpr, macroInstPtr ) { \ - if ((macroExpr) != 0) { \ - if ((macroExpr) == -1) { \ - (macroInstPtr)->ErrorCode = - (NETEQ_OTHER_ERROR); \ - } else { \ - (macroInstPtr)->ErrorCode = -((int16_t) (macroExpr)); \ - } \ - return(-1); \ - } } - -int WebRtcNetEQ_strncpy(char *strDest, int numberOfElements, - const char *strSource, int count) -{ - /* check vector lengths */ - if (count > numberOfElements) - { - strDest[0] = '\0'; - return (-1); - } - else - { - strncpy(strDest, strSource, count); - return (0); - } -} - -/********************************************************** - * NETEQ Functions - */ - -/***************************************** - * Error functions - */ - -int WebRtcNetEQ_GetErrorCode(void *inst) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - return (NetEqMainInst->ErrorCode); -} - -int WebRtcNetEQ_GetErrorName(int errorCode, char *errorName, int maxStrLen) -{ - if ((errorName == NULL) || (maxStrLen <= 0)) - { - return (-1); - } - - if (errorCode < 0) - { - errorCode = -errorCode; // absolute value - } - - switch (errorCode) - { - case 1: // could be -1 - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "OTHER_ERROR", maxStrLen); - break; - } - case 1001: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_INSTRUCTION", maxStrLen); - break; - } - case 1002: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_NETWORK_TYPE", maxStrLen); - break; - } - case 1003: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_DELAYVALUE", maxStrLen); - break; - } - case 1004: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_PLAYOUTMODE", maxStrLen); - break; - } - case 1005: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CORRUPT_INSTANCE", maxStrLen); - break; - } - case 1006: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "ILLEGAL_MASTER_SLAVE_SWITCH", maxStrLen); - break; - } - case 1007: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "MASTER_SLAVE_ERROR", maxStrLen); - break; - } - case 2001: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_BUFSTAT_DECISION", maxStrLen); - break; - } - case 2002: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_DECODING", maxStrLen); - break; - } - case 2003: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_SAMPLEUNDERRUN", maxStrLen); - break; - } - case 2004: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_DECODED_TOO_MUCH", - maxStrLen); - break; - } - case 3001: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_CNG_ERROR", maxStrLen); - break; - } - case 3002: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_UNKNOWNPAYLOAD", maxStrLen); - break; - } - case 3003: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_BUFFERINSERT_ERROR", maxStrLen); - break; - } - case 4001: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_INIT_ERROR", maxStrLen); - break; - } - case 4002: - case 4003: - case 4004: - case 4005: - case 4006: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_INSERT_ERROR1", maxStrLen); - break; - } - case 4007: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_G723_HEADER", maxStrLen); - break; - } - case 4008: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_NONEXISTING_PACKET", maxStrLen); - break; - } - case 4009: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_NOT_INITIALIZED", maxStrLen); - break; - } - case 4010: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "AMBIGUOUS_ILBC_FRAME_SIZE", maxStrLen); - break; - } - case 5001: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_FULL", maxStrLen); - break; - } - case 5002: - case 5003: - case 5004: - case 5005: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_NOT_EXIST", maxStrLen); - break; - } - case 5006: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNKNOWN_CODEC", maxStrLen); - break; - } - case 5007: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_PAYLOAD_TAKEN", maxStrLen); - break; - } - case 5008: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNSUPPORTED_CODEC", maxStrLen); - break; - } - case 5009: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNSUPPORTED_FS", maxStrLen); - break; - } - case 6001: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_DEC_PARAMETER_ERROR", maxStrLen); - break; - } - case 6002: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_INSERT_ERROR", maxStrLen); - break; - } - case 6003: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_GEN_UNKNOWN_SAMP_FREQ", maxStrLen); - break; - } - case 6004: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_NOT_SUPPORTED", maxStrLen); - break; - } - case 7001: - case 7002: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RED_SPLIT_ERROR", maxStrLen); - break; - } - case 7003: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RTP_TOO_SHORT_PACKET", maxStrLen); - break; - } - case 7004: - { - WebRtcNetEQ_strncpy(errorName, maxStrLen, "RTP_CORRUPT_PACKET", maxStrLen); - break; - } - default: - { - /* check for decoder error ranges */ - if (errorCode >= 6010 && errorCode <= 6810) - { - /* iSAC error code */ - WebRtcNetEQ_strncpy(errorName, maxStrLen, "iSAC ERROR", maxStrLen); - break; - } - - WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_ERROR", maxStrLen); - return (-1); - } - } - - return (0); -} - -/* Assign functions (create not allowed in order to avoid malloc in lib) */ -int WebRtcNetEQ_AssignSize(int *sizeinbytes) -{ - *sizeinbytes = (sizeof(MainInst_t) * 2) / sizeof(int16_t); - return (0); -} - -int WebRtcNetEQ_Assign(void **inst, void *NETEQ_inst_Addr) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) NETEQ_inst_Addr; - *inst = NETEQ_inst_Addr; - if (*inst == NULL) return (-1); - - WebRtcSpl_Init(); - - /* Clear memory */ - WebRtcSpl_MemSetW16((int16_t*) NetEqMainInst, 0, - (sizeof(MainInst_t) / sizeof(int16_t))); - ok = WebRtcNetEQ_McuReset(&NetEqMainInst->MCUinst); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (0); -} - -int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecoder *codec, - int noOfCodecs, enum WebRtcNetEQNetworkType nwType, - int *MaxNoOfPackets, int *sizeinbytes, - int* per_packet_overhead_bytes) -{ - int ok = 0; - int multiplier; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - *MaxNoOfPackets = 0; - *sizeinbytes = 0; - ok = WebRtcNetEQ_GetDefaultCodecSettings(codec, noOfCodecs, sizeinbytes, - MaxNoOfPackets, - per_packet_overhead_bytes); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - if (nwType == kUDPNormal) - { - multiplier = 1; - } - else if (nwType == kUDPVideoSync) - { - multiplier = 4; - } - else if (nwType == kTCPNormal) - { - multiplier = 4; - } - else if (nwType == kTCPLargeJitter) - { - multiplier = 8; - } - else if (nwType == kTCPXLargeJitter) - { - multiplier = 12; - } - else - { - NetEqMainInst->ErrorCode = -FAULTY_NETWORK_TYPE; - return (-1); - } - *MaxNoOfPackets = (*MaxNoOfPackets) * multiplier; - *sizeinbytes = (*sizeinbytes) * multiplier; - return 0; -} - -int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr, - int sizeinbytes) -{ - int ok; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - ok = WebRtcNetEQ_PacketBufferInit(&NetEqMainInst->MCUinst.PacketBuffer_inst, - MaxNoOfPackets, (int16_t*) NETEQ_Buffer_Addr, (sizeinbytes >> 1)); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -/************************************************ - * Init functions - */ - -/**************************************************************************** - * WebRtcNetEQ_Init(...) - * - * Initialize NetEQ. - * - * Input: - * - inst : NetEQ instance - * - fs : Initial sample rate in Hz (may change with payload) - * - * Output: - * - inst : Initialized NetEQ instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_Init(void *inst, uint16_t fs) -{ - int ok = 0; - - /* Typecast inst to internal instance format */ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - if (NetEqMainInst == NULL) - { - return (-1); - } - -#ifdef NETEQ_VAD - /* Start out with no PostDecode VAD instance */ - NetEqMainInst->DSPinst.VADInst.VADState = NULL; - /* Also set all VAD function pointers to NULL */ - NetEqMainInst->DSPinst.VADInst.initFunction = NULL; - NetEqMainInst->DSPinst.VADInst.setmodeFunction = NULL; - NetEqMainInst->DSPinst.VADInst.VADFunction = NULL; -#endif /* NETEQ_VAD */ - - ok = WebRtcNetEQ_DSPinit(NetEqMainInst); /* Init addresses between MCU and DSP */ - RETURN_ON_ERROR(ok, NetEqMainInst); - - ok = WebRtcNetEQ_DSPInit(&NetEqMainInst->DSPinst, fs); /* Init dsp side */ - RETURN_ON_ERROR(ok, NetEqMainInst); - /* set BGN mode to default, since it is not cleared by DSP init function */ - NetEqMainInst->DSPinst.BGNInst.bgnMode = BGN_ON; - - /* init statistics functions and counters */ - ok = WebRtcNetEQ_ClearInCallStats(&NetEqMainInst->DSPinst); - RETURN_ON_ERROR(ok, NetEqMainInst); - ok = WebRtcNetEQ_ClearPostCallStats(&NetEqMainInst->DSPinst); - RETURN_ON_ERROR(ok, NetEqMainInst); - ok = WebRtcNetEQ_ResetMcuJitterStat(&NetEqMainInst->MCUinst); - RETURN_ON_ERROR(ok, NetEqMainInst); - - /* flush packet buffer */ - ok = WebRtcNetEQ_PacketBufferFlush(&NetEqMainInst->MCUinst.PacketBuffer_inst); - RETURN_ON_ERROR(ok, NetEqMainInst); - - /* set some variables to initial values */ - NetEqMainInst->MCUinst.current_Codec = -1; - NetEqMainInst->MCUinst.current_Payload = -1; - NetEqMainInst->MCUinst.first_packet = 1; - NetEqMainInst->MCUinst.one_desc = 0; - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = 0; - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms = 0; - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.maximum_delay_ms = - 10000; - NetEqMainInst->MCUinst.NoOfExpandCalls = 0; - NetEqMainInst->MCUinst.fs = fs; - - /* Not in AV-sync by default. */ - NetEqMainInst->MCUinst.av_sync = 0; - -#ifdef NETEQ_ATEVENT_DECODE - /* init DTMF decoder */ - ok = WebRtcNetEQ_DtmfDecoderInit(&(NetEqMainInst->MCUinst.DTMF_inst),fs,560); - RETURN_ON_ERROR(ok, NetEqMainInst); -#endif - - /* init RTCP statistics */ - WebRtcNetEQ_RTCPInit(&(NetEqMainInst->MCUinst.RTCP_inst), 0); - - /* set BufferStat struct to zero */ - WebRtcSpl_MemSetW16((int16_t*) &(NetEqMainInst->MCUinst.BufferStat_inst), 0, - sizeof(BufstatsInst_t) / sizeof(int16_t)); - - /* reset automode */ - WebRtcNetEQ_ResetAutomode(&(NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst), - NetEqMainInst->MCUinst.PacketBuffer_inst.maxInsertPositions); - - NetEqMainInst->ErrorCode = 0; - -#ifdef NETEQ_STEREO - /* set master/slave info to undecided */ - NetEqMainInst->masterSlave = 0; -#endif - - /* Set to an invalid value. */ - NetEqMainInst->MCUinst.decoded_packet_sequence_number = -1; - NetEqMainInst->MCUinst.decoded_packet_timestamp = 0; - - return (ok); -} - -int WebRtcNetEQ_FlushBuffers(void *inst) -{ - int ok = 0; - - /* Typecast inst to internal instance format */ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - if (NetEqMainInst == NULL) - { - return (-1); - } - - /* Flush packet buffer */ - ok = WebRtcNetEQ_PacketBufferFlush(&NetEqMainInst->MCUinst.PacketBuffer_inst); - RETURN_ON_ERROR(ok, NetEqMainInst); - - /* Set MCU to wait for new codec */ - NetEqMainInst->MCUinst.first_packet = 1; - - /* Flush speech buffer */ - ok = WebRtcNetEQ_FlushSpeechBuffer(&NetEqMainInst->DSPinst); - RETURN_ON_ERROR(ok, NetEqMainInst); - - return 0; -} - -int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); -#ifdef NETEQ_ATEVENT_DECODE - NetEqMainInst->MCUinst.AVT_PlayoutOn = PlayoutAVTon; - return(0); -#else - if (PlayoutAVTon != 0) - { - NetEqMainInst->ErrorCode = -DTMF_NOT_SUPPORTED; - return (-1); - } - else - { - return (0); - } -#endif -} - -int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs) { - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - if ((DelayInMs < 0) || (DelayInMs > 10000)) { - NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE; - return (-1); - } - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = DelayInMs; - return (0); -} - -int WebRtcNetEQ_SetMinimumDelay(void *inst, int minimum_delay_ms) { - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return -1; - if (minimum_delay_ms < 0 || minimum_delay_ms > 10000) { - NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE; - return -1; - } - if ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.maximum_delay_ms > - 0) && (minimum_delay_ms > - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.maximum_delay_ms)) { - NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE; - return -1; - } - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms = - minimum_delay_ms; - return 0; -} - -int WebRtcNetEQ_SetMaximumDelay(void *inst, int maximum_delay_ms) { - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return -1; - if (maximum_delay_ms < 0 || maximum_delay_ms > 10000) { - NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE; - return -1; - } - if (maximum_delay_ms < - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms) { - NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE; - return -1; - } - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.maximum_delay_ms = - maximum_delay_ms; - return 0; -} - -int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - if ((playoutMode != kPlayoutOn) && (playoutMode != kPlayoutOff) && (playoutMode - != kPlayoutFax) && (playoutMode != kPlayoutStreaming)) - { - NetEqMainInst->ErrorCode = -FAULTY_PLAYOUTMODE; - return (-1); - } - else - { - NetEqMainInst->MCUinst.NetEqPlayoutMode = playoutMode; - return (0); - } -} - -int WebRtcNetEQ_SetBGNMode(void *inst, enum WebRtcNetEQBGNMode bgnMode) -{ - - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - /* Instance sanity */ - if (NetEqMainInst == NULL) return (-1); - - /* Check for corrupt/cleared instance */ - if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst) - { - /* Instance is corrupt */ - NetEqMainInst->ErrorCode = CORRUPT_INSTANCE; - return (-1); - } - - NetEqMainInst->DSPinst.BGNInst.bgnMode = (enum BGNMode) bgnMode; - - return (0); -} - -int WebRtcNetEQ_GetBGNMode(const void *inst, enum WebRtcNetEQBGNMode *bgnMode) -{ - - const MainInst_t *NetEqMainInst = (const MainInst_t*) inst; - - /* Instance sanity */ - if (NetEqMainInst == NULL) return (-1); - - *bgnMode = (enum WebRtcNetEQBGNMode) NetEqMainInst->DSPinst.BGNInst.bgnMode; - - return (0); -} - -/************************************************ - * CodecDB functions - */ - -int WebRtcNetEQ_CodecDbReset(void *inst) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - ok = WebRtcNetEQ_DbReset(&NetEqMainInst->MCUinst.codec_DB_inst); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - - /* set function pointers to NULL to prevent RecOut from using the codec */ - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeRCU = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcAddLatePkt = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeInit = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodePLC = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcGetMDinfo = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcUpdBWEst = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcGetErrorCode = NULL; - - return (0); -} - -int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, int16_t *UsedEntries, - int16_t *MaxEntries) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - *MaxEntries = NUM_CODECS; - *UsedEntries = NetEqMainInst->MCUinst.codec_DB_inst.nrOfCodecs; - return (0); -} - -int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, int16_t Entry, - enum WebRtcNetEQDecoder *codec) -{ - int i; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - *codec = (enum WebRtcNetEQDecoder) 0; - if ((Entry >= 0) && (Entry < NetEqMainInst->MCUinst.codec_DB_inst.nrOfCodecs)) - { - for (i = 0; i < NUM_TOTAL_CODECS; i++) - { - if (NetEqMainInst->MCUinst.codec_DB_inst.position[i] == Entry) - { - *codec = (enum WebRtcNetEQDecoder) i; - } - } - } - else - { - NetEqMainInst->ErrorCode = -(CODEC_DB_NOT_EXIST1); - return (-1); - } - return (0); -} - -int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - ok = WebRtcNetEQ_DbAdd(&NetEqMainInst->MCUinst.codec_DB_inst, codecInst->codec, - codecInst->payloadType, codecInst->funcDecode, codecInst->funcDecodeRCU, - codecInst->funcDecodePLC, codecInst->funcDecodeInit, codecInst->funcAddLatePkt, - codecInst->funcGetMDinfo, codecInst->funcGetPitch, codecInst->funcUpdBWEst, - codecInst->funcDurationEst, codecInst->funcGetErrorCode, - codecInst->codec_state, codecInst->codec_fs); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - - /* check if currently used codec is being removed */ - if (NetEqMainInst->MCUinst.current_Codec == (int16_t) codec) - { - /* set function pointers to NULL to prevent RecOut from using the codec */ - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeRCU = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcAddLatePkt = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeInit = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodePLC = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcGetMDinfo = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcUpdBWEst = NULL; - NetEqMainInst->DSPinst.codec_ptr_inst.funcGetErrorCode = NULL; - } - - ok = WebRtcNetEQ_DbRemove(&NetEqMainInst->MCUinst.codec_DB_inst, codec); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -/********************************* - * Real-time functions - */ - -int WebRtcNetEQ_RecIn(void *inst, int16_t *p_w16datagramstart, int16_t w16_RTPlen, - uint32_t uw32_timeRec) -{ - int ok = 0; - RTPPacket_t RTPpacket; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - - /* Check for corrupt/cleared instance */ - if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst) - { - /* Instance is corrupt */ - NetEqMainInst->ErrorCode = CORRUPT_INSTANCE; - return (-1); - } - - /* Parse RTP header */ - ok = WebRtcNetEQ_RTPPayloadInfo(p_w16datagramstart, w16_RTPlen, &RTPpacket); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - - ok = WebRtcNetEQ_RecInInternal(&NetEqMainInst->MCUinst, &RTPpacket, uw32_timeRec); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -/**************************************************************************** - * WebRtcNetEQ_RecInRTPStruct(...) - * - * Alternative RecIn function, used when the RTP data has already been - * parsed into an RTP info struct (WebRtcNetEQ_RTPInfo). - * - * Input: - * - inst : NetEQ instance - * - rtpInfo : Pointer to RTP info - * - payloadPtr : Pointer to the RTP payload (first byte after header) - * - payloadLenBytes : Length (in bytes) of the payload in payloadPtr - * - timeRec : Receive time (in timestamps of the used codec) - * - * Return value : 0 - Ok - * -1 - Error - */ -int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo, - const uint8_t *payloadPtr, int16_t payloadLenBytes, - uint32_t uw32_timeRec) -{ - int ok = 0; - RTPPacket_t RTPpacket; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) - { - return (-1); - } - - /* Check for corrupt/cleared instance */ - if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst) - { - /* Instance is corrupt */ - NetEqMainInst->ErrorCode = CORRUPT_INSTANCE; - return (-1); - } - - /* Load NetEQ's RTP struct from Module RTP struct */ - RTPpacket.payloadType = rtpInfo->payloadType; - RTPpacket.seqNumber = rtpInfo->sequenceNumber; - RTPpacket.timeStamp = rtpInfo->timeStamp; - RTPpacket.ssrc = rtpInfo->SSRC; - RTPpacket.payload = (const int16_t*) payloadPtr; - RTPpacket.payloadLen = payloadLenBytes; - RTPpacket.starts_byte1 = 0; - - ok = WebRtcNetEQ_RecInInternal(&NetEqMainInst->MCUinst, &RTPpacket, uw32_timeRec); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -int WebRtcNetEQ_RecOut(void *inst, int16_t *pw16_outData, int16_t *pw16_len) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; -#ifdef NETEQ_STEREO - MasterSlaveInfo msInfo; - msInfo.msMode = NETEQ_MONO; -#endif - - if (NetEqMainInst == NULL) return (-1); - - /* Check for corrupt/cleared instance */ - if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst) - { - /* Instance is corrupt */ - NetEqMainInst->ErrorCode = CORRUPT_INSTANCE; - return (-1); - } - -#ifdef NETEQ_STEREO - NetEqMainInst->DSPinst.msInfo = &msInfo; -#endif - - ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData, - pw16_len, 0 /* not BGN only */, NetEqMainInst->MCUinst.av_sync); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -/**************************************************************************** - * WebRtcNetEQ_RecOutMasterSlave(...) - * - * RecOut function for running several NetEQ instances in master/slave mode. - * One master can be used to control several slaves. - * - * Input: - * - inst : NetEQ instance - * - isMaster : Non-zero indicates that this is the master channel - * - msInfo : (slave only) Information from master - * - * Output: - * - inst : Updated NetEQ instance - * - pw16_outData : Pointer to vector where output should be written - * - pw16_len : Pointer to variable where output length is returned - * - msInfo : (master only) Information to slave(s) - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_RecOutMasterSlave(void *inst, int16_t *pw16_outData, - int16_t *pw16_len, void *msInfo, - int16_t isMaster) -{ -#ifndef NETEQ_STEREO - /* Stereo not supported */ - return(-1); -#else - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - if (NetEqMainInst == NULL) return (-1); - - /* Check for corrupt/cleared instance */ - if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst) - { - /* Instance is corrupt */ - NetEqMainInst->ErrorCode = CORRUPT_INSTANCE; - return (-1); - } - - if (msInfo == NULL) - { - /* msInfo not provided */ - NetEqMainInst->ErrorCode = NETEQ_OTHER_ERROR; - return (-1); - } - - /* translate from external to internal Master/Slave information */ - NetEqMainInst->DSPinst.msInfo = (MasterSlaveInfo *) msInfo; - - /* check that we have not done a master/slave switch without first re-initializing */ - if ((NetEqMainInst->masterSlave == 1 && !isMaster) || /* switch from master to slave */ - (NetEqMainInst->masterSlave == 2 && isMaster)) /* switch from slave to master */ - { - NetEqMainInst->ErrorCode = ILLEGAL_MASTER_SLAVE_SWITCH; - return (-1); - } - - if (!isMaster) - { - /* this is the slave */ - NetEqMainInst->masterSlave = 2; - NetEqMainInst->DSPinst.msInfo->msMode = NETEQ_SLAVE; - } - else - { - NetEqMainInst->DSPinst.msInfo->msMode = NETEQ_MASTER; - } - - ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData, - pw16_len, 0 /* not BGN only */, NetEqMainInst->MCUinst.av_sync); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - - if (isMaster) - { - /* this is the master */ - NetEqMainInst->masterSlave = 1; - } - - return (ok); -#endif -} - -int WebRtcNetEQ_GetMasterSlaveInfoSize() -{ -#ifdef NETEQ_STEREO - return (sizeof(MasterSlaveInfo)); -#else - return(-1); -#endif -} - -/* Special RecOut that does not do any decoding. */ -int WebRtcNetEQ_RecOutNoDecode(void *inst, int16_t *pw16_outData, - int16_t *pw16_len) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; -#ifdef NETEQ_STEREO - MasterSlaveInfo msInfo; -#endif - - if (NetEqMainInst == NULL) return (-1); - - /* Check for corrupt/cleared instance */ - if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst) - { - /* Instance is corrupt */ - NetEqMainInst->ErrorCode = CORRUPT_INSTANCE; - return (-1); - } - -#ifdef NETEQ_STEREO - /* keep same mode as before */ - switch (NetEqMainInst->masterSlave) - { - case 1: - { - msInfo.msMode = NETEQ_MASTER; - break; - } - case 2: - { - msInfo.msMode = NETEQ_SLAVE; - break; - } - default: - { - msInfo.msMode = NETEQ_MONO; - break; - } - } - - NetEqMainInst->DSPinst.msInfo = &msInfo; -#endif - - ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData, - pw16_len, 1 /* BGN only */, NetEqMainInst->MCUinst.av_sync); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - ok = WebRtcNetEQ_RTCPGetStats(&NetEqMainInst->MCUinst.RTCP_inst, - &RTCP_inst->fraction_lost, &RTCP_inst->cum_lost, &RTCP_inst->ext_max, - &RTCP_inst->jitter, 0); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst) -{ - int ok = 0; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - ok = WebRtcNetEQ_RTCPGetStats(&NetEqMainInst->MCUinst.RTCP_inst, - &RTCP_inst->fraction_lost, &RTCP_inst->cum_lost, &RTCP_inst->ext_max, - &RTCP_inst->jitter, 1); - if (ok != 0) - { - NetEqMainInst->ErrorCode = -ok; - return (-1); - } - return (ok); -} - -int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, uint32_t *timestamp) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - - if (NetEqMainInst->MCUinst.TSscalingInitialized) - { - *timestamp = WebRtcNetEQ_ScaleTimestampInternalToExternal(&NetEqMainInst->MCUinst, - NetEqMainInst->DSPinst.videoSyncTimestamp); - } - else - { - *timestamp = NetEqMainInst->DSPinst.videoSyncTimestamp; - } - - return (0); -} - -/**************************************************************************** - * WebRtcNetEQ_GetSpeechOutputType(...) - * - * Get the output type for the audio provided by the latest call to - * WebRtcNetEQ_RecOut(). - * - * kOutputNormal = normal audio (possibly processed) - * kOutputPLC = loss concealment through stretching audio - * kOutputCNG = comfort noise (codec-internal or RFC3389) - * kOutputPLCtoCNG = background noise only due to long expand or error - * kOutputVADPassive = PostDecode VAD signalling passive speaker - * - * Input: - * - inst : NetEQ instance - * - * Output: - * - outputType : Output type from enum list WebRtcNetEQOutputType - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType) -{ - /* Typecast to internal instance type */ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - if (NetEqMainInst == NULL) - { - return (-1); - } - - if ((NetEqMainInst->DSPinst.w16_mode & MODE_BGN_ONLY) != 0) - { - /* If last mode was background noise only */ - *outputType = kOutputPLCtoCNG; - - } - else if ((NetEqMainInst->DSPinst.w16_mode == MODE_CODEC_INTERNAL_CNG) - || (NetEqMainInst->DSPinst.w16_mode == MODE_RFC3389CNG)) - { - /* If CN or internal CNG */ - *outputType = kOutputCNG; - - } - else if ((NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND) - && (NetEqMainInst->DSPinst.ExpandInst.w16_expandMuteFactor == 0)) - { - /* Expand mode has faded down to background noise only (very long expand) */ - *outputType = kOutputPLCtoCNG; - - } - else if (NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND) - { - /* PLC mode */ - *outputType = kOutputPLC; - -#ifdef NETEQ_VAD - } - else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 ) - { - /* post-decode VAD says passive speaker */ - *outputType = kOutputVADPassive; -#endif /* NETEQ_VAD */ - - } - else - { - /* Normal speech output type (can still be manipulated, e.g., accelerated) */ - *outputType = kOutputNormal; - } - - return (0); -} - -/********************************** - * Functions related to VQmon - */ - -#define WEBRTC_NETEQ_CONCEALMENTFLAG_LOST 0x01 -#define WEBRTC_NETEQ_CONCEALMENTFLAG_DISCARDED 0x02 -#define WEBRTC_NETEQ_CONCEALMENTFLAG_SUPRESS 0x04 -#define WEBRTC_NETEQ_CONCEALMENTFLAG_CNGACTIVE 0x80 - -int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, uint16_t *validVoiceDurationMs, - uint16_t *concealedVoiceDurationMs, - uint8_t *concealedVoiceFlags) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - int16_t fs_mult; - int16_t ms_lost; - if (NetEqMainInst == NULL) return (-1); - fs_mult = WebRtcSpl_DivW32W16ResW16(NetEqMainInst->MCUinst.fs, 8000); - - ms_lost = WebRtcSpl_DivW32W16ResW16( - (int32_t) NetEqMainInst->DSPinst.w16_concealedTS, (int16_t) (8 * fs_mult)); - if (ms_lost > NetEqMainInst->DSPinst.millisecondsPerCall) ms_lost - = NetEqMainInst->DSPinst.millisecondsPerCall; - - *validVoiceDurationMs = NetEqMainInst->DSPinst.millisecondsPerCall - ms_lost; - *concealedVoiceDurationMs = ms_lost; - if (ms_lost > 0) - { - *concealedVoiceFlags = WEBRTC_NETEQ_CONCEALMENTFLAG_LOST; - } - else - { - *concealedVoiceFlags = 0; - } - NetEqMainInst->DSPinst.w16_concealedTS -= ms_lost * (8 * fs_mult); - - return (0); -} - -int WebRtcNetEQ_VQmonGetConfiguration(void *inst, uint16_t *absMaxDelayMs, - uint8_t *adaptationRate) -{ - /* Dummy check the inst, just to avoid compiler warnings. */ - if (inst == NULL) - { - /* Do nothing. */ - } - - /* Hardcoded variables that are used for VQmon as jitter buffer parameters */ - *absMaxDelayMs = 240; - *adaptationRate = 1; - return (0); -} - -int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, uint16_t *avgDelayMs, - uint16_t *maxDelayMs) -{ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) return (-1); - *avgDelayMs = (uint16_t) (NetEqMainInst->MCUinst.BufferStat_inst.avgDelayMsQ8 >> 8); - *maxDelayMs = (uint16_t) NetEqMainInst->MCUinst.BufferStat_inst.maxDelayMs; - return (0); -} - -/************************************* - * Statistics functions - */ - -/* Get the "in-call" statistics from NetEQ. - * The statistics are reset after the query. */ -int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *stats) - -{ - - uint16_t tempU16; - uint32_t tempU32, tempU32_2; - int numShift; - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - /* Instance sanity */ - if (NetEqMainInst == NULL) return (-1); - - stats->addedSamples = NetEqMainInst->DSPinst.statInst.addedSamples; - - /*******************/ - /* Get buffer size */ - /*******************/ - - if (NetEqMainInst->MCUinst.fs != 0) - { - int32_t temp32; - /* Query packet buffer for number of samples. */ - temp32 = WebRtcNetEQ_PacketBufferGetSize( - &NetEqMainInst->MCUinst.PacketBuffer_inst, - &NetEqMainInst->MCUinst.codec_DB_inst, - NetEqMainInst->MCUinst.av_sync); - - /* Divide by sample rate. - * Calculate temp32 * 1000 / fs to get result in ms. */ - stats->currentBufferSize = (uint16_t) - WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs); - - /* Add number of samples yet to play in sync buffer. */ - temp32 = (int32_t) (NetEqMainInst->DSPinst.endPosition - - NetEqMainInst->DSPinst.curPosition); - stats->currentBufferSize += (uint16_t) - WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs); - } - else - { - /* Sample rate not initialized. */ - stats->currentBufferSize = 0; - } - - /***************************/ - /* Get optimal buffer size */ - /***************************/ - - if (NetEqMainInst->MCUinst.fs != 0) - { - /* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */ - stats->preferredBufferSize - = (uint16_t) WEBRTC_SPL_MUL_16_16( - (int16_t) ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.optBufLevel) >> 8), /* optimal buffer level in packets shifted to Q0 */ - WebRtcSpl_DivW32W16ResW16( - (int32_t) NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.packetSpeechLenSamp, /* samples per packet */ - WebRtcSpl_DivW32W16ResW16( (int32_t) NetEqMainInst->MCUinst.fs, (int16_t) 1000 ) /* samples per ms */ - ) ); - - /* add extra delay */ - if (NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs > 0) - { - stats->preferredBufferSize - += NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs; - } - } - else - { - /* sample rate not initialized */ - stats->preferredBufferSize = 0; - } - - /***********************************/ - /* Check if jitter peaks are found */ - /***********************************/ - - stats->jitterPeaksFound = - NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.peakFound; - - /***********************/ - /* Calculate loss rate */ - /***********************/ - - /* timestamps elapsed since last report */ - tempU32 = NetEqMainInst->MCUinst.lastReportTS; - - if (NetEqMainInst->MCUinst.lostTS == 0) - { - /* no losses */ - stats->currentPacketLossRate = 0; - } - else if (NetEqMainInst->MCUinst.lostTS < tempU32) - { - /* calculate shifts; we want the result in Q14 */ - numShift = WebRtcSpl_NormU32(NetEqMainInst->MCUinst.lostTS); /* numerator shift for normalize */ - - if (numShift < 14) - { - /* cannot shift numerator 14 steps; shift denominator too */ - tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */ - } - else - { - /* shift no more than 14 steps */ - numShift = 14; - } - - if (tempU32 == 0) - { - /* check for zero denominator; result should be zero in this case */ - stats->currentPacketLossRate = 0; - } - else - { - /* check that denominator fits in signed 16-bit */ - while (tempU32 > WEBRTC_SPL_WORD16_MAX) - { - tempU32 >>= 1; /* right-shift 1 step */ - numShift--; /* compensate in numerator */ - } - tempU16 = (uint16_t) tempU32; - - /* do the shift of numerator */ - tempU32 - = WEBRTC_SPL_SHIFT_W32( (uint32_t) NetEqMainInst->MCUinst.lostTS, numShift); - - stats->currentPacketLossRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, - tempU16); - } - } - else - { - /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */ - /* set loss rate = 1 */ - stats->currentPacketLossRate = 1 << 14; /* 1 in Q14 */ - } - - /**************************/ - /* Calculate discard rate */ - /**************************/ - - /* timestamps elapsed since last report */ - tempU32 = NetEqMainInst->MCUinst.lastReportTS; - - /* number of discarded samples */ - tempU32_2 - = WEBRTC_SPL_MUL_16_U16( (int16_t) NetEqMainInst->MCUinst.PacketBuffer_inst.packSizeSamples, - NetEqMainInst->MCUinst.PacketBuffer_inst.discardedPackets); - - if (tempU32_2 == 0) - { - /* no discarded samples */ - stats->currentDiscardRate = 0; - } - else if (tempU32_2 < tempU32) - { - /* calculate shifts; we want the result in Q14 */ - numShift = WebRtcSpl_NormU32(tempU32_2); /* numerator shift for normalize */ - - if (numShift < 14) - { - /* cannot shift numerator 14 steps; shift denominator too */ - tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */ - } - else - { - /* shift no more than 14 steps */ - numShift = 14; - } - - if (tempU32 == 0) - { - /* check for zero denominator; result should be zero in this case */ - stats->currentDiscardRate = 0; - } - else - { - /* check that denominator fits in signed 16-bit */ - while (tempU32 > WEBRTC_SPL_WORD16_MAX) - { - tempU32 >>= 1; /* right-shift 1 step */ - numShift--; /* compensate in numerator */ - } - tempU16 = (uint16_t) tempU32; - - /* do the shift of numerator */ - tempU32 = WEBRTC_SPL_SHIFT_W32( tempU32_2, numShift); - - stats->currentDiscardRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, tempU16); - } - } - else - { - /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */ - /* set loss rate = 1 */ - stats->currentDiscardRate = 1 << 14; /* 1 in Q14 */ - } - - /*************************************************************/ - /* Calculate Accelerate, Expand and Pre-emptive Expand rates */ - /*************************************************************/ - - /* timestamps elapsed since last report */ - tempU32 = NetEqMainInst->MCUinst.lastReportTS; - - if (NetEqMainInst->DSPinst.statInst.accelerateLength == 0) - { - /* no accelerate */ - stats->currentAccelerateRate = 0; - } - else if (NetEqMainInst->DSPinst.statInst.accelerateLength < tempU32) - { - /* calculate shifts; we want the result in Q14 */ - numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.accelerateLength); /* numerator shift for normalize */ - - if (numShift < 14) - { - /* cannot shift numerator 14 steps; shift denominator too */ - tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */ - } - else - { - /* shift no more than 14 steps */ - numShift = 14; - } - - if (tempU32 == 0) - { - /* check for zero denominator; result should be zero in this case */ - stats->currentAccelerateRate = 0; - } - else - { - /* check that denominator fits in signed 16-bit */ - while (tempU32 > WEBRTC_SPL_WORD16_MAX) - { - tempU32 >>= 1; /* right-shift 1 step */ - numShift--; /* compensate in numerator */ - } - tempU16 = (uint16_t) tempU32; - - /* do the shift of numerator */ - tempU32 - = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.accelerateLength, numShift); - - stats->currentAccelerateRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, - tempU16); - } - } - else - { - /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */ - /* set loss rate = 1 */ - stats->currentAccelerateRate = 1 << 14; /* 1 in Q14 */ - } - - /* timestamps elapsed since last report */ - tempU32 = NetEqMainInst->MCUinst.lastReportTS; - - if (NetEqMainInst->DSPinst.statInst.expandLength == 0) - { - /* no expand */ - stats->currentExpandRate = 0; - } - else if (NetEqMainInst->DSPinst.statInst.expandLength < tempU32) - { - /* calculate shifts; we want the result in Q14 */ - numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.expandLength); /* numerator shift for normalize */ - - if (numShift < 14) - { - /* cannot shift numerator 14 steps; shift denominator too */ - tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */ - } - else - { - /* shift no more than 14 steps */ - numShift = 14; - } - - if (tempU32 == 0) - { - /* check for zero denominator; result should be zero in this case */ - stats->currentExpandRate = 0; - } - else - { - /* check that denominator fits in signed 16-bit */ - while (tempU32 > WEBRTC_SPL_WORD16_MAX) - { - tempU32 >>= 1; /* right-shift 1 step */ - numShift--; /* compensate in numerator */ - } - tempU16 = (uint16_t) tempU32; - - /* do the shift of numerator */ - tempU32 - = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.expandLength, numShift); - - stats->currentExpandRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, tempU16); - } - } - else - { - /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */ - /* set loss rate = 1 */ - stats->currentExpandRate = 1 << 14; /* 1 in Q14 */ - } - - /* timestamps elapsed since last report */ - tempU32 = NetEqMainInst->MCUinst.lastReportTS; - - if (NetEqMainInst->DSPinst.statInst.preemptiveLength == 0) - { - /* no pre-emptive expand */ - stats->currentPreemptiveRate = 0; - } - else if (NetEqMainInst->DSPinst.statInst.preemptiveLength < tempU32) - { - /* calculate shifts; we want the result in Q14 */ - numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.preemptiveLength); /* numerator shift for normalize */ - - if (numShift < 14) - { - /* cannot shift numerator 14 steps; shift denominator too */ - tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */ - } - else - { - /* shift no more than 14 steps */ - numShift = 14; - } - - if (tempU32 == 0) - { - /* check for zero denominator; result should be zero in this case */ - stats->currentPreemptiveRate = 0; - } - else - { - /* check that denominator fits in signed 16-bit */ - while (tempU32 > WEBRTC_SPL_WORD16_MAX) - { - tempU32 >>= 1; /* right-shift 1 step */ - numShift--; /* compensate in numerator */ - } - tempU16 = (uint16_t) tempU32; - - /* do the shift of numerator */ - tempU32 - = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.preemptiveLength, numShift); - - stats->currentPreemptiveRate = (uint16_t) WebRtcSpl_DivU32U16(tempU32, - tempU16); - } - } - else - { - /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */ - /* set loss rate = 1 */ - stats->currentPreemptiveRate = 1 << 14; /* 1 in Q14 */ - } - - stats->clockDriftPPM = WebRtcNetEQ_AverageIAT( - &NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst); - - /* reset counters */ - WebRtcNetEQ_ResetMcuInCallStats(&(NetEqMainInst->MCUinst)); - WebRtcNetEQ_ClearInCallStats(&(NetEqMainInst->DSPinst)); - - return (0); -} - -int WebRtcNetEQ_GetRawFrameWaitingTimes(void *inst, - int max_length, - int* waiting_times_ms) { - int i = 0; - MainInst_t *main_inst = (MainInst_t*) inst; - if (main_inst == NULL) return -1; - - while ((i < max_length) && (i < main_inst->MCUinst.len_waiting_times)) { - waiting_times_ms[i] = main_inst->MCUinst.waiting_times[i] * - main_inst->DSPinst.millisecondsPerCall; - ++i; - } - assert(i <= kLenWaitingTimes); - WebRtcNetEQ_ResetWaitingTimeStats(&main_inst->MCUinst); - return i; -} - -/**************************************************************************** - * WebRtcNetEQ_SetVADInstance(...) - * - * Provide a pointer to an allocated VAD instance. If function is never - * called or it is called with NULL pointer as VAD_inst, the post-decode - * VAD functionality is disabled. Also provide pointers to init, setmode - * and VAD functions. These are typically pointers to WebRtcVad_Init, - * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in the - * interface file webrtc_vad.h. - * - * Input: - * - NetEQ_inst : NetEQ instance - * - VADinst : VAD instance - * - initFunction : Pointer to VAD init function - * - setmodeFunction : Pointer to VAD setmode function - * - VADfunction : Pointer to VAD function - * - * Output: - * - NetEQ_inst : Updated NetEQ instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_SetVADInstance(void *NetEQ_inst, void *VAD_inst, - WebRtcNetEQ_VADInitFunction initFunction, - WebRtcNetEQ_VADSetmodeFunction setmodeFunction, - WebRtcNetEQ_VADFunction VADFunction) -{ - - /* Typecast to internal instance type */ - MainInst_t *NetEqMainInst = (MainInst_t*) NetEQ_inst; - if (NetEqMainInst == NULL) - { - return (-1); - } - -#ifdef NETEQ_VAD - - /* Store pointer in PostDecode VAD struct */ - NetEqMainInst->DSPinst.VADInst.VADState = VAD_inst; - - /* Store function pointers */ - NetEqMainInst->DSPinst.VADInst.initFunction = initFunction; - NetEqMainInst->DSPinst.VADInst.setmodeFunction = setmodeFunction; - NetEqMainInst->DSPinst.VADInst.VADFunction = VADFunction; - - /* Call init function and return the result (ok or fail) */ - return(WebRtcNetEQ_InitVAD(&NetEqMainInst->DSPinst.VADInst, NetEqMainInst->DSPinst.fs)); - -#else /* NETEQ_VAD not defined */ - return (-1); -#endif /* NETEQ_VAD */ - -} - -/**************************************************************************** - * WebRtcNetEQ_SetVADMode(...) - * - * Pass an aggressiveness mode parameter to the post-decode VAD instance. - * If this function is never called, mode 0 (quality mode) is used as default. - * - * Input: - * - inst : NetEQ instance - * - mode : mode parameter (same range as WebRtc VAD mode) - * - * Output: - * - inst : Updated NetEQ instance - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcNetEQ_SetVADMode(void *inst, int mode) -{ - - /* Typecast to internal instance type */ - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst == NULL) - { - return (-1); - } - -#ifdef NETEQ_VAD - - /* Set mode and return result */ - return(WebRtcNetEQ_SetVADModeInternal(&NetEqMainInst->DSPinst.VADInst, mode)); - -#else /* NETEQ_VAD not defined */ - return (-1); -#endif /* NETEQ_VAD */ - -} - -void WebRtcNetEQ_GetProcessingActivity(void *inst, - WebRtcNetEQ_ProcessingActivity *stats) { - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - - stats->accelerate_bgn_samples = - NetEqMainInst->DSPinst.activity_stats.accelerate_bgn_samples; - stats->accelerate_normal_samples = - NetEqMainInst->DSPinst.activity_stats.accelarate_normal_samples; - - stats->expand_bgn_sampels = - NetEqMainInst->DSPinst.activity_stats.expand_bgn_samples; - stats->expand_normal_samples = - NetEqMainInst->DSPinst.activity_stats.expand_normal_samples; - - stats->preemptive_expand_bgn_samples = - NetEqMainInst->DSPinst.activity_stats.preemptive_expand_bgn_samples; - stats->preemptive_expand_normal_samples = - NetEqMainInst->DSPinst.activity_stats.preemptive_expand_normal_samples; - - stats->merge_expand_bgn_samples = - NetEqMainInst->DSPinst.activity_stats.merge_expand_bgn_samples; - stats->merge_expand_normal_samples = - NetEqMainInst->DSPinst.activity_stats.merge_expand_normal_samples; - - WebRtcNetEQ_ClearActivityStats(&NetEqMainInst->DSPinst); -} - -void WebRtcNetEQ_EnableAVSync(void* inst, int enable) { - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - NetEqMainInst->MCUinst.av_sync = (enable != 0) ? 1 : 0; -} - -int WebRtcNetEQ_RecInSyncRTP(void* inst, WebRtcNetEQ_RTPInfo* rtp_info, - uint32_t receive_timestamp) { - MainInst_t *NetEqMainInst = (MainInst_t*) inst; - if (NetEqMainInst->MCUinst.av_sync == 0) - return -1; - if (WebRtcNetEQ_RecInRTPStruct(inst, rtp_info, kSyncPayload, - SYNC_PAYLOAD_LEN_BYTES, - receive_timestamp) < 0) { - return -1; - } - return SYNC_PAYLOAD_LEN_BYTES; -} - -int WebRtcNetEQ_GetRequiredDelayMs(const void* inst) { - const MainInst_t* NetEqMainInst = (MainInst_t*)inst; - const AutomodeInst_t* auto_mode = (NetEqMainInst == NULL) ? NULL : - &NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst; - - /* Instance sanity */ - if (NetEqMainInst == NULL || auto_mode == NULL) - return 0; - - if (NetEqMainInst->MCUinst.fs == 0) - return 0; // Sampling rate not initialized. - - /* |required_delay_q8| has the unit of packets in Q8 domain, therefore, - * the corresponding delay is - * required_delay_ms = (1000 * required_delay_q8 * samples_per_packet / - * sample_rate_hz) / 256; - */ - return (auto_mode->required_delay_q8 * - ((auto_mode->packetSpeechLenSamp * 1000) / NetEqMainInst->MCUinst.fs) + - 128) >> 8; -} - -int WebRtcNetEQ_DecodedRtpInfo(const void* inst, - int* sequence_number, - uint32_t* timestamp) { - const MainInst_t *NetEqMainInst = (inst == NULL) ? NULL : - (const MainInst_t*) inst; - if (NetEqMainInst->MCUinst.decoded_packet_sequence_number < 0) - return -1; - *sequence_number = NetEqMainInst->MCUinst.decoded_packet_sequence_number; - *timestamp = NetEqMainInst->MCUinst.decoded_packet_timestamp; - return 0; -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc deleted file mode 100644 index c37f8990a8b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc +++ /dev/null @@ -1,778 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file includes unit tests for NetEQ. - */ - -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" - -#include <stdlib.h> -#include <string.h> // memset - -#include <set> -#include <sstream> -#include <string> -#include <vector> - -#include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h" -#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h" -#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h" -#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h" -#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -class RefFiles { - public: - RefFiles(const std::string& input_file, const std::string& output_file); - ~RefFiles(); - template<class T> void ProcessReference(const T& test_results); - template<typename T, size_t n> void ProcessReference( - const T (&test_results)[n], - size_t length); - template<typename T, size_t n> void WriteToFile( - const T (&test_results)[n], - size_t length); - template<typename T, size_t n> void ReadFromFileAndCompare( - const T (&test_results)[n], - size_t length); - void WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats); - void ReadFromFileAndCompare(const WebRtcNetEQ_NetworkStatistics& stats); - void WriteToFile(const WebRtcNetEQ_RTCPStat& stats); - void ReadFromFileAndCompare(const WebRtcNetEQ_RTCPStat& stats); - - FILE* input_fp_; - FILE* output_fp_; -}; - -RefFiles::RefFiles(const std::string &input_file, - const std::string &output_file) - : input_fp_(NULL), - output_fp_(NULL) { - if (!input_file.empty()) { - input_fp_ = fopen(input_file.c_str(), "rb"); - EXPECT_TRUE(input_fp_ != NULL); - } - if (!output_file.empty()) { - output_fp_ = fopen(output_file.c_str(), "wb"); - EXPECT_TRUE(output_fp_ != NULL); - } -} - -RefFiles::~RefFiles() { - if (input_fp_) { - EXPECT_EQ(EOF, fgetc(input_fp_)); // Make sure that we reached the end. - fclose(input_fp_); - } - if (output_fp_) fclose(output_fp_); -} - -template<class T> -void RefFiles::ProcessReference(const T& test_results) { - WriteToFile(test_results); - ReadFromFileAndCompare(test_results); -} - -template<typename T, size_t n> -void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) { - WriteToFile(test_results, length); - ReadFromFileAndCompare(test_results, length); -} - -template<typename T, size_t n> -void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) { - if (output_fp_) { - ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_)); - } -} - -template<typename T, size_t n> -void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n], - size_t length) { - if (input_fp_) { - // Read from ref file. - T* ref = new T[length]; - ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_)); - // Compare - ASSERT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length)); - delete [] ref; - } -} - -void RefFiles::WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats) { - if (output_fp_) { - ASSERT_EQ(1u, fwrite(&stats, sizeof(WebRtcNetEQ_NetworkStatistics), 1, - output_fp_)); - } -} - -void RefFiles::ReadFromFileAndCompare( - const WebRtcNetEQ_NetworkStatistics& stats) { - if (input_fp_) { - // Read from ref file. - size_t stat_size = sizeof(WebRtcNetEQ_NetworkStatistics); - WebRtcNetEQ_NetworkStatistics ref_stats; - ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_)); - // Compare - EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size)); - } -} - -void RefFiles::WriteToFile(const WebRtcNetEQ_RTCPStat& stats) { - if (output_fp_) { - ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1, - output_fp_)); - ASSERT_EQ(1u, fwrite(&(stats.cum_lost), sizeof(stats.cum_lost), 1, - output_fp_)); - ASSERT_EQ(1u, fwrite(&(stats.ext_max), sizeof(stats.ext_max), 1, - output_fp_)); - ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1, - output_fp_)); - } -} - -void RefFiles::ReadFromFileAndCompare( - const WebRtcNetEQ_RTCPStat& stats) { - if (input_fp_) { - // Read from ref file. - WebRtcNetEQ_RTCPStat ref_stats; - ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost), - sizeof(ref_stats.fraction_lost), 1, input_fp_)); - ASSERT_EQ(1u, fread(&(ref_stats.cum_lost), sizeof(ref_stats.cum_lost), 1, - input_fp_)); - ASSERT_EQ(1u, fread(&(ref_stats.ext_max), sizeof(ref_stats.ext_max), 1, - input_fp_)); - ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1, - input_fp_)); - // Compare - EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost); - EXPECT_EQ(ref_stats.cum_lost, stats.cum_lost); - EXPECT_EQ(ref_stats.ext_max, stats.ext_max); - EXPECT_EQ(ref_stats.jitter, stats.jitter); - } -} - -class NetEqDecodingTest : public ::testing::Test { - protected: - // NetEQ must be polled for data once every 10 ms. Thus, neither of the - // constants below can be changed. - static const int kTimeStepMs = 10; - static const int kBlockSize8kHz = kTimeStepMs * 8; - static const int kBlockSize16kHz = kTimeStepMs * 16; - static const int kBlockSize32kHz = kTimeStepMs * 32; - static const int kMaxBlockSize = kBlockSize32kHz; - - NetEqDecodingTest(); - virtual void SetUp(); - virtual void TearDown(); - void SelectDecoders(WebRtcNetEQDecoder* used_codec); - void LoadDecoders(); - void OpenInputFile(const std::string &rtp_file); - void Process(NETEQTEST_RTPpacket* rtp_ptr, int16_t* out_len); - void DecodeAndCompare(const std::string &rtp_file, - const std::string &ref_file); - void DecodeAndCheckStats(const std::string &rtp_file, - const std::string &stat_ref_file, - const std::string &rtcp_ref_file); - static void PopulateRtpInfo(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info); - static void PopulateCng(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info, - uint8_t* payload, - int* payload_len); - void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp, - const std::set<uint16_t>& drop_seq_numbers); - - NETEQTEST_NetEQClass* neteq_inst_; - std::vector<NETEQTEST_Decoder*> dec_; - FILE* rtp_fp_; - unsigned int sim_clock_; - int16_t out_data_[kMaxBlockSize]; -}; - -NetEqDecodingTest::NetEqDecodingTest() - : neteq_inst_(NULL), - rtp_fp_(NULL), - sim_clock_(0) { - memset(out_data_, 0, sizeof(out_data_)); -} - -void NetEqDecodingTest::SetUp() { - WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1]; - - SelectDecoders(usedCodec); - neteq_inst_ = new NETEQTEST_NetEQClass(usedCodec, dec_.size(), 8000, - kTCPLargeJitter); - ASSERT_TRUE(neteq_inst_); - LoadDecoders(); -} - -void NetEqDecodingTest::TearDown() { - if (neteq_inst_) - delete neteq_inst_; - for (size_t i = 0; i < dec_.size(); ++i) { - if (dec_[i]) - delete dec_[i]; - } - if (rtp_fp_) - fclose(rtp_fp_); -} - -void NetEqDecodingTest::SelectDecoders(WebRtcNetEQDecoder* used_codec) { - *used_codec++ = kDecoderPCMu; - dec_.push_back(new decoder_PCMU(0)); - *used_codec++ = kDecoderPCMa; - dec_.push_back(new decoder_PCMA(8)); - *used_codec++ = kDecoderILBC; - dec_.push_back(new decoder_ILBC(102)); - *used_codec++ = kDecoderISAC; - dec_.push_back(new decoder_iSAC(103)); - *used_codec++ = kDecoderISACswb; - dec_.push_back(new decoder_iSACSWB(104)); - *used_codec++ = kDecoderISACfb; - dec_.push_back(new decoder_iSACFB(105)); - *used_codec++ = kDecoderPCM16B; - dec_.push_back(new decoder_PCM16B_NB(93)); - *used_codec++ = kDecoderPCM16Bwb; - dec_.push_back(new decoder_PCM16B_WB(94)); - *used_codec++ = kDecoderPCM16Bswb32kHz; - dec_.push_back(new decoder_PCM16B_SWB32(95)); - *used_codec++ = kDecoderCNG; - dec_.push_back(new decoder_CNG(13, 8000)); - *used_codec++ = kDecoderCNG; - dec_.push_back(new decoder_CNG(98, 16000)); -} - -void NetEqDecodingTest::LoadDecoders() { - for (size_t i = 0; i < dec_.size(); ++i) { - ASSERT_EQ(0, dec_[i]->loadToNetEQ(*neteq_inst_)); - } -} - -void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) { - rtp_fp_ = fopen(rtp_file.c_str(), "rb"); - ASSERT_TRUE(rtp_fp_ != NULL); - ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp_)); -} - -void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int16_t* out_len) { - // Check if time to receive. - while ((sim_clock_ >= rtp->time()) && - (rtp->dataLen() >= 0)) { - if (rtp->dataLen() > 0) { - ASSERT_EQ(0, neteq_inst_->recIn(*rtp)); - } - // Get next packet. - ASSERT_NE(-1, rtp->readFromFile(rtp_fp_)); - } - - // RecOut - *out_len = neteq_inst_->recOut(out_data_); - ASSERT_TRUE((*out_len == kBlockSize8kHz) || - (*out_len == kBlockSize16kHz) || - (*out_len == kBlockSize32kHz)); - - // Increase time. - sim_clock_ += kTimeStepMs; -} - -void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file, - const std::string &ref_file) { - OpenInputFile(rtp_file); - - std::string ref_out_file = ""; - if (ref_file.empty()) { - ref_out_file = webrtc::test::OutputPath() + "neteq_out.pcm"; - } - RefFiles ref_files(ref_file, ref_out_file); - - NETEQTEST_RTPpacket rtp; - ASSERT_GT(rtp.readFromFile(rtp_fp_), 0); - int i = 0; - while (rtp.dataLen() >= 0) { - std::ostringstream ss; - ss << "Lap number " << i++ << " in DecodeAndCompare while loop"; - SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. - int16_t out_len; - ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len)); - ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len)); - } -} - -void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file, - const std::string &stat_ref_file, - const std::string &rtcp_ref_file) { - OpenInputFile(rtp_file); - std::string stat_out_file = ""; - if (stat_ref_file.empty()) { - stat_out_file = webrtc::test::OutputPath() + - "neteq_network_stats.dat"; - } - RefFiles network_stat_files(stat_ref_file, stat_out_file); - - std::string rtcp_out_file = ""; - if (rtcp_ref_file.empty()) { - rtcp_out_file = webrtc::test::OutputPath() + - "neteq_rtcp_stats.dat"; - } - RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file); - - NETEQTEST_RTPpacket rtp; - ASSERT_GT(rtp.readFromFile(rtp_fp_), 0); - while (rtp.dataLen() >= 0) { - int16_t out_len; - Process(&rtp, &out_len); - - // Query the network statistics API once per second - if (sim_clock_ % 1000 == 0) { - // Process NetworkStatistics. - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - network_stat_files.ProcessReference(network_stats); - - // Process RTCPstat. - WebRtcNetEQ_RTCPStat rtcp_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetRTCPStats(neteq_inst_->instance(), - &rtcp_stats)); - rtcp_stat_files.ProcessReference(rtcp_stats); - } - } -} - -void NetEqDecodingTest::PopulateRtpInfo(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info) { - rtp_info->sequenceNumber = frame_index; - rtp_info->timeStamp = timestamp; - rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC. - rtp_info->payloadType = 94; // PCM16b WB codec. - rtp_info->markerBit = 0; -} - -void NetEqDecodingTest::PopulateCng(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info, - uint8_t* payload, - int* payload_len) { - rtp_info->sequenceNumber = frame_index; - rtp_info->timeStamp = timestamp; - rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC. - rtp_info->payloadType = 98; // WB CNG. - rtp_info->markerBit = 0; - payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen. - *payload_len = 1; // Only noise level, no spectral parameters. -} - -#if (defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)) || defined(WEBRTC_ANDROID) -// Disabled for Windows 64-bit until webrtc:1460 is fixed. -#define MAYBE_TestBitExactness DISABLED_TestBitExactness -#else -#define MAYBE_TestBitExactness TestBitExactness -#endif - -TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { - const std::string kInputRtpFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_universal.rtp"; -#if defined(_MSC_VER) && (_MSC_VER >= 1700) - // For Visual Studio 2012 and later, we will have to use the generic reference - // file, rather than the windows-specific one. - const std::string kInputRefFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_universal_ref.pcm"; -#else - const std::string kInputRefFile = - webrtc::test::ResourcePath("audio_coding/neteq_universal_ref", "pcm"); -#endif - DecodeAndCompare(kInputRtpFile, kInputRefFile); -} - -TEST_F(NetEqDecodingTest, TestNetworkStatistics) { - const std::string kInputRtpFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_universal.rtp"; -#if defined(_MSC_VER) && (_MSC_VER >= 1700) - // For Visual Studio 2012 and later, we will have to use the generic reference - // file, rather than the windows-specific one. - const std::string kNetworkStatRefFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_network_stats.dat"; -#else - const std::string kNetworkStatRefFile = - webrtc::test::ResourcePath("audio_coding/neteq_network_stats", "dat"); -#endif - const std::string kRtcpStatRefFile = - webrtc::test::ResourcePath("audio_coding/neteq_rtcp_stats", "dat"); - DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile); -} - -TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) { - // Use fax mode to avoid time-scaling. This is to simplify the testing of - // packet waiting times in the packet buffer. - ASSERT_EQ(0, - WebRtcNetEQ_SetPlayoutMode(neteq_inst_->instance(), kPlayoutFax)); - // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio. - int num_frames = 30; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - for (int i = 0; i < num_frames; ++i) { - uint16_t payload[kSamples] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - rtp_info.sequenceNumber = i; - rtp_info.timeStamp = i * kSamples; - rtp_info.SSRC = 0x1234; // Just an arbitrary SSRC. - rtp_info.payloadType = 94; // PCM16b WB codec. - rtp_info.markerBit = 0; - ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info, - reinterpret_cast<uint8_t*>(payload), - kPayloadBytes, 0)); - } - // Pull out all data. - for (int i = 0; i < num_frames; ++i) { - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - const int kVecLen = 110; // More than kLenWaitingTimes in mcu.h. - int waiting_times[kVecLen]; - int len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(), - kVecLen, waiting_times); - EXPECT_EQ(num_frames, len); - // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms - // spacing (per definition), we expect the delay to increase with 10 ms for - // each packet. - for (int i = 0; i < len; ++i) { - EXPECT_EQ((i + 1) * 10, waiting_times[i]); - } - - // Check statistics again and make sure it's been reset. - EXPECT_EQ(0, WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(), - kVecLen, waiting_times)); - - // Process > 100 frames, and make sure that that we get statistics - // only for 100 frames. Note the new SSRC, causing NetEQ to reset. - num_frames = 110; - for (int i = 0; i < num_frames; ++i) { - uint16_t payload[kSamples] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - rtp_info.sequenceNumber = i; - rtp_info.timeStamp = i * kSamples; - rtp_info.SSRC = 0x1235; // Just an arbitrary SSRC. - rtp_info.payloadType = 94; // PCM16b WB codec. - rtp_info.markerBit = 0; - ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info, - reinterpret_cast<uint8_t*>(payload), - kPayloadBytes, 0)); - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(), - kVecLen, waiting_times); - EXPECT_EQ(100, len); -} - -TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) { - const int kNumFrames = 3000; // Needed for convergence. - int frame_index = 0; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - while (frame_index < kNumFrames) { - // Insert one packet each time, except every 10th time where we insert two - // packets at once. This will create a negative clock-drift of approx. 10%. - int num_packets = (frame_index % 10 == 0 ? 2 : 1); - for (int n = 0; n < num_packets; ++n) { - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++frame_index; - } - - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - EXPECT_EQ(-103196, network_stats.clockDriftPPM); -} - -TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) { - const int kNumFrames = 5000; // Needed for convergence. - int frame_index = 0; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - for (int i = 0; i < kNumFrames; ++i) { - // Insert one packet each time, except every 10th time where we don't insert - // any packet. This will create a positive clock-drift of approx. 11%. - int num_packets = (i % 10 == 9 ? 0 : 1); - for (int n = 0; n < num_packets; ++n) { - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++frame_index; - } - - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - EXPECT_EQ(110946, network_stats.clockDriftPPM); -} - -TEST_F(NetEqDecodingTest, LongCngWithClockDrift) { - uint16_t seq_no = 0; - uint32_t timestamp = 0; - const int kFrameSizeMs = 30; - const int kSamples = kFrameSizeMs * 16; - const int kPayloadBytes = kSamples * 2; - // Apply a clock drift of -25 ms / s (sender faster than receiver). - const double kDriftFactor = 1000.0 / (1000.0 + 25.0); - double next_input_time_ms = 0.0; - double t_ms; - - // Insert speech for 5 seconds. - const int kSpeechDurationMs = 5000; - for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one 30 ms speech frame. - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(seq_no, timestamp, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++seq_no; - timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor; - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - EXPECT_EQ(kOutputNormal, neteq_inst_->getOutputType()); - int32_t delay_before = timestamp - neteq_inst_->getSpeechTimeStamp(); - - // Insert CNG for 1 minute (= 60000 ms). - const int kCngPeriodMs = 100; - const int kCngPeriodSamples = kCngPeriodMs * 16; // Period in 16 kHz samples. - const int kCngDurationMs = 60000; - for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one CNG frame each 100 ms. - uint8_t payload[kPayloadBytes]; - int payload_len; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - payload_len, 0)); - ++seq_no; - timestamp += kCngPeriodSamples; - next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor; - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - EXPECT_EQ(kOutputCNG, neteq_inst_->getOutputType()); - - // Insert speech again until output type is speech. - while (neteq_inst_->getOutputType() != kOutputNormal) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one 30 ms speech frame. - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(seq_no, timestamp, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++seq_no; - timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor; - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - // Increase clock. - t_ms += 10; - } - - int32_t delay_after = timestamp - neteq_inst_->getSpeechTimeStamp(); - // Compare delay before and after, and make sure it differs less than 20 ms. - EXPECT_LE(delay_after, delay_before + 20 * 16); - EXPECT_GE(delay_after, delay_before - 20 * 16); -} - -TEST_F(NetEqDecodingTest, NoInputDataStereo) { - void *ms_info; - ms_info = malloc(WebRtcNetEQ_GetMasterSlaveInfoSize()); - neteq_inst_->setMaster(); - - // Slave instance without decoders (because it is easier). - WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1]; - usedCodec[0] = kDecoderPCMu; - NETEQTEST_NetEQClass* slave_inst = - new NETEQTEST_NetEQClass(usedCodec, 1, 8000, kTCPLargeJitter); - ASSERT_TRUE(slave_inst); - NETEQTEST_Decoder* dec = new decoder_PCMU(0); - ASSERT_TRUE(dec != NULL); - dec->loadToNetEQ(*slave_inst); - slave_inst->setSlave(); - - // Pull out data. - const int kNumFrames = 100; - for (int i = 0; i < kNumFrames; ++i) { - ASSERT_TRUE(kBlockSize8kHz == neteq_inst_->recOut(out_data_, ms_info)); - ASSERT_TRUE(kBlockSize8kHz == slave_inst->recOut(out_data_, ms_info)); - } - - delete dec; - delete slave_inst; - free(ms_info); -} - -TEST_F(NetEqDecodingTest, TestExtraDelay) { - static const int kNumFrames = 120000; // Needed for convergence. - int frame_index = 0; - static const int kFrameSizeSamples = 30 * 16; - static const int kPayloadBytes = kFrameSizeSamples * 2; - test::InputAudioFile input_file( - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm")); - int16_t input[kFrameSizeSamples]; - // Buffers of NetEq cannot accommodate larger delays for PCM16. - static const int kExtraDelayMs = 3200; - ASSERT_EQ(0, WebRtcNetEQ_SetExtraDelay(neteq_inst_->instance(), - kExtraDelayMs)); - for (int i = 0; i < kNumFrames; ++i) { - ASSERT_TRUE(input_file.Read(kFrameSizeSamples, input)); - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(frame_index, frame_index * kFrameSizeSamples, &rtp_info); - uint8_t* payload = reinterpret_cast<uint8_t*>(input); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++frame_index; - // Pull out data. - for (int j = 0; j < 3; ++j) { - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - if (i % 100 == 0) { - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - const int expected_lower_limit = - std::min(i * 0.083 - 210, 0.9 * network_stats.preferredBufferSize); - EXPECT_GE(network_stats.currentBufferSize, expected_lower_limit); - const int expected_upper_limit = - std::min(i * 0.083 + 255, 1.2 * network_stats.preferredBufferSize); - EXPECT_LE(network_stats.currentBufferSize, expected_upper_limit); - } - } -} - -void NetEqDecodingTest::WrapTest(uint16_t start_seq_no, - uint32_t start_timestamp, - const std::set<uint16_t>& drop_seq_numbers) { - uint16_t seq_no = start_seq_no; - uint32_t timestamp = start_timestamp; - const int kFrameSizeMs = 30; - const int kSamples = kFrameSizeMs * 16; - const int kPayloadBytes = kSamples * 2; - double next_input_time_ms = 0.0; - - // Insert speech for 1 second. - const int kSpeechDurationMs = 1000; - for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one 30 ms speech frame. - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(seq_no, timestamp, &rtp_info); - if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) { - // This sequence number was not in the set to drop. Insert it. - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - } - ++seq_no; - timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs); - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - // Expect preferred and actual buffer size to be no more than 2 frames. - EXPECT_LE(network_stats.preferredBufferSize, kFrameSizeMs * 2); - EXPECT_LE(network_stats.currentBufferSize, kFrameSizeMs * 2); - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - // Expect delay (in samples) to be less than 2 packets. - EXPECT_LE(timestamp - neteq_inst_->getSpeechTimeStamp(), - static_cast<uint32_t>(kSamples * 2)); - } -} - -TEST_F(NetEqDecodingTest, SequenceNumberWrap) { - // Start with a sequence number that will soon wrap. - std::set<uint16_t> drop_seq_numbers; // Don't drop any packets. - WrapTest(0xFFFF - 5, 0, drop_seq_numbers); -} - -TEST_F(NetEqDecodingTest, SequenceNumberWrapAndDrop) { - // Start with a sequence number that will soon wrap. - std::set<uint16_t> drop_seq_numbers; - drop_seq_numbers.insert(0xFFFF); - drop_seq_numbers.insert(0x0); - WrapTest(0xFFFF - 5, 0, drop_seq_numbers); -} - -TEST_F(NetEqDecodingTest, TimestampWrap) { - // Start with a timestamp that will soon wrap. - std::set<uint16_t> drop_seq_numbers; - WrapTest(0, 0xFFFFFFFF - 1000, drop_seq_numbers); -} - -TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) { - // Start with a timestamp and a sequence number that will wrap at the same - // time. - std::set<uint16_t> drop_seq_numbers; - WrapTest(0xFFFF - 2, 0xFFFFFFFF - 1000, drop_seq_numbers); -} - -} // namespace diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/OWNERS b/chromium/third_party/webrtc/modules/audio_coding/neteq4/OWNERS deleted file mode 100644 index d54559c4ad8..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/OWNERS +++ /dev/null @@ -1,4 +0,0 @@ -henrik.lundin@webrtc.org -tina.legrand@webrtc.org -turaj@webrtc.org -minyue@webrtc.org diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.h deleted file mode 100644 index d08b64f4921..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.h +++ /dev/null @@ -1,116 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_BUFFER_H_ - -#include <list> -#include <string> // size_t - -#include "webrtc/system_wrappers/interface/constructor_magic.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -struct DtmfEvent { - uint32_t timestamp; - int event_no; - int volume; - int duration; - bool end_bit; - - // Constructors - DtmfEvent() - : timestamp(0), - event_no(0), - volume(0), - duration(0), - end_bit(false) { - } - DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end) - : timestamp(ts), - event_no(ev), - volume(vol), - duration(dur), - end_bit(end) { - } -}; - -// This is the buffer holding DTMF events while waiting for them to be played. -class DtmfBuffer { - public: - enum BufferReturnCodes { - kOK = 0, - kInvalidPointer, - kPayloadTooShort, - kInvalidEventParameters, - kInvalidSampleRate - }; - - // Set up the buffer for use at sample rate |fs_hz|. - explicit DtmfBuffer(int fs_hz) { - SetSampleRate(fs_hz); - } - - virtual ~DtmfBuffer() {} - - // Flushes the buffer. - virtual void Flush() { buffer_.clear(); } - - // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733) - // and write the parsed information into the struct |event|. Input variable - // |rtp_timestamp| is simply copied into the struct. - static int ParseEvent(uint32_t rtp_timestamp, - const uint8_t* payload, - int payload_length_bytes, - DtmfEvent* event); - - // Inserts |event| into the buffer. The method looks for a matching event and - // merges the two if a match is found. - virtual int InsertEvent(const DtmfEvent& event); - - // Checks if a DTMF event should be played at time |current_timestamp|. If so, - // the method returns true; otherwise false. The parameters of the event to - // play will be written to |event|. - virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event); - - // Number of events in the buffer. - virtual size_t Length() const { return buffer_.size(); } - - virtual bool Empty() const { return buffer_.empty(); } - - // Set a new sample rate. - virtual int SetSampleRate(int fs_hz); - - private: - typedef std::list<DtmfEvent> DtmfList; - - int max_extrapolation_samples_; - int frame_len_samples_; // TODO(hlundin): Remove this later. - - // Compares two events and returns true if they are the same. - static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b); - - // Merges |event| to the event pointed out by |it|. The method checks that - // the two events are the same (using the SameEvent method), and merges them - // if that was the case, returning true. If the events are not the same, false - // is returned. - bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event); - - // Method used by the sort algorithm to rank events in the buffer. - static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b); - - DtmfList buffer_; - - DISALLOW_COPY_AND_ASSIGN(DtmfBuffer); -}; - -} // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq.cc deleted file mode 100644 index 1ec71a2a6fb..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq.cc +++ /dev/null @@ -1,52 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" - -#include "webrtc/modules/audio_coding/neteq4/buffer_level_filter.h" -#include "webrtc/modules/audio_coding/neteq4/decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/delay_peak_detector.h" -#include "webrtc/modules/audio_coding/neteq4/dtmf_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h" -#include "webrtc/modules/audio_coding/neteq4/neteq_impl.h" -#include "webrtc/modules/audio_coding/neteq4/packet_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/payload_splitter.h" -#include "webrtc/modules/audio_coding/neteq4/timestamp_scaler.h" - -namespace webrtc { - -// Creates all classes needed and inject them into a new NetEqImpl object. -// Return the new object. -NetEq* NetEq::Create(int sample_rate_hz) { - BufferLevelFilter* buffer_level_filter = new BufferLevelFilter; - DecoderDatabase* decoder_database = new DecoderDatabase; - DelayPeakDetector* delay_peak_detector = new DelayPeakDetector; - DelayManager* delay_manager = new DelayManager(kMaxNumPacketsInBuffer, - delay_peak_detector); - DtmfBuffer* dtmf_buffer = new DtmfBuffer(sample_rate_hz); - DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator; - PacketBuffer* packet_buffer = new PacketBuffer(kMaxNumPacketsInBuffer, - kMaxBytesInBuffer); - PayloadSplitter* payload_splitter = new PayloadSplitter; - TimestampScaler* timestamp_scaler = new TimestampScaler(*decoder_database); - return new NetEqImpl(sample_rate_hz, - buffer_level_filter, - decoder_database, - delay_manager, - delay_peak_detector, - dtmf_buffer, - dtmf_tone_generator, - packet_buffer, - payload_splitter, - timestamp_scaler); -} - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq.gypi b/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq.gypi deleted file mode 100644 index 41fdb31ea6b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq.gypi +++ /dev/null @@ -1,220 +0,0 @@ -# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -{ - 'variables': { - 'neteq_dependencies': [ - 'G711', - 'G722', - 'PCM16B', - 'iLBC', - 'iSAC', - 'iSACFix', - 'CNG', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', - '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', - ], - 'neteq_defines': [], - 'conditions': [ - ['include_opus==1', { - 'neteq_dependencies': ['webrtc_opus',], - 'neteq_defines': ['WEBRTC_CODEC_OPUS',], - }], - ], - }, - 'targets': [ - { - 'target_name': 'NetEq4', - 'type': 'static_library', - 'dependencies': [ - '<@(neteq_dependencies)', - ], - 'defines': [ - '<@(neteq_defines)', - ], - 'include_dirs': [ - 'interface', - '<(webrtc_root)', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - 'interface', - '<(webrtc_root)', - ], - }, - 'sources': [ - 'interface/audio_decoder.h', - 'interface/neteq.h', - 'accelerate.cc', - 'accelerate.h', - 'audio_decoder_impl.cc', - 'audio_decoder_impl.h', - 'audio_decoder.cc', - 'audio_multi_vector.cc', - 'audio_multi_vector.h', - 'audio_vector.cc', - 'audio_vector.h', - 'background_noise.cc', - 'background_noise.h', - 'buffer_level_filter.cc', - 'buffer_level_filter.h', - 'comfort_noise.cc', - 'comfort_noise.h', - 'decision_logic.cc', - 'decision_logic.h', - 'decision_logic_fax.cc', - 'decision_logic_fax.h', - 'decision_logic_normal.cc', - 'decision_logic_normal.h', - 'decoder_database.cc', - 'decoder_database.h', - 'defines.h', - 'delay_manager.cc', - 'delay_manager.h', - 'delay_peak_detector.cc', - 'delay_peak_detector.h', - 'dsp_helper.cc', - 'dsp_helper.h', - 'dtmf_buffer.cc', - 'dtmf_buffer.h', - 'dtmf_tone_generator.cc', - 'dtmf_tone_generator.h', - 'expand.cc', - 'expand.h', - 'merge.cc', - 'merge.h', - 'neteq_impl.cc', - 'neteq_impl.h', - 'neteq.cc', - 'statistics_calculator.cc', - 'statistics_calculator.h', - 'normal.cc', - 'normal.h', - 'packet_buffer.cc', - 'packet_buffer.h', - 'payload_splitter.cc', - 'payload_splitter.h', - 'post_decode_vad.cc', - 'post_decode_vad.h', - 'preemptive_expand.cc', - 'preemptive_expand.h', - 'random_vector.cc', - 'random_vector.h', - 'rtcp.cc', - 'rtcp.h', - 'sync_buffer.cc', - 'sync_buffer.h', - 'timestamp_scaler.cc', - 'timestamp_scaler.h', - 'time_stretch.cc', - 'time_stretch.h', - ], - }, - ], # targets - 'conditions': [ - ['include_tests==1', { - 'includes': ['neteq_tests.gypi',], - 'targets': [ - { - 'target_name': 'audio_decoder_unittests', - 'type': '<(gtest_target_type)', - 'dependencies': [ - '<@(neteq_dependencies)', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', - '<(webrtc_root)/test/test.gyp:test_support_main', - ], - 'defines': [ - 'AUDIO_DECODER_UNITTEST', - 'WEBRTC_CODEC_G722', - 'WEBRTC_CODEC_ILBC', - 'WEBRTC_CODEC_ISACFX', - 'WEBRTC_CODEC_ISAC', - 'WEBRTC_CODEC_PCM16', - '<@(neteq_defines)', - ], - 'sources': [ - 'audio_decoder_impl.cc', - 'audio_decoder_impl.h', - 'audio_decoder_unittest.cc', - 'audio_decoder.cc', - 'interface/audio_decoder.h', - ], - 'conditions': [ - # TODO(henrike): remove build_with_chromium==1 when the bots are - # using Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { - 'dependencies': [ - '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', - ], - }], - ], - }, # audio_decoder_unittests - - { - 'target_name': 'neteq_unittest_tools', - 'type': 'static_library', - 'dependencies': [ - '<(DEPTH)/testing/gmock.gyp:gmock', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/test/test.gyp:test_support_main', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - 'tools', - ], - }, - 'include_dirs': [ - 'tools', - ], - 'sources': [ - 'tools/audio_loop.cc', - 'tools/audio_loop.h', - 'tools/input_audio_file.cc', - 'tools/input_audio_file.h', - 'tools/rtp_generator.cc', - 'tools/rtp_generator.h', - ], - }, # neteq_unittest_tools - ], # targets - 'conditions': [ - # TODO(henrike): remove build_with_chromium==1 when the bots are using - # Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { - 'targets': [ - { - 'target_name': 'audio_decoder_unittests_apk_target', - 'type': 'none', - 'dependencies': [ - '<(apk_tests_path):audio_decoder_unittests_apk', - ], - }, - ], - }], - ['test_isolation_mode != "noop"', { - 'targets': [ - { - 'target_name': 'audio_decoder_unittests_run', - 'type': 'none', - 'dependencies': [ - 'audio_decoder_unittests', - ], - 'includes': [ - '../../../build/isolate.gypi', - 'audio_decoder_unittests.isolate', - ], - 'sources': [ - 'audio_decoder_unittests.isolate', - ], - }, - ], - }], - ], - }], # include_tests - ], # conditions -} diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc deleted file mode 100644 index 7a82053918c..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc +++ /dev/null @@ -1,229 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/neteq_impl.h" - -#include "gmock/gmock.h" -#include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_buffer_level_filter.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_delay_manager.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_delay_peak_detector.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_packet_buffer.h" -#include "webrtc/modules/audio_coding/neteq4/mock/mock_payload_splitter.h" -#include "webrtc/modules/audio_coding/neteq4/timestamp_scaler.h" - -using ::testing::Return; -using ::testing::ReturnNull; -using ::testing::_; -using ::testing::SetArgPointee; -using ::testing::InSequence; -using ::testing::Invoke; -using ::testing::WithArg; - -namespace webrtc { - -// This function is called when inserting a packet list into the mock packet -// buffer. The purpose is to delete all inserted packets properly, to avoid -// memory leaks in the test. -int DeletePacketsAndReturnOk(PacketList* packet_list) { - PacketBuffer::DeleteAllPackets(packet_list); - return PacketBuffer::kOK; -} - -class NetEqImplTest : public ::testing::Test { - protected: - static const int kInitSampleRateHz = 8000; - NetEqImplTest() { - buffer_level_filter_ = new MockBufferLevelFilter; - decoder_database_ = new MockDecoderDatabase; - delay_peak_detector_ = new MockDelayPeakDetector; - EXPECT_CALL(*delay_peak_detector_, Reset()).Times(1); - delay_manager_ = new MockDelayManager(NetEq::kMaxNumPacketsInBuffer, - delay_peak_detector_); - dtmf_buffer_ = new MockDtmfBuffer(kInitSampleRateHz); - dtmf_tone_generator_ = new MockDtmfToneGenerator; - packet_buffer_ = new MockPacketBuffer(NetEq::kMaxNumPacketsInBuffer, - NetEq::kMaxBytesInBuffer); - payload_splitter_ = new MockPayloadSplitter; - timestamp_scaler_ = new TimestampScaler(*decoder_database_); - EXPECT_CALL(*decoder_database_, GetActiveCngDecoder()) - .WillOnce(ReturnNull()); - neteq_ = new NetEqImpl(kInitSampleRateHz, - buffer_level_filter_, - decoder_database_, - delay_manager_, - delay_peak_detector_, - dtmf_buffer_, - dtmf_tone_generator_, - packet_buffer_, - payload_splitter_, - timestamp_scaler_); - } - - virtual ~NetEqImplTest() { - EXPECT_CALL(*buffer_level_filter_, Die()).Times(1); - EXPECT_CALL(*decoder_database_, Die()).Times(1); - EXPECT_CALL(*delay_manager_, Die()).Times(1); - EXPECT_CALL(*delay_peak_detector_, Die()).Times(1); - EXPECT_CALL(*dtmf_buffer_, Die()).Times(1); - EXPECT_CALL(*dtmf_tone_generator_, Die()).Times(1); - EXPECT_CALL(*packet_buffer_, Die()).Times(1); - delete neteq_; - } - - NetEqImpl* neteq_; - MockBufferLevelFilter* buffer_level_filter_; - MockDecoderDatabase* decoder_database_; - MockDelayPeakDetector* delay_peak_detector_; - MockDelayManager* delay_manager_; - MockDtmfBuffer* dtmf_buffer_; - MockDtmfToneGenerator* dtmf_tone_generator_; - MockPacketBuffer* packet_buffer_; - MockPayloadSplitter* payload_splitter_; - TimestampScaler* timestamp_scaler_; -}; - - -// This tests the interface class NetEq. -// TODO(hlundin): Move to separate file? -TEST(NetEq, CreateAndDestroy) { - NetEq* neteq = NetEq::Create(8000); - delete neteq; -} - -TEST_F(NetEqImplTest, RegisterPayloadType) { - uint8_t rtp_payload_type = 0; - NetEqDecoder codec_type = kDecoderPCMu; - EXPECT_CALL(*decoder_database_, - RegisterPayload(rtp_payload_type, codec_type)); - neteq_->RegisterPayloadType(codec_type, rtp_payload_type); -} - -TEST_F(NetEqImplTest, RemovePayloadType) { - uint8_t rtp_payload_type = 0; - EXPECT_CALL(*decoder_database_, - Remove(rtp_payload_type)) - .WillOnce(Return(DecoderDatabase::kDecoderNotFound)); - // Check that kFail is returned when database returns kDecoderNotFound. - EXPECT_EQ(NetEq::kFail, neteq_->RemovePayloadType(rtp_payload_type)); -} - -TEST_F(NetEqImplTest, InsertPacket) { - const int kPayloadLength = 100; - const uint8_t kPayloadType = 0; - const uint16_t kFirstSequenceNumber = 0x1234; - const uint32_t kFirstTimestamp = 0x12345678; - const uint32_t kSsrc = 0x87654321; - const uint32_t kFirstReceiveTime = 17; - uint8_t payload[kPayloadLength] = {0}; - WebRtcRTPHeader rtp_header; - rtp_header.header.payloadType = kPayloadType; - rtp_header.header.sequenceNumber = kFirstSequenceNumber; - rtp_header.header.timestamp = kFirstTimestamp; - rtp_header.header.ssrc = kSsrc; - - // Create a mock decoder object. - MockAudioDecoder mock_decoder; - // BWE update function called with first packet. - EXPECT_CALL(mock_decoder, IncomingPacket(_, - kPayloadLength, - kFirstSequenceNumber, - kFirstTimestamp, - kFirstReceiveTime)); - // BWE update function called with second packet. - EXPECT_CALL(mock_decoder, IncomingPacket(_, - kPayloadLength, - kFirstSequenceNumber + 1, - kFirstTimestamp + 160, - kFirstReceiveTime + 155)); - EXPECT_CALL(mock_decoder, Die()).Times(1); // Called when deleted. - - // Expectations for decoder database. - EXPECT_CALL(*decoder_database_, IsRed(kPayloadType)) - .WillRepeatedly(Return(false)); // This is not RED. - EXPECT_CALL(*decoder_database_, CheckPayloadTypes(_)) - .Times(2) - .WillRepeatedly(Return(DecoderDatabase::kOK)); // Payload type is valid. - EXPECT_CALL(*decoder_database_, IsDtmf(kPayloadType)) - .WillRepeatedly(Return(false)); // This is not DTMF. - EXPECT_CALL(*decoder_database_, GetDecoder(kPayloadType)) - .Times(3) - .WillRepeatedly(Return(&mock_decoder)); - EXPECT_CALL(*decoder_database_, IsComfortNoise(kPayloadType)) - .WillRepeatedly(Return(false)); // This is not CNG. - DecoderDatabase::DecoderInfo info; - info.codec_type = kDecoderPCMu; - EXPECT_CALL(*decoder_database_, GetDecoderInfo(kPayloadType)) - .WillRepeatedly(Return(&info)); - - // Expectations for packet buffer. - EXPECT_CALL(*packet_buffer_, NumPacketsInBuffer()) - .WillOnce(Return(0)) // First packet. - .WillOnce(Return(1)) // Second packet. - .WillOnce(Return(2)); // Second packet, checking after it was inserted. - EXPECT_CALL(*packet_buffer_, Flush()) - .Times(1); - EXPECT_CALL(*packet_buffer_, InsertPacketList(_, _, _, _)) - .Times(2) - .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType), - WithArg<0>(Invoke(DeletePacketsAndReturnOk)))); - // SetArgPointee<2>(kPayloadType) means that the third argument (zero-based - // index) is a pointer, and the variable pointed to is set to kPayloadType. - // Also invoke the function DeletePacketsAndReturnOk to properly delete all - // packets in the list (to avoid memory leaks in the test). - EXPECT_CALL(*packet_buffer_, NextRtpHeader()) - .Times(1) - .WillOnce(Return(&rtp_header.header)); - - // Expectations for DTMF buffer. - EXPECT_CALL(*dtmf_buffer_, Flush()) - .Times(1); - - // Expectations for delay manager. - { - // All expectations within this block must be called in this specific order. - InSequence sequence; // Dummy variable. - // Expectations when the first packet is inserted. - EXPECT_CALL(*delay_manager_, LastDecoderType(kDecoderPCMu)) - .Times(1); - EXPECT_CALL(*delay_manager_, last_pack_cng_or_dtmf()) - .Times(2) - .WillRepeatedly(Return(-1)); - EXPECT_CALL(*delay_manager_, set_last_pack_cng_or_dtmf(0)) - .Times(1); - EXPECT_CALL(*delay_manager_, ResetPacketIatCount()).Times(1); - // Expectations when the second packet is inserted. Slightly different. - EXPECT_CALL(*delay_manager_, LastDecoderType(kDecoderPCMu)) - .Times(1); - EXPECT_CALL(*delay_manager_, last_pack_cng_or_dtmf()) - .WillOnce(Return(0)); - } - - // Expectations for payload splitter. - EXPECT_CALL(*payload_splitter_, SplitAudio(_, _)) - .Times(2) - .WillRepeatedly(Return(PayloadSplitter::kOK)); - - // Insert first packet. - neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime); - - // Insert second packet. - rtp_header.header.timestamp += 160; - rtp_header.header.sequenceNumber += 1; - neteq_->InsertPacket(rtp_header, payload, kPayloadLength, - kFirstReceiveTime + 155); -} - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.h deleted file mode 100644 index e964c28f207..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.h +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PACKET_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PACKET_BUFFER_H_ - -#include "webrtc/modules/audio_coding/neteq4/packet.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -// Forward declaration. -class DecoderDatabase; - -// This is the actual buffer holding the packets before decoding. -class PacketBuffer { - public: - enum BufferReturnCodes { - kOK = 0, - kFlushed, - kNotFound, - kBufferEmpty, - kInvalidPacket, - kInvalidPointer, - kOversizePacket - }; - - // Constructor creates a buffer which can hold a maximum of - // |max_number_of_packets| packets and |max_payload_memory| bytes of payload, - // excluding RTP headers. - PacketBuffer(size_t max_number_of_packets, size_t max_payload_memory); - - // Deletes all packets in the buffer before destroying the buffer. - virtual ~PacketBuffer(); - - // Flushes the buffer and deletes all packets in it. - virtual void Flush(); - - // Returns true for an empty buffer. - virtual bool Empty() const { return buffer_.empty(); } - - // Inserts |packet| into the buffer. The buffer will take over ownership of - // the packet object. - // Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer - // was flushed due to overfilling. - virtual int InsertPacket(Packet* packet); - - // Inserts a list of packets into the buffer. The buffer will take over - // ownership of the packet objects. - // Returns PacketBuffer::kOK if all packets were inserted successfully. - // If the buffer was flushed due to overfilling, only a subset of the list is - // inserted, and PacketBuffer::kFlushed is returned. - // The last three parameters are included for legacy compatibility. - // TODO(hlundin): Redesign to not use current_*_payload_type and - // decoder_database. - virtual int InsertPacketList(PacketList* packet_list, - const DecoderDatabase& decoder_database, - uint8_t* current_rtp_payload_type, - uint8_t* current_cng_rtp_payload_type); - - // Gets the timestamp for the first packet in the buffer and writes it to the - // output variable |next_timestamp|. - // Returns PacketBuffer::kBufferEmpty if the buffer is empty, - // PacketBuffer::kOK otherwise. - virtual int NextTimestamp(uint32_t* next_timestamp) const; - - // Gets the timestamp for the first packet in the buffer with a timestamp no - // lower than the input limit |timestamp|. The result is written to the output - // variable |next_timestamp|. - // Returns PacketBuffer::kBufferEmpty if the buffer is empty, - // PacketBuffer::kOK otherwise. - virtual int NextHigherTimestamp(uint32_t timestamp, - uint32_t* next_timestamp) const; - - // Returns a (constant) pointer the RTP header of the first packet in the - // buffer. Returns NULL if the buffer is empty. - virtual const RTPHeader* NextRtpHeader() const; - - // Extracts the first packet in the buffer and returns a pointer to it. - // Returns NULL if the buffer is empty. The caller is responsible for deleting - // the packet. - // Subsequent packets with the same timestamp as the one extracted will be - // discarded and properly deleted. The number of discarded packets will be - // written to the output variable |discard_count|. - virtual Packet* GetNextPacket(int* discard_count); - - // Discards the first packet in the buffer. The packet is deleted. - // Returns PacketBuffer::kBufferEmpty if the buffer is empty, - // PacketBuffer::kOK otherwise. - virtual int DiscardNextPacket(); - - // Discards all packets that are (strictly) older than |timestamp_limit|. - // Returns number of packets discarded. - virtual int DiscardOldPackets(uint32_t timestamp_limit); - - // Returns the number of packets in the buffer, including duplicates and - // redundant packets. - virtual int NumPacketsInBuffer() const { - return static_cast<int>(buffer_.size()); - } - - // Returns the number of samples in the buffer, including samples carried in - // duplicate and redundant packets. - virtual int NumSamplesInBuffer(DecoderDatabase* decoder_database, - int last_decoded_length) const; - - // Increase the waiting time counter for every packet in the buffer by |inc|. - // The default value for |inc| is 1. - virtual void IncrementWaitingTimes(int inc = 1); - - virtual void BufferStat(int* num_packets, - int* max_num_packets, - int* current_memory_bytes, - int* max_memory_bytes) const; - - virtual int current_memory_bytes() const { return current_memory_bytes_; } - - // Static method that properly deletes the first packet, and its payload - // array, in |packet_list|. Returns false if |packet_list| already was empty, - // otherwise true. - static bool DeleteFirstPacket(PacketList* packet_list); - - // Static method that properly deletes all packets, and their payload arrays, - // in |packet_list|. - static void DeleteAllPackets(PacketList* packet_list); - - private: - size_t max_number_of_packets_; - size_t max_memory_bytes_; - int current_memory_bytes_; - PacketList buffer_; - DISALLOW_COPY_AND_ASSIGN(PacketBuffer); -}; - -} // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_PACKET_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/rtcp.h b/chromium/third_party/webrtc/modules/audio_coding/neteq4/rtcp.h deleted file mode 100644 index 00cbbd15836..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/rtcp.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_RTCP_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_RTCP_H_ - -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -// Forward declaration. -struct RTPHeader; - -class Rtcp { - public: - Rtcp() { - Init(0); - } - - ~Rtcp() {} - - // Resets the RTCP statistics, and sets the first received sequence number. - void Init(uint16_t start_sequence_number); - - // Updates the RTCP statistics with a new received packet. - void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp); - - // Returns the current RTCP statistics. If |no_reset| is true, the statistics - // are not reset, otherwise they are. - void GetStatistics(bool no_reset, RtcpStatistics* stats); - - private: - uint16_t cycles_; // The number of wrap-arounds for the sequence number. - uint16_t max_seq_no_; // The maximum sequence number received. Starts over - // from 0 after wrap-around. - uint16_t base_seq_no_; // The sequence number of the first received packet. - uint32_t received_packets_; // The number of packets that have been received. - uint32_t received_packets_prior_; // Number of packets received when last - // report was generated. - uint32_t expected_prior_; // Expected number of packets, at the time of the - // last report. - uint32_t jitter_; // Current jitter value. - int32_t transit_; // Clock difference for previous packet. - - DISALLOW_COPY_AND_ASSIGN(Rtcp); -}; - -} // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_RTCP_H_ diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc deleted file mode 100644 index cf8131f3a06..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc +++ /dev/null @@ -1,31 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Unit tests for Accelerate and PreemptiveExpand classes. - -#include "webrtc/modules/audio_coding/neteq4/accelerate.h" -#include "webrtc/modules/audio_coding/neteq4/preemptive_expand.h" - -#include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq4/background_noise.h" - -namespace webrtc { - -TEST(TimeStretch, CreateAndDestroy) { - int sample_rate = 8000; - size_t num_channels = 1; - BackgroundNoise bgn(num_channels); - Accelerate accelerate(sample_rate, num_channels, bgn); - PreemptiveExpand preemptive_expand(sample_rate, num_channels, bgn); -} - -// TODO(hlundin): Write more tests. - -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/chromium/third_party/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h index 352537d6ef7..2969ecebe66 100644 --- a/chromium/third_party/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h +++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h @@ -57,7 +57,7 @@ public: // Add/remove participants as candidates for mixing. virtual int32_t SetMixabilityStatus(MixerParticipant& participant, - const bool mixable) = 0; + bool mixable) = 0; // mixable is set to true if a participant is a candidate for mixing. virtual int32_t MixabilityStatus(MixerParticipant& participant, bool& mixable) = 0; diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/OWNERS b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc index c5cf137f53c..26ef3e881f2 100644 --- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc @@ -19,6 +19,13 @@ namespace webrtc { namespace { +struct ParticipantFramePair { + MixerParticipant* participant; + AudioFrame* audioFrame; +}; + +typedef std::list<ParticipantFramePair*> ParticipantFramePairList; + // Mix |frame| into |mixed_frame|, with saturation protection and upmixing. // These effects are applied to |frame| itself prior to mixing. Assumes that // |mixed_frame| always has at least as many channels as |frame|. Supports @@ -40,20 +47,18 @@ void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame) { } // Return the max number of channels from a |list| composed of AudioFrames. -int MaxNumChannels(const ListWrapper& list) { - ListItem* item = list.First(); +int MaxNumChannels(const AudioFrameList* list) { int max_num_channels = 1; - while (item) { - AudioFrame* frame = static_cast<AudioFrame*>(item->GetItem()); - max_num_channels = std::max(max_num_channels, frame->num_channels_); - item = list.Next(item); + for (AudioFrameList::const_iterator iter = list->begin(); + iter != list->end(); + ++iter) { + max_num_channels = std::max(max_num_channels, (*iter)->num_channels_); } return max_num_channels; } void SetParticipantStatistics(ParticipantStatistics* stats, - const AudioFrame& frame) -{ + const AudioFrame& frame) { stats->participant = frame.id_; stats->level = 0; // TODO(andrew): to what should this be set? } @@ -61,58 +66,47 @@ void SetParticipantStatistics(ParticipantStatistics* stats, } // namespace MixerParticipant::MixerParticipant() - : _mixHistory(new MixHistory()) -{ + : _mixHistory(new MixHistory()) { } -MixerParticipant::~MixerParticipant() -{ +MixerParticipant::~MixerParticipant() { delete _mixHistory; } -int32_t MixerParticipant::IsMixed(bool& mixed) const -{ +int32_t MixerParticipant::IsMixed(bool& mixed) const { return _mixHistory->IsMixed(mixed); } MixHistory::MixHistory() - : _isMixed(0) -{ + : _isMixed(0) { } -MixHistory::~MixHistory() -{ +MixHistory::~MixHistory() { } -int32_t MixHistory::IsMixed(bool& mixed) const -{ +int32_t MixHistory::IsMixed(bool& mixed) const { mixed = _isMixed; return 0; } -int32_t MixHistory::WasMixed(bool& wasMixed) const -{ +int32_t MixHistory::WasMixed(bool& wasMixed) const { // Was mixed is the same as is mixed depending on perspective. This function // is for the perspective of AudioConferenceMixerImpl. return IsMixed(wasMixed); } -int32_t MixHistory::SetIsMixed(const bool mixed) -{ +int32_t MixHistory::SetIsMixed(const bool mixed) { _isMixed = mixed; return 0; } -void MixHistory::ResetMixedStatus() -{ +void MixHistory::ResetMixedStatus() { _isMixed = false; } -AudioConferenceMixer* AudioConferenceMixer::Create(int id) -{ +AudioConferenceMixer* AudioConferenceMixer::Create(int id) { AudioConferenceMixerImpl* mixer = new AudioConferenceMixerImpl(id); - if(!mixer->Init()) - { + if(!mixer->Init()) { delete mixer; return NULL; } @@ -140,11 +134,9 @@ AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id) _timeStamp(0), _timeScheduler(kProcessPeriodicityInMs), _mixedAudioLevel(), - _processCalls(0) -{} + _processCalls(0) {} -bool AudioConferenceMixerImpl::Init() -{ +bool AudioConferenceMixerImpl::Init() { _crit.reset(CriticalSectionWrapper::CreateCriticalSection()); if (_crit.get() == NULL) return false; @@ -153,8 +145,10 @@ bool AudioConferenceMixerImpl::Init() if(_cbCrit.get() == NULL) return false; - _limiter.reset(AudioProcessing::Create(_id)); - if(_limiter.get() == NULL) + Config config; + config.Set<ExperimentalAgc>(new ExperimentalAgc(false)); + _limiter.reset(AudioProcessing::Create(config)); + if(!_limiter.get()) return false; MemoryPool<AudioFrame>::CreateMemoryPool(_audioFramePool, @@ -165,10 +159,6 @@ bool AudioConferenceMixerImpl::Init() if(SetOutputFrequency(kDefaultFrequency) == -1) return false; - // Assume mono. - if (!SetNumLimiterChannels(1)) - return false; - if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) != _limiter->kNoError) return false; @@ -192,25 +182,21 @@ bool AudioConferenceMixerImpl::Init() return true; } -AudioConferenceMixerImpl::~AudioConferenceMixerImpl() -{ +AudioConferenceMixerImpl::~AudioConferenceMixerImpl() { MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool); assert(_audioFramePool == NULL); } -int32_t AudioConferenceMixerImpl::ChangeUniqueId(const int32_t id) -{ +int32_t AudioConferenceMixerImpl::ChangeUniqueId(const int32_t id) { _id = id; return 0; } // Process should be called every kProcessPeriodicityInMs ms -int32_t AudioConferenceMixerImpl::TimeUntilNextProcess() -{ +int32_t AudioConferenceMixerImpl::TimeUntilNextProcess() { int32_t timeUntilNextProcess = 0; CriticalSectionScoped cs(_crit.get()); - if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) - { + if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) { WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, "failed in TimeToNextUpdate() call"); // Sanity check @@ -220,9 +206,8 @@ int32_t AudioConferenceMixerImpl::TimeUntilNextProcess() return timeUntilNextProcess; } -int32_t AudioConferenceMixerImpl::Process() -{ - uint32_t remainingParticipantsAllowedToMix = +int32_t AudioConferenceMixerImpl::Process() { + size_t remainingParticipantsAllowedToMix = kMaximumAmountOfMixedParticipants; { CriticalSectionScoped cs(_crit.get()); @@ -233,9 +218,9 @@ int32_t AudioConferenceMixerImpl::Process() _timeScheduler.UpdateScheduler(); } - ListWrapper mixList; - ListWrapper rampOutList; - ListWrapper additionalFramesList; + AudioFrameList mixList; + AudioFrameList rampOutList; + AudioFrameList additionalFramesList; std::map<int, MixerParticipant*> mixedParticipantsMap; { CriticalSectionScoped cs(_cbCrit.get()); @@ -246,41 +231,34 @@ int32_t AudioConferenceMixerImpl::Process() // information. // TODO(henrike): this is probably more appropriate to do in // GetLowestMixingFrequency(). - if (lowFreq == 12000) - { + if (lowFreq == 12000) { lowFreq = 16000; } else if (lowFreq == 24000) { lowFreq = 32000; } - if(lowFreq <= 0) - { + if(lowFreq <= 0) { CriticalSectionScoped cs(_crit.get()); _processCalls--; return 0; - } else { - switch(lowFreq) - { + } else { + switch(lowFreq) { case 8000: - if(OutputFrequency() != kNbInHz) - { + if(OutputFrequency() != kNbInHz) { SetOutputFrequency(kNbInHz); } break; case 16000: - if(OutputFrequency() != kWbInHz) - { + if(OutputFrequency() != kWbInHz) { SetOutputFrequency(kWbInHz); } break; case 32000: - if(OutputFrequency() != kSwbInHz) - { + if(OutputFrequency() != kSwbInHz) { SetOutputFrequency(kSwbInHz); } break; case 48000: - if(OutputFrequency() != kFbInHz) - { + if(OutputFrequency() != kFbInHz) { SetOutputFrequency(kFbInHz); } break; @@ -293,19 +271,17 @@ int32_t AudioConferenceMixerImpl::Process() } } - UpdateToMix(mixList, rampOutList, &mixedParticipantsMap, + UpdateToMix(&mixList, &rampOutList, &mixedParticipantsMap, remainingParticipantsAllowedToMix); - GetAdditionalAudio(additionalFramesList); + GetAdditionalAudio(&additionalFramesList); UpdateMixedStatus(mixedParticipantsMap); - _scratchParticipantsToMixAmount = - static_cast<uint32_t>(mixedParticipantsMap.size()); + _scratchParticipantsToMixAmount = mixedParticipantsMap.size(); } // Get an AudioFrame for mixing from the memory pool. AudioFrame* mixedAudio = NULL; - if(_audioFramePool->PopMemory(mixedAudio) == -1) - { + if(_audioFramePool->PopMemory(mixedAudio) == -1) { WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id, "failed PopMemory() call"); assert(false); @@ -322,12 +298,9 @@ int32_t AudioConferenceMixerImpl::Process() // with an API instead of dynamically. // Find the max channels over all mixing lists. - const int num_mixed_channels = std::max(MaxNumChannels(mixList), - std::max(MaxNumChannels(additionalFramesList), - MaxNumChannels(rampOutList))); - - if (!SetNumLimiterChannels(num_mixed_channels)) - retval = -1; + const int num_mixed_channels = std::max(MaxNumChannels(&mixList), + std::max(MaxNumChannels(&additionalFramesList), + MaxNumChannels(&rampOutList))); mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency, AudioFrame::kNormalSpeech, @@ -335,18 +308,15 @@ int32_t AudioConferenceMixerImpl::Process() _timeStamp += _sampleSize; - MixFromList(*mixedAudio, mixList); - MixAnonomouslyFromList(*mixedAudio, additionalFramesList); - MixAnonomouslyFromList(*mixedAudio, rampOutList); + MixFromList(*mixedAudio, &mixList); + MixAnonomouslyFromList(*mixedAudio, &additionalFramesList); + MixAnonomouslyFromList(*mixedAudio, &rampOutList); - if(mixedAudio->samples_per_channel_ == 0) - { + if(mixedAudio->samples_per_channel_ == 0) { // Nothing was mixed, set the audio samples to silence. mixedAudio->samples_per_channel_ = _sampleSize; mixedAudio->Mute(); - } - else - { + } else { // Only call the limiter if we have something to mix. if(!LimitMixedAudio(*mixedAudio)) retval = -1; @@ -355,12 +325,10 @@ int32_t AudioConferenceMixerImpl::Process() _mixedAudioLevel.ComputeLevel(mixedAudio->data_,_sampleSize); audioLevel = _mixedAudioLevel.GetLevel(); - if(_mixerStatusCb) - { + if(_mixerStatusCb) { _scratchVadPositiveParticipantsAmount = 0; - UpdateVADPositiveParticipants(mixList); - if(_amountOf10MsUntilNextCallback-- == 0) - { + UpdateVADPositiveParticipants(&mixList); + if(_amountOf10MsUntilNextCallback-- == 0) { _amountOf10MsUntilNextCallback = _amountOf10MsBetweenCallbacks; timeForMixerCallback = true; } @@ -369,8 +337,7 @@ int32_t AudioConferenceMixerImpl::Process() { CriticalSectionScoped cs(_cbCrit.get()); - if(_mixReceiver != NULL) - { + if(_mixReceiver != NULL) { const AudioFrame** dummy = NULL; _mixReceiver->NewMixedAudio( _id, @@ -380,12 +347,11 @@ int32_t AudioConferenceMixerImpl::Process() } if((_mixerStatusCallback != NULL) && - timeForMixerCallback) - { + timeForMixerCallback) { _mixerStatusCallback->MixedParticipants( _id, _scratchMixedParticipants, - _scratchParticipantsToMixAmount); + static_cast<uint32_t>(_scratchParticipantsToMixAmount)); _mixerStatusCallback->VADPositiveParticipants( _id, @@ -397,9 +363,9 @@ int32_t AudioConferenceMixerImpl::Process() // Reclaim all outstanding memory. _audioFramePool->PushMemory(mixedAudio); - ClearAudioFrameList(mixList); - ClearAudioFrameList(rampOutList); - ClearAudioFrameList(additionalFramesList); + ClearAudioFrameList(&mixList); + ClearAudioFrameList(&rampOutList); + ClearAudioFrameList(&additionalFramesList); { CriticalSectionScoped cs(_crit.get()); _processCalls--; @@ -408,22 +374,18 @@ int32_t AudioConferenceMixerImpl::Process() } int32_t AudioConferenceMixerImpl::RegisterMixedStreamCallback( - AudioMixerOutputReceiver& mixReceiver) -{ + AudioMixerOutputReceiver& mixReceiver) { CriticalSectionScoped cs(_cbCrit.get()); - if(_mixReceiver != NULL) - { + if(_mixReceiver != NULL) { return -1; } _mixReceiver = &mixReceiver; return 0; } -int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() -{ +int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() { CriticalSectionScoped cs(_cbCrit.get()); - if(_mixReceiver == NULL) - { + if(_mixReceiver == NULL) { return -1; } _mixReceiver = NULL; @@ -431,16 +393,8 @@ int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() } int32_t AudioConferenceMixerImpl::SetOutputFrequency( - const Frequency frequency) -{ + const Frequency frequency) { CriticalSectionScoped cs(_crit.get()); - const int error = _limiter->set_sample_rate_hz(frequency); - if(error != _limiter->kNoError) - { - WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, - "Error from AudioProcessing: %d", error); - return -1; - } _outputFrequency = frequency; _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000; @@ -449,36 +403,15 @@ int32_t AudioConferenceMixerImpl::SetOutputFrequency( } AudioConferenceMixer::Frequency -AudioConferenceMixerImpl::OutputFrequency() const -{ +AudioConferenceMixerImpl::OutputFrequency() const { CriticalSectionScoped cs(_crit.get()); return _outputFrequency; } -bool AudioConferenceMixerImpl::SetNumLimiterChannels(int numChannels) -{ - if(_limiter->num_input_channels() != numChannels) - { - const int error = _limiter->set_num_channels(numChannels, - numChannels); - if(error != _limiter->kNoError) - { - WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, - "Error from AudioProcessing: %d", error); - assert(false); - return false; - } - } - - return true; -} - int32_t AudioConferenceMixerImpl::RegisterMixerStatusCallback( AudioMixerStatusReceiver& mixerStatusCallback, - const uint32_t amountOf10MsBetweenCallbacks) -{ - if(amountOf10MsBetweenCallbacks == 0) - { + const uint32_t amountOf10MsBetweenCallbacks) { + if(amountOf10MsBetweenCallbacks == 0) { WEBRTC_TRACE( kTraceWarning, kTraceAudioMixerServer, @@ -488,8 +421,7 @@ int32_t AudioConferenceMixerImpl::RegisterMixerStatusCallback( } { CriticalSectionScoped cs(_cbCrit.get()); - if(_mixerStatusCallback != NULL) - { + if(_mixerStatusCallback != NULL) { WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, "Mixer status callback already registered"); return -1; @@ -505,8 +437,7 @@ int32_t AudioConferenceMixerImpl::RegisterMixerStatusCallback( return 0; } -int32_t AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() -{ +int32_t AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() { { CriticalSectionScoped cs(_crit.get()); if(!_mixerStatusCb) @@ -526,38 +457,31 @@ int32_t AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() int32_t AudioConferenceMixerImpl::SetMixabilityStatus( MixerParticipant& participant, - const bool mixable) -{ - if (!mixable) - { + bool mixable) { + if (!mixable) { // Anonymous participants are in a separate list. Make sure that the // participant is in the _participantList if it is being mixed. SetAnonymousMixabilityStatus(participant, false); } - uint32_t numMixedParticipants; + size_t numMixedParticipants; { CriticalSectionScoped cs(_cbCrit.get()); const bool isMixed = - IsParticipantInList(participant,_participantList); + IsParticipantInList(participant, &_participantList); // API must be called with a new state. - if(!(mixable ^ isMixed)) - { + if(!(mixable ^ isMixed)) { WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, "Mixable is aready %s", isMixed ? "ON" : "off"); return -1; } bool success = false; - if(mixable) - { - success = AddParticipantToList(participant,_participantList); - } - else - { - success = RemoveParticipantFromList(participant,_participantList); + if(mixable) { + success = AddParticipantToList(participant, &_participantList); + } else { + success = RemoveParticipantFromList(participant, &_participantList); } - if(!success) - { + if(!success) { WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, "failed to %s participant", mixable ? "add" : "remove"); @@ -565,13 +489,12 @@ int32_t AudioConferenceMixerImpl::SetMixabilityStatus( return -1; } - int numMixedNonAnonymous = _participantList.GetSize(); - if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants) - { + size_t numMixedNonAnonymous = _participantList.size(); + if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants) { numMixedNonAnonymous = kMaximumAmountOfMixedParticipants; } - numMixedParticipants = numMixedNonAnonymous + - _additionalParticipantList.GetSize(); + numMixedParticipants = + numMixedNonAnonymous + _additionalParticipantList.size(); } // A MixerParticipant was added or removed. Make sure the scratch // buffer is updated if necessary. @@ -583,40 +506,34 @@ int32_t AudioConferenceMixerImpl::SetMixabilityStatus( int32_t AudioConferenceMixerImpl::MixabilityStatus( MixerParticipant& participant, - bool& mixable) -{ + bool& mixable) { CriticalSectionScoped cs(_cbCrit.get()); - mixable = IsParticipantInList(participant, _participantList); + mixable = IsParticipantInList(participant, &_participantList); return 0; } int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus( - MixerParticipant& participant, const bool anonymous) -{ + MixerParticipant& participant, const bool anonymous) { CriticalSectionScoped cs(_cbCrit.get()); - if(IsParticipantInList(participant, _additionalParticipantList)) - { - if(anonymous) - { + if(IsParticipantInList(participant, &_additionalParticipantList)) { + if(anonymous) { return 0; } - if(!RemoveParticipantFromList(participant, _additionalParticipantList)) - { + if(!RemoveParticipantFromList(participant, + &_additionalParticipantList)) { WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, "unable to remove participant from anonymous list"); assert(false); return -1; } - return AddParticipantToList(participant, _participantList) ? 0 : -1; + return AddParticipantToList(participant, &_participantList) ? 0 : -1; } - if(!anonymous) - { + if(!anonymous) { return 0; } const bool mixable = RemoveParticipantFromList(participant, - _participantList); - if(!mixable) - { + &_participantList); + if(!mixable) { WEBRTC_TRACE( kTraceWarning, kTraceAudioMixerServer, @@ -626,39 +543,33 @@ int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus( // already registered. return -1; } - return AddParticipantToList(participant, _additionalParticipantList) ? + return AddParticipantToList(participant, &_additionalParticipantList) ? 0 : -1; } int32_t AudioConferenceMixerImpl::AnonymousMixabilityStatus( - MixerParticipant& participant, bool& mixable) -{ + MixerParticipant& participant, bool& mixable) { CriticalSectionScoped cs(_cbCrit.get()); mixable = IsParticipantInList(participant, - _additionalParticipantList); + &_additionalParticipantList); return 0; } int32_t AudioConferenceMixerImpl::SetMinimumMixingFrequency( - Frequency freq) -{ + Frequency freq) { // Make sure that only allowed sampling frequencies are used. Use closest // higher sampling frequency to avoid losing information. - if (static_cast<int>(freq) == 12000) - { + if (static_cast<int>(freq) == 12000) { freq = kWbInHz; } else if (static_cast<int>(freq) == 24000) { freq = kSwbInHz; } if((freq == kNbInHz) || (freq == kWbInHz) || (freq == kSwbInHz) || - (freq == kLowestPossible)) - { + (freq == kLowestPossible)) { _minimumMixingFreq=freq; return 0; - } - else - { + } else { WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, "SetMinimumMixingFrequency incorrect frequency: %i",freq); assert(false); @@ -668,20 +579,17 @@ int32_t AudioConferenceMixerImpl::SetMinimumMixingFrequency( // Check all AudioFrames that are to be mixed. The highest sampling frequency // found is the lowest that can be used without losing information. -int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() -{ +int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() { const int participantListFrequency = - GetLowestMixingFrequencyFromList(_participantList); + GetLowestMixingFrequencyFromList(&_participantList); const int anonymousListFrequency = - GetLowestMixingFrequencyFromList(_additionalParticipantList); + GetLowestMixingFrequencyFromList(&_additionalParticipantList); const int highestFreq = (participantListFrequency > anonymousListFrequency) ? participantListFrequency : anonymousListFrequency; // Check if the user specified a lowest mixing frequency. - if(_minimumMixingFreq != kLowestPossible) - { - if(_minimumMixingFreq > highestFreq) - { + if(_minimumMixingFreq != kLowestPossible) { + if(_minimumMixingFreq > highestFreq) { return _minimumMixingFreq; } } @@ -689,60 +597,47 @@ int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() } int32_t AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList( - ListWrapper& mixList) -{ + MixerParticipantList* mixList) { int32_t highestFreq = 8000; - ListItem* item = mixList.First(); - while(item) - { - MixerParticipant* participant = - static_cast<MixerParticipant*>(item->GetItem()); - const int32_t neededFrequency = participant->NeededFrequency(_id); - if(neededFrequency > highestFreq) - { + for (MixerParticipantList::iterator iter = mixList->begin(); + iter != mixList->end(); + ++iter) { + const int32_t neededFrequency = (*iter)->NeededFrequency(_id); + if(neededFrequency > highestFreq) { highestFreq = neededFrequency; } - item = mixList.Next(item); } return highestFreq; } void AudioConferenceMixerImpl::UpdateToMix( - ListWrapper& mixList, - ListWrapper& rampOutList, + AudioFrameList* mixList, + AudioFrameList* rampOutList, std::map<int, MixerParticipant*>* mixParticipantList, - uint32_t& maxAudioFrameCounter) { + size_t& maxAudioFrameCounter) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "UpdateToMix(mixList,rampOutList,mixParticipantList,%d)", maxAudioFrameCounter); - const uint32_t mixListStartSize = mixList.GetSize(); - ListWrapper activeList; // Elements are AudioFrames + const size_t mixListStartSize = mixList->size(); + AudioFrameList activeList; // Struct needed by the passive lists to keep track of which AudioFrame // belongs to which MixerParticipant. - struct ParticipantFramePair - { - MixerParticipant* participant; - AudioFrame* audioFrame; - }; - ListWrapper passiveWasNotMixedList; // Elements are MixerParticipant - ListWrapper passiveWasMixedList; // Elements are MixerParticipant - ListItem* item = _participantList.First(); - while(item) - { + ParticipantFramePairList passiveWasNotMixedList; + ParticipantFramePairList passiveWasMixedList; + for (MixerParticipantList::iterator participant = _participantList.begin(); + participant != _participantList.end(); + ++participant) { // Stop keeping track of passive participants if there are already // enough participants available (they wont be mixed anyway). bool mustAddToPassiveList = (maxAudioFrameCounter > - (activeList.GetSize() + - passiveWasMixedList.GetSize() + - passiveWasNotMixedList.GetSize())); + (activeList.size() + + passiveWasMixedList.size() + + passiveWasNotMixedList.size())); - MixerParticipant* participant = static_cast<MixerParticipant*>( - item->GetItem()); bool wasMixed = false; - participant->_mixHistory->WasMixed(wasMixed); + (*participant)->_mixHistory->WasMixed(wasMixed); AudioFrame* audioFrame = NULL; - if(_audioFramePool->PopMemory(audioFrame) == -1) - { + if(_audioFramePool->PopMemory(audioFrame) == -1) { WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id, "failed PopMemory() call"); assert(false); @@ -750,56 +645,51 @@ void AudioConferenceMixerImpl::UpdateToMix( } audioFrame->sample_rate_hz_ = _outputFrequency; - if(participant->GetAudioFrame(_id,*audioFrame) != 0) - { + if((*participant)->GetAudioFrame(_id,*audioFrame) != 0) { WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, "failed to GetAudioFrame() from participant"); _audioFramePool->PushMemory(audioFrame); - item = _participantList.Next(item); continue; } + if (_participantList.size() != 1) { + // TODO(wu): Issue 3390, add support for multiple participants case. + audioFrame->ntp_time_ms_ = -1; + } + // TODO(henrike): this assert triggers in some test cases where SRTP is // used which prevents NetEQ from making a VAD. Temporarily disable this // assert until the problem is fixed on a higher level. // assert(audioFrame->vad_activity_ != AudioFrame::kVadUnknown); - if (audioFrame->vad_activity_ == AudioFrame::kVadUnknown) - { + if (audioFrame->vad_activity_ == AudioFrame::kVadUnknown) { WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, "invalid VAD state from participant"); } - if(audioFrame->vad_activity_ == AudioFrame::kVadActive) - { - if(!wasMixed) - { + if(audioFrame->vad_activity_ == AudioFrame::kVadActive) { + if(!wasMixed) { RampIn(*audioFrame); } - if(activeList.GetSize() >= maxAudioFrameCounter) - { + if(activeList.size() >= maxAudioFrameCounter) { // There are already more active participants than should be // mixed. Only keep the ones with the highest energy. - ListItem* replaceItem = NULL; + AudioFrameList::iterator replaceItem; CalculateEnergy(*audioFrame); uint32_t lowestEnergy = audioFrame->energy_; - ListItem* activeItem = activeList.First(); - while(activeItem) - { - AudioFrame* replaceFrame = static_cast<AudioFrame*>( - activeItem->GetItem()); - CalculateEnergy(*replaceFrame); - if(replaceFrame->energy_ < lowestEnergy) - { - replaceItem = activeItem; - lowestEnergy = replaceFrame->energy_; + bool found_replace_item = false; + for (AudioFrameList::iterator iter = activeList.begin(); + iter != activeList.end(); + ++iter) { + CalculateEnergy(**iter); + if((*iter)->energy_ < lowestEnergy) { + replaceItem = iter; + lowestEnergy = (*iter)->energy_; + found_replace_item = true; } - activeItem = activeList.Next(activeItem); } - if(replaceItem != NULL) - { - AudioFrame* replaceFrame = static_cast<AudioFrame*>( - replaceItem->GetItem()); + if(found_replace_item) { + AudioFrame* replaceFrame = *replaceItem; bool replaceWasMixed = false; std::map<int, MixerParticipant*>::iterator it = @@ -812,255 +702,219 @@ void AudioConferenceMixerImpl::UpdateToMix( it->second->_mixHistory->WasMixed(replaceWasMixed); mixParticipantList->erase(replaceFrame->id_); - activeList.Erase(replaceItem); + activeList.erase(replaceItem); - activeList.PushFront(static_cast<void*>(audioFrame)); - (*mixParticipantList)[audioFrame->id_] = participant; + activeList.push_front(audioFrame); + (*mixParticipantList)[audioFrame->id_] = *participant; assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants); if (replaceWasMixed) { RampOut(*replaceFrame); - rampOutList.PushBack(static_cast<void*>(replaceFrame)); - assert(rampOutList.GetSize() <= + rampOutList->push_back(replaceFrame); + assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants); } else { _audioFramePool->PushMemory(replaceFrame); } } else { - if(wasMixed) - { + if(wasMixed) { RampOut(*audioFrame); - rampOutList.PushBack(static_cast<void*>(audioFrame)); - assert(rampOutList.GetSize() <= + rampOutList->push_back(audioFrame); + assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants); } else { _audioFramePool->PushMemory(audioFrame); } } } else { - activeList.PushFront(static_cast<void*>(audioFrame)); - (*mixParticipantList)[audioFrame->id_] = participant; + activeList.push_front(audioFrame); + (*mixParticipantList)[audioFrame->id_] = *participant; assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants); } } else { - if(wasMixed) - { + if(wasMixed) { ParticipantFramePair* pair = new ParticipantFramePair; pair->audioFrame = audioFrame; - pair->participant = participant; - passiveWasMixedList.PushBack(static_cast<void*>(pair)); + pair->participant = *participant; + passiveWasMixedList.push_back(pair); } else if(mustAddToPassiveList) { RampIn(*audioFrame); ParticipantFramePair* pair = new ParticipantFramePair; pair->audioFrame = audioFrame; - pair->participant = participant; - passiveWasNotMixedList.PushBack(static_cast<void*>(pair)); + pair->participant = *participant; + passiveWasNotMixedList.push_back(pair); } else { _audioFramePool->PushMemory(audioFrame); } } - item = _participantList.Next(item); } - assert(activeList.GetSize() <= maxAudioFrameCounter); + assert(activeList.size() <= maxAudioFrameCounter); // At this point it is known which participants should be mixed. Transfer // this information to this functions output parameters. - while(!activeList.Empty()) - { - ListItem* mixItem = activeList.First(); - mixList.PushBack(mixItem->GetItem()); - activeList.Erase(mixItem); + for (AudioFrameList::iterator iter = activeList.begin(); + iter != activeList.end(); + ++iter) { + mixList->push_back(*iter); } + activeList.clear(); // Always mix a constant number of AudioFrames. If there aren't enough // active participants mix passive ones. Starting with those that was mixed // last iteration. - while(!passiveWasMixedList.Empty()) - { - ListItem* mixItem = passiveWasMixedList.First(); - ParticipantFramePair* pair = static_cast<ParticipantFramePair*>( - mixItem->GetItem()); - if(mixList.GetSize() < maxAudioFrameCounter + mixListStartSize) - { - mixList.PushBack(pair->audioFrame); - (*mixParticipantList)[pair->audioFrame->id_] = - pair->participant; + for (ParticipantFramePairList::iterator iter = passiveWasMixedList.begin(); + iter != passiveWasMixedList.end(); + ++iter) { + if(mixList->size() < maxAudioFrameCounter + mixListStartSize) { + mixList->push_back((*iter)->audioFrame); + (*mixParticipantList)[(*iter)->audioFrame->id_] = + (*iter)->participant; assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants); + } else { + _audioFramePool->PushMemory((*iter)->audioFrame); } - else - { - _audioFramePool->PushMemory(pair->audioFrame); - } - delete pair; - passiveWasMixedList.Erase(mixItem); + delete *iter; } // And finally the ones that have not been mixed for a while. - while(!passiveWasNotMixedList.Empty()) - { - ListItem* mixItem = passiveWasNotMixedList.First(); - ParticipantFramePair* pair = static_cast<ParticipantFramePair*>( - mixItem->GetItem()); - if(mixList.GetSize() < maxAudioFrameCounter + mixListStartSize) - { - mixList.PushBack(pair->audioFrame); - (*mixParticipantList)[pair->audioFrame->id_] = pair->participant; + for (ParticipantFramePairList::iterator iter = + passiveWasNotMixedList.begin(); + iter != passiveWasNotMixedList.end(); + ++iter) { + if(mixList->size() < maxAudioFrameCounter + mixListStartSize) { + mixList->push_back((*iter)->audioFrame); + (*mixParticipantList)[(*iter)->audioFrame->id_] = + (*iter)->participant; assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants); + } else { + _audioFramePool->PushMemory((*iter)->audioFrame); } - else - { - _audioFramePool->PushMemory(pair->audioFrame); - } - delete pair; - passiveWasNotMixedList.Erase(mixItem); + delete *iter; } - assert(maxAudioFrameCounter + mixListStartSize >= mixList.GetSize()); - maxAudioFrameCounter += mixListStartSize - mixList.GetSize(); + assert(maxAudioFrameCounter + mixListStartSize >= mixList->size()); + maxAudioFrameCounter += mixListStartSize - mixList->size(); } void AudioConferenceMixerImpl::GetAdditionalAudio( - ListWrapper& additionalFramesList) -{ + AudioFrameList* additionalFramesList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "GetAdditionalAudio(additionalFramesList)"); - ListItem* item = _additionalParticipantList.First(); - while(item) - { - // The GetAudioFrame() callback may remove the current item. Store the - // next item just in case that happens. - ListItem* nextItem = _additionalParticipantList.Next(item); - - MixerParticipant* participant = static_cast<MixerParticipant*>( - item->GetItem()); + // The GetAudioFrame() callback may result in the participant being removed + // from additionalParticipantList_. If that happens it will invalidate any + // iterators. Create a copy of the participants list such that the list of + // participants can be traversed safely. + MixerParticipantList additionalParticipantList; + additionalParticipantList.insert(additionalParticipantList.begin(), + _additionalParticipantList.begin(), + _additionalParticipantList.end()); + + for (MixerParticipantList::iterator participant = + additionalParticipantList.begin(); + participant != additionalParticipantList.end(); + ++participant) { AudioFrame* audioFrame = NULL; - if(_audioFramePool->PopMemory(audioFrame) == -1) - { + if(_audioFramePool->PopMemory(audioFrame) == -1) { WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id, "failed PopMemory() call"); assert(false); return; } audioFrame->sample_rate_hz_ = _outputFrequency; - if(participant->GetAudioFrame(_id, *audioFrame) != 0) - { + if((*participant)->GetAudioFrame(_id, *audioFrame) != 0) { WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, "failed to GetAudioFrame() from participant"); _audioFramePool->PushMemory(audioFrame); - item = nextItem; continue; } - if(audioFrame->samples_per_channel_ == 0) - { + if(audioFrame->samples_per_channel_ == 0) { // Empty frame. Don't use it. _audioFramePool->PushMemory(audioFrame); - item = nextItem; continue; } - additionalFramesList.PushBack(static_cast<void*>(audioFrame)); - item = nextItem; + additionalFramesList->push_back(audioFrame); } } void AudioConferenceMixerImpl::UpdateMixedStatus( - std::map<int, MixerParticipant*>& mixedParticipantsMap) -{ + std::map<int, MixerParticipant*>& mixedParticipantsMap) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "UpdateMixedStatus(mixedParticipantsMap)"); assert(mixedParticipantsMap.size() <= kMaximumAmountOfMixedParticipants); // Loop through all participants. If they are in the mix map they // were mixed. - ListItem* participantItem = _participantList.First(); - while(participantItem != NULL) - { + for (MixerParticipantList::iterator participant = _participantList.begin(); + participant != _participantList.end(); + ++participant) { bool isMixed = false; - MixerParticipant* participant = - static_cast<MixerParticipant*>(participantItem->GetItem()); - for (std::map<int, MixerParticipant*>::iterator it = mixedParticipantsMap.begin(); it != mixedParticipantsMap.end(); ++it) { - if (it->second == participant) { + if (it->second == *participant) { isMixed = true; break; } } - participant->_mixHistory->SetIsMixed(isMixed); - participantItem = _participantList.Next(participantItem); + (*participant)->_mixHistory->SetIsMixed(isMixed); } } -void AudioConferenceMixerImpl::ClearAudioFrameList(ListWrapper& audioFrameList) -{ +void AudioConferenceMixerImpl::ClearAudioFrameList( + AudioFrameList* audioFrameList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "ClearAudioFrameList(audioFrameList)"); - ListItem* item = audioFrameList.First(); - while(item) - { - AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem()); - _audioFramePool->PushMemory(audioFrame); - audioFrameList.Erase(item); - item = audioFrameList.First(); + for (AudioFrameList::iterator iter = audioFrameList->begin(); + iter != audioFrameList->end(); + ++iter) { + _audioFramePool->PushMemory(*iter); } + audioFrameList->clear(); } void AudioConferenceMixerImpl::UpdateVADPositiveParticipants( - ListWrapper& mixList) -{ + AudioFrameList* mixList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "UpdateVADPositiveParticipants(mixList)"); - ListItem* item = mixList.First(); - while(item != NULL) - { - AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem()); - CalculateEnergy(*audioFrame); - if(audioFrame->vad_activity_ == AudioFrame::kVadActive) - { + for (AudioFrameList::iterator iter = mixList->begin(); + iter != mixList->end(); + ++iter) { + CalculateEnergy(**iter); + if((*iter)->vad_activity_ == AudioFrame::kVadActive) { _scratchVadPositiveParticipants[ _scratchVadPositiveParticipantsAmount].participant = - audioFrame->id_; + (*iter)->id_; // TODO(andrew): to what should this be set? _scratchVadPositiveParticipants[ _scratchVadPositiveParticipantsAmount].level = 0; _scratchVadPositiveParticipantsAmount++; } - item = mixList.Next(item); } } bool AudioConferenceMixerImpl::IsParticipantInList( MixerParticipant& participant, - ListWrapper& participantList) -{ + MixerParticipantList* participantList) const { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "IsParticipantInList(participant,participantList)"); - ListItem* item = participantList.First(); - while(item != NULL) - { - MixerParticipant* rhsParticipant = - static_cast<MixerParticipant*>(item->GetItem()); - if(&participant == rhsParticipant) - { + for (MixerParticipantList::const_iterator iter = participantList->begin(); + iter != participantList->end(); + ++iter) { + if(&participant == *iter) { return true; } - item = participantList.Next(item); } return false; } bool AudioConferenceMixerImpl::AddParticipantToList( MixerParticipant& participant, - ListWrapper& participantList) -{ + MixerParticipantList* participantList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "AddParticipantToList(participant, participantList)"); - if(participantList.PushBack(static_cast<void*>(&participant)) == -1) - { - return false; - } + participantList->push_back(&participant); // Make sure that the mixed status is correct for new MixerParticipant. participant._mixHistory->ResetMixedStatus(); return true; @@ -1068,52 +922,53 @@ bool AudioConferenceMixerImpl::AddParticipantToList( bool AudioConferenceMixerImpl::RemoveParticipantFromList( MixerParticipant& participant, - ListWrapper& participantList) -{ + MixerParticipantList* participantList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "RemoveParticipantFromList(participant, participantList)"); - ListItem* item = participantList.First(); - while(item) - { - if(item->GetItem() == &participant) - { - participantList.Erase(item); + for (MixerParticipantList::iterator iter = participantList->begin(); + iter != participantList->end(); + ++iter) { + if(*iter == &participant) { + participantList->erase(iter); // Participant is no longer mixed, reset to default. participant._mixHistory->ResetMixedStatus(); return true; } - item = participantList.Next(item); } return false; } int32_t AudioConferenceMixerImpl::MixFromList( AudioFrame& mixedAudio, - const ListWrapper& audioFrameList) -{ + const AudioFrameList* audioFrameList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "MixFromList(mixedAudio, audioFrameList)"); - uint32_t position = 0; - ListItem* item = audioFrameList.First(); - if(item == NULL) - { - return 0; - } + if(audioFrameList->empty()) return 0; - if(_numMixedParticipants == 1) - { + uint32_t position = 0; + if(_numMixedParticipants == 1) { // No mixing required here; skip the saturation protection. - AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem()); + AudioFrame* audioFrame = audioFrameList->front(); mixedAudio.CopyFrom(*audioFrame); SetParticipantStatistics(&_scratchMixedParticipants[position], *audioFrame); return 0; } - while(item != NULL) - { - if(position >= kMaximumAmountOfMixedParticipants) - { + if (audioFrameList->size() == 1) { + mixedAudio.timestamp_ = audioFrameList->front()->timestamp_; + mixedAudio.elapsed_time_ms_ = audioFrameList->front()->elapsed_time_ms_; + } else { + // TODO(wu): Issue 3390. + // Audio frame timestamp is only supported in one channel case. + mixedAudio.timestamp_ = 0; + mixedAudio.elapsed_time_ms_ = -1; + } + + for (AudioFrameList::const_iterator iter = audioFrameList->begin(); + iter != audioFrameList->end(); + ++iter) { + if(position >= kMaximumAmountOfMixedParticipants) { WEBRTC_TRACE( kTraceMemory, kTraceAudioMixerServer, @@ -1124,14 +979,12 @@ int32_t AudioConferenceMixerImpl::MixFromList( assert(false); position = 0; } - AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem()); - MixFrames(&mixedAudio, audioFrame); + MixFrames(&mixedAudio, (*iter)); SetParticipantStatistics(&_scratchMixedParticipants[position], - *audioFrame); + **iter); position++; - item = audioFrameList.Next(item); } return 0; @@ -1140,35 +993,29 @@ int32_t AudioConferenceMixerImpl::MixFromList( // TODO(andrew): consolidate this function with MixFromList. int32_t AudioConferenceMixerImpl::MixAnonomouslyFromList( AudioFrame& mixedAudio, - const ListWrapper& audioFrameList) -{ + const AudioFrameList* audioFrameList) { WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, "MixAnonomouslyFromList(mixedAudio, audioFrameList)"); - ListItem* item = audioFrameList.First(); - if(item == NULL) - return 0; - if(_numMixedParticipants == 1) - { + if(audioFrameList->empty()) return 0; + + if(_numMixedParticipants == 1) { // No mixing required here; skip the saturation protection. - AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem()); + AudioFrame* audioFrame = audioFrameList->front(); mixedAudio.CopyFrom(*audioFrame); return 0; } - while(item != NULL) - { - AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem()); - MixFrames(&mixedAudio, audioFrame); - item = audioFrameList.Next(item); + for (AudioFrameList::const_iterator iter = audioFrameList->begin(); + iter != audioFrameList->end(); + ++iter) { + MixFrames(&mixedAudio, *iter); } return 0; } -bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) -{ - if(_numMixedParticipants == 1) - { +bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) { + if(_numMixedParticipants == 1) { return true; } @@ -1187,8 +1034,7 @@ bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) // negative value is undefined). mixedAudio += mixedAudio; - if(error != _limiter->kNoError) - { + if(error != _limiter->kNoError) { WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, "Error from AudioProcessing: %d", error); assert(false); diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h index 737acbb0ba4..31dc71e5dce 100644 --- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h +++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h @@ -11,6 +11,7 @@ #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_ +#include <list> #include <map> #include "webrtc/engine_configurations.h" @@ -19,13 +20,15 @@ #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h" #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h" #include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/interface/list_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { class AudioProcessing; class CriticalSectionWrapper; +typedef std::list<AudioFrame*> AudioFrameList; +typedef std::list<MixerParticipant*> MixerParticipantList; + // Cheshire cat implementation of MixerParticipant's non virtual functions. class MixHistory { @@ -74,7 +77,7 @@ public: const uint32_t amountOf10MsBetweenCallbacks); virtual int32_t UnRegisterMixerStatusCallback(); virtual int32_t SetMixabilityStatus(MixerParticipant& participant, - const bool mixable); + bool mixable); virtual int32_t MixabilityStatus(MixerParticipant& participant, bool& mixable); virtual int32_t SetMinimumMixingFrequency(Frequency freq); @@ -89,10 +92,6 @@ private: int32_t SetOutputFrequency(const Frequency frequency); Frequency OutputFrequency() const; - // Must be called whenever an audio frame indicates the number of channels - // has changed. - bool SetNumLimiterChannels(int numChannels); - // Fills mixList with the AudioFrames pointers that should be used when // mixing. Fills mixParticipantList with ParticipantStatistics for the // participants who's AudioFrames are inside mixList. @@ -102,18 +101,18 @@ private: // used to be mixed but shouldn't be mixed any longer. These AudioFrames // should be ramped out over this AudioFrame to avoid audio discontinuities. void UpdateToMix( - ListWrapper& mixList, - ListWrapper& rampOutList, + AudioFrameList* mixList, + AudioFrameList* rampOutList, std::map<int, MixerParticipant*>* mixParticipantList, - uint32_t& maxAudioFrameCounter); + size_t& maxAudioFrameCounter); // Return the lowest mixing frequency that can be used without having to // downsample any audio. int32_t GetLowestMixingFrequency(); - int32_t GetLowestMixingFrequencyFromList(ListWrapper& mixList); + int32_t GetLowestMixingFrequencyFromList(MixerParticipantList* mixList); // Return the AudioFrames that should be mixed anonymously. - void GetAdditionalAudio(ListWrapper& additionalFramesList); + void GetAdditionalAudio(AudioFrameList* additionalFramesList); // Update the MixHistory of all MixerParticipants. mixedParticipantsList // should contain a map of MixerParticipants that have been mixed. @@ -121,44 +120,44 @@ private: std::map<int, MixerParticipant*>& mixedParticipantsList); // Clears audioFrameList and reclaims all memory associated with it. - void ClearAudioFrameList(ListWrapper& audioFrameList); + void ClearAudioFrameList(AudioFrameList* audioFrameList); // Update the list of MixerParticipants who have a positive VAD. mixList // should be a list of AudioFrames void UpdateVADPositiveParticipants( - ListWrapper& mixList); + AudioFrameList* mixList); // This function returns true if it finds the MixerParticipant in the // specified list of MixerParticipants. bool IsParticipantInList( MixerParticipant& participant, - ListWrapper& participantList); + MixerParticipantList* participantList) const; // Add/remove the MixerParticipant to the specified // MixerParticipant list. bool AddParticipantToList( MixerParticipant& participant, - ListWrapper& participantList); + MixerParticipantList* participantList); bool RemoveParticipantFromList( MixerParticipant& removeParticipant, - ListWrapper& participantList); + MixerParticipantList* participantList); // Mix the AudioFrames stored in audioFrameList into mixedAudio. int32_t MixFromList( AudioFrame& mixedAudio, - const ListWrapper& audioFrameList); + const AudioFrameList* audioFrameList); // Mix the AudioFrames stored in audioFrameList into mixedAudio. No // record will be kept of this mix (e.g. the corresponding MixerParticipants // will not be marked as IsMixed() int32_t MixAnonomouslyFromList(AudioFrame& mixedAudio, - const ListWrapper& audioFrameList); + const AudioFrameList* audioFrameList); bool LimitMixedAudio(AudioFrame& mixedAudio); // Scratch memory // Note that the scratch memory may only be touched in the scope of // Process(). - uint32_t _scratchParticipantsToMixAmount; + size_t _scratchParticipantsToMixAmount; ParticipantStatistics _scratchMixedParticipants[ kMaximumAmountOfMixedParticipants]; uint32_t _scratchVadPositiveParticipantsAmount; @@ -176,9 +175,9 @@ private: AudioMixerOutputReceiver* _mixReceiver; AudioMixerStatusReceiver* _mixerStatusCallback; - uint32_t _amountOf10MsBetweenCallbacks; - uint32_t _amountOf10MsUntilNextCallback; - bool _mixerStatusCb; + uint32_t _amountOf10MsBetweenCallbacks; + uint32_t _amountOf10MsUntilNextCallback; + bool _mixerStatusCb; // The current sample frequency and sample size when mixing. Frequency _outputFrequency; @@ -188,10 +187,11 @@ private: MemoryPool<AudioFrame>* _audioFramePool; // List of all participants. Note all lists are disjunct - ListWrapper _participantList; // May be mixed. - ListWrapper _additionalParticipantList; // Always mixed, anonomously. + MixerParticipantList _participantList; // May be mixed. + // Always mixed, anonomously. + MixerParticipantList _additionalParticipantList; - uint32_t _numMixedParticipants; + size_t _numMixedParticipants; uint32_t _timeStamp; diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc index 679d608f50f..3dce5c8bea6 100644 --- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc +++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc @@ -41,10 +41,6 @@ const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]); namespace webrtc { void CalculateEnergy(AudioFrame& audioFrame) { - if(audioFrame.energy_ != 0xffffffff) - { - return; - } audioFrame.energy_ = 0; for(int position = 0; position < audioFrame.samples_per_channel_; position++) diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h index 6d4dccf8ae3..04e7cd52254 100644 --- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h +++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h @@ -12,9 +12,9 @@ #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_ #include <assert.h> +#include <list> #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/list_wrapper.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -40,7 +40,7 @@ private: bool _terminate; - ListWrapper _memoryPool; + std::list<MemoryType*> _memoryPool; uint32_t _initialPoolSize; uint32_t _createdMemory; @@ -51,7 +51,6 @@ template<class MemoryType> MemoryPoolImpl<MemoryType>::MemoryPoolImpl(int32_t initialPoolSize) : _crit(CriticalSectionWrapper::CreateCriticalSection()), _terminate(false), - _memoryPool(), _initialPoolSize(initialPoolSize), _createdMemory(0), _outstandingMemory(0) @@ -76,20 +75,17 @@ int32_t MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory) memory = NULL; return -1; } - ListItem* item = _memoryPool.First(); - if(item == NULL) - { + if (_memoryPool.empty()) { // _memoryPool empty create new memory. CreateMemory(_initialPoolSize); - item = _memoryPool.First(); - if(item == NULL) + if(_memoryPool.empty()) { memory = NULL; return -1; } } - memory = static_cast<MemoryType*>(item->GetItem()); - _memoryPool.Erase(item); + memory = _memoryPool.front(); + _memoryPool.pop_front(); _outstandingMemory++; return 0; } @@ -103,7 +99,7 @@ int32_t MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory) } CriticalSectionScoped cs(_crit); _outstandingMemory--; - if(_memoryPool.GetSize() > (_initialPoolSize << 1)) + if(_memoryPool.size() > (_initialPoolSize << 1)) { // Reclaim memory if less than half of the pool is unused. _createdMemory--; @@ -111,7 +107,7 @@ int32_t MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory) memory = NULL; return 0; } - _memoryPool.PushBack(static_cast<void*>(memory)); + _memoryPool.push_back(memory); memory = NULL; return 0; } @@ -127,21 +123,15 @@ template<class MemoryType> int32_t MemoryPoolImpl<MemoryType>::Terminate() { CriticalSectionScoped cs(_crit); - assert(_createdMemory == _outstandingMemory + _memoryPool.GetSize()); + assert(_createdMemory == _outstandingMemory + _memoryPool.size()); _terminate = true; // Reclaim all memory. while(_createdMemory > 0) { - ListItem* item = _memoryPool.First(); - if(item == NULL) - { - // There is memory that hasn't been returned yet. - return -1; - } - MemoryType* memory = static_cast<MemoryType*>(item->GetItem()); + MemoryType* memory = _memoryPool.front(); + _memoryPool.pop_front(); delete memory; - _memoryPool.Erase(item); _createdMemory--; } return 0; @@ -158,7 +148,7 @@ int32_t MemoryPoolImpl<MemoryType>::CreateMemory( { return -1; } - _memoryPool.PushBack(static_cast<void*>(memory)); + _memoryPool.push_back(memory); _createdMemory++; } return 0; diff --git a/chromium/third_party/webrtc/modules/audio_device/Android.mk b/chromium/third_party/webrtc/modules/audio_device/Android.mk index affa5e1c2d4..4b3b9124d7b 100644 --- a/chromium/third_party/webrtc/modules/audio_device/Android.mk +++ b/chromium/third_party/webrtc/modules/audio_device/Android.mk @@ -25,7 +25,8 @@ LOCAL_SRC_FILES := \ android/audio_device_android_opensles.cc \ android/audio_device_utility_android.cc \ dummy/audio_device_utility_dummy.cc \ - dummy/audio_device_dummy.cc + dummy/audio_device_dummy.cc \ + dummy/file_audio_device.cc # Flags passed to both C and C++ files. LOCAL_CFLAGS := \ diff --git a/chromium/third_party/webrtc/modules/audio_device/OWNERS b/chromium/third_party/webrtc/modules/audio_device/OWNERS index a07ced37b30..7bb3cd52378 100644 --- a/chromium/third_party/webrtc/modules/audio_device/OWNERS +++ b/chromium/third_party/webrtc/modules/audio_device/OWNERS @@ -2,3 +2,10 @@ henrikg@webrtc.org henrika@webrtc.org niklas.enbom@webrtc.org xians@webrtc.org + +per-file *.isolate=kjellander@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_device/android/audio_device_template.h b/chromium/third_party/webrtc/modules/audio_device/android/audio_device_template.h index 28112698259..f851f4703d2 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/audio_device_template.h +++ b/chromium/third_party/webrtc/modules/audio_device/android/audio_device_template.h @@ -30,6 +30,12 @@ class AudioDeviceTemplate : public AudioDeviceGeneric { } return InputType::SetAndroidAudioDeviceObjects(javaVM, env, context); } + + static void ClearAndroidAudioDeviceObjects() { + OutputType::ClearAndroidAudioDeviceObjects(); + InputType::ClearAndroidAudioDeviceObjects(); + } + explicit AudioDeviceTemplate(const int32_t id) : output_(id), input_(id, &output_) { @@ -169,11 +175,6 @@ class AudioDeviceTemplate : public AudioDeviceGeneric { return -1; } - int32_t SpeakerIsAvailable( - bool& available) { // NOLINT - return output_.SpeakerIsAvailable(available); - } - int32_t InitSpeaker() { return output_.InitSpeaker(); } @@ -182,11 +183,6 @@ class AudioDeviceTemplate : public AudioDeviceGeneric { return output_.SpeakerIsInitialized(); } - int32_t MicrophoneIsAvailable( - bool& available) { // NOLINT - return input_.MicrophoneIsAvailable(available); - } - int32_t InitMicrophone() { return input_.InitMicrophone(); } diff --git a/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer.h b/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer.h index 597b8aaa389..e577b72fd4b 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer.h @@ -56,7 +56,7 @@ class FineAudioBuffer { int bytes_per_10_ms_; // Storage for samples that are not yet asked for. - scoped_array<int8_t> cache_buffer_; + scoped_ptr<int8_t[]> cache_buffer_; int cached_buffer_start_; // Location of first unread sample. int cached_bytes_; // Number of bytes stored in cache. }; diff --git a/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc index 69ba741d18a..e1f03f8f3c3 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc @@ -80,7 +80,7 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) { FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes, sample_rate); - scoped_array<int8_t> out_buffer; + scoped_ptr<int8_t[]> out_buffer; out_buffer.reset( new int8_t[fine_buffer.RequiredBufferSizeBytes()]); for (int i = 0; i < kNumberOfFrames; ++i) { diff --git a/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.cc b/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.cc index e276fcc5230..f22d8bf7ef8 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.cc +++ b/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.cc @@ -75,6 +75,9 @@ int32_t OpenSlesInput::SetAndroidAudioDeviceObjects(void* javaVM, return 0; } +void OpenSlesInput::ClearAndroidAudioDeviceObjects() { +} + int32_t OpenSlesInput::Init() { assert(!initialized_); @@ -177,11 +180,6 @@ int32_t OpenSlesInput::SetAGC(bool enable) { return 0; } -int32_t OpenSlesInput::MicrophoneIsAvailable(bool& available) { // NOLINT - available = true; - return 0; -} - int32_t OpenSlesInput::InitMicrophone() { assert(initialized_); assert(!recording_); @@ -291,7 +289,7 @@ void OpenSlesInput::AllocateBuffers() { fifo_.reset(new SingleRwFifo(num_fifo_buffers_needed_)); // Allocate the memory area to be used. - rec_buf_.reset(new scoped_array<int8_t>[TotalBuffersUsed()]); + rec_buf_.reset(new scoped_ptr<int8_t[]>[TotalBuffersUsed()]); for (int i = 0; i < TotalBuffersUsed(); ++i) { rec_buf_[i].reset(new int8_t[buffer_size_bytes()]); } diff --git a/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.h b/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.h index ea5144b70fe..d27d82435db 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.h +++ b/chromium/third_party/webrtc/modules/audio_device/android/opensles_input.h @@ -41,6 +41,7 @@ class OpenSlesInput { static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* env, void* context); + static void ClearAndroidAudioDeviceObjects(); // Main initializaton and termination int32_t Init(); @@ -76,7 +77,6 @@ class OpenSlesInput { bool AGC() const { return agc_enabled_; } // Audio mixer initialization - int32_t MicrophoneIsAvailable(bool& available); // NOLINT int32_t InitMicrophone(); bool MicrophoneIsInitialized() const { return mic_initialized_; } @@ -205,7 +205,7 @@ class OpenSlesInput { // Audio buffers AudioDeviceBuffer* audio_buffer_; // Holds all allocated memory such that it is deallocated properly. - scoped_array<scoped_array<int8_t> > rec_buf_; + scoped_ptr<scoped_ptr<int8_t[]>[]> rec_buf_; // Index in |rec_buf_| pointing to the audio buffer that will be ready the // next time RecorderSimpleBufferQueueCallbackHandler is invoked. // Ready means buffer contains audio data from the device. diff --git a/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.cc b/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.cc index 3823305532d..377789b2371 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.cc +++ b/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.cc @@ -76,6 +76,10 @@ int32_t OpenSlesOutput::SetAndroidAudioDeviceObjects(void* javaVM, return 0; } +void OpenSlesOutput::ClearAndroidAudioDeviceObjects() { + AudioManagerJni::ClearAndroidAudioDeviceObjects(); +} + int32_t OpenSlesOutput::Init() { assert(!initialized_); @@ -184,11 +188,6 @@ int32_t OpenSlesOutput::StopPlayout() { return 0; } -int32_t OpenSlesOutput::SpeakerIsAvailable(bool& available) { // NOLINT - available = true; - return 0; -} - int32_t OpenSlesOutput::InitSpeaker() { assert(!playing_); speaker_initialized_ = true; @@ -341,7 +340,7 @@ void OpenSlesOutput::AllocateBuffers() { fifo_.reset(new SingleRwFifo(num_fifo_buffers_needed_)); // Allocate the memory area to be used. - play_buf_.reset(new scoped_array<int8_t>[TotalBuffersUsed()]); + play_buf_.reset(new scoped_ptr<int8_t[]>[TotalBuffersUsed()]); int required_buffer_size = fine_buffer_->RequiredBufferSizeBytes(); for (int i = 0; i < TotalBuffersUsed(); ++i) { play_buf_[i].reset(new int8_t[required_buffer_size]); diff --git a/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.h b/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.h index 38ca969a019..aa9b5bf1213 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.h +++ b/chromium/third_party/webrtc/modules/audio_device/android/opensles_output.h @@ -41,6 +41,7 @@ class OpenSlesOutput : public PlayoutDelayProvider { static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* env, void* context); + static void ClearAndroidAudioDeviceObjects(); // Main initializaton and termination int32_t Init(); @@ -73,7 +74,6 @@ class OpenSlesOutput : public PlayoutDelayProvider { bool Playing() const { return playing_; } // Audio mixer initialization - int32_t SpeakerIsAvailable(bool& available); // NOLINT int32_t InitSpeaker(); bool SpeakerIsInitialized() const { return speaker_initialized_; } @@ -223,7 +223,7 @@ class OpenSlesOutput : public PlayoutDelayProvider { // Audio buffers AudioDeviceBuffer* audio_buffer_; scoped_ptr<FineAudioBuffer> fine_buffer_; - scoped_array<scoped_array<int8_t> > play_buf_; + scoped_ptr<scoped_ptr<int8_t[]>[]> play_buf_; // Index in |rec_buf_| pointing to the audio buffer that will be ready the // next time PlayerSimpleBufferQueueCallbackHandler is invoked. // Ready means buffer is ready to be played out to device. diff --git a/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.cc b/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.cc index d65ab9fbb63..73d4d61dd3f 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.cc +++ b/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.cc @@ -10,6 +10,8 @@ #include "webrtc/modules/audio_device/android/single_rw_fifo.h" +#include <assert.h> + static int UpdatePos(int pos, int capacity) { return (pos + 1) % capacity; } @@ -18,14 +20,16 @@ namespace webrtc { namespace subtle { -#if defined(__ARMEL__) +#if defined(__aarch64__) +// From http://http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_arm64_gcc.h +inline void MemoryBarrier() { + __asm__ __volatile__ ("dmb ish" ::: "memory"); +} + +#elif defined(__ARMEL__) // From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_arm_gcc.h -// Note that it is only the MemoryBarrier function that makes this class arm -// specific. Borrowing other MemoryBarrier implementations, this class could -// be extended to more platforms. inline void MemoryBarrier() { - // Note: This is a function call, which is also an implicit compiler - // barrier. + // Note: This is a function call, which is also an implicit compiler barrier. typedef void (*KernelMemoryBarrierFunc)(); ((KernelMemoryBarrierFunc)0xffff0fa0)(); } diff --git a/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.h b/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.h index a1fcfaab417..092b1d5e090 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.h +++ b/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo.h @@ -35,7 +35,7 @@ class SingleRwFifo { int capacity() const { return capacity_; } private: - scoped_array<int8_t*> queue_; + scoped_ptr<int8_t*[]> queue_; int capacity_; Atomic32 size_; diff --git a/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc b/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc index c722c2756cd..9925baaa88f 100644 --- a/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc @@ -90,7 +90,7 @@ class SingleRwFifoTest : public testing::Test { protected: SingleRwFifo fifo_; // Memory area for proper de-allocation. - scoped_array<int8_t> buffer_[kCapacity]; + scoped_ptr<int8_t[]> buffer_[kCapacity]; std::list<int8_t*> memory_queue_; int pushed_; diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi b/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi index 1e14747f3df..a64856b5da5 100644 --- a/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi +++ b/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi @@ -20,7 +20,7 @@ '.', '../interface', 'include', - 'dummy', # dummy audio device + 'dummy', # Contains dummy audio device implementations. ], 'direct_dependent_settings': { 'include_dirs': [ @@ -45,6 +45,8 @@ 'dummy/audio_device_dummy.h', 'dummy/audio_device_utility_dummy.cc', 'dummy/audio_device_utility_dummy.h', + 'dummy/file_audio_device.cc', + 'dummy/file_audio_device.h', ], 'conditions': [ ['OS=="linux"', { @@ -77,6 +79,13 @@ 'WEBRTC_DUMMY_AUDIO_BUILD', ], }], + ['build_with_chromium==0', { + 'sources': [ + # Don't link these into Chrome since they contain static data. + 'dummy/file_audio_device_factory.cc', + 'dummy/file_audio_device_factory.h', + ], + }], ['include_internal_audio_device==1', { 'sources': [ 'linux/alsasymboltable_linux.cc', @@ -246,66 +255,6 @@ }, ], }], - ['OS=="android"', { - 'targets': [ - { - 'target_name': 'libopensl-demo-jni', - 'type': 'loadable_module', - 'dependencies': [ - 'audio_device', - ], - 'sources': [ - 'android/test/jni/opensl_runner.cc', - 'android/test/fake_audio_device_buffer.cc', - ], - 'link_settings': { - 'libraries': [ - '-llog', - '-lOpenSLES', - ], - }, - }, - { - 'target_name': 'OpenSlDemo', - 'type': 'none', - 'dependencies': [ - 'libopensl-demo-jni', - '<(modules_java_gyp_path):*', - ], - 'actions': [ - { - # TODO(henrik): Convert building of the demo to a proper GYP - # target so this action is not needed once chromium's - # apk-building machinery can be used. (crbug.com/225101) - 'action_name': 'build_opensldemo_apk', - 'variables': { - 'android_opensl_demo_root': '<(webrtc_root)/modules/audio_device/android/test', - }, - 'inputs' : [ - '<(PRODUCT_DIR)/lib.java/audio_device_module_java.jar', - '<(PRODUCT_DIR)/libopensl-demo-jni.so', - '<!@(find <(android_opensl_demo_root)/src -name "*.java")', - '<!@(find <(android_opensl_demo_root)/res -name "*.xml")', - '<!@(find <(android_opensl_demo_root)/res -name "*.png")', - '<(android_opensl_demo_root)/AndroidManifest.xml', - '<(android_opensl_demo_root)/build.xml', - '<(android_opensl_demo_root)/project.properties', - ], - 'outputs': ['<(PRODUCT_DIR)/OpenSlDemo-debug.apk'], - 'action': ['bash', '-ec', - 'rm -f <(_outputs) && ' - 'mkdir -p <(android_opensl_demo_root)/libs/<(android_app_abi) && ' - '<(android_strip) -o <(android_opensl_demo_root)/libs/<(android_app_abi)/libopensl-demo-jni.so <(PRODUCT_DIR)/libopensl-demo-jni.so && ' - 'cp <(PRODUCT_DIR)/lib.java/audio_device_module_java.jar <(android_opensl_demo_root)/libs/ &&' - 'cd <(android_opensl_demo_root) && ' - 'ant debug && ' - 'cd - && ' - 'cp <(android_opensl_demo_root)/bin/OpenSlDemo-debug.apk <(_outputs)' - ], - }, - ], - }], - }], ['OS=="android" and enable_android_opensl==1', { 'targets': [ { diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_buffer.cc b/chromium/third_party/webrtc/modules/audio_device/audio_device_buffer.cc index db5cc322f98..42fdaad22cb 100644 --- a/chromium/third_party/webrtc/modules/audio_device/audio_device_buffer.cc +++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_buffer.cc @@ -548,13 +548,16 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples) if (_ptrCbAudioTransport) { uint32_t res(0); - + int64_t elapsed_time_ms = -1; + int64_t ntp_time_ms = -1; res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples, playBytesPerSample, playChannels, playSampleRate, &_playBuffer[0], - nSamplesOut); + nSamplesOut, + &elapsed_time_ms, + &ntp_time_ms); if (res != 0) { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed"); diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_generic.h b/chromium/third_party/webrtc/modules/audio_device/audio_device_generic.h index 8038028029b..a4c320eaf75 100644 --- a/chromium/third_party/webrtc/modules/audio_device/audio_device_generic.h +++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_generic.h @@ -76,10 +76,8 @@ class AudioDeviceGeneric uint16_t& volumeRight) const = 0; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available) = 0; virtual int32_t InitSpeaker() = 0; virtual bool SpeakerIsInitialized() const = 0; - virtual int32_t MicrophoneIsAvailable(bool& available) = 0; virtual int32_t InitMicrophone() = 0; virtual bool MicrophoneIsInitialized() const = 0; @@ -137,7 +135,7 @@ class AudioDeviceGeneric // CPU load virtual int32_t CPULoad(uint16_t& load) const = 0; - + // Native sample rate controls (samples/sec) virtual int32_t SetRecordingSampleRate( const uint32_t samplesPerSec); @@ -147,7 +145,7 @@ class AudioDeviceGeneric // Speaker audio routing (for mobile devices) virtual int32_t SetLoudspeakerStatus(bool enable); virtual int32_t GetLoudspeakerStatus(bool& enable) const; - + // Reset Audio Device (for mobile devices) virtual int32_t ResetAudioDevice(); diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc index 05201c0e2a5..58411e3b94d 100644 --- a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc @@ -45,8 +45,14 @@ #include "audio_device_utility_mac.h" #include "audio_device_mac.h" #endif + +#if defined(WEBRTC_DUMMY_FILE_DEVICES) +#include "webrtc/modules/audio_device/dummy/file_audio_device_factory.h" +#endif + #include "webrtc/modules/audio_device/dummy/audio_device_dummy.h" #include "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h" +#include "webrtc/modules/audio_device/dummy/file_audio_device.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" @@ -203,6 +209,14 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() { ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id()); } +#elif defined(WEBRTC_DUMMY_FILE_DEVICES) + ptrAudioDevice = FileAudioDeviceFactory::CreateFileAudioDevice(Id()); + WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, + "Will use file-playing dummy device."); + if (ptrAudioDevice != NULL) + { + ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id()); + } #else const AudioLayer audioLayer(PlatformAudioLayer()); @@ -641,27 +655,6 @@ bool AudioDeviceModuleImpl::Initialized() const } // ---------------------------------------------------------------------------- -// SpeakerIsAvailable -// ---------------------------------------------------------------------------- - -int32_t AudioDeviceModuleImpl::SpeakerIsAvailable(bool* available) -{ - CHECK_INITIALIZED(); - - bool isAvailable(0); - - if (_ptrAudioDevice->SpeakerIsAvailable(isAvailable) == -1) - { - return -1; - } - - *available = isAvailable; - - WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", available); - return (0); -} - -// ---------------------------------------------------------------------------- // InitSpeaker // ---------------------------------------------------------------------------- @@ -672,27 +665,6 @@ int32_t AudioDeviceModuleImpl::InitSpeaker() } // ---------------------------------------------------------------------------- -// MicrophoneIsAvailable -// ---------------------------------------------------------------------------- - -int32_t AudioDeviceModuleImpl::MicrophoneIsAvailable(bool* available) -{ - CHECK_INITIALIZED(); - - bool isAvailable(0); - - if (_ptrAudioDevice->MicrophoneIsAvailable(isAvailable) == -1) - { - return -1; - } - - *available = isAvailable; - - WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available); - return (0); -} - -// ---------------------------------------------------------------------------- // InitMicrophone // ---------------------------------------------------------------------------- @@ -1750,8 +1722,6 @@ int32_t AudioDeviceModuleImpl::StopRawOutputFileRecording() CHECK_INITIALIZED(); return (_audioDeviceBuffer.StopOutputFileRecording()); - - return 0; } // ---------------------------------------------------------------------------- diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h index 26bd54399b2..a545d580f4b 100644 --- a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h +++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h @@ -115,10 +115,8 @@ public: uint16_t* volumeRight) const OVERRIDE; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool* available) OVERRIDE; virtual int32_t InitSpeaker() OVERRIDE; virtual bool SpeakerIsInitialized() const OVERRIDE; - virtual int32_t MicrophoneIsAvailable(bool* available) OVERRIDE; virtual int32_t InitMicrophone() OVERRIDE; virtual bool MicrophoneIsInitialized() const OVERRIDE; diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_tests.isolate b/chromium/third_party/webrtc/modules/audio_device/audio_device_tests.isolate index 69e877c14f1..ebe8bfb40b9 100644 --- a/chromium/third_party/webrtc/modules/audio_device/audio_device_tests.isolate +++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_tests.isolate @@ -8,27 +8,25 @@ { 'conditions': [ ['OS=="android"', { - # When doing Android builds, the WebRTC code is put in third_party/webrtc - # of a Chromium checkout, this is one level above the standalone build. 'variables': { 'isolate_dependency_untracked': [ - '../../../../data/', - '../../../../resources/', + '<(DEPTH)/data/', + '<(DEPTH)/resources/', ], }, }], ['OS=="linux" or OS=="mac" or OS=="win"', { 'variables': { 'command': [ - '../../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/audio_device_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_tracked': [ - '../../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/audio_device_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_untracked': [ - '../../../tools/swarming_client/', + '<(DEPTH)/tools/swarming_client/', ], }, }], diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.cc b/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.cc index 379b25766d8..92199f6d295 100644 --- a/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.cc +++ b/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.cc @@ -91,14 +91,10 @@ int32_t AudioDeviceDummy::WaveOutVolume(uint16_t& volumeLeft, return -1; } -int32_t AudioDeviceDummy::SpeakerIsAvailable(bool& available) { return -1; } - int32_t AudioDeviceDummy::InitSpeaker() { return -1; } bool AudioDeviceDummy::SpeakerIsInitialized() const { return false; } -int32_t AudioDeviceDummy::MicrophoneIsAvailable(bool& available) { return -1; } - int32_t AudioDeviceDummy::InitMicrophone() { return -1; } bool AudioDeviceDummy::MicrophoneIsInitialized() const { return false; } diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.h b/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.h index 47d7aab79e4..41e52e95443 100644 --- a/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.h +++ b/chromium/third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.h @@ -76,10 +76,8 @@ class AudioDeviceDummy : public AudioDeviceGeneric { uint16_t& volumeRight) const OVERRIDE; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available) OVERRIDE; virtual int32_t InitSpeaker() OVERRIDE; virtual bool SpeakerIsInitialized() const OVERRIDE; - virtual int32_t MicrophoneIsAvailable(bool& available) OVERRIDE; virtual int32_t InitMicrophone() OVERRIDE; virtual bool MicrophoneIsInitialized() const OVERRIDE; diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device.cc b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device.cc new file mode 100644 index 00000000000..e7771c66de8 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device.cc @@ -0,0 +1,586 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include <iostream> +#include "webrtc/modules/audio_device/dummy/file_audio_device.h" +#include "webrtc/system_wrappers/interface/sleep.h" +#include "webrtc/system_wrappers/interface/thread_wrapper.h" + +namespace webrtc { + +int kRecordingFixedSampleRate = 48000; +int kRecordingNumChannels = 2; +int kPlayoutFixedSampleRate = 48000; +int kPlayoutNumChannels = 2; +int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100 + * kPlayoutNumChannels * 2; +int kRecordingBufferSize = kRecordingFixedSampleRate / 100 + * kRecordingNumChannels * 2; + +FileAudioDevice::FileAudioDevice(const int32_t id, + const char* inputFilename, + const char* outputFile): + _ptrAudioBuffer(NULL), + _recordingBuffer(NULL), + _playoutBuffer(NULL), + _recordingFramesLeft(0), + _playoutFramesLeft(0), + _critSect(*CriticalSectionWrapper::CreateCriticalSection()), + _recordingBufferSizeIn10MS(0), + _recordingFramesIn10MS(0), + _playoutFramesIn10MS(0), + _ptrThreadRec(NULL), + _ptrThreadPlay(NULL), + _recThreadID(0), + _playThreadID(0), + _playing(false), + _recording(false), + _lastCallPlayoutMillis(0), + _lastCallRecordMillis(0), + _outputFile(*FileWrapper::Create()), + _inputFile(*FileWrapper::Create()), + _outputFilename(outputFile), + _inputFilename(inputFilename), + _clock(Clock::GetRealTimeClock()) { +} + +FileAudioDevice::~FileAudioDevice() { + _outputFile.Flush(); + _outputFile.CloseFile(); + delete &_outputFile; + _inputFile.Flush(); + _inputFile.CloseFile(); + delete &_inputFile; +} + +int32_t FileAudioDevice::ActiveAudioLayer( + AudioDeviceModule::AudioLayer& audioLayer) const { + return -1; +} + +int32_t FileAudioDevice::Init() { return 0; } + +int32_t FileAudioDevice::Terminate() { return 0; } + +bool FileAudioDevice::Initialized() const { return true; } + +int16_t FileAudioDevice::PlayoutDevices() { + return 1; +} + +int16_t FileAudioDevice::RecordingDevices() { + return 1; +} + +int32_t FileAudioDevice::PlayoutDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) { + const char* kName = "dummy_device"; + const char* kGuid = "dummy_device_unique_id"; + if (index < 1) { + memset(name, 0, kAdmMaxDeviceNameSize); + memset(guid, 0, kAdmMaxGuidSize); + memcpy(name, kName, strlen(kName)); + memcpy(guid, kGuid, strlen(guid)); + return 0; + } + return -1; +} + +int32_t FileAudioDevice::RecordingDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) { + const char* kName = "dummy_device"; + const char* kGuid = "dummy_device_unique_id"; + if (index < 1) { + memset(name, 0, kAdmMaxDeviceNameSize); + memset(guid, 0, kAdmMaxGuidSize); + memcpy(name, kName, strlen(kName)); + memcpy(guid, kGuid, strlen(guid)); + return 0; + } + return -1; +} + +int32_t FileAudioDevice::SetPlayoutDevice(uint16_t index) { + if (index == 0) { + _playout_index = index; + return 0; + } + return -1; +} + +int32_t FileAudioDevice::SetPlayoutDevice( + AudioDeviceModule::WindowsDeviceType device) { + return -1; +} + +int32_t FileAudioDevice::SetRecordingDevice(uint16_t index) { + if (index == 0) { + _record_index = index; + return _record_index; + } + return -1; +} + +int32_t FileAudioDevice::SetRecordingDevice( + AudioDeviceModule::WindowsDeviceType device) { + return -1; +} + +int32_t FileAudioDevice::PlayoutIsAvailable(bool& available) { + if (_playout_index == 0) { + available = true; + return _playout_index; + } + available = false; + return -1; +} + +int32_t FileAudioDevice::InitPlayout() { + if (_ptrAudioBuffer) + { + // Update webrtc audio buffer with the selected parameters + _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate); + _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels); + } + return 0; +} + +bool FileAudioDevice::PlayoutIsInitialized() const { + return true; +} + +int32_t FileAudioDevice::RecordingIsAvailable(bool& available) { + if (_record_index == 0) { + available = true; + return _record_index; + } + available = false; + return -1; +} + +int32_t FileAudioDevice::InitRecording() { + CriticalSectionScoped lock(&_critSect); + + if (_recording) { + return -1; + } + + _recordingFramesIn10MS = kRecordingFixedSampleRate/100; + + if (_ptrAudioBuffer) { + _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate); + _ptrAudioBuffer->SetRecordingChannels(kRecordingNumChannels); + } + return 0; +} + +bool FileAudioDevice::RecordingIsInitialized() const { + return true; +} + +int32_t FileAudioDevice::StartPlayout() { + if (_playing) + { + return 0; + } + + _playing = true; + _playoutFramesLeft = 0; + + if (!_playoutBuffer) + _playoutBuffer = new int8_t[2 * + kPlayoutNumChannels * + kPlayoutFixedSampleRate/100]; + if (!_playoutBuffer) + { + _playing = false; + return -1; + } + + // PLAYOUT + const char* threadName = "webrtc_audio_module_play_thread"; + _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, + this, + kRealtimePriority, + threadName); + if (_ptrThreadPlay == NULL) + { + _playing = false; + delete [] _playoutBuffer; + _playoutBuffer = NULL; + return -1; + } + + if (_outputFile.OpenFile(_outputFilename.c_str(), + false, false, false) == -1) { + printf("Failed to open playout file %s!", _outputFilename.c_str()); + _playing = false; + delete [] _playoutBuffer; + _playoutBuffer = NULL; + return -1; + } + + unsigned int threadID(0); + if (!_ptrThreadPlay->Start(threadID)) + { + _playing = false; + delete _ptrThreadPlay; + _ptrThreadPlay = NULL; + delete [] _playoutBuffer; + _playoutBuffer = NULL; + return -1; + } + _playThreadID = threadID; + + return 0; +} + +int32_t FileAudioDevice::StopPlayout() { + { + CriticalSectionScoped lock(&_critSect); + _playing = false; + } + + // stop playout thread first + if (_ptrThreadPlay && !_ptrThreadPlay->Stop()) + { + return -1; + } + else { + delete _ptrThreadPlay; + _ptrThreadPlay = NULL; + } + + CriticalSectionScoped lock(&_critSect); + + _playoutFramesLeft = 0; + delete [] _playoutBuffer; + _playoutBuffer = NULL; + _outputFile.Flush(); + _outputFile.CloseFile(); + return 0; +} + +bool FileAudioDevice::Playing() const { + return true; +} + +int32_t FileAudioDevice::StartRecording() { + _recording = true; + + // Make sure we only create the buffer once. + _recordingBufferSizeIn10MS = _recordingFramesIn10MS * + kRecordingNumChannels * + 2; + if (!_recordingBuffer) { + _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS]; + } + + if (_inputFile.OpenFile(_inputFilename.c_str(), true, + true, false) == -1) { + printf("Failed to open audio input file %s!\n", + _inputFilename.c_str()); + _recording = false; + delete[] _recordingBuffer; + _recordingBuffer = NULL; + return -1; + } + + const char* threadName = "webrtc_audio_module_capture_thread"; + _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, + this, + kRealtimePriority, + threadName); + if (_ptrThreadRec == NULL) + { + _recording = false; + delete [] _recordingBuffer; + _recordingBuffer = NULL; + return -1; + } + + unsigned int threadID(0); + if (!_ptrThreadRec->Start(threadID)) + { + _recording = false; + delete _ptrThreadRec; + _ptrThreadRec = NULL; + delete [] _recordingBuffer; + _recordingBuffer = NULL; + return -1; + } + _recThreadID = threadID; + + return 0; +} + + +int32_t FileAudioDevice::StopRecording() { + { + CriticalSectionScoped lock(&_critSect); + _recording = false; + } + + if (_ptrThreadRec && !_ptrThreadRec->Stop()) + { + return -1; + } + else { + delete _ptrThreadRec; + _ptrThreadRec = NULL; + } + + CriticalSectionScoped lock(&_critSect); + _recordingFramesLeft = 0; + if (_recordingBuffer) + { + delete [] _recordingBuffer; + _recordingBuffer = NULL; + } + return 0; +} + +bool FileAudioDevice::Recording() const { + return _recording; +} + +int32_t FileAudioDevice::SetAGC(bool enable) { return -1; } + +bool FileAudioDevice::AGC() const { return false; } + +int32_t FileAudioDevice::SetWaveOutVolume(uint16_t volumeLeft, + uint16_t volumeRight) { + return -1; +} + +int32_t FileAudioDevice::WaveOutVolume(uint16_t& volumeLeft, + uint16_t& volumeRight) const { + return -1; +} + +int32_t FileAudioDevice::InitSpeaker() { return -1; } + +bool FileAudioDevice::SpeakerIsInitialized() const { return false; } + +int32_t FileAudioDevice::InitMicrophone() { return 0; } + +bool FileAudioDevice::MicrophoneIsInitialized() const { return true; } + +int32_t FileAudioDevice::SpeakerVolumeIsAvailable(bool& available) { + return -1; +} + +int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) { return -1; } + +int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const { return -1; } + +int32_t FileAudioDevice::MaxSpeakerVolume(uint32_t& maxVolume) const { + return -1; +} + +int32_t FileAudioDevice::MinSpeakerVolume(uint32_t& minVolume) const { + return -1; +} + +int32_t FileAudioDevice::SpeakerVolumeStepSize(uint16_t& stepSize) const { + return -1; +} + +int32_t FileAudioDevice::MicrophoneVolumeIsAvailable(bool& available) { + return -1; +} + +int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) { return -1; } + +int32_t FileAudioDevice::MicrophoneVolume(uint32_t& volume) const { + return -1; +} + +int32_t FileAudioDevice::MaxMicrophoneVolume(uint32_t& maxVolume) const { + return -1; +} + +int32_t FileAudioDevice::MinMicrophoneVolume(uint32_t& minVolume) const { + return -1; +} + +int32_t FileAudioDevice::MicrophoneVolumeStepSize(uint16_t& stepSize) const { + return -1; +} + +int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) { return -1; } + +int32_t FileAudioDevice::SetSpeakerMute(bool enable) { return -1; } + +int32_t FileAudioDevice::SpeakerMute(bool& enabled) const { return -1; } + +int32_t FileAudioDevice::MicrophoneMuteIsAvailable(bool& available) { + return -1; +} + +int32_t FileAudioDevice::SetMicrophoneMute(bool enable) { return -1; } + +int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const { return -1; } + +int32_t FileAudioDevice::MicrophoneBoostIsAvailable(bool& available) { + return -1; +} + +int32_t FileAudioDevice::SetMicrophoneBoost(bool enable) { return -1; } + +int32_t FileAudioDevice::MicrophoneBoost(bool& enabled) const { return -1; } + +int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) { + available = true; + return 0; +} +int32_t FileAudioDevice::SetStereoPlayout(bool enable) { + return 0; +} + +int32_t FileAudioDevice::StereoPlayout(bool& enabled) const { + enabled = true; + return 0; +} + +int32_t FileAudioDevice::StereoRecordingIsAvailable(bool& available) { + available = true; + return 0; +} + +int32_t FileAudioDevice::SetStereoRecording(bool enable) { + return 0; +} + +int32_t FileAudioDevice::StereoRecording(bool& enabled) const { + enabled = true; + return 0; +} + +int32_t FileAudioDevice::SetPlayoutBuffer( + const AudioDeviceModule::BufferType type, + uint16_t sizeMS) { + return 0; +} + +int32_t FileAudioDevice::PlayoutBuffer(AudioDeviceModule::BufferType& type, + uint16_t& sizeMS) const { + type = _playBufType; + return 0; +} + +int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const { + return 0; +} + +int32_t FileAudioDevice::RecordingDelay(uint16_t& delayMS) const { return -1; } + +int32_t FileAudioDevice::CPULoad(uint16_t& load) const { return -1; } + +bool FileAudioDevice::PlayoutWarning() const { return false; } + +bool FileAudioDevice::PlayoutError() const { return false; } + +bool FileAudioDevice::RecordingWarning() const { return false; } + +bool FileAudioDevice::RecordingError() const { return false; } + +void FileAudioDevice::ClearPlayoutWarning() {} + +void FileAudioDevice::ClearPlayoutError() {} + +void FileAudioDevice::ClearRecordingWarning() {} + +void FileAudioDevice::ClearRecordingError() {} + +void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { + CriticalSectionScoped lock(&_critSect); + + _ptrAudioBuffer = audioBuffer; + + // Inform the AudioBuffer about default settings for this implementation. + // Set all values to zero here since the actual settings will be done by + // InitPlayout and InitRecording later. + _ptrAudioBuffer->SetRecordingSampleRate(0); + _ptrAudioBuffer->SetPlayoutSampleRate(0); + _ptrAudioBuffer->SetRecordingChannels(0); + _ptrAudioBuffer->SetPlayoutChannels(0); +} + +bool FileAudioDevice::PlayThreadFunc(void* pThis) +{ + return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess()); +} + +bool FileAudioDevice::RecThreadFunc(void* pThis) +{ + return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess()); +} + +bool FileAudioDevice::PlayThreadProcess() +{ + if(!_playing) + return false; + + uint64_t currentTime = _clock->CurrentNtpInMilliseconds(); + _critSect.Enter(); + + if (_lastCallPlayoutMillis == 0 || + currentTime - _lastCallPlayoutMillis >= 10) + { + _critSect.Leave(); + _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS); + _critSect.Enter(); + + _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer); + assert(_playoutFramesLeft == _playoutFramesIn10MS); + if (_outputFile.Open()) { + _outputFile.Write(_playoutBuffer, kPlayoutBufferSize); + _outputFile.Flush(); + } + _lastCallPlayoutMillis = currentTime; + } + _playoutFramesLeft = 0; + _critSect.Leave(); + SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); + return true; +} + +bool FileAudioDevice::RecThreadProcess() +{ + if (!_recording) + return false; + + uint64_t currentTime = _clock->CurrentNtpInMilliseconds(); + _critSect.Enter(); + + if (_lastCallRecordMillis == 0 || + currentTime - _lastCallRecordMillis >= 10) { + if (_inputFile.Open()) { + if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) { + _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer, + _recordingFramesIn10MS); + } else { + _inputFile.Rewind(); + } + _lastCallRecordMillis = currentTime; + _critSect.Leave(); + _ptrAudioBuffer->DeliverRecordedData(); + _critSect.Enter(); + } + } + + _critSect.Leave(); + SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); + return true; +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device.h b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device.h new file mode 100644 index 00000000000..6f417eb2e0b --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device.h @@ -0,0 +1,202 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H +#define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H + +#include <stdio.h> + +#include <string> + +#include "webrtc/modules/audio_device/audio_device_generic.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/file_wrapper.h" +#include "webrtc/system_wrappers/interface/clock.h" + +namespace webrtc { +class EventWrapper; +class ThreadWrapper; + +// This is a fake audio device which plays audio from a file as its microphone +// and plays out into a file. +class FileAudioDevice : public AudioDeviceGeneric { + public: + // Constructs a file audio device with |id|. It will read audio from + // |inputFilename| and record output audio to |outputFilename|. + // + // The input file should be a readable 48k stereo raw file, and the output + // file should point to a writable location. The output format will also be + // 48k stereo raw audio. + FileAudioDevice(const int32_t id, + const char* inputFilename, + const char* outputFilename); + virtual ~FileAudioDevice(); + + // Retrieve the currently utilized audio layer + virtual int32_t ActiveAudioLayer( + AudioDeviceModule::AudioLayer& audioLayer) const OVERRIDE; + + // Main initializaton and termination + virtual int32_t Init() OVERRIDE; + virtual int32_t Terminate() OVERRIDE; + virtual bool Initialized() const OVERRIDE; + + // Device enumeration + virtual int16_t PlayoutDevices() OVERRIDE; + virtual int16_t RecordingDevices() OVERRIDE; + virtual int32_t PlayoutDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) OVERRIDE; + virtual int32_t RecordingDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) OVERRIDE; + + // Device selection + virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE; + virtual int32_t SetPlayoutDevice( + AudioDeviceModule::WindowsDeviceType device) OVERRIDE; + virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE; + virtual int32_t SetRecordingDevice( + AudioDeviceModule::WindowsDeviceType device) OVERRIDE; + + // Audio transport initialization + virtual int32_t PlayoutIsAvailable(bool& available) OVERRIDE; + virtual int32_t InitPlayout() OVERRIDE; + virtual bool PlayoutIsInitialized() const OVERRIDE; + virtual int32_t RecordingIsAvailable(bool& available) OVERRIDE; + virtual int32_t InitRecording() OVERRIDE; + virtual bool RecordingIsInitialized() const OVERRIDE; + + // Audio transport control + virtual int32_t StartPlayout() OVERRIDE; + virtual int32_t StopPlayout() OVERRIDE; + virtual bool Playing() const OVERRIDE; + virtual int32_t StartRecording() OVERRIDE; + virtual int32_t StopRecording() OVERRIDE; + virtual bool Recording() const OVERRIDE; + + // Microphone Automatic Gain Control (AGC) + virtual int32_t SetAGC(bool enable) OVERRIDE; + virtual bool AGC() const OVERRIDE; + + // Volume control based on the Windows Wave API (Windows only) + virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, + uint16_t volumeRight) OVERRIDE; + virtual int32_t WaveOutVolume(uint16_t& volumeLeft, + uint16_t& volumeRight) const OVERRIDE; + + // Audio mixer initialization + virtual int32_t InitSpeaker() OVERRIDE; + virtual bool SpeakerIsInitialized() const OVERRIDE; + virtual int32_t InitMicrophone() OVERRIDE; + virtual bool MicrophoneIsInitialized() const OVERRIDE; + + // Speaker volume controls + virtual int32_t SpeakerVolumeIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE; + virtual int32_t SpeakerVolume(uint32_t& volume) const OVERRIDE; + virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const OVERRIDE; + virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const OVERRIDE; + virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const OVERRIDE; + + // Microphone volume controls + virtual int32_t MicrophoneVolumeIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE; + virtual int32_t MicrophoneVolume(uint32_t& volume) const OVERRIDE; + virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const OVERRIDE; + virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const OVERRIDE; + virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const OVERRIDE; + + // Speaker mute control + virtual int32_t SpeakerMuteIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetSpeakerMute(bool enable) OVERRIDE; + virtual int32_t SpeakerMute(bool& enabled) const OVERRIDE; + + // Microphone mute control + virtual int32_t MicrophoneMuteIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE; + virtual int32_t MicrophoneMute(bool& enabled) const OVERRIDE; + + // Microphone boost control + virtual int32_t MicrophoneBoostIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE; + virtual int32_t MicrophoneBoost(bool& enabled) const OVERRIDE; + + // Stereo support + virtual int32_t StereoPlayoutIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetStereoPlayout(bool enable) OVERRIDE; + virtual int32_t StereoPlayout(bool& enabled) const OVERRIDE; + virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE; + virtual int32_t SetStereoRecording(bool enable) OVERRIDE; + virtual int32_t StereoRecording(bool& enabled) const OVERRIDE; + + // Delay information and control + virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type, + uint16_t sizeMS) OVERRIDE; + virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, + uint16_t& sizeMS) const OVERRIDE; + virtual int32_t PlayoutDelay(uint16_t& delayMS) const OVERRIDE; + virtual int32_t RecordingDelay(uint16_t& delayMS) const OVERRIDE; + + // CPU load + virtual int32_t CPULoad(uint16_t& load) const OVERRIDE; + + virtual bool PlayoutWarning() const OVERRIDE; + virtual bool PlayoutError() const OVERRIDE; + virtual bool RecordingWarning() const OVERRIDE; + virtual bool RecordingError() const OVERRIDE; + virtual void ClearPlayoutWarning() OVERRIDE; + virtual void ClearPlayoutError() OVERRIDE; + virtual void ClearRecordingWarning() OVERRIDE; + virtual void ClearRecordingError() OVERRIDE; + + virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE; + + private: + static bool RecThreadFunc(void*); + static bool PlayThreadFunc(void*); + bool RecThreadProcess(); + bool PlayThreadProcess(); + + int32_t _playout_index; + int32_t _record_index; + AudioDeviceModule::BufferType _playBufType; + AudioDeviceBuffer* _ptrAudioBuffer; + int8_t* _recordingBuffer; // In bytes. + int8_t* _playoutBuffer; // In bytes. + uint32_t _recordingFramesLeft; + uint32_t _playoutFramesLeft; + CriticalSectionWrapper& _critSect; + + uint32_t _recordingBufferSizeIn10MS; + uint32_t _recordingFramesIn10MS; + uint32_t _playoutFramesIn10MS; + + ThreadWrapper* _ptrThreadRec; + ThreadWrapper* _ptrThreadPlay; + uint32_t _recThreadID; + uint32_t _playThreadID; + + bool _playing; + bool _recording; + uint64_t _lastCallPlayoutMillis; + uint64_t _lastCallRecordMillis; + + FileWrapper& _outputFile; + FileWrapper& _inputFile; + std::string _outputFilename; + std::string _inputFilename; + + Clock* _clock; +}; + +} // namespace webrtc + +#endif // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc new file mode 100644 index 00000000000..db35bf111b1 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc @@ -0,0 +1,43 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_device/dummy/file_audio_device_factory.h" + +#include <cstring> + +#include "webrtc/modules/audio_device/dummy/file_audio_device.h" + +namespace webrtc { + +char FileAudioDeviceFactory::_inputAudioFilename[MAX_FILENAME_LEN] = ""; +char FileAudioDeviceFactory::_outputAudioFilename[MAX_FILENAME_LEN] = ""; + +FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice( + const int32_t id) { + // Bail out here if the files aren't set. + if (strlen(_inputAudioFilename) == 0 || strlen(_outputAudioFilename) == 0) { + printf("Was compiled with WEBRTC_DUMMY_AUDIO_PLAY_STATIC_FILE " + "but did not set input/output files to use. Bailing out.\n"); + exit(1); + } + return new FileAudioDevice(id, _inputAudioFilename, _outputAudioFilename); +} + +void FileAudioDeviceFactory::SetFilenamesToUse( + const char* inputAudioFilename, const char* outputAudioFilename) { + assert(strlen(inputAudioFilename) < MAX_FILENAME_LEN && + strlen(outputAudioFilename) < MAX_FILENAME_LEN); + + // Copy the strings since we don't know the lifetime of the input pointers. + strncpy(_inputAudioFilename, inputAudioFilename, MAX_FILENAME_LEN); + strncpy(_outputAudioFilename, outputAudioFilename, MAX_FILENAME_LEN); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.h b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.h new file mode 100644 index 00000000000..9975d7b90e0 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.h @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H +#define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H + +#include "webrtc/common_types.h" + +namespace webrtc { + +class FileAudioDevice; + +// This class is used by audio_device_impl.cc when WebRTC is compiled with +// WEBRTC_DUMMY_FILE_DEVICES. The application must include this file and set the +// filenames to use before the audio device module is initialized. This is +// intended for test tools which use the audio device module. +class FileAudioDeviceFactory { + public: + static FileAudioDevice* CreateFileAudioDevice(const int32_t id); + + // The input file must be a readable 48k stereo raw file. The output + // file must be writable. The strings will be copied. + static void SetFilenamesToUse(const char* inputAudioFilename, + const char* outputAudioFilename); + + private: + static const uint32_t MAX_FILENAME_LEN = 256; + static char _inputAudioFilename[MAX_FILENAME_LEN]; + static char _outputAudioFilename[MAX_FILENAME_LEN]; +}; + +} // namespace webrtc + +#endif // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H diff --git a/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h b/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h index eb403aa9b98..ec40274de49 100644 --- a/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h +++ b/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h @@ -107,10 +107,8 @@ class AudioDeviceModule : public RefCountedModule { uint16_t* volumeRight) const = 0; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool* available) = 0; virtual int32_t InitSpeaker() = 0; virtual bool SpeakerIsInitialized() const = 0; - virtual int32_t MicrophoneIsAvailable(bool* available) = 0; virtual int32_t InitMicrophone() = 0; virtual bool MicrophoneIsInitialized() const = 0; diff --git a/chromium/third_party/webrtc/modules/audio_device/include/audio_device_defines.h b/chromium/third_party/webrtc/modules/audio_device/include/audio_device_defines.h index c37c4b13955..56a584ef9ea 100644 --- a/chromium/third_party/webrtc/modules/audio_device/include/audio_device_defines.h +++ b/chromium/third_party/webrtc/modules/audio_device/include/audio_device_defines.h @@ -63,14 +63,16 @@ public: const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, - uint32_t& newMicLevel) = 0; + uint32_t& newMicLevel) = 0; virtual int32_t NeedMorePlayData(const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t samplesPerSec, void* audioSamples, - uint32_t& nSamplesOut) = 0; + uint32_t& nSamplesOut, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms) = 0; // Method to pass captured data directly and unmixed to network channels. // |channel_ids| contains a list of VoE channels which are the @@ -85,8 +87,8 @@ public: // will be ignored. // The return value is the new microphone volume, in the range of |0, 255]. // When the volume does not need to be updated, it returns 0. - // TODO(xians): Make the interface pure virtual after libjingle has its - // implementation. + // TODO(xians): Remove this interface after Chrome and Libjingle switches + // to OnData(). virtual int OnDataAvailable(const int voe_channels[], int number_of_voe_channels, const int16_t* audio_data, @@ -98,6 +100,37 @@ public: bool key_pressed, bool need_audio_processing) { return 0; } + // Method to pass the captured audio data to the specific VoE channel. + // |voe_channel| is the id of the VoE channel which is the sink to the + // capture data. + // TODO(xians): Remove this interface after Libjingle switches to + // PushCaptureData(). + virtual void OnData(int voe_channel, const void* audio_data, + int bits_per_sample, int sample_rate, + int number_of_channels, + int number_of_frames) {} + + // Method to push the captured audio data to the specific VoE channel. + // The data will not undergo audio processing. + // |voe_channel| is the id of the VoE channel which is the sink to the + // capture data. + // TODO(xians): Make the interface pure virtual after Libjingle + // has its implementation. + virtual void PushCaptureData(int voe_channel, const void* audio_data, + int bits_per_sample, int sample_rate, + int number_of_channels, + int number_of_frames) {} + + // Method to pull mixed render audio data from all active VoE channels. + // The data will not be passed as reference for audio processing internally. + // TODO(xians): Support getting the unmixed render data from specific VoE + // channel. + virtual void PullRenderData(int bits_per_sample, int sample_rate, + int number_of_channels, int number_of_frames, + void* audio_data, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms) {} + protected: virtual ~AudioTransport() {} }; diff --git a/chromium/third_party/webrtc/modules/audio_device/include/fake_audio_device.h b/chromium/third_party/webrtc/modules/audio_device/include/fake_audio_device.h index 0248317550d..5cdf54fe8cd 100644 --- a/chromium/third_party/webrtc/modules/audio_device/include/fake_audio_device.h +++ b/chromium/third_party/webrtc/modules/audio_device/include/fake_audio_device.h @@ -25,19 +25,11 @@ class FakeAudioDeviceModule : public AudioDeviceModule { return 0; } virtual int32_t Init() { return 0; } - virtual int32_t SpeakerIsAvailable(bool* available) { - *available = true; - return 0; - } virtual int32_t InitSpeaker() { return 0; } virtual int32_t SetPlayoutDevice(uint16_t index) { return 0; } virtual int32_t SetPlayoutDevice(WindowsDeviceType device) { return 0; } virtual int32_t SetStereoPlayout(bool enable) { return 0; } virtual int32_t StopPlayout() { return 0; } - virtual int32_t MicrophoneIsAvailable(bool* available) { - *available = true; - return 0; - } virtual int32_t InitMicrophone() { return 0; } virtual int32_t SetRecordingDevice(uint16_t index) { return 0; } virtual int32_t SetRecordingDevice(WindowsDeviceType device) { return 0; } diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.cc b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.cc index bad3915f14f..7a7189a2b7f 100644 --- a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.cc +++ b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.cc @@ -175,15 +175,6 @@ bool AudioDeviceIPhone::Initialized() const { return (_initialized); } -int32_t AudioDeviceIPhone::SpeakerIsAvailable(bool& available) { - WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "%s", __FUNCTION__); - - // speaker is always available in IOS - available = true; - return 0; -} - int32_t AudioDeviceIPhone::InitSpeaker() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -214,30 +205,6 @@ int32_t AudioDeviceIPhone::InitSpeaker() { return 0; } -int32_t AudioDeviceIPhone::MicrophoneIsAvailable(bool& available) { - WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, - "%s", __FUNCTION__); - - available = false; - - OSStatus result = -1; - UInt32 channel = 0; - UInt32 size = sizeof(channel); - result = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable, - &size, &channel); - if (channel != 0) { - // API is not supported on this platform, we return available = true - WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, - _id, " API call not supported on this version"); - available = true; - return 0; - } - - available = (channel == 0) ? false : true; - - return 0; -} - int32_t AudioDeviceIPhone::InitMicrophone() { WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__); @@ -1332,7 +1299,7 @@ int32_t AudioDeviceIPhone::InitPlayOrRecord() { // todo: Add 48 kHz (increase buffer sizes). Other fs? if ((playoutDesc.mSampleRate > 44090.0) && (playoutDesc.mSampleRate < 44110.0)) { - _adbSampFreq = 44000; + _adbSampFreq = 44100; } else if ((playoutDesc.mSampleRate > 15990.0) && (playoutDesc.mSampleRate < 16010.0)) { _adbSampFreq = 16000; @@ -1716,7 +1683,10 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() { if (_playoutDelayMeasurementCounter >= 100) { // Update HW and OS delay every second, unlikely to change - _playoutDelay = 0; + // Since this is eventually rounded to integral ms, add 0.5ms + // here to get round-to-nearest-int behavior instead of + // truncation. + float totalDelaySeconds = 0.0005; // HW output latency Float32 f32(0); @@ -1727,7 +1697,8 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error HW latency (result=%d)", result); } - _playoutDelay += static_cast<int>(f32 * 1000000); + assert(f32 >= 0); + totalDelaySeconds += f32; // HW buffer duration f32 = 0; @@ -1737,7 +1708,8 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error HW buffer duration (result=%d)", result); } - _playoutDelay += static_cast<int>(f32 * 1000000); + assert(f32 >= 0); + totalDelaySeconds += f32; // AU latency Float64 f64(0); @@ -1748,16 +1720,17 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error AU latency (result=%d)", result); } - _playoutDelay += static_cast<int>(f64 * 1000000); + assert(f64 >= 0); + totalDelaySeconds += f64; // To ms - _playoutDelay = (_playoutDelay - 500) / 1000; + _playoutDelay = static_cast<uint32_t>(totalDelaySeconds / 1000); // Reset counter _playoutDelayMeasurementCounter = 0; } - // todo: Add playout buffer? (Only used for 44.1 kHz) + // todo: Add playout buffer? } void AudioDeviceIPhone::UpdateRecordingDelay() { @@ -1766,7 +1739,10 @@ void AudioDeviceIPhone::UpdateRecordingDelay() { if (_recordingDelayMeasurementCounter >= 100) { // Update HW and OS delay every second, unlikely to change - _recordingDelayHWAndOS = 0; + // Since this is eventually rounded to integral ms, add 0.5ms + // here to get round-to-nearest-int behavior instead of + // truncation. + float totalDelaySeconds = 0.0005; // HW input latency Float32 f32(0); @@ -1777,7 +1753,8 @@ void AudioDeviceIPhone::UpdateRecordingDelay() { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error HW latency (result=%d)", result); } - _recordingDelayHWAndOS += static_cast<int>(f32 * 1000000); + assert(f32 >= 0); + totalDelaySeconds += f32; // HW buffer duration f32 = 0; @@ -1787,7 +1764,8 @@ void AudioDeviceIPhone::UpdateRecordingDelay() { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error HW buffer duration (result=%d)", result); } - _recordingDelayHWAndOS += static_cast<int>(f32 * 1000000); + assert(f32 >= 0); + totalDelaySeconds += f32; // AU latency Float64 f64(0); @@ -1799,10 +1777,12 @@ void AudioDeviceIPhone::UpdateRecordingDelay() { WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "error AU latency (result=%d)", result); } - _recordingDelayHWAndOS += static_cast<int>(f64 * 1000000); + assert(f64 >= 0); + totalDelaySeconds += f64; // To ms - _recordingDelayHWAndOS = (_recordingDelayHWAndOS - 500) / 1000; + _recordingDelayHWAndOS = + static_cast<uint32_t>(totalDelaySeconds / 1000); // Reset counter _recordingDelayMeasurementCounter = 0; diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h index fdaf94d6a60..011b6acf481 100644 --- a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h +++ b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h @@ -90,10 +90,8 @@ public: uint16_t& volumeRight) const; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available); virtual int32_t InitSpeaker(); virtual bool SpeakerIsInitialized() const; - virtual int32_t MicrophoneIsAvailable(bool& available); virtual int32_t InitMicrophone(); virtual bool MicrophoneIsInitialized() const; diff --git a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc index caa1efed8ff..67a845c845b 100644 --- a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc +++ b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc @@ -180,7 +180,7 @@ int32_t AudioDeviceLinuxALSA::Init() { return 0; } - +#if defined(USE_X11) //Get X display handle for typing detection _XDisplay = XOpenDisplay(NULL); if (!_XDisplay) @@ -188,7 +188,7 @@ int32_t AudioDeviceLinuxALSA::Init() WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " failed to open X display, typing detection will not work"); } - +#endif _playWarning = 0; _playError = 0; _recWarning = 0; @@ -254,13 +254,13 @@ int32_t AudioDeviceLinuxALSA::Terminate() _critSect.Enter(); } - +#if defined(USE_X11) if (_XDisplay) { XCloseDisplay(_XDisplay); _XDisplay = NULL; } - +#endif _initialized = false; _outputDeviceIsSpecified = false; _inputDeviceIsSpecified = false; @@ -273,34 +273,6 @@ bool AudioDeviceLinuxALSA::Initialized() const return (_initialized); } -int32_t AudioDeviceLinuxALSA::SpeakerIsAvailable(bool& available) -{ - - bool wasInitialized = _mixerManager.SpeakerIsInitialized(); - - // Make an attempt to open up the - // output mixer corresponding to the currently selected output device. - // - if (!wasInitialized && InitSpeaker() == -1) - { - available = false; - return 0; - } - - // Given that InitSpeaker was successful, we know that a valid speaker - // exists - available = true; - - // Close the initialized output mixer - // - if (!wasInitialized) - { - _mixerManager.CloseSpeaker(); - } - - return 0; -} - int32_t AudioDeviceLinuxALSA::InitSpeaker() { @@ -316,34 +288,6 @@ int32_t AudioDeviceLinuxALSA::InitSpeaker() return _mixerManager.OpenSpeaker(devName); } -int32_t AudioDeviceLinuxALSA::MicrophoneIsAvailable(bool& available) -{ - - bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); - - // Make an attempt to open up the - // input mixer corresponding to the currently selected output device. - // - if (!wasInitialized && InitMicrophone() == -1) - { - available = false; - return 0; - } - - // Given that InitMicrophone was successful, we know that a valid - // microphone exists - available = true; - - // Close the initialized input mixer - // - if (!wasInitialized) - { - _mixerManager.CloseMicrophone(); - } - - return 0; -} - int32_t AudioDeviceLinuxALSA::InitMicrophone() { @@ -2342,7 +2286,7 @@ bool AudioDeviceLinuxALSA::RecThreadProcess() bool AudioDeviceLinuxALSA::KeyPressed() const{ - +#if defined(USE_X11) char szKey[32]; unsigned int i = 0; char state = 0; @@ -2360,5 +2304,8 @@ bool AudioDeviceLinuxALSA::KeyPressed() const{ // Save old state memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState)); return (state != 0); +#else + return false; +#endif } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h index 35abc152fc8..1d75c8e94a0 100644 --- a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h +++ b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h @@ -15,8 +15,9 @@ #include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" - +#if defined(USE_X11) #include <X11/Xlib.h> +#endif #include <alsa/asoundlib.h> #include <sys/ioctl.h> #include <sys/soundcard.h> @@ -89,10 +90,8 @@ public: uint16_t& volumeRight) const OVERRIDE; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available) OVERRIDE; virtual int32_t InitSpeaker() OVERRIDE; virtual bool SpeakerIsInitialized() const OVERRIDE; - virtual int32_t MicrophoneIsAvailable(bool& available) OVERRIDE; virtual int32_t InitMicrophone() OVERRIDE; virtual bool MicrophoneIsInitialized() const OVERRIDE; @@ -135,7 +134,7 @@ public: virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE; virtual int32_t SetStereoRecording(bool enable) OVERRIDE; virtual int32_t StereoRecording(bool& enabled) const OVERRIDE; - + // Delay information and control virtual int32_t SetPlayoutBuffer( const AudioDeviceModule::BufferType type, @@ -174,8 +173,8 @@ private: bool KeyPressed() const; private: - void Lock() { _critSect.Enter(); }; - void UnLock() { _critSect.Leave(); }; + void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); }; + void UnLock() UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); }; private: inline int32_t InputSanityCheckAfterUnlockedPeriod() const; inline int32_t OutputSanityCheckAfterUnlockedPeriod() const; @@ -188,7 +187,7 @@ private: private: AudioDeviceBuffer* _ptrAudioBuffer; - + CriticalSectionWrapper& _critSect; ThreadWrapper* _ptrThreadRec; @@ -250,7 +249,9 @@ private: uint16_t _playBufDelayFixed; // fixed playback delay char _oldKeyState[32]; +#if defined(USE_X11) Display* _XDisplay; +#endif }; } diff --git a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc index e095eed0d5b..00d4afed892 100644 --- a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc +++ b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc @@ -341,34 +341,6 @@ bool AudioDeviceLinuxPulse::Initialized() const return (_initialized); } -int32_t AudioDeviceLinuxPulse::SpeakerIsAvailable(bool& available) -{ - - bool wasInitialized = _mixerManager.SpeakerIsInitialized(); - - // Make an attempt to open up the - // output mixer corresponding to the currently selected output device. - // - if (!wasInitialized && InitSpeaker() == -1) - { - available = false; - return 0; - } - - // Given that InitSpeaker was successful, we know that a valid speaker exists - // - available = true; - - // Close the initialized output mixer - // - if (!wasInitialized) - { - _mixerManager.CloseSpeaker(); - } - - return 0; -} - int32_t AudioDeviceLinuxPulse::InitSpeaker() { @@ -414,34 +386,6 @@ int32_t AudioDeviceLinuxPulse::InitSpeaker() return 0; } -int32_t AudioDeviceLinuxPulse::MicrophoneIsAvailable(bool& available) -{ - - bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); - - // Make an attempt to open up the - // input mixer corresponding to the currently selected output device. - // - if (!wasInitialized && InitMicrophone() == -1) - { - available = false; - return 0; - } - - // Given that InitMicrophone was successful, we know that a valid microphone - // exists - available = true; - - // Close the initialized input mixer - // - if (!wasInitialized) - { - _mixerManager.CloseMicrophone(); - } - - return 0; -} - int32_t AudioDeviceLinuxPulse::InitMicrophone() { @@ -2613,7 +2557,7 @@ int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData, int32_t AudioDeviceLinuxPulse::ProcessRecordedData( int8_t *bufferData, uint32_t bufferSizeInSamples, - uint32_t recDelay) + uint32_t recDelay) EXCLUSIVE_LOCKS_REQUIRED(_critSect) { uint32_t currentMicLevel(0); uint32_t newMicLevel(0); diff --git a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h index 43228a1babf..cde7e464390 100644 --- a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h +++ b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h @@ -151,10 +151,8 @@ public: uint16_t& volumeRight) const OVERRIDE; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available) OVERRIDE; virtual int32_t InitSpeaker() OVERRIDE; virtual bool SpeakerIsInitialized() const OVERRIDE; - virtual int32_t MicrophoneIsAvailable(bool& available) OVERRIDE; virtual int32_t InitMicrophone() OVERRIDE; virtual bool MicrophoneIsInitialized() const OVERRIDE; @@ -224,16 +222,12 @@ public: virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE; private: - void Lock() - { + void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); } - ; - void UnLock() - { + void UnLock() UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); } - ; void WaitForOperationCompletion(pa_operation* paOperation) const; void WaitForSuccess(pa_operation* paOperation) const; diff --git a/chromium/third_party/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h b/chromium/third_party/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h index b5186fa7bf2..052390a658e 100644 --- a/chromium/third_party/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h +++ b/chromium/third_party/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h @@ -32,7 +32,7 @@ #include <stddef.h> // for NULL #include <string.h> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/system_wrappers/interface/trace.h" // This file provides macros for creating "symbol table" classes to simplify the diff --git a/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.cc b/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.cc index b07c94dd11c..2c875796f18 100644 --- a/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.cc +++ b/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.cc @@ -571,7 +571,6 @@ int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available) return 0; } - int32_t AudioDeviceMac::InitMicrophone() { diff --git a/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.h b/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.h index a2662239286..fae4041133b 100644 --- a/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.h +++ b/chromium/third_party/webrtc/modules/audio_device/mac/audio_device_mac.h @@ -108,10 +108,8 @@ public: uint16_t& volumeRight) const; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available); virtual int32_t InitSpeaker(); virtual bool SpeakerIsInitialized() const; - virtual int32_t MicrophoneIsAvailable(bool& available); virtual int32_t InitMicrophone(); virtual bool MicrophoneIsInitialized() const; @@ -167,7 +165,6 @@ public: // CPU load virtual int32_t CPULoad(uint16_t& load) const; -public: virtual bool PlayoutWarning() const; virtual bool PlayoutError() const; virtual bool RecordingWarning() const; @@ -177,10 +174,12 @@ public: virtual void ClearRecordingWarning(); virtual void ClearRecordingError(); -public: virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); private: + virtual int32_t MicrophoneIsAvailable(bool& available); + virtual int32_t SpeakerIsAvailable(bool& available); + void Lock() { _critSect.Enter(); @@ -236,7 +235,6 @@ private: int32_t HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress); -private: static OSStatus deviceIOProc(AudioDeviceID device, const AudioTimeStamp *now, const AudioBufferList *inputData, @@ -284,10 +282,8 @@ private: bool CaptureWorkerThread(); bool RenderWorkerThread(); -private: bool KeyPressed(); -private: AudioDeviceBuffer* _ptrAudioBuffer; CriticalSectionWrapper& _critSect; @@ -325,7 +321,6 @@ private: AudioDeviceModule::BufferType _playBufType; -private: bool _initialized; bool _isShutDown; bool _recording; @@ -361,7 +356,6 @@ private: int32_t _renderDelayOffsetSamples; -private: uint16_t _playBufDelayFixed; // fixed playback delay uint16_t _playWarning; @@ -378,7 +372,6 @@ private: int _captureBufSizeSamples; int _renderBufSizeSamples; -private: // Typing detection // 0x5c is key "9", after that comes function keys. bool prev_key_state_[0x5d]; diff --git a/chromium/third_party/webrtc/modules/audio_device/main/source/OWNERS b/chromium/third_party/webrtc/modules/audio_device/main/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_device/main/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.cc b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.cc index a71b821189a..32b5e49af1f 100644 --- a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.cc +++ b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.cc @@ -751,25 +751,6 @@ bool AudioDeviceWindowsCore::Initialized() const } // ---------------------------------------------------------------------------- -// SpeakerIsAvailable -// ---------------------------------------------------------------------------- - -int32_t AudioDeviceWindowsCore::SpeakerIsAvailable(bool& available) -{ - - CriticalSectionScoped lock(&_critSect); - - if (_ptrDeviceOut == NULL) - { - return -1; - } - - available = true; - - return 0; -} - -// ---------------------------------------------------------------------------- // InitSpeaker // ---------------------------------------------------------------------------- @@ -852,25 +833,6 @@ int32_t AudioDeviceWindowsCore::InitSpeaker() } // ---------------------------------------------------------------------------- -// MicrophoneIsAvailable -// ---------------------------------------------------------------------------- - -int32_t AudioDeviceWindowsCore::MicrophoneIsAvailable(bool& available) -{ - - CriticalSectionScoped lock(&_critSect); - - if (_ptrDeviceIn == NULL) - { - return -1; - } - - available = true; - - return 0; -} - -// ---------------------------------------------------------------------------- // InitMicrophone // ---------------------------------------------------------------------------- diff --git a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.h b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.h index 7a9a5245ca9..4d30928c5ed 100644 --- a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.h +++ b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_core_win.h @@ -138,10 +138,8 @@ public: virtual int32_t WaveOutVolume(uint16_t& volumeLeft, uint16_t& volumeRight) const; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available); virtual int32_t InitSpeaker(); virtual bool SpeakerIsInitialized() const; - virtual int32_t MicrophoneIsAvailable(bool& available); virtual int32_t InitMicrophone(); virtual bool MicrophoneIsInitialized() const; diff --git a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc index c05d0a29a7a..e2e515b1004 100644 --- a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc +++ b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc @@ -490,33 +490,6 @@ bool AudioDeviceWindowsWave::Initialized() const } // ---------------------------------------------------------------------------- -// SpeakerIsAvailable -// ---------------------------------------------------------------------------- - -int32_t AudioDeviceWindowsWave::SpeakerIsAvailable(bool& available) -{ - - // Enumerate all avaliable speakers and make an attempt to open up the - // output mixer corresponding to the currently selected output device. - // - if (InitSpeaker() == -1) - { - available = false; - return 0; - } - - // Given that InitSpeaker was successful, we know that a valid speaker exists - // - available = true; - - // Close the initialized output mixer - // - _mixerManager.CloseSpeaker(); - - return 0; -} - -// ---------------------------------------------------------------------------- // InitSpeaker // ---------------------------------------------------------------------------- @@ -555,33 +528,6 @@ int32_t AudioDeviceWindowsWave::InitSpeaker() } // ---------------------------------------------------------------------------- -// MicrophoneIsAvailable -// ---------------------------------------------------------------------------- - -int32_t AudioDeviceWindowsWave::MicrophoneIsAvailable(bool& available) -{ - - // Enumerate all avaliable microphones and make an attempt to open up the - // input mixer corresponding to the currently selected output device. - // - if (InitMicrophone() == -1) - { - available = false; - return 0; - } - - // Given that InitMicrophone was successful, we know that a valid microphone exists - // - available = true; - - // Close the initialized input mixer - // - _mixerManager.CloseMicrophone(); - - return 0; -} - -// ---------------------------------------------------------------------------- // InitMicrophone // ---------------------------------------------------------------------------- diff --git a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.h b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.h index 480bbcbc778..1767b90a858 100644 --- a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.h +++ b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.h @@ -94,10 +94,8 @@ public: virtual int32_t WaveOutVolume(uint16_t& volumeLeft, uint16_t& volumeRight) const; // Audio mixer initialization - virtual int32_t SpeakerIsAvailable(bool& available); virtual int32_t InitSpeaker(); virtual bool SpeakerIsInitialized() const; - virtual int32_t MicrophoneIsAvailable(bool& available); virtual int32_t InitMicrophone(); virtual bool MicrophoneIsInitialized() const; diff --git a/chromium/third_party/webrtc/modules/audio_processing/OWNERS b/chromium/third_party/webrtc/modules/audio_processing/OWNERS index 5a2563444b6..7c1f7881c2c 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/OWNERS +++ b/chromium/third_party/webrtc/modules/audio_processing/OWNERS @@ -1,2 +1,8 @@ +aluebs@webrtc.org andrew@webrtc.org bjornv@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/Android.mk b/chromium/third_party/webrtc/modules/audio_processing/aec/Android.mk index 3ad52b96625..181e87d9a76 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/Android.mk +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/Android.mk @@ -47,3 +47,12 @@ ifndef NDK_ROOT include external/stlport/libstlport.mk endif include $(BUILD_STATIC_LIBRARY) + +######################### +# Build the neon library. +ifeq ($(WEBRTC_BUILD_NEON_LIBS),true) + +LOCAL_SRC_FILES += \ + aec_core_neon.c + +endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true) diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.c index bfa087c23de..207c6dc3bfc 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.c +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.c @@ -67,7 +67,7 @@ static const float sqrtHanning[65] = { // Matlab code to produce table: // weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1]; // fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve); -const float WebRtcAec_weightCurve[65] = { +ALIGN16_BEG const float ALIGN16_END WebRtcAec_weightCurve[65] = { 0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f, 0.1845f, 0.1926f, 0.2000f, 0.2069f, 0.2134f, 0.2195f, 0.2254f, 0.2309f, 0.2363f, 0.2414f, 0.2464f, 0.2512f, 0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f, @@ -81,7 +81,7 @@ const float WebRtcAec_weightCurve[65] = { // Matlab code to produce table: // overDriveCurve = [sqrt(linspace(0,1,65))' + 1]; // fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve); -const float WebRtcAec_overDriveCurve[65] = { +ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65] = { 1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f, 1.3062f, 1.3307f, 1.3536f, 1.3750f, 1.3953f, 1.4146f, 1.4330f, 1.4507f, 1.4677f, 1.4841f, 1.5000f, 1.5154f, 1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f, @@ -116,7 +116,7 @@ extern int webrtc_aec_instance_count; // "Private" function prototypes. static void ProcessBlock(AecCore* aec); -static void NonLinearProcessing(AecCore* aec, short* output, short* outputH); +static void NonLinearProcessing(AecCore* aec, float* output, float* outputH); static void GetHighbandGain(const float* lambda, float* nlpGainHband); @@ -160,28 +160,28 @@ int WebRtcAec_CreateAec(AecCore** aecInst) { return -1; } - aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); + aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float)); if (!aec->nearFrBuf) { WebRtcAec_FreeAec(aec); aec = NULL; return -1; } - aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); + aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float)); if (!aec->outFrBuf) { WebRtcAec_FreeAec(aec); aec = NULL; return -1; } - aec->nearFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); + aec->nearFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float)); if (!aec->nearFrBufH) { WebRtcAec_FreeAec(aec); aec = NULL; return -1; } - aec->outFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); + aec->outFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float)); if (!aec->outFrBufH) { WebRtcAec_FreeAec(aec); aec = NULL; @@ -419,6 +419,7 @@ WebRtcAec_FilterFar_t WebRtcAec_FilterFar; WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal; WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation; WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress; +WebRtcAec_ComfortNoise_t WebRtcAec_ComfortNoise; int WebRtcAec_InitAec(AecCore* aec, int sampFreq) { int i; @@ -472,9 +473,21 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) { aec->delay_logging_enabled = 0; memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram)); + aec->reported_delay_enabled = 1; aec->extended_filter_enabled = 0; aec->num_partitions = kNormalNumPartitions; + // Update the delay estimator with filter length. We use half the + // |num_partitions| to take the echo path into account. In practice we say + // that the echo has a duration of maximum half |num_partitions|, which is not + // true, but serves as a crude measure. + WebRtc_set_allowed_offset(aec->delay_estimator, aec->num_partitions / 2); + // TODO(bjornv): I currently hard coded the enable. Once we've established + // that AECM has no performance regression, robust_validation will be enabled + // all the time and the APIs to turn it on/off will be removed. Hence, remove + // this line then. + WebRtc_enable_robust_validation(aec->delay_estimator, 1); + // Default target suppression mode. aec->nlp_mode = 1; @@ -557,6 +570,7 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) { WebRtcAec_ScaleErrorSignal = ScaleErrorSignal; WebRtcAec_FilterAdaptation = FilterAdaptation; WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress; + WebRtcAec_ComfortNoise = ComfortNoise; #if defined(WEBRTC_ARCH_X86_FAMILY) if (WebRtc_GetCPUInfo(kSSE2)) { @@ -564,6 +578,14 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) { } #endif +#if defined(MIPS_FPU_LE) + WebRtcAec_InitAec_mips(); +#endif + +#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON) + WebRtcAec_InitAec_neon(); +#endif + aec_rdft_init(); return 0; @@ -599,11 +621,11 @@ int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) { } void WebRtcAec_ProcessFrame(AecCore* aec, - const short* nearend, - const short* nearendH, + const float* nearend, + const float* nearendH, int knownDelay, - int16_t* out, - int16_t* outH) { + float* out, + float* outH) { int out_elements = 0; // For each frame the process is as follows: @@ -724,7 +746,7 @@ int WebRtcAec_GetDelayMetricsCore(AecCore* self, int* median, int* std) { // Calculate the L1 norm, with median value as central moment. for (i = 0; i < kHistorySizeBlocks; i++) { - l1_norm += (float)(fabs(i - my_median) * self->delay_histogram[i]); + l1_norm += (float)abs(i - my_median) * self->delay_histogram[i]; } *std = (int)(l1_norm / (float)num_delay_values + 0.5f) * kMsPerBlock; @@ -768,9 +790,19 @@ void WebRtcAec_SetConfigCore(AecCore* self, } } +void WebRtcAec_enable_reported_delay(AecCore* self, int enable) { + self->reported_delay_enabled = enable; +} + +int WebRtcAec_reported_delay_enabled(AecCore* self) { + return self->reported_delay_enabled; +} + void WebRtcAec_enable_delay_correction(AecCore* self, int enable) { self->extended_filter_enabled = enable; self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions; + // Update the delay estimator with filter length. See InitAEC() for details. + WebRtc_set_allowed_offset(self->delay_estimator, self->num_partitions / 2); } int WebRtcAec_delay_correction_enabled(AecCore* self) { @@ -786,7 +818,7 @@ void WebRtcAec_SetSystemDelay(AecCore* self, int delay) { static void ProcessBlock(AecCore* aec) { int i; - float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN]; + float y[PART_LEN], e[PART_LEN]; float scale; float fft[PART_LEN2]; @@ -805,30 +837,22 @@ static void ProcessBlock(AecCore* aec) { const float ramp = 1.0002f; const float gInitNoise[2] = {0.999f, 0.001f}; - int16_t nearend[PART_LEN]; - int16_t* nearend_ptr = NULL; - int16_t output[PART_LEN]; - int16_t outputH[PART_LEN]; + float nearend[PART_LEN]; + float* nearend_ptr = NULL; + float output[PART_LEN]; + float outputH[PART_LEN]; float* xf_ptr = NULL; - memset(dH, 0, sizeof(dH)); + // Concatenate old and new nearend blocks. if (aec->sampFreq == 32000) { - // Get the upper band first so we can reuse |nearend|. WebRtc_ReadBuffer(aec->nearFrBufH, (void**)&nearend_ptr, nearend, PART_LEN); - for (i = 0; i < PART_LEN; i++) { - dH[i] = (float)(nearend_ptr[i]); - } - memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN); + memcpy(aec->dBufH + PART_LEN, nearend_ptr, sizeof(nearend)); } WebRtc_ReadBuffer(aec->nearFrBuf, (void**)&nearend_ptr, nearend, PART_LEN); + memcpy(aec->dBuf + PART_LEN, nearend_ptr, sizeof(nearend)); // ---------- Ooura fft ---------- - // Concatenate old and new nearend blocks. - for (i = 0; i < PART_LEN; i++) { - d[i] = (float)(nearend_ptr[i]); - } - memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN); #ifdef WEBRTC_AEC_DEBUG_DUMP { @@ -940,7 +964,7 @@ static void ProcessBlock(AecCore* aec) { } for (i = 0; i < PART_LEN; i++) { - e[i] = d[i] - y[i]; + e[i] = nearend_ptr[i] - y[i]; } // Error fft @@ -999,7 +1023,7 @@ static void ProcessBlock(AecCore* aec) { #endif } -static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) { +static void NonLinearProcessing(AecCore* aec, float* output, float* outputH) { float efw[2][PART_LEN1], dfw[2][PART_LEN1], xfw[2][PART_LEN1]; complex_t comfortNoiseHband[PART_LEN1]; float fft[PART_LEN2]; @@ -1266,7 +1290,7 @@ static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) { WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw); // Add comfort noise. - ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl); + WebRtcAec_ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl); // TODO(bjornv): Investigate how to take the windowing below into account if // needed. @@ -1293,12 +1317,12 @@ static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) { fft[i] *= scale; // fft scaling fft[i] = fft[i] * sqrtHanning[i] + aec->outBuf[i]; - // Saturation protection - output[i] = (short)WEBRTC_SPL_SAT( - WEBRTC_SPL_WORD16_MAX, fft[i], WEBRTC_SPL_WORD16_MIN); - fft[PART_LEN + i] *= scale; // fft scaling aec->outBuf[i] = fft[PART_LEN + i] * sqrtHanning[PART_LEN - i]; + + // Saturate output to keep it in the allowed range. + output[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, fft[i], WEBRTC_SPL_WORD16_MIN); } // For H band @@ -1323,8 +1347,8 @@ static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) { // compute gain factor for (i = 0; i < PART_LEN; i++) { - dtmp = (float)aec->dBufH[i]; - dtmp = (float)dtmp * nlpGainHband; // for variable gain + dtmp = aec->dBufH[i]; + dtmp = dtmp * nlpGainHband; // for variable gain // add some comfort noise where Hband is attenuated if (flagHbandCn == 1) { @@ -1332,8 +1356,8 @@ static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) { dtmp += cnScaleHband * fft[i]; } - // Saturation protection - outputH[i] = (short)WEBRTC_SPL_SAT( + // Saturate output to keep it in the allowed range. + outputH[i] = WEBRTC_SPL_SAT( WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN); } } diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h index d3c6d7e2b2e..93bfed46688 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h @@ -22,17 +22,6 @@ #define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients #define PART_LEN2 (PART_LEN * 2) // Length of partition * 2 -// Delay estimator constants, used for logging. -enum { - kMaxDelayBlocks = 60 -}; -enum { - kLookaheadBlocks = 15 -}; -enum { - kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks -}; - typedef float complex_t[2]; // For performance reasons, some arrays of complex numbers are replaced by twice // as long arrays of float, all the real parts followed by all the imaginary @@ -65,14 +54,20 @@ int WebRtcAec_CreateAec(AecCore** aec); int WebRtcAec_FreeAec(AecCore* aec); int WebRtcAec_InitAec(AecCore* aec, int sampFreq); void WebRtcAec_InitAec_SSE2(void); +#if defined(MIPS_FPU_LE) +void WebRtcAec_InitAec_mips(void); +#endif +#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON) +void WebRtcAec_InitAec_neon(void); +#endif void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend); void WebRtcAec_ProcessFrame(AecCore* aec, - const short* nearend, - const short* nearendH, + const float* nearend, + const float* nearendH, int knownDelay, - int16_t* out, - int16_t* outH); + float* out, + float* outH); // A helper function to call WebRtc_MoveReadPtr() for all far-end buffers. // Returns the number of elements moved, and adjusts |system_delay| by the @@ -101,6 +96,12 @@ void WebRtcAec_SetConfigCore(AecCore* self, int metrics_mode, int delay_logging); +// Non-zero enables, zero disables. +void WebRtcAec_enable_reported_delay(AecCore* self, int enable); + +// Returns non-zero if reported delay is enabled and zero if disabled. +int WebRtcAec_reported_delay_enabled(AecCore* self); + // We now interpret delay correction to mean an extended filter length feature. // We reuse the delay correction infrastructure to avoid changes through to // libjingle. See details along with |DelayCorrection| in diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h index 193369382ca..1c560f91c9c 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h @@ -26,6 +26,17 @@ enum { }; static const int kNormalNumPartitions = 12; +// Delay estimator constants, used for logging. +enum { + kMaxDelayBlocks = 60 +}; +enum { + kLookaheadBlocks = 15 +}; +enum { + kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks +}; + // Extended filter adaptation parameters. // TODO(ajm): No narrowband tuning yet. static const float kExtendedMu = 0.4f; @@ -122,6 +133,7 @@ struct AecCore { void* delay_estimator_farend; void* delay_estimator; + int reported_delay_enabled; // 0 = disabled, otherwise enabled. // 1 = extended filter mode enabled, 0 = disabled. int extended_filter_enabled; // Runtime selection of number of filter partitions. @@ -151,4 +163,11 @@ typedef void (*WebRtcAec_OverdriveAndSuppress_t)(AecCore* aec, float efw[2][PART_LEN1]); extern WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress; +typedef void (*WebRtcAec_ComfortNoise_t)(AecCore* aec, + float efw[2][PART_LEN1], + complex_t* comfortNoiseHband, + const float* noisePow, + const float* lambda); +extern WebRtcAec_ComfortNoise_t WebRtcAec_ComfortNoise; + #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.c new file mode 100644 index 00000000000..d861e10f908 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.c @@ -0,0 +1,774 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/* + * The core AEC algorithm, which is presented with time-aligned signals. + */ + +#include "webrtc/modules/audio_processing/aec/aec_core.h" + +#include <math.h> + +#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" +#include "webrtc/modules/audio_processing/aec/aec_core_internal.h" +#include "webrtc/modules/audio_processing/aec/aec_rdft.h" + +static const int flagHbandCn = 1; // flag for adding comfort noise in H band +extern const float WebRtcAec_weightCurve[65]; +extern const float WebRtcAec_overDriveCurve[65]; + +void WebRtcAec_ComfortNoise_mips(AecCore* aec, + float efw[2][PART_LEN1], + complex_t* comfortNoiseHband, + const float* noisePow, + const float* lambda) { + int i, num; + float rand[PART_LEN]; + float noise, noiseAvg, tmp, tmpAvg; + int16_t randW16[PART_LEN]; + complex_t u[PART_LEN1]; + + const float pi2 = 6.28318530717959f; + const float pi2t = pi2 / 32768; + + // Generate a uniform random array on [0 1] + WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed); + + int16_t *randWptr = randW16; + float randTemp, randTemp2, randTemp3, randTemp4; + short tmp1s, tmp2s, tmp3s, tmp4s; + + for (i = 0; i < PART_LEN; i+=4) { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lh %[tmp1s], 0(%[randWptr]) \n\t" + "lh %[tmp2s], 2(%[randWptr]) \n\t" + "lh %[tmp3s], 4(%[randWptr]) \n\t" + "lh %[tmp4s], 6(%[randWptr]) \n\t" + "mtc1 %[tmp1s], %[randTemp] \n\t" + "mtc1 %[tmp2s], %[randTemp2] \n\t" + "mtc1 %[tmp3s], %[randTemp3] \n\t" + "mtc1 %[tmp4s], %[randTemp4] \n\t" + "cvt.s.w %[randTemp], %[randTemp] \n\t" + "cvt.s.w %[randTemp2], %[randTemp2] \n\t" + "cvt.s.w %[randTemp3], %[randTemp3] \n\t" + "cvt.s.w %[randTemp4], %[randTemp4] \n\t" + "addiu %[randWptr], %[randWptr], 8 \n\t" + "mul.s %[randTemp], %[randTemp], %[pi2t] \n\t" + "mul.s %[randTemp2], %[randTemp2], %[pi2t] \n\t" + "mul.s %[randTemp3], %[randTemp3], %[pi2t] \n\t" + "mul.s %[randTemp4], %[randTemp4], %[pi2t] \n\t" + ".set pop \n\t" + : [randWptr] "+r" (randWptr), [randTemp] "=&f" (randTemp), + [randTemp2] "=&f" (randTemp2), [randTemp3] "=&f" (randTemp3), + [randTemp4] "=&f" (randTemp4), [tmp1s] "=&r" (tmp1s), + [tmp2s] "=&r" (tmp2s), [tmp3s] "=&r" (tmp3s), + [tmp4s] "=&r" (tmp4s) + : [pi2t] "f" (pi2t) + : "memory" + ); + + u[i+1][0] = (float)cos(randTemp); + u[i+1][1] = (float)sin(randTemp); + u[i+2][0] = (float)cos(randTemp2); + u[i+2][1] = (float)sin(randTemp2); + u[i+3][0] = (float)cos(randTemp3); + u[i+3][1] = (float)sin(randTemp3); + u[i+4][0] = (float)cos(randTemp4); + u[i+4][1] = (float)sin(randTemp4); + } + + // Reject LF noise + float *u_ptr = &u[1][0]; + float noise2, noise3, noise4; + float tmp1f, tmp2f, tmp3f, tmp4f, tmp5f, tmp6f, tmp7f, tmp8f; + + u[0][0] = 0; + u[0][1] = 0; + for (i = 1; i < PART_LEN1; i+=4) { + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lwc1 %[noise], 4(%[noisePow]) \n\t" + "lwc1 %[noise2], 8(%[noisePow]) \n\t" + "lwc1 %[noise3], 12(%[noisePow]) \n\t" + "lwc1 %[noise4], 16(%[noisePow]) \n\t" + "sqrt.s %[noise], %[noise] \n\t" + "sqrt.s %[noise2], %[noise2] \n\t" + "sqrt.s %[noise3], %[noise3] \n\t" + "sqrt.s %[noise4], %[noise4] \n\t" + "lwc1 %[tmp1f], 0(%[u_ptr]) \n\t" + "lwc1 %[tmp2f], 4(%[u_ptr]) \n\t" + "lwc1 %[tmp3f], 8(%[u_ptr]) \n\t" + "lwc1 %[tmp4f], 12(%[u_ptr]) \n\t" + "lwc1 %[tmp5f], 16(%[u_ptr]) \n\t" + "lwc1 %[tmp6f], 20(%[u_ptr]) \n\t" + "lwc1 %[tmp7f], 24(%[u_ptr]) \n\t" + "lwc1 %[tmp8f], 28(%[u_ptr]) \n\t" + "addiu %[noisePow], %[noisePow], 16 \n\t" + "mul.s %[tmp1f], %[tmp1f], %[noise] \n\t" + "mul.s %[tmp2f], %[tmp2f], %[noise] \n\t" + "mul.s %[tmp3f], %[tmp3f], %[noise2] \n\t" + "mul.s %[tmp4f], %[tmp4f], %[noise2] \n\t" + "mul.s %[tmp5f], %[tmp5f], %[noise3] \n\t" + "mul.s %[tmp6f], %[tmp6f], %[noise3] \n\t" + "swc1 %[tmp1f], 0(%[u_ptr]) \n\t" + "swc1 %[tmp3f], 8(%[u_ptr]) \n\t" + "mul.s %[tmp8f], %[tmp8f], %[noise4] \n\t" + "mul.s %[tmp7f], %[tmp7f], %[noise4] \n\t" + "neg.s %[tmp2f] \n\t" + "neg.s %[tmp4f] \n\t" + "neg.s %[tmp6f] \n\t" + "neg.s %[tmp8f] \n\t" + "swc1 %[tmp5f], 16(%[u_ptr]) \n\t" + "swc1 %[tmp7f], 24(%[u_ptr]) \n\t" + "swc1 %[tmp2f], 4(%[u_ptr]) \n\t" + "swc1 %[tmp4f], 12(%[u_ptr]) \n\t" + "swc1 %[tmp6f], 20(%[u_ptr]) \n\t" + "swc1 %[tmp8f], 28(%[u_ptr]) \n\t" + "addiu %[u_ptr], %[u_ptr], 32 \n\t" + ".set pop \n\t" + : [u_ptr] "+r" (u_ptr), [noisePow] "+r" (noisePow), + [noise] "=&f" (noise), [noise2] "=&f" (noise2), + [noise3] "=&f" (noise3), [noise4] "=&f" (noise4), + [tmp1f] "=&f" (tmp1f), [tmp2f] "=&f" (tmp2f), + [tmp3f] "=&f" (tmp3f), [tmp4f] "=&f" (tmp4f), + [tmp5f] "=&f" (tmp5f), [tmp6f] "=&f" (tmp6f), + [tmp7f] "=&f" (tmp7f), [tmp8f] "=&f" (tmp8f) + : + : "memory" + ); + } + u[PART_LEN][1] = 0; + noisePow -= PART_LEN; + + u_ptr = &u[0][0]; + float *u_ptr_end = &u[PART_LEN][0]; + float *efw_ptr_0 = &efw[0][0]; + float *efw_ptr_1 = &efw[1][0]; + float tmp9f, tmp10f; + const float tmp1c = 1.0; + const float tmp2c = 0.0; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "lwc1 %[tmp1f], 0(%[lambda]) \n\t" + "lwc1 %[tmp6f], 4(%[lambda]) \n\t" + "addiu %[lambda], %[lambda], 8 \n\t" + "c.lt.s %[tmp1f], %[tmp1c] \n\t" + "bc1f 4f \n\t" + " nop \n\t" + "c.lt.s %[tmp6f], %[tmp1c] \n\t" + "bc1f 3f \n\t" + " nop \n\t" + "2: \n\t" + "mul.s %[tmp1f], %[tmp1f], %[tmp1f] \n\t" + "mul.s %[tmp6f], %[tmp6f], %[tmp6f] \n\t" + "sub.s %[tmp1f], %[tmp1c], %[tmp1f] \n\t" + "sub.s %[tmp6f], %[tmp1c], %[tmp6f] \n\t" + "sqrt.s %[tmp1f], %[tmp1f] \n\t" + "sqrt.s %[tmp6f], %[tmp6f] \n\t" + "lwc1 %[tmp2f], 0(%[efw_ptr_0]) \n\t" + "lwc1 %[tmp3f], 0(%[u_ptr]) \n\t" + "lwc1 %[tmp7f], 4(%[efw_ptr_0]) \n\t" + "lwc1 %[tmp8f], 8(%[u_ptr]) \n\t" + "lwc1 %[tmp4f], 0(%[efw_ptr_1]) \n\t" + "lwc1 %[tmp5f], 4(%[u_ptr]) \n\t" + "lwc1 %[tmp9f], 4(%[efw_ptr_1]) \n\t" + "lwc1 %[tmp10f], 12(%[u_ptr]) \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[tmp3f], %[tmp1f], %[tmp3f] \n\t" + "add.s %[tmp2f], %[tmp2f], %[tmp3f] \n\t" + "mul.s %[tmp3f], %[tmp1f], %[tmp5f] \n\t" + "add.s %[tmp4f], %[tmp4f], %[tmp3f] \n\t" + "mul.s %[tmp3f], %[tmp6f], %[tmp8f] \n\t" + "add.s %[tmp7f], %[tmp7f], %[tmp3f] \n\t" + "mul.s %[tmp3f], %[tmp6f], %[tmp10f] \n\t" + "add.s %[tmp9f], %[tmp9f], %[tmp3f] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[tmp2f], %[tmp2f], %[tmp1f], %[tmp3f] \n\t" + "madd.s %[tmp4f], %[tmp4f], %[tmp1f], %[tmp5f] \n\t" + "madd.s %[tmp7f], %[tmp7f], %[tmp6f], %[tmp8f] \n\t" + "madd.s %[tmp9f], %[tmp9f], %[tmp6f], %[tmp10f] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[tmp2f], 0(%[efw_ptr_0]) \n\t" + "swc1 %[tmp4f], 0(%[efw_ptr_1]) \n\t" + "swc1 %[tmp7f], 4(%[efw_ptr_0]) \n\t" + "b 5f \n\t" + " swc1 %[tmp9f], 4(%[efw_ptr_1]) \n\t" + "3: \n\t" + "mul.s %[tmp1f], %[tmp1f], %[tmp1f] \n\t" + "sub.s %[tmp1f], %[tmp1c], %[tmp1f] \n\t" + "sqrt.s %[tmp1f], %[tmp1f] \n\t" + "lwc1 %[tmp2f], 0(%[efw_ptr_0]) \n\t" + "lwc1 %[tmp3f], 0(%[u_ptr]) \n\t" + "lwc1 %[tmp4f], 0(%[efw_ptr_1]) \n\t" + "lwc1 %[tmp5f], 4(%[u_ptr]) \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[tmp3f], %[tmp1f], %[tmp3f] \n\t" + "add.s %[tmp2f], %[tmp2f], %[tmp3f] \n\t" + "mul.s %[tmp3f], %[tmp1f], %[tmp5f] \n\t" + "add.s %[tmp4f], %[tmp4f], %[tmp3f] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[tmp2f], %[tmp2f], %[tmp1f], %[tmp3f] \n\t" + "madd.s %[tmp4f], %[tmp4f], %[tmp1f], %[tmp5f] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[tmp2f], 0(%[efw_ptr_0]) \n\t" + "b 5f \n\t" + " swc1 %[tmp4f], 0(%[efw_ptr_1]) \n\t" + "4: \n\t" + "c.lt.s %[tmp6f], %[tmp1c] \n\t" + "bc1f 5f \n\t" + " nop \n\t" + "mul.s %[tmp6f], %[tmp6f], %[tmp6f] \n\t" + "sub.s %[tmp6f], %[tmp1c], %[tmp6f] \n\t" + "sqrt.s %[tmp6f], %[tmp6f] \n\t" + "lwc1 %[tmp7f], 4(%[efw_ptr_0]) \n\t" + "lwc1 %[tmp8f], 8(%[u_ptr]) \n\t" + "lwc1 %[tmp9f], 4(%[efw_ptr_1]) \n\t" + "lwc1 %[tmp10f], 12(%[u_ptr]) \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[tmp3f], %[tmp6f], %[tmp8f] \n\t" + "add.s %[tmp7f], %[tmp7f], %[tmp3f] \n\t" + "mul.s %[tmp3f], %[tmp6f], %[tmp10f] \n\t" + "add.s %[tmp9f], %[tmp9f], %[tmp3f] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[tmp7f], %[tmp7f], %[tmp6f], %[tmp8f] \n\t" + "madd.s %[tmp9f], %[tmp9f], %[tmp6f], %[tmp10f] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[tmp7f], 4(%[efw_ptr_0]) \n\t" + "swc1 %[tmp9f], 4(%[efw_ptr_1]) \n\t" + "5: \n\t" + "addiu %[u_ptr], %[u_ptr], 16 \n\t" + "addiu %[efw_ptr_0], %[efw_ptr_0], 8 \n\t" + "bne %[u_ptr], %[u_ptr_end], 1b \n\t" + " addiu %[efw_ptr_1], %[efw_ptr_1], 8 \n\t" + ".set pop \n\t" + : [lambda] "+r" (lambda), [u_ptr] "+r" (u_ptr), + [efw_ptr_0] "+r" (efw_ptr_0), [efw_ptr_1] "+r" (efw_ptr_1), + [tmp1f] "=&f" (tmp1f), [tmp2f] "=&f" (tmp2f), [tmp3f] "=&f" (tmp3f), + [tmp4f] "=&f" (tmp4f), [tmp5f] "=&f" (tmp5f), + [tmp6f] "=&f" (tmp6f), [tmp7f] "=&f" (tmp7f), [tmp8f] "=&f" (tmp8f), + [tmp9f] "=&f" (tmp9f), [tmp10f] "=&f" (tmp10f) + : [tmp1c] "f" (tmp1c), [tmp2c] "f" (tmp2c), [u_ptr_end] "r" (u_ptr_end) + : "memory" + ); + + lambda -= PART_LEN; + tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[PART_LEN] * lambda[PART_LEN], 0)); + //tmp = 1 - lambda[i]; + efw[0][PART_LEN] += tmp * u[PART_LEN][0]; + efw[1][PART_LEN] += tmp * u[PART_LEN][1]; + + // For H band comfort noise + // TODO: don't compute noise and "tmp" twice. Use the previous results. + noiseAvg = 0.0; + tmpAvg = 0.0; + num = 0; + if (aec->sampFreq == 32000 && flagHbandCn == 1) { + for (i = 0; i < PART_LEN; i++) { + rand[i] = ((float)randW16[i]) / 32768; + } + + // average noise scale + // average over second half of freq spectrum (i.e., 4->8khz) + // TODO: we shouldn't need num. We know how many elements we're summing. + for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) { + num++; + noiseAvg += sqrtf(noisePow[i]); + } + noiseAvg /= (float)num; + + // average nlp scale + // average over second half of freq spectrum (i.e., 4->8khz) + // TODO: we shouldn't need num. We know how many elements we're summing. + num = 0; + for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) { + num++; + tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0)); + } + tmpAvg /= (float)num; + + // Use average noise for H band + // TODO: we should probably have a new random vector here. + // Reject LF noise + u[0][0] = 0; + u[0][1] = 0; + for (i = 1; i < PART_LEN1; i++) { + tmp = pi2 * rand[i - 1]; + + // Use average noise for H band + u[i][0] = noiseAvg * (float)cos(tmp); + u[i][1] = -noiseAvg * (float)sin(tmp); + } + u[PART_LEN][1] = 0; + + for (i = 0; i < PART_LEN1; i++) { + // Use average NLP weight for H band + comfortNoiseHband[i][0] = tmpAvg * u[i][0]; + comfortNoiseHband[i][1] = tmpAvg * u[i][1]; + } + } +} + +void WebRtcAec_FilterFar_mips(AecCore *aec, float yf[2][PART_LEN1]) { + int i; + for (i = 0; i < aec->num_partitions; i++) { + int xPos = (i + aec->xfBufBlockPos) * PART_LEN1; + int pos = i * PART_LEN1; + // Check for wrap + if (i + aec->xfBufBlockPos >= aec->num_partitions) { + xPos -= aec->num_partitions * (PART_LEN1); + } + float *yf0 = yf[0]; + float *yf1 = yf[1]; + float *aRe = aec->xfBuf[0] + xPos; + float *aIm = aec->xfBuf[1] + xPos; + float *bRe = aec->wfBuf[0] + pos; + float *bIm = aec->wfBuf[1] + pos; + float f0, f1, f2, f3, f4, f5, f6, f7, f8, f9, f10, f11, f12, f13; + int len = PART_LEN1 >> 1; + int len1 = PART_LEN1 & 1; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "lwc1 %[f0], 0(%[aRe]) \n\t" + "lwc1 %[f1], 0(%[bRe]) \n\t" + "lwc1 %[f2], 0(%[bIm]) \n\t" + "lwc1 %[f3], 0(%[aIm]) \n\t" + "lwc1 %[f4], 4(%[aRe]) \n\t" + "lwc1 %[f5], 4(%[bRe]) \n\t" + "lwc1 %[f6], 4(%[bIm]) \n\t" + "mul.s %[f8], %[f0], %[f1] \n\t" + "mul.s %[f0], %[f0], %[f2] \n\t" + "mul.s %[f9], %[f4], %[f5] \n\t" + "mul.s %[f4], %[f4], %[f6] \n\t" + "lwc1 %[f7], 4(%[aIm]) \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[f12], %[f2], %[f3] \n\t" + "mul.s %[f1], %[f3], %[f1] \n\t" + "mul.s %[f11], %[f6], %[f7] \n\t" + "addiu %[aRe], %[aRe], 8 \n\t" + "addiu %[aIm], %[aIm], 8 \n\t" + "addiu %[len], %[len], -1 \n\t" + "sub.s %[f8], %[f8], %[f12] \n\t" + "mul.s %[f12], %[f7], %[f5] \n\t" + "lwc1 %[f2], 0(%[yf0]) \n\t" + "add.s %[f1], %[f0], %[f1] \n\t" + "lwc1 %[f3], 0(%[yf1]) \n\t" + "sub.s %[f9], %[f9], %[f11] \n\t" + "lwc1 %[f6], 4(%[yf0]) \n\t" + "add.s %[f4], %[f4], %[f12] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "addiu %[aRe], %[aRe], 8 \n\t" + "addiu %[aIm], %[aIm], 8 \n\t" + "addiu %[len], %[len], -1 \n\t" + "nmsub.s %[f8], %[f8], %[f2], %[f3] \n\t" + "lwc1 %[f2], 0(%[yf0]) \n\t" + "madd.s %[f1], %[f0], %[f3], %[f1] \n\t" + "lwc1 %[f3], 0(%[yf1]) \n\t" + "nmsub.s %[f9], %[f9], %[f6], %[f7] \n\t" + "lwc1 %[f6], 4(%[yf0]) \n\t" + "madd.s %[f4], %[f4], %[f7], %[f5] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "lwc1 %[f5], 4(%[yf1]) \n\t" + "add.s %[f2], %[f2], %[f8] \n\t" + "addiu %[bRe], %[bRe], 8 \n\t" + "addiu %[bIm], %[bIm], 8 \n\t" + "add.s %[f3], %[f3], %[f1] \n\t" + "add.s %[f6], %[f6], %[f9] \n\t" + "add.s %[f5], %[f5], %[f4] \n\t" + "swc1 %[f2], 0(%[yf0]) \n\t" + "swc1 %[f3], 0(%[yf1]) \n\t" + "swc1 %[f6], 4(%[yf0]) \n\t" + "swc1 %[f5], 4(%[yf1]) \n\t" + "addiu %[yf0], %[yf0], 8 \n\t" + "bgtz %[len], 1b \n\t" + " addiu %[yf1], %[yf1], 8 \n\t" + "lwc1 %[f0], 0(%[aRe]) \n\t" + "lwc1 %[f1], 0(%[bRe]) \n\t" + "lwc1 %[f2], 0(%[bIm]) \n\t" + "lwc1 %[f3], 0(%[aIm]) \n\t" + "mul.s %[f8], %[f0], %[f1] \n\t" + "mul.s %[f0], %[f0], %[f2] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[f12], %[f2], %[f3] \n\t" + "mul.s %[f1], %[f3], %[f1] \n\t" + "sub.s %[f8], %[f8], %[f12] \n\t" + "lwc1 %[f2], 0(%[yf0]) \n\t" + "add.s %[f1], %[f0], %[f1] \n\t" + "lwc1 %[f3], 0(%[yf1]) \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmsub.s %[f8], %[f8], %[f2], %[f3] \n\t" + "lwc1 %[f2], 0(%[yf0]) \n\t" + "madd.s %[f1], %[f0], %[f3], %[f1] \n\t" + "lwc1 %[f3], 0(%[yf1]) \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "add.s %[f2], %[f2], %[f8] \n\t" + "add.s %[f3], %[f3], %[f1] \n\t" + "swc1 %[f2], 0(%[yf0]) \n\t" + "swc1 %[f3], 0(%[yf1]) \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), + [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5), + [f6] "=&f" (f6), [f7] "=&f" (f7), [f8] "=&f" (f8), + [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11), + [f12] "=&f" (f12), [f13] "=&f" (f13), [aRe] "+r" (aRe), + [aIm] "+r" (aIm), [bRe] "+r" (bRe), [bIm] "+r" (bIm), + [yf0] "+r" (yf0), [yf1] "+r" (yf1), [len] "+r" (len) + : [len1] "r" (len1) + : "memory" + ); + } +} + +void WebRtcAec_FilterAdaptation_mips(AecCore *aec, + float *fft, + float ef[2][PART_LEN1]) { + int i; + for (i = 0; i < aec->num_partitions; i++) { + int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1); + int pos; + // Check for wrap + if (i + aec->xfBufBlockPos >= aec->num_partitions) { + xPos -= aec->num_partitions * PART_LEN1; + } + + pos = i * PART_LEN1; + float *aRe = aec->xfBuf[0] + xPos; + float *aIm = aec->xfBuf[1] + xPos; + float *bRe = ef[0]; + float *bIm = ef[1]; + float *fft_tmp = fft; + + float f0, f1, f2, f3, f4, f5, f6 ,f7, f8, f9, f10, f11, f12; + int len = PART_LEN >> 1; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "lwc1 %[f0], 0(%[aRe]) \n\t" + "lwc1 %[f1], 0(%[bRe]) \n\t" + "lwc1 %[f2], 0(%[bIm]) \n\t" + "lwc1 %[f4], 4(%[aRe]) \n\t" + "lwc1 %[f5], 4(%[bRe]) \n\t" + "lwc1 %[f6], 4(%[bIm]) \n\t" + "addiu %[aRe], %[aRe], 8 \n\t" + "addiu %[bRe], %[bRe], 8 \n\t" + "mul.s %[f8], %[f0], %[f1] \n\t" + "mul.s %[f0], %[f0], %[f2] \n\t" + "lwc1 %[f3], 0(%[aIm]) \n\t" + "mul.s %[f9], %[f4], %[f5] \n\t" + "lwc1 %[f7], 4(%[aIm]) \n\t" + "mul.s %[f4], %[f4], %[f6] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[f10], %[f3], %[f2] \n\t" + "mul.s %[f1], %[f3], %[f1] \n\t" + "mul.s %[f11], %[f7], %[f6] \n\t" + "mul.s %[f5], %[f7], %[f5] \n\t" + "addiu %[aIm], %[aIm], 8 \n\t" + "addiu %[bIm], %[bIm], 8 \n\t" + "addiu %[len], %[len], -1 \n\t" + "add.s %[f8], %[f8], %[f10] \n\t" + "sub.s %[f1], %[f0], %[f1] \n\t" + "add.s %[f9], %[f9], %[f11] \n\t" + "sub.s %[f5], %[f4], %[f5] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "addiu %[aIm], %[aIm], 8 \n\t" + "addiu %[bIm], %[bIm], 8 \n\t" + "addiu %[len], %[len], -1 \n\t" + "madd.s %[f8], %[f8], %[f3], %[f2] \n\t" + "nmsub.s %[f1], %[f0], %[f3], %[f1] \n\t" + "madd.s %[f9], %[f9], %[f7], %[f6] \n\t" + "nmsub.s %[f5], %[f4], %[f7], %[f5] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[f8], 0(%[fft_tmp]) \n\t" + "swc1 %[f1], 4(%[fft_tmp]) \n\t" + "swc1 %[f9], 8(%[fft_tmp]) \n\t" + "swc1 %[f5], 12(%[fft_tmp]) \n\t" + "bgtz %[len], 1b \n\t" + " addiu %[fft_tmp], %[fft_tmp], 16 \n\t" + "lwc1 %[f0], 0(%[aRe]) \n\t" + "lwc1 %[f1], 0(%[bRe]) \n\t" + "lwc1 %[f2], 0(%[bIm]) \n\t" + "lwc1 %[f3], 0(%[aIm]) \n\t" + "mul.s %[f8], %[f0], %[f1] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[f10], %[f3], %[f2] \n\t" + "add.s %[f8], %[f8], %[f10] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[f8], %[f8], %[f3], %[f2] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[f8], 4(%[fft]) \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), + [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5), + [f6] "=&f" (f6), [f7] "=&f" (f7), [f8] "=&f" (f8), + [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11), + [f12] "=&f" (f12), [aRe] "+r" (aRe), [aIm] "+r" (aIm), + [bRe] "+r" (bRe), [bIm] "+r" (bIm), [fft_tmp] "+r" (fft_tmp), + [len] "+r" (len), [fft] "=&r" (fft) + : + : "memory" + ); + + aec_rdft_inverse_128(fft); + memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN); + + // fft scaling + { + float scale = 2.0f / PART_LEN2; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[fft_tmp], %[fft], 0 \n\t" + "addiu %[len], $zero, 8 \n\t" + "1: \n\t" + "addiu %[len], %[len], -1 \n\t" + "lwc1 %[f0], 0(%[fft_tmp]) \n\t" + "lwc1 %[f1], 4(%[fft_tmp]) \n\t" + "lwc1 %[f2], 8(%[fft_tmp]) \n\t" + "lwc1 %[f3], 12(%[fft_tmp]) \n\t" + "mul.s %[f0], %[f0], %[scale] \n\t" + "mul.s %[f1], %[f1], %[scale] \n\t" + "mul.s %[f2], %[f2], %[scale] \n\t" + "mul.s %[f3], %[f3], %[scale] \n\t" + "lwc1 %[f4], 16(%[fft_tmp]) \n\t" + "lwc1 %[f5], 20(%[fft_tmp]) \n\t" + "lwc1 %[f6], 24(%[fft_tmp]) \n\t" + "lwc1 %[f7], 28(%[fft_tmp]) \n\t" + "mul.s %[f4], %[f4], %[scale] \n\t" + "mul.s %[f5], %[f5], %[scale] \n\t" + "mul.s %[f6], %[f6], %[scale] \n\t" + "mul.s %[f7], %[f7], %[scale] \n\t" + "swc1 %[f0], 0(%[fft_tmp]) \n\t" + "swc1 %[f1], 4(%[fft_tmp]) \n\t" + "swc1 %[f2], 8(%[fft_tmp]) \n\t" + "swc1 %[f3], 12(%[fft_tmp]) \n\t" + "swc1 %[f4], 16(%[fft_tmp]) \n\t" + "swc1 %[f5], 20(%[fft_tmp]) \n\t" + "swc1 %[f6], 24(%[fft_tmp]) \n\t" + "swc1 %[f7], 28(%[fft_tmp]) \n\t" + "bgtz %[len], 1b \n\t" + " addiu %[fft_tmp], %[fft_tmp], 32 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), + [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5), + [f6] "=&f" (f6), [f7] "=&f" (f7), [len] "=&r" (len), + [fft_tmp] "=&r" (fft_tmp) + : [scale] "f" (scale), [fft] "r" (fft) + : "memory" + ); + } + aec_rdft_forward_128(fft); + aRe = aec->wfBuf[0] + pos; + aIm = aec->wfBuf[1] + pos; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[fft_tmp], %[fft], 0 \n\t" + "addiu %[len], $zero, 31 \n\t" + "lwc1 %[f0], 0(%[aRe]) \n\t" + "lwc1 %[f1], 0(%[fft_tmp]) \n\t" + "lwc1 %[f2], 256(%[aRe]) \n\t" + "lwc1 %[f3], 4(%[fft_tmp]) \n\t" + "lwc1 %[f4], 4(%[aRe]) \n\t" + "lwc1 %[f5], 8(%[fft_tmp]) \n\t" + "lwc1 %[f6], 4(%[aIm]) \n\t" + "lwc1 %[f7], 12(%[fft_tmp]) \n\t" + "add.s %[f0], %[f0], %[f1] \n\t" + "add.s %[f2], %[f2], %[f3] \n\t" + "add.s %[f4], %[f4], %[f5] \n\t" + "add.s %[f6], %[f6], %[f7] \n\t" + "addiu %[fft_tmp], %[fft_tmp], 16 \n\t" + "swc1 %[f0], 0(%[aRe]) \n\t" + "swc1 %[f2], 256(%[aRe]) \n\t" + "swc1 %[f4], 4(%[aRe]) \n\t" + "addiu %[aRe], %[aRe], 8 \n\t" + "swc1 %[f6], 4(%[aIm]) \n\t" + "addiu %[aIm], %[aIm], 8 \n\t" + "1: \n\t" + "lwc1 %[f0], 0(%[aRe]) \n\t" + "lwc1 %[f1], 0(%[fft_tmp]) \n\t" + "lwc1 %[f2], 0(%[aIm]) \n\t" + "lwc1 %[f3], 4(%[fft_tmp]) \n\t" + "lwc1 %[f4], 4(%[aRe]) \n\t" + "lwc1 %[f5], 8(%[fft_tmp]) \n\t" + "lwc1 %[f6], 4(%[aIm]) \n\t" + "lwc1 %[f7], 12(%[fft_tmp]) \n\t" + "add.s %[f0], %[f0], %[f1] \n\t" + "add.s %[f2], %[f2], %[f3] \n\t" + "add.s %[f4], %[f4], %[f5] \n\t" + "add.s %[f6], %[f6], %[f7] \n\t" + "addiu %[len], %[len], -1 \n\t" + "addiu %[fft_tmp], %[fft_tmp], 16 \n\t" + "swc1 %[f0], 0(%[aRe]) \n\t" + "swc1 %[f2], 0(%[aIm]) \n\t" + "swc1 %[f4], 4(%[aRe]) \n\t" + "addiu %[aRe], %[aRe], 8 \n\t" + "swc1 %[f6], 4(%[aIm]) \n\t" + "bgtz %[len], 1b \n\t" + " addiu %[aIm], %[aIm], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), + [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5), + [f6] "=&f" (f6), [f7] "=&f" (f7), [len] "=&r" (len), + [fft_tmp] "=&r" (fft_tmp) + : [aRe] "r" (aRe), [aIm] "r" (aIm), [fft] "r" (fft) + : "memory" + ); + } +} + +void WebRtcAec_OverdriveAndSuppress_mips(AecCore *aec, + float hNl[PART_LEN1], + const float hNlFb, + float efw[2][PART_LEN1]) { + int i; + const float one = 1.0; + float *p_hNl, *p_efw0, *p_efw1; + float *p_WebRtcAec_wC; + float temp1, temp2, temp3, temp4; + + p_hNl = &hNl[0]; + p_efw0 = &efw[0][0]; + p_efw1 = &efw[1][0]; + p_WebRtcAec_wC = (float*)&WebRtcAec_weightCurve[0]; + + for (i = 0; i < PART_LEN1; i++) { + // Weight subbands + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lwc1 %[temp1], 0(%[p_hNl]) \n\t" + "lwc1 %[temp2], 0(%[p_wC]) \n\t" + "c.lt.s %[hNlFb], %[temp1] \n\t" + "bc1f 1f \n\t" + " mul.s %[temp3], %[temp2], %[hNlFb] \n\t" + "sub.s %[temp4], %[one], %[temp2] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[temp1], %[temp1], %[temp4] \n\t" + "add.s %[temp1], %[temp3], %[temp1] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[temp1], %[temp3], %[temp1], %[temp4] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[temp1], 0(%[p_hNl]) \n\t" + "1: \n\t" + "addiu %[p_wC], %[p_wC], 4 \n\t" + ".set pop \n\t" + : [temp1] "=&f" (temp1), [temp2] "=&f" (temp2), [temp3] "=&f" (temp3), + [temp4] "=&f" (temp4), [p_wC] "+r" (p_WebRtcAec_wC) + : [hNlFb] "f" (hNlFb), [one] "f" (one), [p_hNl] "r" (p_hNl) + : "memory" + ); + + hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]); + + __asm __volatile ( + "lwc1 %[temp1], 0(%[p_hNl]) \n\t" + "lwc1 %[temp3], 0(%[p_efw1]) \n\t" + "lwc1 %[temp2], 0(%[p_efw0]) \n\t" + "addiu %[p_hNl], %[p_hNl], 4 \n\t" + "mul.s %[temp3], %[temp3], %[temp1] \n\t" + "mul.s %[temp2], %[temp2], %[temp1] \n\t" + "addiu %[p_efw0], %[p_efw0], 4 \n\t" + "addiu %[p_efw1], %[p_efw1], 4 \n\t" + "neg.s %[temp4], %[temp3] \n\t" + "swc1 %[temp2], -4(%[p_efw0]) \n\t" + "swc1 %[temp4], -4(%[p_efw1]) \n\t" + : [temp1] "=&f" (temp1), [temp2] "=&f" (temp2), [temp3] "=&f" (temp3), + [temp4] "=&f" (temp4), [p_efw0] "+r" (p_efw0), [p_efw1] "+r" (p_efw1), + [p_hNl] "+r" (p_hNl) + : + : "memory" + ); + } +} + +void WebRtcAec_ScaleErrorSignal_mips(AecCore *aec, float ef[2][PART_LEN1]) { + const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu; + const float error_threshold = aec->extended_filter_enabled + ? kExtendedErrorThreshold + : aec->normal_error_threshold; + int len = (PART_LEN1); + float *ef0 = ef[0]; + float *ef1 = ef[1]; + float *xPow = aec->xPow; + float fac1 = 1e-10f; + float err_th2 = error_threshold * error_threshold; + float f0, f1, f2; +#if !defined(MIPS32_R2_LE) + float f3; +#endif + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "lwc1 %[f0], 0(%[xPow]) \n\t" + "lwc1 %[f1], 0(%[ef0]) \n\t" + "lwc1 %[f2], 0(%[ef1]) \n\t" + "add.s %[f0], %[f0], %[fac1] \n\t" + "div.s %[f1], %[f1], %[f0] \n\t" + "div.s %[f2], %[f2], %[f0] \n\t" + "mul.s %[f0], %[f1], %[f1] \n\t" +#if defined(MIPS32_R2_LE) + "madd.s %[f0], %[f0], %[f2], %[f2] \n\t" +#else + "mul.s %[f3], %[f2], %[f2] \n\t" + "add.s %[f0], %[f0], %[f3] \n\t" +#endif + "c.le.s %[f0], %[err_th2] \n\t" + "nop \n\t" + "bc1t 2f \n\t" + " nop \n\t" + "sqrt.s %[f0], %[f0] \n\t" + "add.s %[f0], %[f0], %[fac1] \n\t" + "div.s %[f0], %[err_th], %[f0] \n\t" + "mul.s %[f1], %[f1], %[f0] \n\t" + "mul.s %[f2], %[f2], %[f0] \n\t" + "2: \n\t" + "mul.s %[f1], %[f1], %[mu] \n\t" + "mul.s %[f2], %[f2], %[mu] \n\t" + "swc1 %[f1], 0(%[ef0]) \n\t" + "swc1 %[f2], 0(%[ef1]) \n\t" + "addiu %[len], %[len], -1 \n\t" + "addiu %[xPow], %[xPow], 4 \n\t" + "addiu %[ef0], %[ef0], 4 \n\t" + "bgtz %[len], 1b \n\t" + " addiu %[ef1], %[ef1], 4 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), +#if !defined(MIPS32_R2_LE) + [f3] "=&f" (f3), +#endif + [xPow] "+r" (xPow), [ef0] "+r" (ef0), [ef1] "+r" (ef1), + [len] "+r" (len) + : [fac1] "f" (fac1), [err_th2] "f" (err_th2), [mu] "f" (mu), + [err_th] "f" (error_threshold) + : "memory" + ); +} + +void WebRtcAec_InitAec_mips(void) +{ + WebRtcAec_FilterFar = WebRtcAec_FilterFar_mips; + WebRtcAec_FilterAdaptation = WebRtcAec_FilterAdaptation_mips; + WebRtcAec_ScaleErrorSignal = WebRtcAec_ScaleErrorSignal_mips; + WebRtcAec_ComfortNoise = WebRtcAec_ComfortNoise_mips; + WebRtcAec_OverdriveAndSuppress = WebRtcAec_OverdriveAndSuppress_mips; +} + diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.c new file mode 100644 index 00000000000..cec0a7e3379 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.c @@ -0,0 +1,304 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/* + * The core AEC algorithm, neon version of speed-critical functions. + * + * Based on aec_core_sse2.c. + */ + +#include "webrtc/modules/audio_processing/aec/aec_core.h" + +#include <arm_neon.h> +#include <math.h> +#include <string.h> // memset + +#include "webrtc/modules/audio_processing/aec/aec_core_internal.h" +#include "webrtc/modules/audio_processing/aec/aec_rdft.h" + +enum { kShiftExponentIntoTopMantissa = 8 }; +enum { kFloatExponentShift = 23 }; + +__inline static float MulRe(float aRe, float aIm, float bRe, float bIm) { + return aRe * bRe - aIm * bIm; +} + +static void FilterAdaptationNEON(AecCore* aec, + float* fft, + float ef[2][PART_LEN1]) { + int i; + const int num_partitions = aec->num_partitions; + for (i = 0; i < num_partitions; i++) { + int xPos = (i + aec->xfBufBlockPos) * PART_LEN1; + int pos = i * PART_LEN1; + int j; + // Check for wrap + if (i + aec->xfBufBlockPos >= num_partitions) { + xPos -= num_partitions * PART_LEN1; + } + + // Process the whole array... + for (j = 0; j < PART_LEN; j += 4) { + // Load xfBuf and ef. + const float32x4_t xfBuf_re = vld1q_f32(&aec->xfBuf[0][xPos + j]); + const float32x4_t xfBuf_im = vld1q_f32(&aec->xfBuf[1][xPos + j]); + const float32x4_t ef_re = vld1q_f32(&ef[0][j]); + const float32x4_t ef_im = vld1q_f32(&ef[1][j]); + // Calculate the product of conjugate(xfBuf) by ef. + // re(conjugate(a) * b) = aRe * bRe + aIm * bIm + // im(conjugate(a) * b)= aRe * bIm - aIm * bRe + const float32x4_t a = vmulq_f32(xfBuf_re, ef_re); + const float32x4_t e = vmlaq_f32(a, xfBuf_im, ef_im); + const float32x4_t c = vmulq_f32(xfBuf_re, ef_im); + const float32x4_t f = vmlsq_f32(c, xfBuf_im, ef_re); + // Interleave real and imaginary parts. + const float32x4x2_t g_n_h = vzipq_f32(e, f); + // Store + vst1q_f32(&fft[2 * j + 0], g_n_h.val[0]); + vst1q_f32(&fft[2 * j + 4], g_n_h.val[1]); + } + // ... and fixup the first imaginary entry. + fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN], + -aec->xfBuf[1][xPos + PART_LEN], + ef[0][PART_LEN], + ef[1][PART_LEN]); + + aec_rdft_inverse_128(fft); + memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN); + + // fft scaling + { + const float scale = 2.0f / PART_LEN2; + const float32x4_t scale_ps = vmovq_n_f32(scale); + for (j = 0; j < PART_LEN; j += 4) { + const float32x4_t fft_ps = vld1q_f32(&fft[j]); + const float32x4_t fft_scale = vmulq_f32(fft_ps, scale_ps); + vst1q_f32(&fft[j], fft_scale); + } + } + aec_rdft_forward_128(fft); + + { + const float wt1 = aec->wfBuf[1][pos]; + aec->wfBuf[0][pos + PART_LEN] += fft[1]; + for (j = 0; j < PART_LEN; j += 4) { + float32x4_t wtBuf_re = vld1q_f32(&aec->wfBuf[0][pos + j]); + float32x4_t wtBuf_im = vld1q_f32(&aec->wfBuf[1][pos + j]); + const float32x4_t fft0 = vld1q_f32(&fft[2 * j + 0]); + const float32x4_t fft4 = vld1q_f32(&fft[2 * j + 4]); + const float32x4x2_t fft_re_im = vuzpq_f32(fft0, fft4); + wtBuf_re = vaddq_f32(wtBuf_re, fft_re_im.val[0]); + wtBuf_im = vaddq_f32(wtBuf_im, fft_re_im.val[1]); + + vst1q_f32(&aec->wfBuf[0][pos + j], wtBuf_re); + vst1q_f32(&aec->wfBuf[1][pos + j], wtBuf_im); + } + aec->wfBuf[1][pos] = wt1; + } + } +} + +extern const float WebRtcAec_weightCurve[65]; +extern const float WebRtcAec_overDriveCurve[65]; + +static float32x4_t vpowq_f32(float32x4_t a, float32x4_t b) { + // a^b = exp2(b * log2(a)) + // exp2(x) and log2(x) are calculated using polynomial approximations. + float32x4_t log2_a, b_log2_a, a_exp_b; + + // Calculate log2(x), x = a. + { + // To calculate log2(x), we decompose x like this: + // x = y * 2^n + // n is an integer + // y is in the [1.0, 2.0) range + // + // log2(x) = log2(y) + n + // n can be evaluated by playing with float representation. + // log2(y) in a small range can be approximated, this code uses an order + // five polynomial approximation. The coefficients have been + // estimated with the Remez algorithm and the resulting + // polynomial has a maximum relative error of 0.00086%. + + // Compute n. + // This is done by masking the exponent, shifting it into the top bit of + // the mantissa, putting eight into the biased exponent (to shift/ + // compensate the fact that the exponent has been shifted in the top/ + // fractional part and finally getting rid of the implicit leading one + // from the mantissa by substracting it out. + const uint32x4_t vec_float_exponent_mask = vdupq_n_u32(0x7F800000); + const uint32x4_t vec_eight_biased_exponent = vdupq_n_u32(0x43800000); + const uint32x4_t vec_implicit_leading_one = vdupq_n_u32(0x43BF8000); + const uint32x4_t two_n = vandq_u32(vreinterpretq_u32_f32(a), + vec_float_exponent_mask); + const uint32x4_t n_1 = vshrq_n_u32(two_n, kShiftExponentIntoTopMantissa); + const uint32x4_t n_0 = vorrq_u32(n_1, vec_eight_biased_exponent); + const float32x4_t n = + vsubq_f32(vreinterpretq_f32_u32(n_0), + vreinterpretq_f32_u32(vec_implicit_leading_one)); + // Compute y. + const uint32x4_t vec_mantissa_mask = vdupq_n_u32(0x007FFFFF); + const uint32x4_t vec_zero_biased_exponent_is_one = vdupq_n_u32(0x3F800000); + const uint32x4_t mantissa = vandq_u32(vreinterpretq_u32_f32(a), + vec_mantissa_mask); + const float32x4_t y = + vreinterpretq_f32_u32(vorrq_u32(mantissa, + vec_zero_biased_exponent_is_one)); + // Approximate log2(y) ~= (y - 1) * pol5(y). + // pol5(y) = C5 * y^5 + C4 * y^4 + C3 * y^3 + C2 * y^2 + C1 * y + C0 + const float32x4_t C5 = vdupq_n_f32(-3.4436006e-2f); + const float32x4_t C4 = vdupq_n_f32(3.1821337e-1f); + const float32x4_t C3 = vdupq_n_f32(-1.2315303f); + const float32x4_t C2 = vdupq_n_f32(2.5988452f); + const float32x4_t C1 = vdupq_n_f32(-3.3241990f); + const float32x4_t C0 = vdupq_n_f32(3.1157899f); + float32x4_t pol5_y = C5; + pol5_y = vmlaq_f32(C4, y, pol5_y); + pol5_y = vmlaq_f32(C3, y, pol5_y); + pol5_y = vmlaq_f32(C2, y, pol5_y); + pol5_y = vmlaq_f32(C1, y, pol5_y); + pol5_y = vmlaq_f32(C0, y, pol5_y); + const float32x4_t y_minus_one = + vsubq_f32(y, vreinterpretq_f32_u32(vec_zero_biased_exponent_is_one)); + const float32x4_t log2_y = vmulq_f32(y_minus_one, pol5_y); + + // Combine parts. + log2_a = vaddq_f32(n, log2_y); + } + + // b * log2(a) + b_log2_a = vmulq_f32(b, log2_a); + + // Calculate exp2(x), x = b * log2(a). + { + // To calculate 2^x, we decompose x like this: + // x = n + y + // n is an integer, the value of x - 0.5 rounded down, therefore + // y is in the [0.5, 1.5) range + // + // 2^x = 2^n * 2^y + // 2^n can be evaluated by playing with float representation. + // 2^y in a small range can be approximated, this code uses an order two + // polynomial approximation. The coefficients have been estimated + // with the Remez algorithm and the resulting polynomial has a + // maximum relative error of 0.17%. + // To avoid over/underflow, we reduce the range of input to ]-127, 129]. + const float32x4_t max_input = vdupq_n_f32(129.f); + const float32x4_t min_input = vdupq_n_f32(-126.99999f); + const float32x4_t x_min = vminq_f32(b_log2_a, max_input); + const float32x4_t x_max = vmaxq_f32(x_min, min_input); + // Compute n. + const float32x4_t half = vdupq_n_f32(0.5f); + const float32x4_t x_minus_half = vsubq_f32(x_max, half); + const int32x4_t x_minus_half_floor = vcvtq_s32_f32(x_minus_half); + + // Compute 2^n. + const int32x4_t float_exponent_bias = vdupq_n_s32(127); + const int32x4_t two_n_exponent = + vaddq_s32(x_minus_half_floor, float_exponent_bias); + const float32x4_t two_n = + vreinterpretq_f32_s32(vshlq_n_s32(two_n_exponent, kFloatExponentShift)); + // Compute y. + const float32x4_t y = vsubq_f32(x_max, vcvtq_f32_s32(x_minus_half_floor)); + + // Approximate 2^y ~= C2 * y^2 + C1 * y + C0. + const float32x4_t C2 = vdupq_n_f32(3.3718944e-1f); + const float32x4_t C1 = vdupq_n_f32(6.5763628e-1f); + const float32x4_t C0 = vdupq_n_f32(1.0017247f); + float32x4_t exp2_y = C2; + exp2_y = vmlaq_f32(C1, y, exp2_y); + exp2_y = vmlaq_f32(C0, y, exp2_y); + + // Combine parts. + a_exp_b = vmulq_f32(exp2_y, two_n); + } + + return a_exp_b; +} + +static void OverdriveAndSuppressNEON(AecCore* aec, + float hNl[PART_LEN1], + const float hNlFb, + float efw[2][PART_LEN1]) { + int i; + const float32x4_t vec_hNlFb = vmovq_n_f32(hNlFb); + const float32x4_t vec_one = vdupq_n_f32(1.0f); + const float32x4_t vec_minus_one = vdupq_n_f32(-1.0f); + const float32x4_t vec_overDriveSm = vmovq_n_f32(aec->overDriveSm); + + // vectorized code (four at once) + for (i = 0; i + 3 < PART_LEN1; i += 4) { + // Weight subbands + float32x4_t vec_hNl = vld1q_f32(&hNl[i]); + const float32x4_t vec_weightCurve = vld1q_f32(&WebRtcAec_weightCurve[i]); + const uint32x4_t bigger = vcgtq_f32(vec_hNl, vec_hNlFb); + const float32x4_t vec_weightCurve_hNlFb = vmulq_f32(vec_weightCurve, + vec_hNlFb); + const float32x4_t vec_one_weightCurve = vsubq_f32(vec_one, vec_weightCurve); + const float32x4_t vec_one_weightCurve_hNl = vmulq_f32(vec_one_weightCurve, + vec_hNl); + const uint32x4_t vec_if0 = vandq_u32(vmvnq_u32(bigger), + vreinterpretq_u32_f32(vec_hNl)); + const float32x4_t vec_one_weightCurve_add = + vaddq_f32(vec_weightCurve_hNlFb, vec_one_weightCurve_hNl); + const uint32x4_t vec_if1 = + vandq_u32(bigger, vreinterpretq_u32_f32(vec_one_weightCurve_add)); + + vec_hNl = vreinterpretq_f32_u32(vorrq_u32(vec_if0, vec_if1)); + + { + const float32x4_t vec_overDriveCurve = + vld1q_f32(&WebRtcAec_overDriveCurve[i]); + const float32x4_t vec_overDriveSm_overDriveCurve = + vmulq_f32(vec_overDriveSm, vec_overDriveCurve); + vec_hNl = vpowq_f32(vec_hNl, vec_overDriveSm_overDriveCurve); + vst1q_f32(&hNl[i], vec_hNl); + } + + // Suppress error signal + { + float32x4_t vec_efw_re = vld1q_f32(&efw[0][i]); + float32x4_t vec_efw_im = vld1q_f32(&efw[1][i]); + vec_efw_re = vmulq_f32(vec_efw_re, vec_hNl); + vec_efw_im = vmulq_f32(vec_efw_im, vec_hNl); + + // Ooura fft returns incorrect sign on imaginary component. It matters + // here because we are making an additive change with comfort noise. + vec_efw_im = vmulq_f32(vec_efw_im, vec_minus_one); + vst1q_f32(&efw[0][i], vec_efw_re); + vst1q_f32(&efw[1][i], vec_efw_im); + } + } + + // scalar code for the remaining items. + for (; i < PART_LEN1; i++) { + // Weight subbands + if (hNl[i] > hNlFb) { + hNl[i] = WebRtcAec_weightCurve[i] * hNlFb + + (1 - WebRtcAec_weightCurve[i]) * hNl[i]; + } + + hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]); + + // Suppress error signal + efw[0][i] *= hNl[i]; + efw[1][i] *= hNl[i]; + + // Ooura fft returns incorrect sign on imaginary component. It matters + // here because we are making an additive change with comfort noise. + efw[1][i] *= -1; + } +} + +void WebRtcAec_InitAec_neon(void) { + WebRtcAec_FilterAdaptation = FilterAdaptationNEON; + WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressNEON; +} + diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.c index a19e8877bbc..7731b37b224 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.c +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.c @@ -116,7 +116,7 @@ static void bitrv2_32(int* ip, float* a) { } } -static void bitrv2_128(float* a) { +static void bitrv2_128_C(float* a) { /* Following things have been attempted but are no faster: (a) Storing the swap indexes in a LUT (index calculations are done @@ -512,7 +512,7 @@ static void cftmdl_128_C(float* a) { } } -static void cftfsub_128(float* a) { +static void cftfsub_128_C(float* a) { int j, j1, j2, j3, l; float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i; @@ -542,7 +542,7 @@ static void cftfsub_128(float* a) { } } -static void cftbsub_128(float* a) { +static void cftbsub_128_C(float* a) { int j, j1, j2, j3, l; float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i; @@ -640,17 +640,26 @@ rft_sub_128_t cft1st_128; rft_sub_128_t cftmdl_128; rft_sub_128_t rftfsub_128; rft_sub_128_t rftbsub_128; +rft_sub_128_t cftfsub_128; +rft_sub_128_t cftbsub_128; +rft_sub_128_t bitrv2_128; void aec_rdft_init(void) { cft1st_128 = cft1st_128_C; cftmdl_128 = cftmdl_128_C; rftfsub_128 = rftfsub_128_C; rftbsub_128 = rftbsub_128_C; + cftfsub_128 = cftfsub_128_C; + cftbsub_128 = cftbsub_128_C; + bitrv2_128 = bitrv2_128_C; #if defined(WEBRTC_ARCH_X86_FAMILY) if (WebRtc_GetCPUInfo(kSSE2)) { aec_rdft_init_sse2(); } #endif +#if defined(MIPS_FPU_LE) + aec_rdft_init_mips(); +#endif // init library constants. makewt_32(); makect_32(); diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h index 3380633ce6c..795c57d44c2 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h @@ -33,13 +33,13 @@ extern float rdft_w[64]; extern float rdft_wk3ri_first[32]; extern float rdft_wk3ri_second[32]; // constants used by SSE2 but initialized in C path. -extern float rdft_wk1r[32]; -extern float rdft_wk2r[32]; -extern float rdft_wk3r[32]; -extern float rdft_wk1i[32]; -extern float rdft_wk2i[32]; -extern float rdft_wk3i[32]; -extern float cftmdl_wk1r[4]; +extern ALIGN16_BEG float ALIGN16_END rdft_wk1r[32]; +extern ALIGN16_BEG float ALIGN16_END rdft_wk2r[32]; +extern ALIGN16_BEG float ALIGN16_END rdft_wk3r[32]; +extern ALIGN16_BEG float ALIGN16_END rdft_wk1i[32]; +extern ALIGN16_BEG float ALIGN16_END rdft_wk2i[32]; +extern ALIGN16_BEG float ALIGN16_END rdft_wk3i[32]; +extern ALIGN16_BEG float ALIGN16_END cftmdl_wk1r[4]; // code path selection function pointers typedef void (*rft_sub_128_t)(float* a); @@ -47,6 +47,9 @@ extern rft_sub_128_t rftfsub_128; extern rft_sub_128_t rftbsub_128; extern rft_sub_128_t cft1st_128; extern rft_sub_128_t cftmdl_128; +extern rft_sub_128_t cftfsub_128; +extern rft_sub_128_t cftbsub_128; +extern rft_sub_128_t bitrv2_128; // entry points void aec_rdft_init(void); @@ -54,4 +57,8 @@ void aec_rdft_init_sse2(void); void aec_rdft_forward_128(float* a); void aec_rdft_inverse_128(float* a); +#if defined(MIPS_FPU_LE) +void aec_rdft_init_mips(void); +#endif + #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.c new file mode 100644 index 00000000000..a0dac5f135c --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.c @@ -0,0 +1,1213 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_processing/aec/aec_rdft.h" +#include "webrtc/typedefs.h" + +static void bitrv2_128_mips(float *a) { + // n is 128 + float xr, xi, yr, yi; + + xr = a[8]; + xi = a[9]; + yr = a[16]; + yi = a[17]; + a[8] = yr; + a[9] = yi; + a[16] = xr; + a[17] = xi; + + xr = a[64]; + xi = a[65]; + yr = a[2]; + yi = a[3]; + a[64] = yr; + a[65] = yi; + a[2] = xr; + a[3] = xi; + + xr = a[72]; + xi = a[73]; + yr = a[18]; + yi = a[19]; + a[72] = yr; + a[73] = yi; + a[18] = xr; + a[19] = xi; + + xr = a[80]; + xi = a[81]; + yr = a[10]; + yi = a[11]; + a[80] = yr; + a[81] = yi; + a[10] = xr; + a[11] = xi; + + xr = a[88]; + xi = a[89]; + yr = a[26]; + yi = a[27]; + a[88] = yr; + a[89] = yi; + a[26] = xr; + a[27] = xi; + + xr = a[74]; + xi = a[75]; + yr = a[82]; + yi = a[83]; + a[74] = yr; + a[75] = yi; + a[82] = xr; + a[83] = xi; + + xr = a[32]; + xi = a[33]; + yr = a[4]; + yi = a[5]; + a[32] = yr; + a[33] = yi; + a[4] = xr; + a[5] = xi; + + xr = a[40]; + xi = a[41]; + yr = a[20]; + yi = a[21]; + a[40] = yr; + a[41] = yi; + a[20] = xr; + a[21] = xi; + + xr = a[48]; + xi = a[49]; + yr = a[12]; + yi = a[13]; + a[48] = yr; + a[49] = yi; + a[12] = xr; + a[13] = xi; + + xr = a[56]; + xi = a[57]; + yr = a[28]; + yi = a[29]; + a[56] = yr; + a[57] = yi; + a[28] = xr; + a[29] = xi; + + xr = a[34]; + xi = a[35]; + yr = a[68]; + yi = a[69]; + a[34] = yr; + a[35] = yi; + a[68] = xr; + a[69] = xi; + + xr = a[42]; + xi = a[43]; + yr = a[84]; + yi = a[85]; + a[42] = yr; + a[43] = yi; + a[84] = xr; + a[85] = xi; + + xr = a[50]; + xi = a[51]; + yr = a[76]; + yi = a[77]; + a[50] = yr; + a[51] = yi; + a[76] = xr; + a[77] = xi; + + xr = a[58]; + xi = a[59]; + yr = a[92]; + yi = a[93]; + a[58] = yr; + a[59] = yi; + a[92] = xr; + a[93] = xi; + + xr = a[44]; + xi = a[45]; + yr = a[52]; + yi = a[53]; + a[44] = yr; + a[45] = yi; + a[52] = xr; + a[53] = xi; + + xr = a[96]; + xi = a[97]; + yr = a[6]; + yi = a[7]; + a[96] = yr; + a[97] = yi; + a[6] = xr; + a[7] = xi; + + xr = a[104]; + xi = a[105]; + yr = a[22]; + yi = a[23]; + a[104] = yr; + a[105] = yi; + a[22] = xr; + a[23] = xi; + + xr = a[112]; + xi = a[113]; + yr = a[14]; + yi = a[15]; + a[112] = yr; + a[113] = yi; + a[14] = xr; + a[15] = xi; + + xr = a[120]; + xi = a[121]; + yr = a[30]; + yi = a[31]; + a[120] = yr; + a[121] = yi; + a[30] = xr; + a[31] = xi; + + xr = a[98]; + xi = a[99]; + yr = a[70]; + yi = a[71]; + a[98] = yr; + a[99] = yi; + a[70] = xr; + a[71] = xi; + + xr = a[106]; + xi = a[107]; + yr = a[86]; + yi = a[87]; + a[106] = yr; + a[107] = yi; + a[86] = xr; + a[87] = xi; + + xr = a[114]; + xi = a[115]; + yr = a[78]; + yi = a[79]; + a[114] = yr; + a[115] = yi; + a[78] = xr; + a[79] = xi; + + xr = a[122]; + xi = a[123]; + yr = a[94]; + yi = a[95]; + a[122] = yr; + a[123] = yi; + a[94] = xr; + a[95] = xi; + + xr = a[100]; + xi = a[101]; + yr = a[38]; + yi = a[39]; + a[100] = yr; + a[101] = yi; + a[38] = xr; + a[39] = xi; + + xr = a[108]; + xi = a[109]; + yr = a[54]; + yi = a[55]; + a[108] = yr; + a[109] = yi; + a[54] = xr; + a[55] = xi; + + xr = a[116]; + xi = a[117]; + yr = a[46]; + yi = a[47]; + a[116] = yr; + a[117] = yi; + a[46] = xr; + a[47] = xi; + + xr = a[124]; + xi = a[125]; + yr = a[62]; + yi = a[63]; + a[124] = yr; + a[125] = yi; + a[62] = xr; + a[63] = xi; + + xr = a[110]; + xi = a[111]; + yr = a[118]; + yi = a[119]; + a[110] = yr; + a[111] = yi; + a[118] = xr; + a[119] = xi; +} + +static void cft1st_128_mips(float *a) { + float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i; + float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i; + + float f0, f1, f2, f3, f4, f5, f6, f7; + int a_ptr, p1_rdft, p2_rdft, count; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + // first 16 + "lwc1 %[f0], 0(%[a]) \n\t" + "lwc1 %[f1], 4(%[a]) \n\t" + "lwc1 %[f2], 8(%[a]) \n\t" + "lwc1 %[f3], 12(%[a]) \n\t" + "lwc1 %[f4], 16(%[a]) \n\t" + "lwc1 %[f5], 20(%[a]) \n\t" + "lwc1 %[f6], 24(%[a]) \n\t" + "lwc1 %[f7], 28(%[a]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "sub.s %[f4], %[x0r], %[x2r] \n\t" + "sub.s %[f5], %[x0i], %[x2i] \n\t" + "sub.s %[f2], %[x1r], %[x3i] \n\t" + "add.s %[f3], %[x1i], %[x3r] \n\t" + "add.s %[f6], %[x1r], %[x3i] \n\t" + "sub.s %[f7], %[x1i], %[x3r] \n\t" + "swc1 %[f0], 0(%[a]) \n\t" + "swc1 %[f1], 4(%[a]) \n\t" + "swc1 %[f2], 8(%[a]) \n\t" + "swc1 %[f3], 12(%[a]) \n\t" + "swc1 %[f4], 16(%[a]) \n\t" + "swc1 %[f5], 20(%[a]) \n\t" + "swc1 %[f6], 24(%[a]) \n\t" + "swc1 %[f7], 28(%[a]) \n\t" + "lwc1 %[f0], 32(%[a]) \n\t" + "lwc1 %[f1], 36(%[a]) \n\t" + "lwc1 %[f2], 40(%[a]) \n\t" + "lwc1 %[f3], 44(%[a]) \n\t" + "lwc1 %[f4], 48(%[a]) \n\t" + "lwc1 %[f5], 52(%[a]) \n\t" + "lwc1 %[f6], 56(%[a]) \n\t" + "lwc1 %[f7], 60(%[a]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "lwc1 %[wk2r], 8(%[rdft_w]) \n\t" + "add.s %[f3], %[x1i], %[x3r] \n\t" + "sub.s %[f2], %[x1r], %[x3i] \n\t" + "add.s %[f6], %[x3i], %[x1r] \n\t" + "sub.s %[f7], %[x3r], %[x1i] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "sub.s %[x1r], %[f2], %[f3] \n\t" + "add.s %[x1i], %[f3], %[f2] \n\t" + "sub.s %[x3r], %[f7], %[f6] \n\t" + "add.s %[x3i], %[f7], %[f6] \n\t" + "sub.s %[f4], %[x0r], %[x2r] \n\t" + "mul.s %[f2], %[wk2r], %[x1r] \n\t" + "mul.s %[f3], %[wk2r], %[x1i] \n\t" + "mul.s %[f6], %[wk2r], %[x3r] \n\t" + "mul.s %[f7], %[wk2r], %[x3i] \n\t" + "sub.s %[f5], %[x2i], %[x0i] \n\t" + "swc1 %[f0], 32(%[a]) \n\t" + "swc1 %[f1], 36(%[a]) \n\t" + "swc1 %[f2], 40(%[a]) \n\t" + "swc1 %[f3], 44(%[a]) \n\t" + "swc1 %[f5], 48(%[a]) \n\t" + "swc1 %[f4], 52(%[a]) \n\t" + "swc1 %[f6], 56(%[a]) \n\t" + "swc1 %[f7], 60(%[a]) \n\t" + // prepare for loop + "addiu %[a_ptr], %[a], 64 \n\t" + "addiu %[p1_rdft], %[rdft_w], 8 \n\t" + "addiu %[p2_rdft], %[rdft_w], 16 \n\t" + "addiu %[count], $zero, 7 \n\t" + // loop + "1: \n\t" + "lwc1 %[f0], 0(%[a_ptr]) \n\t" + "lwc1 %[f1], 4(%[a_ptr]) \n\t" + "lwc1 %[f2], 8(%[a_ptr]) \n\t" + "lwc1 %[f3], 12(%[a_ptr]) \n\t" + "lwc1 %[f4], 16(%[a_ptr]) \n\t" + "lwc1 %[f5], 20(%[a_ptr]) \n\t" + "lwc1 %[f6], 24(%[a_ptr]) \n\t" + "lwc1 %[f7], 28(%[a_ptr]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "lwc1 %[wk2i], 4(%[p1_rdft]) \n\t" + "sub.s %[f0], %[x0r], %[x2r] \n\t" + "sub.s %[f1], %[x0i], %[x2i] \n\t" + "add.s %[f2], %[x1i], %[x3r] \n\t" + "sub.s %[f3], %[x1r], %[x3i] \n\t" + "lwc1 %[wk1r], 0(%[p2_rdft]) \n\t" + "add.s %[f4], %[x1r], %[x3i] \n\t" + "sub.s %[f5], %[x1i], %[x3r] \n\t" + "lwc1 %[wk3r], 8(%[first]) \n\t" + "mul.s %[x3r], %[wk2r], %[f0] \n\t" + "mul.s %[x3i], %[wk2r], %[f1] \n\t" + "mul.s %[x1r], %[wk1r], %[f3] \n\t" + "mul.s %[x1i], %[wk1r], %[f2] \n\t" + "lwc1 %[wk1i], 4(%[p2_rdft]) \n\t" + "mul.s %[f6], %[wk3r], %[f4] \n\t" + "mul.s %[f7], %[wk3r], %[f5] \n\t" + "lwc1 %[wk3i], 12(%[first]) \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[wk1r], %[wk2i], %[f1] \n\t" + "mul.s %[f0], %[wk2i], %[f0] \n\t" + "sub.s %[x3r], %[x3r], %[wk1r] \n\t" + "add.s %[x3i], %[x3i], %[f0] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "mul.s %[x0r], %[wk1i], %[f2] \n\t" + "mul.s %[f3], %[wk1i], %[f3] \n\t" + "mul.s %[x2r], %[wk3i], %[f5] \n\t" + "mul.s %[f4], %[wk3i], %[f4] \n\t" + "sub.s %[x1r], %[x1r], %[x0r] \n\t" + "add.s %[x1i], %[x1i], %[f3] \n\t" + "sub.s %[f6], %[f6], %[x2r] \n\t" + "add.s %[f7], %[f7], %[f4] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmsub.s %[x3r], %[x3r], %[wk2i], %[f1] \n\t" + "madd.s %[x3i], %[x3i], %[wk2i], %[f0] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "nmsub.s %[x1r], %[x1r], %[wk1i], %[f2] \n\t" + "madd.s %[x1i], %[x1i], %[wk1i], %[f3] \n\t" + "nmsub.s %[f6], %[f6], %[wk3i], %[f5] \n\t" + "madd.s %[f7], %[f7], %[wk3i], %[f4] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[f0], 0(%[a_ptr]) \n\t" + "swc1 %[f1], 4(%[a_ptr]) \n\t" + "swc1 %[x1r], 8(%[a_ptr]) \n\t" + "swc1 %[x1i], 12(%[a_ptr]) \n\t" + "swc1 %[x3r], 16(%[a_ptr]) \n\t" + "swc1 %[x3i], 20(%[a_ptr]) \n\t" + "swc1 %[f6], 24(%[a_ptr]) \n\t" + "swc1 %[f7], 28(%[a_ptr]) \n\t" + "lwc1 %[f0], 32(%[a_ptr]) \n\t" + "lwc1 %[f1], 36(%[a_ptr]) \n\t" + "lwc1 %[f2], 40(%[a_ptr]) \n\t" + "lwc1 %[f3], 44(%[a_ptr]) \n\t" + "lwc1 %[f4], 48(%[a_ptr]) \n\t" + "lwc1 %[f5], 52(%[a_ptr]) \n\t" + "lwc1 %[f6], 56(%[a_ptr]) \n\t" + "lwc1 %[f7], 60(%[a_ptr]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "lwc1 %[wk1r], 8(%[p2_rdft]) \n\t" + "sub.s %[f0], %[x0r], %[x2r] \n\t" + "sub.s %[f1], %[x0i], %[x2i] \n\t" + "add.s %[f2], %[x1i], %[x3r] \n\t" + "sub.s %[f3], %[x1r], %[x3i] \n\t" + "add.s %[f4], %[x1r], %[x3i] \n\t" + "sub.s %[f5], %[x1i], %[x3r] \n\t" + "lwc1 %[wk3r], 8(%[second]) \n\t" + "mul.s %[x3r], %[wk2i], %[f0] \n\t" + "mul.s %[x3i], %[wk2i], %[f1] \n\t" + "mul.s %[x1r], %[wk1r], %[f3] \n\t" + "mul.s %[x1i], %[wk1r], %[f2] \n\t" + "mul.s %[f6], %[wk3r], %[f4] \n\t" + "mul.s %[f7], %[wk3r], %[f5] \n\t" + "lwc1 %[wk1i], 12(%[p2_rdft]) \n\t" + "lwc1 %[wk3i], 12(%[second]) \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[wk1r], %[wk2r], %[f1] \n\t" + "mul.s %[f0], %[wk2r], %[f0] \n\t" + "add.s %[x3r], %[x3r], %[wk1r] \n\t" + "neg.s %[x3r], %[x3r] \n\t" + "sub.s %[x3i], %[f0], %[x3i] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "mul.s %[x0r], %[wk1i], %[f2] \n\t" + "mul.s %[f3], %[wk1i], %[f3] \n\t" + "mul.s %[x2r], %[wk3i], %[f5] \n\t" + "mul.s %[f4], %[wk3i], %[f4] \n\t" + "sub.s %[x1r], %[x1r], %[x0r] \n\t" + "add.s %[x1i], %[x1i], %[f3] \n\t" + "sub.s %[f6], %[f6], %[x2r] \n\t" + "add.s %[f7], %[f7], %[f4] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmadd.s %[x3r], %[x3r], %[wk2r], %[f1] \n\t" + "msub.s %[x3i], %[x3i], %[wk2r], %[f0] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "nmsub.s %[x1r], %[x1r], %[wk1i], %[f2] \n\t" + "madd.s %[x1i], %[x1i], %[wk1i], %[f3] \n\t" + "nmsub.s %[f6], %[f6], %[wk3i], %[f5] \n\t" + "madd.s %[f7], %[f7], %[wk3i], %[f4] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[wk2r], 8(%[p1_rdft]) \n\t" + "addiu %[a_ptr], %[a_ptr], 64 \n\t" + "addiu %[p1_rdft], %[p1_rdft], 8 \n\t" + "addiu %[p2_rdft], %[p2_rdft], 16 \n\t" + "addiu %[first], %[first], 8 \n\t" + "swc1 %[f0], -32(%[a_ptr]) \n\t" + "swc1 %[f1], -28(%[a_ptr]) \n\t" + "swc1 %[x1r], -24(%[a_ptr]) \n\t" + "swc1 %[x1i], -20(%[a_ptr]) \n\t" + "swc1 %[x3r], -16(%[a_ptr]) \n\t" + "swc1 %[x3i], -12(%[a_ptr]) \n\t" + "swc1 %[f6], -8(%[a_ptr]) \n\t" + "swc1 %[f7], -4(%[a_ptr]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[second], %[second], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [x0r] "=&f" (x0r), [x0i] "=&f" (x0i), [x1r] "=&f" (x1r), + [x1i] "=&f" (x1i), [x2r] "=&f" (x2r), [x2i] "=&f" (x2i), + [x3r] "=&f" (x3r), [x3i] "=&f" (x3i), [wk1r] "=&f" (wk1r), + [wk1i] "=&f" (wk1i), [wk2r] "=&f" (wk2r), [wk2i] "=&f" (wk2i), + [wk3r] "=&f" (wk3r), [wk3i] "=&f" (wk3i), [a_ptr] "=&r" (a_ptr), + [p1_rdft] "=&r" (p1_rdft), [p2_rdft] "=&r" (p2_rdft), + [count] "=&r" (count) + : [a] "r" (a), [rdft_w] "r" (rdft_w), [first] "r" (rdft_wk3ri_first), + [second] "r" (rdft_wk3ri_second) + : "memory" + ); +} + +static void cftmdl_128_mips(float *a) { + float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i; + float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i; + float f0, f1, f2, f3, f4, f5, f6, f7; + int tmp_a, count; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmp_a], %[a], 0 \n\t" + "addiu %[count], $zero, 4 \n\t" + "1: \n\t" + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[f0], 0(%[tmp_a]) \n\t" + "lwc1 %[f1], 4(%[tmp_a]) \n\t" + "lwc1 %[f2], 32(%[tmp_a]) \n\t" + "lwc1 %[f3], 36(%[tmp_a]) \n\t" + "lwc1 %[f4], 64(%[tmp_a]) \n\t" + "lwc1 %[f5], 68(%[tmp_a]) \n\t" + "lwc1 %[f6], 96(%[tmp_a]) \n\t" + "lwc1 %[f7], 100(%[tmp_a]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "sub.s %[f4], %[x0r], %[x2r] \n\t" + "sub.s %[f5], %[x0i], %[x2i] \n\t" + "sub.s %[f2], %[x1r], %[x3i] \n\t" + "add.s %[f3], %[x1i], %[x3r] \n\t" + "add.s %[f6], %[x1r], %[x3i] \n\t" + "sub.s %[f7], %[x1i], %[x3r] \n\t" + "swc1 %[f0], 0(%[tmp_a]) \n\t" + "swc1 %[f1], 4(%[tmp_a]) \n\t" + "swc1 %[f2], 32(%[tmp_a]) \n\t" + "swc1 %[f3], 36(%[tmp_a]) \n\t" + "swc1 %[f4], 64(%[tmp_a]) \n\t" + "swc1 %[f5], 68(%[tmp_a]) \n\t" + "swc1 %[f6], 96(%[tmp_a]) \n\t" + "swc1 %[f7], 100(%[tmp_a]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[tmp_a], %[tmp_a], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [x0r] "=&f" (x0r), [x0i] "=&f" (x0i), [x1r] "=&f" (x1r), + [x1i] "=&f" (x1i), [x2r] "=&f" (x2r), [x2i] "=&f" (x2i), + [x3r] "=&f" (x3r), [x3i] "=&f" (x3i), [tmp_a] "=&r" (tmp_a), + [count] "=&r" (count) + : [a] "r" (a) + : "memory" + ); + wk2r = rdft_w[2]; + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmp_a], %[a], 128 \n\t" + "addiu %[count], $zero, 4 \n\t" + "1: \n\t" + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[f0], 0(%[tmp_a]) \n\t" + "lwc1 %[f1], 4(%[tmp_a]) \n\t" + "lwc1 %[f2], 32(%[tmp_a]) \n\t" + "lwc1 %[f3], 36(%[tmp_a]) \n\t" + "lwc1 %[f4], 64(%[tmp_a]) \n\t" + "lwc1 %[f5], 68(%[tmp_a]) \n\t" + "lwc1 %[f6], 96(%[tmp_a]) \n\t" + "lwc1 %[f7], 100(%[tmp_a]) \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[f0], %[x1r], %[x3i] \n\t" + "add.s %[f1], %[x1i], %[x3r] \n\t" + "sub.s %[f2], %[x3r], %[x1i] \n\t" + "add.s %[f3], %[x3i], %[x1r] \n\t" + "add.s %[f4], %[x0r], %[x2r] \n\t" + "add.s %[f5], %[x0i], %[x2i] \n\t" + "sub.s %[f6], %[f0], %[f1] \n\t" + "add.s %[f0], %[f0], %[f1] \n\t" + "sub.s %[f7], %[f2], %[f3] \n\t" + "add.s %[f2], %[f2], %[f3] \n\t" + "sub.s %[f1], %[x2i], %[x0i] \n\t" + "mul.s %[f6], %[f6], %[wk2r] \n\t" + "mul.s %[f0], %[f0], %[wk2r] \n\t" + "sub.s %[f3], %[x0r], %[x2r] \n\t" + "mul.s %[f7], %[f7], %[wk2r] \n\t" + "mul.s %[f2], %[f2], %[wk2r] \n\t" + "swc1 %[f4], 0(%[tmp_a]) \n\t" + "swc1 %[f5], 4(%[tmp_a]) \n\t" + "swc1 %[f6], 32(%[tmp_a]) \n\t" + "swc1 %[f0], 36(%[tmp_a]) \n\t" + "swc1 %[f1], 64(%[tmp_a]) \n\t" + "swc1 %[f3], 68(%[tmp_a]) \n\t" + "swc1 %[f7], 96(%[tmp_a]) \n\t" + "swc1 %[f2], 100(%[tmp_a]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[tmp_a], %[tmp_a], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [x0r] "=&f" (x0r), [x0i] "=&f" (x0i), [x1r] "=&f" (x1r), + [x1i] "=&f" (x1i), [x2r] "=&f" (x2r), [x2i] "=&f" (x2i), + [x3r] "=&f" (x3r), [x3i] "=&f" (x3i), [tmp_a] "=&r" (tmp_a), + [count] "=&r" (count) + : [a] "r" (a), [wk2r] "f" (wk2r) + : "memory" + ); + wk2i = rdft_w[3]; + wk1r = rdft_w[4]; + wk1i = rdft_w[5]; + wk3r = rdft_wk3ri_first[2]; + wk3i = rdft_wk3ri_first[3]; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmp_a], %[a], 256 \n\t" + "addiu %[count], $zero, 4 \n\t" + "1: \n\t" + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[f0], 0(%[tmp_a]) \n\t" + "lwc1 %[f1], 4(%[tmp_a]) \n\t" + "lwc1 %[f2], 32(%[tmp_a]) \n\t" + "lwc1 %[f3], 36(%[tmp_a]) \n\t" + "lwc1 %[f4], 64(%[tmp_a]) \n\t" + "lwc1 %[f5], 68(%[tmp_a]) \n\t" + "lwc1 %[f6], 96(%[tmp_a]) \n\t" + "lwc1 %[f7], 100(%[tmp_a]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "sub.s %[f0], %[x0r], %[x2r] \n\t" + "sub.s %[f1], %[x0i], %[x2i] \n\t" + "add.s %[f2], %[x1i], %[x3r] \n\t" + "sub.s %[f3], %[x1r], %[x3i] \n\t" + "add.s %[f4], %[x1r], %[x3i] \n\t" + "sub.s %[f5], %[x1i], %[x3r] \n\t" + "mul.s %[x3r], %[wk2r], %[f0] \n\t" + "mul.s %[x3i], %[wk2r], %[f1] \n\t" + "mul.s %[x1r], %[wk1r], %[f3] \n\t" + "mul.s %[x1i], %[wk1r], %[f2] \n\t" + "mul.s %[f6], %[wk3r], %[f4] \n\t" + "mul.s %[f7], %[wk3r], %[f5] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[f1], %[wk2i], %[f1] \n\t" + "mul.s %[f0], %[wk2i], %[f0] \n\t" + "sub.s %[x3r], %[x3r], %[f1] \n\t" + "add.s %[x3i], %[x3i], %[f0] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "mul.s %[f2], %[wk1i], %[f2] \n\t" + "mul.s %[f3], %[wk1i], %[f3] \n\t" + "mul.s %[f5], %[wk3i], %[f5] \n\t" + "mul.s %[f4], %[wk3i], %[f4] \n\t" + "sub.s %[x1r], %[x1r], %[f2] \n\t" + "add.s %[x1i], %[x1i], %[f3] \n\t" + "sub.s %[f6], %[f6], %[f5] \n\t" + "add.s %[f7], %[f7], %[f4] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmsub.s %[x3r], %[x3r], %[wk2i], %[f1] \n\t" + "madd.s %[x3i], %[x3i], %[wk2i], %[f0] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "nmsub.s %[x1r], %[x1r], %[wk1i], %[f2] \n\t" + "madd.s %[x1i], %[x1i], %[wk1i], %[f3] \n\t" + "nmsub.s %[f6], %[f6], %[wk3i], %[f5] \n\t" + "madd.s %[f7], %[f7], %[wk3i], %[f4] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[f0], 0(%[tmp_a]) \n\t" + "swc1 %[f1], 4(%[tmp_a]) \n\t" + "swc1 %[x1r], 32(%[tmp_a]) \n\t" + "swc1 %[x1i], 36(%[tmp_a]) \n\t" + "swc1 %[x3r], 64(%[tmp_a]) \n\t" + "swc1 %[x3i], 68(%[tmp_a]) \n\t" + "swc1 %[f6], 96(%[tmp_a]) \n\t" + "swc1 %[f7], 100(%[tmp_a]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[tmp_a], %[tmp_a], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [x0r] "=&f" (x0r), [x0i] "=&f" (x0i), [x1r] "=&f" (x1r), + [x1i] "=&f" (x1i), [x2r] "=&f" (x2r), [x2i] "=&f" (x2i), + [x3r] "=&f" (x3r), [x3i] "=&f" (x3i), [tmp_a] "=&r" (tmp_a), + [count] "=&r" (count) + : [a] "r" (a), [wk1r] "f" (wk1r), [wk1i] "f" (wk1i), [wk2r] "f" (wk2r), + [wk2i] "f" (wk2i), [wk3r] "f" (wk3r), [wk3i] "f" (wk3i) + : "memory" + ); + + wk1r = rdft_w[6]; + wk1i = rdft_w[7]; + wk3r = rdft_wk3ri_second[2]; + wk3i = rdft_wk3ri_second[3]; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmp_a], %[a], 384 \n\t" + "addiu %[count], $zero, 4 \n\t" + "1: \n\t" + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[f0], 0(%[tmp_a]) \n\t" + "lwc1 %[f1], 4(%[tmp_a]) \n\t" + "lwc1 %[f2], 32(%[tmp_a]) \n\t" + "lwc1 %[f3], 36(%[tmp_a]) \n\t" + "lwc1 %[f4], 64(%[tmp_a]) \n\t" + "lwc1 %[f5], 68(%[tmp_a]) \n\t" + "lwc1 %[f6], 96(%[tmp_a]) \n\t" + "lwc1 %[f7], 100(%[tmp_a]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "sub.s %[f0], %[x0r], %[x2r] \n\t" + "sub.s %[f1], %[x0i], %[x2i] \n\t" + "add.s %[f2], %[x1i], %[x3r] \n\t" + "sub.s %[f3], %[x1r], %[x3i] \n\t" + "add.s %[f4], %[x1r], %[x3i] \n\t" + "sub.s %[f5], %[x1i], %[x3r] \n\t" + "mul.s %[x3r], %[wk2i], %[f0] \n\t" + "mul.s %[x3i], %[wk2i], %[f1] \n\t" + "mul.s %[x1r], %[wk1r], %[f3] \n\t" + "mul.s %[x1i], %[wk1r], %[f2] \n\t" + "mul.s %[f6], %[wk3r], %[f4] \n\t" + "mul.s %[f7], %[wk3r], %[f5] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[f1], %[wk2r], %[f1] \n\t" + "mul.s %[f0], %[wk2r], %[f0] \n\t" + "add.s %[x3r], %[x3r], %[f1] \n\t" + "neg.s %[x3r], %[x3r] \n\t" + "sub.s %[x3i], %[f0], %[x3i] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "mul.s %[f2], %[wk1i], %[f2] \n\t" + "mul.s %[f3], %[wk1i], %[f3] \n\t" + "mul.s %[f5], %[wk3i], %[f5] \n\t" + "mul.s %[f4], %[wk3i], %[f4] \n\t" + "sub.s %[x1r], %[x1r], %[f2] \n\t" + "add.s %[x1i], %[x1i], %[f3] \n\t" + "sub.s %[f6], %[f6], %[f5] \n\t" + "add.s %[f7], %[f7], %[f4] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmadd.s %[x3r], %[x3r], %[wk2r], %[f1] \n\t" + "msub.s %[x3i], %[x3i], %[wk2r], %[f0] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "nmsub.s %[x1r], %[x1r], %[wk1i], %[f2] \n\t" + "madd.s %[x1i], %[x1i], %[wk1i], %[f3] \n\t" + "nmsub.s %[f6], %[f6], %[wk3i], %[f5] \n\t" + "madd.s %[f7], %[f7], %[wk3i], %[f4] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "swc1 %[f0], 0(%[tmp_a]) \n\t" + "swc1 %[f1], 4(%[tmp_a]) \n\t" + "swc1 %[x1r], 32(%[tmp_a]) \n\t" + "swc1 %[x1i], 36(%[tmp_a]) \n\t" + "swc1 %[x3r], 64(%[tmp_a]) \n\t" + "swc1 %[x3i], 68(%[tmp_a]) \n\t" + "swc1 %[f6], 96(%[tmp_a]) \n\t" + "swc1 %[f7], 100(%[tmp_a]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[tmp_a], %[tmp_a], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [x0r] "=&f" (x0r), [x0i] "=&f" (x0i), [x1r] "=&f" (x1r), + [x1i] "=&f" (x1i), [x2r] "=&f" (x2r), [x2i] "=&f" (x2i), + [x3r] "=&f" (x3r), [x3i] "=&f" (x3i), [tmp_a] "=&r" (tmp_a), + [count] "=&r" (count) + : [a] "r" (a), [wk1r] "f" (wk1r), [wk1i] "f" (wk1i), [wk2r] "f" (wk2r), + [wk2i] "f" (wk2i), [wk3r] "f" (wk3r), [wk3i] "f" (wk3i) + : "memory" + ); +} + +static void cftfsub_128_mips(float *a) { + float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i; + float f0, f1, f2, f3, f4, f5, f6, f7; + int tmp_a, count; + + cft1st_128(a); + cftmdl_128(a); + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmp_a], %[a], 0 \n\t" + "addiu %[count], $zero, 16 \n\t" + "1: \n\t" + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[f0], 0(%[tmp_a]) \n\t" + "lwc1 %[f1], 4(%[tmp_a]) \n\t" + "lwc1 %[f2], 128(%[tmp_a]) \n\t" + "lwc1 %[f3], 132(%[tmp_a]) \n\t" + "lwc1 %[f4], 256(%[tmp_a]) \n\t" + "lwc1 %[f5], 260(%[tmp_a]) \n\t" + "lwc1 %[f6], 384(%[tmp_a]) \n\t" + "lwc1 %[f7], 388(%[tmp_a]) \n\t" + "add.s %[x0r], %[f0], %[f2] \n\t" + "add.s %[x0i], %[f1], %[f3] \n\t" + "add.s %[x2r], %[f4], %[f6] \n\t" + "add.s %[x2i], %[f5], %[f7] \n\t" + "sub.s %[x1r], %[f0], %[f2] \n\t" + "sub.s %[x1i], %[f1], %[f3] \n\t" + "sub.s %[x3r], %[f4], %[f6] \n\t" + "sub.s %[x3i], %[f5], %[f7] \n\t" + "add.s %[f0], %[x0r], %[x2r] \n\t" + "add.s %[f1], %[x0i], %[x2i] \n\t" + "sub.s %[f4], %[x0r], %[x2r] \n\t" + "sub.s %[f5], %[x0i], %[x2i] \n\t" + "sub.s %[f2], %[x1r], %[x3i] \n\t" + "add.s %[f3], %[x1i], %[x3r] \n\t" + "add.s %[f6], %[x1r], %[x3i] \n\t" + "sub.s %[f7], %[x1i], %[x3r] \n\t" + "swc1 %[f0], 0(%[tmp_a]) \n\t" + "swc1 %[f1], 4(%[tmp_a]) \n\t" + "swc1 %[f2], 128(%[tmp_a]) \n\t" + "swc1 %[f3], 132(%[tmp_a]) \n\t" + "swc1 %[f4], 256(%[tmp_a]) \n\t" + "swc1 %[f5], 260(%[tmp_a]) \n\t" + "swc1 %[f6], 384(%[tmp_a]) \n\t" + "swc1 %[f7], 388(%[tmp_a]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[tmp_a], %[tmp_a], 8 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [x0r] "=&f" (x0r), [x0i] "=&f" (x0i), [x1r] "=&f" (x1r), + [x1i] "=&f" (x1i), [x2r] "=&f" (x2r), [x2i] "=&f" (x2i), + [x3r] "=&f" (x3r), [x3i] "=&f" (x3i), [tmp_a] "=&r" (tmp_a), + [count] "=&r" (count) + : [a] "r" (a) + : "memory" + ); +} + +static void cftbsub_128_mips(float *a) { + float f0, f1, f2, f3, f4, f5, f6, f7; + float f8, f9, f10, f11, f12, f13, f14, f15; + float f16, f17, f18, f19, f20, f21, f22, f23; + int tmp_a, count; + + cft1st_128(a); + cftmdl_128(a); + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "addiu %[tmp_a], %[a], 0 \n\t" + "addiu %[count], $zero, 8 \n\t" + "1: \n\t" + "addiu %[count], %[count], -1 \n\t" + "lwc1 %[f0], 0(%[tmp_a]) \n\t" + "lwc1 %[f1], 4(%[tmp_a]) \n\t" + "lwc1 %[f2], 128(%[tmp_a]) \n\t" + "lwc1 %[f3], 132(%[tmp_a]) \n\t" + "lwc1 %[f4], 256(%[tmp_a]) \n\t" + "lwc1 %[f5], 260(%[tmp_a]) \n\t" + "lwc1 %[f6], 384(%[tmp_a]) \n\t" + "lwc1 %[f7], 388(%[tmp_a]) \n\t" + "lwc1 %[f8], 8(%[tmp_a]) \n\t" + "lwc1 %[f9], 12(%[tmp_a]) \n\t" + "lwc1 %[f10], 136(%[tmp_a]) \n\t" + "lwc1 %[f11], 140(%[tmp_a]) \n\t" + "lwc1 %[f12], 264(%[tmp_a]) \n\t" + "lwc1 %[f13], 268(%[tmp_a]) \n\t" + "lwc1 %[f14], 392(%[tmp_a]) \n\t" + "lwc1 %[f15], 396(%[tmp_a]) \n\t" + "add.s %[f16], %[f0], %[f2] \n\t" + "add.s %[f17], %[f1], %[f3] \n\t" + "add.s %[f18], %[f4], %[f6] \n\t" + "add.s %[f19], %[f5], %[f7] \n\t" + "sub.s %[f20], %[f0], %[f2] \n\t" + "sub.s %[f21], %[f3], %[f1] \n\t" + "sub.s %[f22], %[f4], %[f6] \n\t" + "sub.s %[f23], %[f5], %[f7] \n\t" + "add.s %[f0], %[f8], %[f10] \n\t" + "add.s %[f1], %[f9], %[f11] \n\t" + "add.s %[f2], %[f12], %[f14] \n\t" + "add.s %[f3], %[f13], %[f15] \n\t" + "sub.s %[f4], %[f8], %[f10] \n\t" + "sub.s %[f5], %[f11], %[f9] \n\t" + "sub.s %[f6], %[f12], %[f14] \n\t" + "sub.s %[f7], %[f13], %[f15] \n\t" + "add.s %[f8], %[f16], %[f18] \n\t" + "add.s %[f9], %[f17], %[f19] \n\t" + "sub.s %[f12], %[f16], %[f18] \n\t" + "sub.s %[f13], %[f19], %[f17] \n\t" + "sub.s %[f10], %[f20], %[f23] \n\t" + "sub.s %[f11], %[f21], %[f22] \n\t" + "add.s %[f14], %[f20], %[f23] \n\t" + "add.s %[f15], %[f21], %[f22] \n\t" + "neg.s %[f9], %[f9] \n\t" + "add.s %[f16], %[f0], %[f2] \n\t" + "add.s %[f17], %[f1], %[f3] \n\t" + "sub.s %[f20], %[f0], %[f2] \n\t" + "sub.s %[f21], %[f3], %[f1] \n\t" + "sub.s %[f18], %[f4], %[f7] \n\t" + "sub.s %[f19], %[f5], %[f6] \n\t" + "add.s %[f22], %[f4], %[f7] \n\t" + "add.s %[f23], %[f5], %[f6] \n\t" + "neg.s %[f17], %[f17] \n\t" + "swc1 %[f8], 0(%[tmp_a]) \n\t" + "swc1 %[f10], 128(%[tmp_a]) \n\t" + "swc1 %[f11], 132(%[tmp_a]) \n\t" + "swc1 %[f12], 256(%[tmp_a]) \n\t" + "swc1 %[f13], 260(%[tmp_a]) \n\t" + "swc1 %[f14], 384(%[tmp_a]) \n\t" + "swc1 %[f15], 388(%[tmp_a]) \n\t" + "swc1 %[f9], 4(%[tmp_a]) \n\t" + "swc1 %[f16], 8(%[tmp_a]) \n\t" + "swc1 %[f18], 136(%[tmp_a]) \n\t" + "swc1 %[f19], 140(%[tmp_a]) \n\t" + "swc1 %[f20], 264(%[tmp_a]) \n\t" + "swc1 %[f21], 268(%[tmp_a]) \n\t" + "swc1 %[f22], 392(%[tmp_a]) \n\t" + "swc1 %[f23], 396(%[tmp_a]) \n\t" + "swc1 %[f17], 12(%[tmp_a]) \n\t" + "bgtz %[count], 1b \n\t" + " addiu %[tmp_a], %[tmp_a], 16 \n\t" + ".set pop \n\t" + : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), + [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), + [f8] "=&f" (f8), [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11), + [f12] "=&f" (f12), [f13] "=&f" (f13), [f14] "=&f" (f14), + [f15] "=&f" (f15), [f16] "=&f" (f16), [f17] "=&f" (f17), + [f18] "=&f" (f18), [f19] "=&f" (f19), [f20] "=&f" (f20), + [f21] "=&f" (f21), [f22] "=&f" (f22), [f23] "=&f" (f23), + [tmp_a] "=&r" (tmp_a), [count] "=&r" (count) + : [a] "r" (a) + : "memory" + ); +} + +static void rftfsub_128_mips(float *a) { + const float *c = rdft_w + 32; + float wkr, wki, xr, xi, yr, yi; + const float temp = 0.5f; + float aj20=0, aj21=0, ak20=0, ak21=0, ck1=0; + float *a1 = a; + float *a2 = a; + float *c1 = rdft_w + 33; + float *c2 = c1 + 30; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lwc1 %[aj20], 8(%[a2]) \n\t" + "lwc1 %[ak20], 504(%[a1]) \n\t" + "lwc1 %[ck1], 0(%[c2]) \n\t" + "lwc1 %[aj21], 12(%[a2]) \n\t" + "lwc1 %[ak21], 508(%[a1]) \n\t" + "sub.s %[wkr], %[temp], %[ck1] \n\t" + "sub.s %[xr], %[aj20], %[ak20] \n\t" + "add.s %[xi], %[aj21], %[ak21] \n\t" + "lwc1 %[wki], 0(%[c1]) \n\t" + "addiu %[c2], %[c2],-4 \n\t" + "mul.s %[yr], %[wkr], %[xr] \n\t" + "mul.s %[yi], %[wkr], %[xi] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[xi], %[wki], %[xi] \n\t" + "mul.s %[xr], %[wki], %[xr] \n\t" + "sub.s %[yr], %[yr], %[xi] \n\t" + "add.s %[yi], %[yi], %[xr] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmsub.s %[yr], %[yr], %[wki], %[xi] \n\t" + "madd.s %[yi], %[yi], %[wki], %[xr] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "addiu %[c1], %[c1], 4 \n\t" + "sub.s %[aj20], %[aj20], %[yr] \n\t" + "sub.s %[aj21], %[aj21], %[yi] \n\t" + "add.s %[ak20], %[ak20], %[yr] \n\t" + "sub.s %[ak21], %[ak21], %[yi] \n\t" + "addiu %[a2], %[a2], 8 \n\t" + "swc1 %[aj20], 0(%[a2]) \n\t" + "swc1 %[aj21], 4(%[a2]) \n\t" + "swc1 %[ak20], 504(%[a1]) \n\t" + "swc1 %[ak21], 508(%[a1]) \n\t" + "addiu %[a1], %[a1], -8 \n\t" + //15x2 passes: + "1: \n\t" + "lwc1 %[ck1], 0(%[c2]) \n\t" + "lwc1 %[aj20], 8(%[a2]) \n\t" + "lwc1 %[aj21], 12(%[a2]) \n\t" + "lwc1 %[ak20], 504(%[a1]) \n\t" + "lwc1 %[ak21], 508(%[a1]) \n\t" + "lwc1 $f0, -4(%[c2]) \n\t" + "lwc1 $f2, 16(%[a2]) \n\t" + "lwc1 $f3, 20(%[a2]) \n\t" + "lwc1 $f8, 496(%[a1]) \n\t" + "lwc1 $f7, 500(%[a1]) \n\t" + "sub.s %[wkr], %[temp], %[ck1] \n\t" + "sub.s %[xr], %[aj20], %[ak20] \n\t" + "add.s %[xi], %[aj21], %[ak21] \n\t" + "lwc1 %[wki], 0(%[c1]) \n\t" + "sub.s $f0, %[temp], $f0 \n\t" + "sub.s $f6, $f2, $f8 \n\t" + "add.s $f4, $f3, $f7 \n\t" + "lwc1 $f5, 4(%[c1]) \n\t" + "mul.s %[yr], %[wkr], %[xr] \n\t" + "mul.s %[yi], %[wkr], %[xi] \n\t" + "mul.s $f1, $f0, $f6 \n\t" + "mul.s $f0, $f0, $f4 \n\t" + "addiu %[c2], %[c2], -8 \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[xi], %[wki], %[xi] \n\t" + "mul.s %[xr], %[wki], %[xr] \n\t" + "mul.s $f4, $f5, $f4 \n\t" + "mul.s $f6, $f5, $f6 \n\t" + "sub.s %[yr], %[yr], %[xi] \n\t" + "add.s %[yi], %[yi], %[xr] \n\t" + "sub.s $f1, $f1, $f4 \n\t" + "add.s $f0, $f0, $f6 \n\t" +#else // #if !defined(MIPS32_R2_LE) + "nmsub.s %[yr], %[yr], %[wki], %[xi] \n\t" + "madd.s %[yi], %[yi], %[wki], %[xr] \n\t" + "nmsub.s $f1, $f1, $f5, $f4 \n\t" + "madd.s $f0, $f0, $f5, $f6 \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "addiu %[c1], %[c1], 8 \n\t" + "sub.s %[aj20], %[aj20], %[yr] \n\t" + "sub.s %[aj21], %[aj21], %[yi] \n\t" + "add.s %[ak20], %[ak20], %[yr] \n\t" + "sub.s %[ak21], %[ak21], %[yi] \n\t" + "sub.s $f2, $f2, $f1 \n\t" + "sub.s $f3, $f3, $f0 \n\t" + "add.s $f1, $f8, $f1 \n\t" + "sub.s $f0, $f7, $f0 \n\t" + "swc1 %[aj20], 8(%[a2]) \n\t" + "swc1 %[aj21], 12(%[a2]) \n\t" + "swc1 %[ak20], 504(%[a1]) \n\t" + "swc1 %[ak21], 508(%[a1]) \n\t" + "swc1 $f2, 16(%[a2]) \n\t" + "swc1 $f3, 20(%[a2]) \n\t" + "swc1 $f1, 496(%[a1]) \n\t" + "swc1 $f0, 500(%[a1]) \n\t" + "addiu %[a2], %[a2], 16 \n\t" + "bne %[c2], %[c], 1b \n\t" + " addiu %[a1], %[a1], -16 \n\t" + ".set pop \n\t" + : [a] "+r" (a), [c] "+r" (c), [a1] "+r" (a1), [a2] "+r" (a2), + [c1] "+r" (c1), [c2] "+r" (c2), [wkr] "=&f" (wkr), [wki] "=&f" (wki), + [xr] "=&f" (xr), [xi] "=&f" (xi), [yr] "=&f" (yr), [yi] "=&f" (yi), + [aj20] "=&f" (aj20), [aj21] "=&f" (aj21), [ak20] "=&f" (ak20), + [ak21] "=&f" (ak21), [ck1] "=&f" (ck1) + : [temp] "f" (temp) + : "memory", "$f0", "$f1", "$f2", "$f3", "$f4", "$f5", "$f6", "$f7", "$f8" + ); +} + +static void rftbsub_128_mips(float *a) { + const float *c = rdft_w + 32; + float wkr, wki, xr, xi, yr, yi; + a[1] = -a[1]; + a[65] = -a[65]; + const float temp = 0.5f; + float aj20=0, aj21=0, ak20=0, ak21=0, ck1=0; + float *a1 = a; + float *a2 = a; + float *c1 = rdft_w + 33; + float *c2 = c1 + 30; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "lwc1 %[aj20], 8(%[a2]) \n\t" + "lwc1 %[ak20], 504(%[a1]) \n\t" + "lwc1 %[ck1], 0(%[c2]) \n\t" + "lwc1 %[aj21], 12(%[a2]) \n\t" + "lwc1 %[ak21], 508(%[a1]) \n\t" + "sub.s %[wkr], %[temp], %[ck1] \n\t" + "sub.s %[xr], %[aj20], %[ak20] \n\t" + "add.s %[xi], %[aj21], %[ak21] \n\t" + "lwc1 %[wki], 0(%[c1]) \n\t" + "addiu %[c2], %[c2], -4 \n\t" + "mul.s %[yr], %[wkr], %[xr] \n\t" + "mul.s %[yi], %[wkr], %[xi] \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[xi], %[wki], %[xi] \n\t" + "mul.s %[xr], %[wki], %[xr] \n\t" + "add.s %[yr], %[yr], %[xi] \n\t" + "sub.s %[yi], %[yi], %[xr] \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[yr], %[yr], %[wki], %[xi] \n\t" + "nmsub.s %[yi], %[yi], %[wki], %[xr] \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "addiu %[c1], %[c1],4 \n\t" + "sub.s %[aj20], %[aj20], %[yr] \n\t" + "sub.s %[aj21], %[yi], %[aj21] \n\t" + "add.s %[ak20], %[ak20], %[yr] \n\t" + "sub.s %[ak21], %[yi], %[ak21] \n\t" + "addiu %[a2], %[a2], 8 \n\t" + "swc1 %[aj20], 0(%[a2]) \n\t" + "swc1 %[aj21], 4(%[a2]) \n\t" + "swc1 %[ak20], 504(%[a1]) \n\t" + "swc1 %[ak21], 508(%[a1]) \n\t" + "addiu %[a1], %[a1], -8 \n\t" + //15x2 passes: + "1: \n\t" + "lwc1 %[ck1], 0(%[c2]) \n\t" + "lwc1 %[aj20], 8(%[a2]) \n\t" + "lwc1 %[aj21], 12(%[a2]) \n\t" + "lwc1 %[ak20], 504(%[a1]) \n\t" + "lwc1 %[ak21], 508(%[a1]) \n\t" + "lwc1 $f0, -4(%[c2]) \n\t" + "lwc1 $f2, 16(%[a2]) \n\t" + "lwc1 $f3, 20(%[a2]) \n\t" + "lwc1 $f8, 496(%[a1]) \n\t" + "lwc1 $f7, 500(%[a1]) \n\t" + "sub.s %[wkr], %[temp], %[ck1] \n\t" + "sub.s %[xr], %[aj20], %[ak20] \n\t" + "add.s %[xi], %[aj21], %[ak21] \n\t" + "lwc1 %[wki], 0(%[c1]) \n\t" + "sub.s $f0, %[temp], $f0 \n\t" + "sub.s $f6, $f2, $f8 \n\t" + "add.s $f4, $f3, $f7 \n\t" + "lwc1 $f5, 4(%[c1]) \n\t" + "mul.s %[yr], %[wkr], %[xr] \n\t" + "mul.s %[yi], %[wkr], %[xi] \n\t" + "mul.s $f1, $f0, $f6 \n\t" + "mul.s $f0, $f0, $f4 \n\t" + "addiu %[c2], %[c2], -8 \n\t" +#if !defined(MIPS32_R2_LE) + "mul.s %[xi], %[wki], %[xi] \n\t" + "mul.s %[xr], %[wki], %[xr] \n\t" + "mul.s $f4, $f5, $f4 \n\t" + "mul.s $f6, $f5, $f6 \n\t" + "add.s %[yr], %[yr], %[xi] \n\t" + "sub.s %[yi], %[yi], %[xr] \n\t" + "add.s $f1, $f1, $f4 \n\t" + "sub.s $f0, $f0, $f6 \n\t" +#else // #if !defined(MIPS32_R2_LE) + "madd.s %[yr], %[yr], %[wki], %[xi] \n\t" + "nmsub.s %[yi], %[yi], %[wki], %[xr] \n\t" + "madd.s $f1, $f1, $f5, $f4 \n\t" + "nmsub.s $f0, $f0, $f5, $f6 \n\t" +#endif // #if !defined(MIPS32_R2_LE) + "addiu %[c1], %[c1], 8 \n\t" + "sub.s %[aj20], %[aj20], %[yr] \n\t" + "sub.s %[aj21], %[yi], %[aj21] \n\t" + "add.s %[ak20], %[ak20], %[yr] \n\t" + "sub.s %[ak21], %[yi], %[ak21] \n\t" + "sub.s $f2, $f2, $f1 \n\t" + "sub.s $f3, $f0, $f3 \n\t" + "add.s $f1, $f8, $f1 \n\t" + "sub.s $f0, $f0, $f7 \n\t" + "swc1 %[aj20], 8(%[a2]) \n\t" + "swc1 %[aj21], 12(%[a2]) \n\t" + "swc1 %[ak20], 504(%[a1]) \n\t" + "swc1 %[ak21], 508(%[a1]) \n\t" + "swc1 $f2, 16(%[a2]) \n\t" + "swc1 $f3, 20(%[a2]) \n\t" + "swc1 $f1, 496(%[a1]) \n\t" + "swc1 $f0, 500(%[a1]) \n\t" + "addiu %[a2], %[a2], 16 \n\t" + "bne %[c2], %[c], 1b \n\t" + " addiu %[a1], %[a1], -16 \n\t" + ".set pop \n\t" + : [a] "+r" (a), [c] "+r" (c), [a1] "+r" (a1), [a2] "+r" (a2), + [c1] "+r" (c1), [c2] "+r" (c2), [wkr] "=&f" (wkr), [wki] "=&f" (wki), + [xr] "=&f" (xr), [xi] "=&f" (xi), [yr] "=&f" (yr), [yi] "=&f" (yi), + [aj20] "=&f" (aj20), [aj21] "=&f" (aj21), [ak20] "=&f" (ak20), + [ak21] "=&f" (ak21), [ck1] "=&f" (ck1) + : [temp] "f" (temp) + : "memory", "$f0", "$f1", "$f2", "$f3", "$f4", "$f5", "$f6", "$f7", "$f8" + ); +} + +void aec_rdft_init_mips(void) { + cft1st_128 = cft1st_128_mips; + cftmdl_128 = cftmdl_128_mips; + rftfsub_128 = rftfsub_128_mips; + rftbsub_128 = rftbsub_128_mips; + cftfsub_128 = cftfsub_128_mips; + cftbsub_128 = cftbsub_128_mips; + bitrv2_128 = bitrv2_128_mips; +} diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.c b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.c index bbdd5f628b2..ba3b9243e19 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.c +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.c @@ -104,18 +104,18 @@ int webrtc_aec_instance_count = 0; static void EstBufDelayNormal(aecpc_t* aecInst); static void EstBufDelayExtended(aecpc_t* aecInst); static int ProcessNormal(aecpc_t* self, - const int16_t* near, - const int16_t* near_high, - int16_t* out, - int16_t* out_high, + const float* near, + const float* near_high, + float* out, + float* out_high, int16_t num_samples, int16_t reported_delay_ms, int32_t skew); static void ProcessExtended(aecpc_t* self, - const int16_t* near, - const int16_t* near_high, - int16_t* out, - int16_t* out_high, + const float* near, + const float* near_high, + float* out, + float* out_high, int16_t num_samples, int16_t reported_delay_ms, int32_t skew); @@ -254,7 +254,7 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) { aecpc->checkBuffSize = 1; aecpc->firstVal = 0; - aecpc->startup_phase = 1; + aecpc->startup_phase = WebRtcAec_reported_delay_enabled(aecpc->aec); aecpc->bufSizeStart = 0; aecpc->checkBufSizeCtr = 0; aecpc->msInSndCardBuf = 0; @@ -372,10 +372,10 @@ int32_t WebRtcAec_BufferFarend(void* aecInst, } int32_t WebRtcAec_Process(void* aecInst, - const int16_t* nearend, - const int16_t* nearendH, - int16_t* out, - int16_t* outH, + const float* nearend, + const float* nearendH, + float* out, + float* outH, int16_t nrOfSamples, int16_t msInSndCardBuf, int32_t skew) { @@ -632,10 +632,10 @@ AecCore* WebRtcAec_aec_core(void* handle) { } static int ProcessNormal(aecpc_t* aecpc, - const int16_t* nearend, - const int16_t* nearendH, - int16_t* out, - int16_t* outH, + const float* nearend, + const float* nearendH, + float* out, + float* outH, int16_t nrOfSamples, int16_t msInSndCardBuf, int32_t skew) { @@ -689,10 +689,10 @@ static int ProcessNormal(aecpc_t* aecpc, if (aecpc->startup_phase) { // Only needed if they don't already point to the same place. if (nearend != out) { - memcpy(out, nearend, sizeof(short) * nrOfSamples); + memcpy(out, nearend, sizeof(*out) * nrOfSamples); } if (nearendH != outH) { - memcpy(outH, nearendH, sizeof(short) * nrOfSamples); + memcpy(outH, nearendH, sizeof(*outH) * nrOfSamples); } // The AEC is in the start up mode @@ -766,7 +766,9 @@ static int ProcessNormal(aecpc_t* aecpc, } } else { // AEC is enabled. - EstBufDelayNormal(aecpc); + if (WebRtcAec_reported_delay_enabled(aecpc->aec)) { + EstBufDelayNormal(aecpc); + } // Note that 1 frame is supported for NB and 2 frames for WB. for (i = 0; i < nFrames; i++) { @@ -787,10 +789,10 @@ static int ProcessNormal(aecpc_t* aecpc, } static void ProcessExtended(aecpc_t* self, - const int16_t* near, - const int16_t* near_high, - int16_t* out, - int16_t* out_high, + const float* near, + const float* near_high, + float* out, + float* out_high, int16_t num_samples, int16_t reported_delay_ms, int32_t skew) { @@ -821,10 +823,10 @@ static void ProcessExtended(aecpc_t* self, if (!self->farend_started) { // Only needed if they don't already point to the same place. if (near != out) { - memcpy(out, near, sizeof(short) * num_samples); + memcpy(out, near, sizeof(*out) * num_samples); } if (near_high != out_high) { - memcpy(out_high, near_high, sizeof(short) * num_samples); + memcpy(out_high, near_high, sizeof(*out_high) * num_samples); } return; } @@ -842,7 +844,9 @@ static void ProcessExtended(aecpc_t* self, self->startup_phase = 0; } - EstBufDelayExtended(self); + if (WebRtcAec_reported_delay_enabled(self->aec)) { + EstBufDelayExtended(self); + } { // |delay_diff_offset| gives us the option to manually rewind the delay on diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/include/echo_cancellation.h b/chromium/third_party/webrtc/modules/audio_processing/aec/include/echo_cancellation.h index 4c852cf64bf..dc64a345c3e 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/include/echo_cancellation.h +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/include/echo_cancellation.h @@ -68,7 +68,7 @@ extern "C" { * * Inputs Description * ------------------------------------------------------------------- - * void **aecInst Pointer to the AEC instance to be created + * void** aecInst Pointer to the AEC instance to be created * and initialized * * Outputs Description @@ -83,7 +83,7 @@ int32_t WebRtcAec_Create(void** aecInst); * * Inputs Description * ------------------------------------------------------------------- - * void *aecInst Pointer to the AEC instance + * void* aecInst Pointer to the AEC instance * * Outputs Description * ------------------------------------------------------------------- @@ -97,7 +97,7 @@ int32_t WebRtcAec_Free(void* aecInst); * * Inputs Description * ------------------------------------------------------------------- - * void *aecInst Pointer to the AEC instance + * void* aecInst Pointer to the AEC instance * int32_t sampFreq Sampling frequency of data * int32_t scSampFreq Soundcard sampling frequency * @@ -113,8 +113,8 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq); * * Inputs Description * ------------------------------------------------------------------- - * void *aecInst Pointer to the AEC instance - * int16_t *farend In buffer containing one frame of + * void* aecInst Pointer to the AEC instance + * int16_t* farend In buffer containing one frame of * farend signal for L band * int16_t nrOfSamples Number of samples in farend buffer * @@ -132,10 +132,10 @@ int32_t WebRtcAec_BufferFarend(void* aecInst, * * Inputs Description * ------------------------------------------------------------------- - * void *aecInst Pointer to the AEC instance - * int16_t *nearend In buffer containing one frame of + * void* aecInst Pointer to the AEC instance + * float* nearend In buffer containing one frame of * nearend+echo signal for L band - * int16_t *nearendH In buffer containing one frame of + * float* nearendH In buffer containing one frame of * nearend+echo signal for H band * int16_t nrOfSamples Number of samples in nearend buffer * int16_t msInSndCardBuf Delay estimate for sound card and @@ -146,18 +146,18 @@ int32_t WebRtcAec_BufferFarend(void* aecInst, * * Outputs Description * ------------------------------------------------------------------- - * int16_t *out Out buffer, one frame of processed nearend + * float* out Out buffer, one frame of processed nearend * for L band - * int16_t *outH Out buffer, one frame of processed nearend + * float* outH Out buffer, one frame of processed nearend * for H band * int32_t return 0: OK * -1: error */ int32_t WebRtcAec_Process(void* aecInst, - const int16_t* nearend, - const int16_t* nearendH, - int16_t* out, - int16_t* outH, + const float* nearend, + const float* nearendH, + float* out, + float* outH, int16_t nrOfSamples, int16_t msInSndCardBuf, int32_t skew); @@ -167,7 +167,7 @@ int32_t WebRtcAec_Process(void* aecInst, * * Inputs Description * ------------------------------------------------------------------- - * void *handle Pointer to the AEC instance + * void* handle Pointer to the AEC instance * AecConfig config Config instance that contains all * properties to be set * @@ -183,11 +183,11 @@ int WebRtcAec_set_config(void* handle, AecConfig config); * * Inputs Description * ------------------------------------------------------------------- - * void *handle Pointer to the AEC instance + * void* handle Pointer to the AEC instance * * Outputs Description * ------------------------------------------------------------------- - * int *status 0: Almost certainly nearend single-talk + * int* status 0: Almost certainly nearend single-talk * 1: Might not be neared single-talk * int return 0: OK * -1: error @@ -199,11 +199,11 @@ int WebRtcAec_get_echo_status(void* handle, int* status); * * Inputs Description * ------------------------------------------------------------------- - * void *handle Pointer to the AEC instance + * void* handle Pointer to the AEC instance * * Outputs Description * ------------------------------------------------------------------- - * AecMetrics *metrics Struct which will be filled out with the + * AecMetrics* metrics Struct which will be filled out with the * current echo metrics. * int return 0: OK * -1: error @@ -232,7 +232,7 @@ int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std); * * Inputs Description * ------------------------------------------------------------------- - * void *aecInst Pointer to the AEC instance + * void* aecInst Pointer to the AEC instance * * Outputs Description * ------------------------------------------------------------------- diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc index a19030ae350..a13d47622c9 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc @@ -9,12 +9,12 @@ */ #include "testing/gtest/include/gtest/gtest.h" - extern "C" { #include "webrtc/modules/audio_processing/aec/aec_core.h" } #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h" #include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h" +#include "webrtc/test/testsupport/gtest_disable.h" #include "webrtc/typedefs.h" namespace { @@ -46,16 +46,18 @@ class SystemDelayTest : public ::testing::Test { aecpc_t* self_; int samples_per_frame_; // Dummy input/output speech data. - int16_t far_[160]; - int16_t near_[160]; - int16_t out_[160]; + static const int kSamplesPerChunk = 160; + int16_t far_[kSamplesPerChunk]; + float near_[kSamplesPerChunk]; + float out_[kSamplesPerChunk]; }; SystemDelayTest::SystemDelayTest() : handle_(NULL), self_(NULL), samples_per_frame_(0) { // Dummy input data are set with more or less arbitrary non-zero values. memset(far_, 1, sizeof(far_)); - memset(near_, 2, sizeof(near_)); + for (int i = 0; i < kSamplesPerChunk; i++) + near_[i] = 514.0; memset(out_, 0, sizeof(out_)); } @@ -251,6 +253,9 @@ TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) { // conditions, but with an empty internal far-end buffer. Once that is done we // verify that the system delay is increased correctly until we have reach an // internal buffer size of 75% of what's been reported. + + // This test assumes the reported delays are used. + WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(handle_), 1); for (size_t i = 0; i < kNumSampleRates; i++) { Init(kSampleRateHz[i]); @@ -332,6 +337,9 @@ TEST_F(SystemDelayTest, CorrectDelayDuringDrift) { // device buffer. The drift is simulated by decreasing the reported device // buffer size by 1 ms every 100 ms. If the device buffer size goes below 30 // ms we jump (add) 10 ms to give a repeated pattern. + + // This test assumes the reported delays are used. + WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(handle_), 1); for (size_t i = 0; i < kNumSampleRates; i++) { Init(kSampleRateHz[i]); RunStableStartup(); @@ -365,6 +373,9 @@ TEST_F(SystemDelayTest, ShouldRecoverAfterGlitch) { // the device. // The system is said to be in a non-causal state if the difference between // the device buffer and system delay is less than a block (64 samples). + + // This test assumes the reported delays are used. + WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(handle_), 1); for (size_t i = 0; i < kNumSampleRates; i++) { Init(kSampleRateHz[i]); RunStableStartup(); diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.c index fc94f1b889e..0f34874612d 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.c +++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.c @@ -266,6 +266,9 @@ int WebRtcAecm_CreateCore(AecmCore_t **aecmInst) aecm = NULL; return -1; } + // TODO(bjornv): Explicitly disable robust delay validation until no + // performance regression has been established. Then remove the line. + WebRtc_enable_robust_validation(aecm->delay_estimator, 0); aecm->real_fft = WebRtcSpl_CreateRealFFT(PART_LEN_SHIFT); if (aecm->real_fft == NULL) { diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c index 63d4ac90280..f8491e97378 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c +++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c @@ -260,7 +260,7 @@ static int TimeToFrequencyDomain(AecmCore_t* aecm, __asm __volatile( "smulbb %[tmp32no1], %[real], %[real]\n\t" "smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t" - :[tmp32no1]"+r"(tmp32no1), + :[tmp32no1]"+&r"(tmp32no1), [tmp32no2]"=r"(tmp32no2) :[real]"r"(freq_signal[i].real), [imag]"r"(freq_signal[i].imag) diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c index b896de0a214..088bbf03f74 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c +++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c @@ -443,27 +443,14 @@ int32_t WebRtcAecm_Process(void *aecmInst, const int16_t *nearendNoisy, // Call the AECM /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i], &out[FRAME_LEN * i], aecm->knownDelay);*/ - if (nearendClean == NULL) - { - if (WebRtcAecm_ProcessFrame(aecm->aecmCore, - farend_ptr, - &nearendNoisy[FRAME_LEN * i], - NULL, - &out[FRAME_LEN * i]) == -1) - { - return -1; - } - } else - { - if (WebRtcAecm_ProcessFrame(aecm->aecmCore, - farend_ptr, - &nearendNoisy[FRAME_LEN * i], - &nearendClean[FRAME_LEN * i], - &out[FRAME_LEN * i]) == -1) - { - return -1; - } - } + if (WebRtcAecm_ProcessFrame(aecm->aecmCore, + farend_ptr, + &nearendNoisy[FRAME_LEN * i], + (nearendClean + ? &nearendClean[FRAME_LEN * i] + : NULL), + &out[FRAME_LEN * i]) == -1) + return -1; } } diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h b/chromium/third_party/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h index 8ea2e87e2ff..ac43576dd26 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h +++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h @@ -45,7 +45,7 @@ extern "C" { * * Inputs Description * ------------------------------------------------------------------- - * void **aecmInst Pointer to the AECM instance to be + * void** aecmInst Pointer to the AECM instance to be * created and initialized * * Outputs Description @@ -60,11 +60,11 @@ int32_t WebRtcAecm_Create(void **aecmInst); * * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance + * void* aecmInst Pointer to the AECM instance * * Outputs Description * ------------------------------------------------------------------- - * int32_t return 0: OK + * int32_t return 0: OK * -1: error */ int32_t WebRtcAecm_Free(void *aecmInst); @@ -74,7 +74,7 @@ int32_t WebRtcAecm_Free(void *aecmInst); * * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance + * void* aecmInst Pointer to the AECM instance * int32_t sampFreq Sampling frequency of data * * Outputs Description @@ -89,8 +89,8 @@ int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq); * * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance - * int16_t *farend In buffer containing one frame of + * void* aecmInst Pointer to the AECM instance + * int16_t* farend In buffer containing one frame of * farend signal * int16_t nrOfSamples Number of samples in farend buffer * @@ -106,14 +106,14 @@ int32_t WebRtcAecm_BufferFarend(void* aecmInst, /* * Runs the AECM on an 80 or 160 sample blocks of data. * - * Inputs Description + * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance - * int16_t *nearendNoisy In buffer containing one frame of + * void* aecmInst Pointer to the AECM instance + * int16_t* nearendNoisy In buffer containing one frame of * reference nearend+echo signal. If * noise reduction is active, provide * the noisy signal here. - * int16_t *nearendClean In buffer containing one frame of + * int16_t* nearendClean In buffer containing one frame of * nearend+echo signal. If noise * reduction is active, provide the * clean signal here. Otherwise pass a @@ -122,11 +122,11 @@ int32_t WebRtcAecm_BufferFarend(void* aecmInst, * int16_t msInSndCardBuf Delay estimate for sound card and * system buffers * - * Outputs Description + * Outputs Description * ------------------------------------------------------------------- - * int16_t *out Out buffer, one frame of processed nearend - * int32_t return 0: OK - * -1: error + * int16_t* out Out buffer, one frame of processed nearend + * int32_t return 0: OK + * -1: error */ int32_t WebRtcAecm_Process(void* aecmInst, const int16_t* nearendNoisy, @@ -140,8 +140,8 @@ int32_t WebRtcAecm_Process(void* aecmInst, * * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance - * AecmConfig config Config instance that contains all + * void* aecmInst Pointer to the AECM instance + * AecmConfig config Config instance that contains all * properties to be set * * Outputs Description @@ -156,11 +156,11 @@ int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config); * * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance + * void* aecmInst Pointer to the AECM instance * * Outputs Description * ------------------------------------------------------------------- - * AecmConfig *config Pointer to the config instance that + * AecmConfig* config Pointer to the config instance that * all properties will be written to * int32_t return 0: OK * -1: error @@ -178,7 +178,7 @@ int32_t WebRtcAecm_get_config(void *aecmInst, AecmConfig *config); * * Outputs Description * ------------------------------------------------------------------- - * int32_t return 0: OK + * int32_t return 0: OK * -1: error */ int32_t WebRtcAecm_InitEchoPath(void* aecmInst, @@ -197,7 +197,7 @@ int32_t WebRtcAecm_InitEchoPath(void* aecmInst, * * Outputs Description * ------------------------------------------------------------------- - * int32_t return 0: OK + * int32_t return 0: OK * -1: error */ int32_t WebRtcAecm_GetEchoPath(void* aecmInst, @@ -209,7 +209,7 @@ int32_t WebRtcAecm_GetEchoPath(void* aecmInst, * * Outputs Description * ------------------------------------------------------------------- - * size_t return : size in bytes + * size_t return Size in bytes */ size_t WebRtcAecm_echo_path_size_bytes(); @@ -218,7 +218,7 @@ size_t WebRtcAecm_echo_path_size_bytes(); * * Inputs Description * ------------------------------------------------------------------- - * void *aecmInst Pointer to the AECM instance + * void* aecmInst Pointer to the AECM instance * * Outputs Description * ------------------------------------------------------------------- diff --git a/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.c b/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.c index 1e8e3d86b2d..4f110cc2092 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.c +++ b/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.c @@ -822,10 +822,16 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, if (inMicLevelTmp != stt->micVol) { - // Incoming level mismatch; update our level. - // This could be the case if the volume is changed manually, or if the - // sound device has a low volume resolution. - stt->micVol = inMicLevelTmp; + if (inMicLevel == stt->lastInMicLevel) { + // We requested a volume adjustment, but it didn't occur. This is + // probably due to a coarse quantization of the volume slider. + // Restore the requested value to prevent getting stuck. + inMicLevelTmp = stt->micVol; + } + else { + // As long as the value changed, update to match. + stt->micVol = inMicLevelTmp; + } } if (inMicLevelTmp > stt->maxLevel) @@ -835,6 +841,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, } // Store last value here, after we've taken care of manual updates etc. + stt->lastInMicLevel = inMicLevel; lastMicVol = stt->micVol; /* Checks if the signal is saturated. Also a check if individual samples @@ -1597,6 +1604,7 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel, stt->maxInit = stt->maxLevel; stt->zeroCtrlMax = stt->maxAnalog; + stt->lastInMicLevel = 0; /* Initialize micVol parameter */ stt->micVol = stt->maxAnalog; diff --git a/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.h b/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.h index 16ea29c4961..b036f449c70 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.h +++ b/chromium/third_party/webrtc/modules/audio_processing/agc/analog_agc.h @@ -111,6 +111,7 @@ typedef struct int32_t minLevel; // Minimum possible volume level int32_t minOutput; // Minimum output volume level int32_t zeroCtrlMax; // Remember max gain => don't amp low input + int32_t lastInMicLevel; int16_t scale; // Scale factor for internal volume levels #ifdef MIC_LEVEL_FEEDBACK diff --git a/chromium/third_party/webrtc/modules/audio_processing/agc/digital_agc.c b/chromium/third_party/webrtc/modules/audio_processing/agc/digital_agc.c index 00565dd7230..4b169c180eb 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/agc/digital_agc.c +++ b/chromium/third_party/webrtc/modules/audio_processing/agc/digital_agc.c @@ -118,7 +118,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16 limiterLvlX = analogTarget - limiterOffset; limiterIdx = 2 + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((int32_t)limiterLvlX, 13), - WEBRTC_SPL_RSHIFT_U16(kLog10_2, 1)); + (kLog10_2 / 2)); tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio); limiterLvl = targetLevelDbfs + tmp16no1; @@ -288,12 +288,7 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc_t *stt, int16_t agcMode) int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const int16_t *in_far, int16_t nrSamples) { - // Check for valid pointer - if (&stt->vadFarend == NULL) - { - return -1; - } - + assert(stt != NULL); // VAD for far end WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples); @@ -778,7 +773,7 @@ int16_t WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12); tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm)); tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm); - tmpU16 = WEBRTC_SPL_LSHIFT_U16((uint16_t)13, 12); + tmpU16 = (13 << 12); tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16); tmp32 += WEBRTC_SPL_RSHIFT_W32(tmp32b, 10); diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.cc b/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.cc index 048d048723e..b0f1eb6c1db 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.cc @@ -10,6 +10,8 @@ #include "webrtc/modules/audio_processing/audio_buffer.h" +#include "webrtc/common_audio/include/audio_util.h" +#include "webrtc/common_audio/resampler/push_sinc_resampler.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" namespace webrtc { @@ -21,145 +23,343 @@ enum { kSamplesPer32kHzChannel = 320 }; -void StereoToMono(const int16_t* left, const int16_t* right, - int16_t* out, int samples_per_channel) { - assert(left != NULL && right != NULL && out != NULL); - for (int i = 0; i < samples_per_channel; i++) { - int32_t data32 = (static_cast<int32_t>(left[i]) + - static_cast<int32_t>(right[i])) >> 1; +bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) { + switch (layout) { + case AudioProcessing::kMono: + case AudioProcessing::kStereo: + return false; + case AudioProcessing::kMonoAndKeyboard: + case AudioProcessing::kStereoAndKeyboard: + return true; + } + assert(false); + return false; +} + +int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) { + switch (layout) { + case AudioProcessing::kMono: + case AudioProcessing::kStereo: + assert(false); + return -1; + case AudioProcessing::kMonoAndKeyboard: + return 1; + case AudioProcessing::kStereoAndKeyboard: + return 2; + } + assert(false); + return -1; +} + - out[i] = WebRtcSpl_SatW32ToW16(data32); +void StereoToMono(const float* left, const float* right, float* out, + int samples_per_channel) { + for (int i = 0; i < samples_per_channel; ++i) { + out[i] = (left[i] + right[i]) / 2; } } + +void StereoToMono(const int16_t* left, const int16_t* right, int16_t* out, + int samples_per_channel) { + for (int i = 0; i < samples_per_channel; ++i) { + out[i] = (left[i] + right[i]) >> 1; + } +} + } // namespace -struct AudioChannel { - AudioChannel() { - memset(data, 0, sizeof(data)); +// One int16_t and one float ChannelBuffer that are kept in sync. The sync is +// broken when someone requests write access to either ChannelBuffer, and +// reestablished when someone requests the outdated ChannelBuffer. It is +// therefore safe to use the return value of ibuf() and fbuf() until the next +// call to the other method. +class IFChannelBuffer { + public: + IFChannelBuffer(int samples_per_channel, int num_channels) + : ivalid_(true), + ibuf_(samples_per_channel, num_channels), + fvalid_(true), + fbuf_(samples_per_channel, num_channels) {} + + ChannelBuffer<int16_t>* ibuf() { + RefreshI(); + fvalid_ = false; + return &ibuf_; + } + + ChannelBuffer<float>* fbuf() { + RefreshF(); + ivalid_ = false; + return &fbuf_; } - int16_t data[kSamplesPer32kHzChannel]; + private: + void RefreshF() { + if (!fvalid_) { + assert(ivalid_); + const int16_t* const int_data = ibuf_.data(); + float* const float_data = fbuf_.data(); + const int length = fbuf_.length(); + for (int i = 0; i < length; ++i) + float_data[i] = int_data[i]; + fvalid_ = true; + } + } + + void RefreshI() { + if (!ivalid_) { + assert(fvalid_); + const float* const float_data = fbuf_.data(); + int16_t* const int_data = ibuf_.data(); + const int length = ibuf_.length(); + for (int i = 0; i < length; ++i) + int_data[i] = WEBRTC_SPL_SAT(std::numeric_limits<int16_t>::max(), + float_data[i], + std::numeric_limits<int16_t>::min()); + ivalid_ = true; + } + } + + bool ivalid_; + ChannelBuffer<int16_t> ibuf_; + bool fvalid_; + ChannelBuffer<float> fbuf_; }; -struct SplitAudioChannel { - SplitAudioChannel() { - memset(low_pass_data, 0, sizeof(low_pass_data)); - memset(high_pass_data, 0, sizeof(high_pass_data)); - memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); - memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); - memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); - memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); +class SplitChannelBuffer { + public: + SplitChannelBuffer(int samples_per_split_channel, int num_channels) + : low_(samples_per_split_channel, num_channels), + high_(samples_per_split_channel, num_channels) { } + ~SplitChannelBuffer() {} - int16_t low_pass_data[kSamplesPer16kHzChannel]; - int16_t high_pass_data[kSamplesPer16kHzChannel]; + int16_t* low_channel(int i) { return low_.ibuf()->channel(i); } + int16_t* high_channel(int i) { return high_.ibuf()->channel(i); } + float* low_channel_f(int i) { return low_.fbuf()->channel(i); } + float* high_channel_f(int i) { return high_.fbuf()->channel(i); } - int32_t analysis_filter_state1[6]; - int32_t analysis_filter_state2[6]; - int32_t synthesis_filter_state1[6]; - int32_t synthesis_filter_state2[6]; + private: + IFChannelBuffer low_; + IFChannelBuffer high_; }; -// TODO(andrew): check range of input parameters? -AudioBuffer::AudioBuffer(int max_num_channels, - int samples_per_channel) - : max_num_channels_(max_num_channels), - num_channels_(0), +AudioBuffer::AudioBuffer(int input_samples_per_channel, + int num_input_channels, + int process_samples_per_channel, + int num_process_channels, + int output_samples_per_channel) + : input_samples_per_channel_(input_samples_per_channel), + num_input_channels_(num_input_channels), + proc_samples_per_channel_(process_samples_per_channel), + num_proc_channels_(num_process_channels), + output_samples_per_channel_(output_samples_per_channel), + samples_per_split_channel_(proc_samples_per_channel_), num_mixed_channels_(0), num_mixed_low_pass_channels_(0), - data_was_mixed_(false), - samples_per_channel_(samples_per_channel), - samples_per_split_channel_(samples_per_channel), reference_copied_(false), activity_(AudioFrame::kVadUnknown), - is_muted_(false), - data_(NULL), - channels_(NULL), - split_channels_(NULL), - mixed_channels_(NULL), - mixed_low_pass_channels_(NULL), - low_pass_reference_channels_(NULL) { - if (max_num_channels_ > 1) { - channels_.reset(new AudioChannel[max_num_channels_]); - mixed_channels_.reset(new AudioChannel[max_num_channels_]); - mixed_low_pass_channels_.reset(new AudioChannel[max_num_channels_]); + keyboard_data_(NULL), + channels_(new IFChannelBuffer(proc_samples_per_channel_, + num_proc_channels_)) { + assert(input_samples_per_channel_ > 0); + assert(proc_samples_per_channel_ > 0); + assert(output_samples_per_channel_ > 0); + assert(num_input_channels_ > 0 && num_input_channels_ <= 2); + assert(num_proc_channels_ <= num_input_channels); + + if (num_input_channels_ == 2 && num_proc_channels_ == 1) { + input_buffer_.reset(new ChannelBuffer<float>(input_samples_per_channel_, + num_proc_channels_)); + } + + if (input_samples_per_channel_ != proc_samples_per_channel_ || + output_samples_per_channel_ != proc_samples_per_channel_) { + // Create an intermediate buffer for resampling. + process_buffer_.reset(new ChannelBuffer<float>(proc_samples_per_channel_, + num_proc_channels_)); + } + + if (input_samples_per_channel_ != proc_samples_per_channel_) { + input_resamplers_.reserve(num_proc_channels_); + for (int i = 0; i < num_proc_channels_; ++i) { + input_resamplers_.push_back( + new PushSincResampler(input_samples_per_channel_, + proc_samples_per_channel_)); + } } - low_pass_reference_channels_.reset(new AudioChannel[max_num_channels_]); - if (samples_per_channel_ == kSamplesPer32kHzChannel) { - split_channels_.reset(new SplitAudioChannel[max_num_channels_]); + if (output_samples_per_channel_ != proc_samples_per_channel_) { + output_resamplers_.reserve(num_proc_channels_); + for (int i = 0; i < num_proc_channels_; ++i) { + output_resamplers_.push_back( + new PushSincResampler(proc_samples_per_channel_, + output_samples_per_channel_)); + } + } + + if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) { samples_per_split_channel_ = kSamplesPer16kHzChannel; + split_channels_.reset(new SplitChannelBuffer(samples_per_split_channel_, + num_proc_channels_)); + filter_states_.reset(new SplitFilterStates[num_proc_channels_]); } } AudioBuffer::~AudioBuffer() {} -int16_t* AudioBuffer::data(int channel) const { - assert(channel >= 0 && channel < num_channels_); - if (data_ != NULL) { - return data_; +void AudioBuffer::CopyFrom(const float* const* data, + int samples_per_channel, + AudioProcessing::ChannelLayout layout) { + assert(samples_per_channel == input_samples_per_channel_); + assert(ChannelsFromLayout(layout) == num_input_channels_); + InitForNewData(); + + if (HasKeyboardChannel(layout)) { + keyboard_data_ = data[KeyboardChannelIndex(layout)]; } - return channels_[channel].data; -} + // Downmix. + const float* const* data_ptr = data; + if (num_input_channels_ == 2 && num_proc_channels_ == 1) { + StereoToMono(data[0], + data[1], + input_buffer_->channel(0), + input_samples_per_channel_); + data_ptr = input_buffer_->channels(); + } -int16_t* AudioBuffer::low_pass_split_data(int channel) const { - assert(channel >= 0 && channel < num_channels_); - if (split_channels_.get() == NULL) { - return data(channel); + // Resample. + if (input_samples_per_channel_ != proc_samples_per_channel_) { + for (int i = 0; i < num_proc_channels_; ++i) { + input_resamplers_[i]->Resample(data_ptr[i], + input_samples_per_channel_, + process_buffer_->channel(i), + proc_samples_per_channel_); + } + data_ptr = process_buffer_->channels(); } - return split_channels_[channel].low_pass_data; + // Convert to int16. + for (int i = 0; i < num_proc_channels_; ++i) { + ScaleAndRoundToInt16(data_ptr[i], proc_samples_per_channel_, + channels_->ibuf()->channel(i)); + } } -int16_t* AudioBuffer::high_pass_split_data(int channel) const { - assert(channel >= 0 && channel < num_channels_); - if (split_channels_.get() == NULL) { - return NULL; +void AudioBuffer::CopyTo(int samples_per_channel, + AudioProcessing::ChannelLayout layout, + float* const* data) { + assert(samples_per_channel == output_samples_per_channel_); + assert(ChannelsFromLayout(layout) == num_proc_channels_); + + // Convert to float. + float* const* data_ptr = data; + if (output_samples_per_channel_ != proc_samples_per_channel_) { + // Convert to an intermediate buffer for subsequent resampling. + data_ptr = process_buffer_->channels(); + } + for (int i = 0; i < num_proc_channels_; ++i) { + ScaleToFloat(channels_->ibuf()->channel(i), + proc_samples_per_channel_, + data_ptr[i]); + } + + // Resample. + if (output_samples_per_channel_ != proc_samples_per_channel_) { + for (int i = 0; i < num_proc_channels_; ++i) { + output_resamplers_[i]->Resample(data_ptr[i], + proc_samples_per_channel_, + data[i], + output_samples_per_channel_); + } } +} - return split_channels_[channel].high_pass_data; +void AudioBuffer::InitForNewData() { + keyboard_data_ = NULL; + num_mixed_channels_ = 0; + num_mixed_low_pass_channels_ = 0; + reference_copied_ = false; + activity_ = AudioFrame::kVadUnknown; } -int16_t* AudioBuffer::mixed_data(int channel) const { +const int16_t* AudioBuffer::data(int channel) const { + assert(channel >= 0 && channel < num_proc_channels_); + return channels_->ibuf()->channel(channel); +} + +int16_t* AudioBuffer::data(int channel) { + const AudioBuffer* t = this; + return const_cast<int16_t*>(t->data(channel)); +} + +float* AudioBuffer::data_f(int channel) { + assert(channel >= 0 && channel < num_proc_channels_); + return channels_->fbuf()->channel(channel); +} + +const int16_t* AudioBuffer::low_pass_split_data(int channel) const { + assert(channel >= 0 && channel < num_proc_channels_); + return split_channels_.get() ? split_channels_->low_channel(channel) + : data(channel); +} + +int16_t* AudioBuffer::low_pass_split_data(int channel) { + const AudioBuffer* t = this; + return const_cast<int16_t*>(t->low_pass_split_data(channel)); +} + +float* AudioBuffer::low_pass_split_data_f(int channel) { + assert(channel >= 0 && channel < num_proc_channels_); + return split_channels_.get() ? split_channels_->low_channel_f(channel) + : data_f(channel); +} + +const int16_t* AudioBuffer::high_pass_split_data(int channel) const { + assert(channel >= 0 && channel < num_proc_channels_); + return split_channels_.get() ? split_channels_->high_channel(channel) : NULL; +} + +int16_t* AudioBuffer::high_pass_split_data(int channel) { + const AudioBuffer* t = this; + return const_cast<int16_t*>(t->high_pass_split_data(channel)); +} + +float* AudioBuffer::high_pass_split_data_f(int channel) { + assert(channel >= 0 && channel < num_proc_channels_); + return split_channels_.get() ? split_channels_->high_channel_f(channel) + : NULL; +} + +const int16_t* AudioBuffer::mixed_data(int channel) const { assert(channel >= 0 && channel < num_mixed_channels_); - return mixed_channels_[channel].data; + return mixed_channels_->channel(channel); } -int16_t* AudioBuffer::mixed_low_pass_data(int channel) const { +const int16_t* AudioBuffer::mixed_low_pass_data(int channel) const { assert(channel >= 0 && channel < num_mixed_low_pass_channels_); - return mixed_low_pass_channels_[channel].data; + return mixed_low_pass_channels_->channel(channel); } -int16_t* AudioBuffer::low_pass_reference(int channel) const { - assert(channel >= 0 && channel < num_channels_); +const int16_t* AudioBuffer::low_pass_reference(int channel) const { + assert(channel >= 0 && channel < num_proc_channels_); if (!reference_copied_) { return NULL; } - return low_pass_reference_channels_[channel].data; -} - -int32_t* AudioBuffer::analysis_filter_state1(int channel) const { - assert(channel >= 0 && channel < num_channels_); - return split_channels_[channel].analysis_filter_state1; -} - -int32_t* AudioBuffer::analysis_filter_state2(int channel) const { - assert(channel >= 0 && channel < num_channels_); - return split_channels_[channel].analysis_filter_state2; + return low_pass_reference_channels_->channel(channel); } -int32_t* AudioBuffer::synthesis_filter_state1(int channel) const { - assert(channel >= 0 && channel < num_channels_); - return split_channels_[channel].synthesis_filter_state1; +const float* AudioBuffer::keyboard_data() const { + return keyboard_data_; } -int32_t* AudioBuffer::synthesis_filter_state2(int channel) const { - assert(channel >= 0 && channel < num_channels_); - return split_channels_[channel].synthesis_filter_state2; +SplitFilterStates* AudioBuffer::filter_states(int channel) { + assert(channel >= 0 && channel < num_proc_channels_); + return &filter_states_[channel]; } void AudioBuffer::set_activity(AudioFrame::VADActivity activity) { @@ -170,126 +370,96 @@ AudioFrame::VADActivity AudioBuffer::activity() const { return activity_; } -bool AudioBuffer::is_muted() const { - return is_muted_; -} - int AudioBuffer::num_channels() const { - return num_channels_; + return num_proc_channels_; } int AudioBuffer::samples_per_channel() const { - return samples_per_channel_; + return proc_samples_per_channel_; } int AudioBuffer::samples_per_split_channel() const { return samples_per_split_channel_; } +int AudioBuffer::samples_per_keyboard_channel() const { + // We don't resample the keyboard channel. + return input_samples_per_channel_; +} + // TODO(andrew): Do deinterleaving and mixing in one step? void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) { - assert(frame->num_channels_ <= max_num_channels_); - assert(frame->samples_per_channel_ == samples_per_channel_); - - num_channels_ = frame->num_channels_; - data_was_mixed_ = false; - num_mixed_channels_ = 0; - num_mixed_low_pass_channels_ = 0; - reference_copied_ = false; + assert(proc_samples_per_channel_ == input_samples_per_channel_); + assert(num_proc_channels_ == num_input_channels_); + assert(frame->num_channels_ == num_proc_channels_); + assert(frame->samples_per_channel_ == proc_samples_per_channel_); + InitForNewData(); activity_ = frame->vad_activity_; - is_muted_ = false; - if (frame->energy_ == 0) { - is_muted_ = true; - } - - if (num_channels_ == 1) { - // We can get away with a pointer assignment in this case. - data_ = frame->data_; - return; - } int16_t* interleaved = frame->data_; - for (int i = 0; i < num_channels_; i++) { - int16_t* deinterleaved = channels_[i].data; + for (int i = 0; i < num_proc_channels_; i++) { + int16_t* deinterleaved = channels_->ibuf()->channel(i); int interleaved_idx = i; - for (int j = 0; j < samples_per_channel_; j++) { + for (int j = 0; j < proc_samples_per_channel_; j++) { deinterleaved[j] = interleaved[interleaved_idx]; - interleaved_idx += num_channels_; + interleaved_idx += num_proc_channels_; } } } void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const { - assert(frame->num_channels_ == num_channels_); - assert(frame->samples_per_channel_ == samples_per_channel_); + assert(proc_samples_per_channel_ == output_samples_per_channel_); + assert(num_proc_channels_ == num_input_channels_); + assert(frame->num_channels_ == num_proc_channels_); + assert(frame->samples_per_channel_ == proc_samples_per_channel_); frame->vad_activity_ = activity_; if (!data_changed) { return; } - if (num_channels_ == 1) { - if (data_was_mixed_) { - memcpy(frame->data_, - channels_[0].data, - sizeof(int16_t) * samples_per_channel_); - } else { - // These should point to the same buffer in this case. - assert(data_ == frame->data_); - } - - return; - } - int16_t* interleaved = frame->data_; - for (int i = 0; i < num_channels_; i++) { - int16_t* deinterleaved = channels_[i].data; + for (int i = 0; i < num_proc_channels_; i++) { + int16_t* deinterleaved = channels_->ibuf()->channel(i); int interleaved_idx = i; - for (int j = 0; j < samples_per_channel_; j++) { + for (int j = 0; j < proc_samples_per_channel_; j++) { interleaved[interleaved_idx] = deinterleaved[j]; - interleaved_idx += num_channels_; + interleaved_idx += num_proc_channels_; } } } -// TODO(andrew): would be good to support the no-mix case with pointer -// assignment. -// TODO(andrew): handle mixing to multiple channels? -void AudioBuffer::Mix(int num_mixed_channels) { - // We currently only support the stereo to mono case. - assert(num_channels_ == 2); - assert(num_mixed_channels == 1); - - StereoToMono(channels_[0].data, - channels_[1].data, - channels_[0].data, - samples_per_channel_); - - num_channels_ = num_mixed_channels; - data_was_mixed_ = true; -} - void AudioBuffer::CopyAndMix(int num_mixed_channels) { // We currently only support the stereo to mono case. - assert(num_channels_ == 2); + assert(num_proc_channels_ == 2); assert(num_mixed_channels == 1); + if (!mixed_channels_.get()) { + mixed_channels_.reset( + new ChannelBuffer<int16_t>(proc_samples_per_channel_, + num_mixed_channels)); + } - StereoToMono(channels_[0].data, - channels_[1].data, - mixed_channels_[0].data, - samples_per_channel_); + StereoToMono(channels_->ibuf()->channel(0), + channels_->ibuf()->channel(1), + mixed_channels_->channel(0), + proc_samples_per_channel_); num_mixed_channels_ = num_mixed_channels; } void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) { // We currently only support the stereo to mono case. - assert(num_channels_ == 2); + assert(num_proc_channels_ == 2); assert(num_mixed_channels == 1); + if (!mixed_low_pass_channels_.get()) { + mixed_low_pass_channels_.reset( + new ChannelBuffer<int16_t>(samples_per_split_channel_, + num_mixed_channels)); + } StereoToMono(low_pass_split_data(0), low_pass_split_data(1), - mixed_low_pass_channels_[0].data, + mixed_low_pass_channels_->channel(0), samples_per_split_channel_); num_mixed_low_pass_channels_ = num_mixed_channels; @@ -297,10 +467,14 @@ void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) { void AudioBuffer::CopyLowPassToReference() { reference_copied_ = true; - for (int i = 0; i < num_channels_; i++) { - memcpy(low_pass_reference_channels_[i].data, - low_pass_split_data(i), - sizeof(int16_t) * samples_per_split_channel_); + if (!low_pass_reference_channels_.get()) { + low_pass_reference_channels_.reset( + new ChannelBuffer<int16_t>(samples_per_split_channel_, + num_proc_channels_)); + } + for (int i = 0; i < num_proc_channels_; i++) { + low_pass_reference_channels_->CopyFrom(low_pass_split_data(i), i); } } + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h b/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h index 2638bef6058..67e4f485043 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h @@ -8,75 +8,124 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ +#include <vector> + +#include "webrtc/modules/audio_processing/common.h" +#include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/scoped_vector.h" #include "webrtc/typedefs.h" namespace webrtc { -struct AudioChannel; -struct SplitAudioChannel; +class PushSincResampler; +class SplitChannelBuffer; +class IFChannelBuffer; + +struct SplitFilterStates { + SplitFilterStates() { + memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); + memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); + memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); + memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); + } + + static const int kStateSize = 6; + int analysis_filter_state1[kStateSize]; + int analysis_filter_state2[kStateSize]; + int synthesis_filter_state1[kStateSize]; + int synthesis_filter_state2[kStateSize]; +}; class AudioBuffer { public: - AudioBuffer(int max_num_channels, int samples_per_channel); + // TODO(ajm): Switch to take ChannelLayouts. + AudioBuffer(int input_samples_per_channel, + int num_input_channels, + int process_samples_per_channel, + int num_process_channels, + int output_samples_per_channel); virtual ~AudioBuffer(); int num_channels() const; int samples_per_channel() const; int samples_per_split_channel() const; + int samples_per_keyboard_channel() const; - int16_t* data(int channel) const; - int16_t* low_pass_split_data(int channel) const; - int16_t* high_pass_split_data(int channel) const; - int16_t* mixed_data(int channel) const; - int16_t* mixed_low_pass_data(int channel) const; - int16_t* low_pass_reference(int channel) const; + int16_t* data(int channel); + const int16_t* data(int channel) const; + int16_t* low_pass_split_data(int channel); + const int16_t* low_pass_split_data(int channel) const; + int16_t* high_pass_split_data(int channel); + const int16_t* high_pass_split_data(int channel) const; + const int16_t* mixed_data(int channel) const; + const int16_t* mixed_low_pass_data(int channel) const; + const int16_t* low_pass_reference(int channel) const; - int32_t* analysis_filter_state1(int channel) const; - int32_t* analysis_filter_state2(int channel) const; - int32_t* synthesis_filter_state1(int channel) const; - int32_t* synthesis_filter_state2(int channel) const; + // Float versions of the accessors, with automatic conversion back and forth + // as necessary. The range of the numbers are the same as for int16_t. + float* data_f(int channel); + float* low_pass_split_data_f(int channel); + float* high_pass_split_data_f(int channel); + + const float* keyboard_data() const; + + SplitFilterStates* filter_states(int channel); void set_activity(AudioFrame::VADActivity activity); AudioFrame::VADActivity activity() const; - bool is_muted() const; - + // Use for int16 interleaved data. void DeinterleaveFrom(AudioFrame* audioFrame); void InterleaveTo(AudioFrame* audioFrame) const; // If |data_changed| is false, only the non-audio data members will be copied // to |frame|. void InterleaveTo(AudioFrame* frame, bool data_changed) const; - void Mix(int num_mixed_channels); + + // Use for float deinterleaved data. + void CopyFrom(const float* const* data, + int samples_per_channel, + AudioProcessing::ChannelLayout layout); + void CopyTo(int samples_per_channel, + AudioProcessing::ChannelLayout layout, + float* const* data); + void CopyAndMix(int num_mixed_channels); void CopyAndMixLowPass(int num_mixed_channels); void CopyLowPassToReference(); private: - const int max_num_channels_; - int num_channels_; + // Called from DeinterleaveFrom() and CopyFrom(). + void InitForNewData(); + + const int input_samples_per_channel_; + const int num_input_channels_; + const int proc_samples_per_channel_; + const int num_proc_channels_; + const int output_samples_per_channel_; + int samples_per_split_channel_; int num_mixed_channels_; int num_mixed_low_pass_channels_; - // Whether the original data was replaced with mixed data. - bool data_was_mixed_; - const int samples_per_channel_; - int samples_per_split_channel_; bool reference_copied_; AudioFrame::VADActivity activity_; - bool is_muted_; - - int16_t* data_; - scoped_array<AudioChannel> channels_; - scoped_array<SplitAudioChannel> split_channels_; - scoped_array<AudioChannel> mixed_channels_; - // TODO(andrew): improve this, we don't need the full 32 kHz space here. - scoped_array<AudioChannel> mixed_low_pass_channels_; - scoped_array<AudioChannel> low_pass_reference_channels_; + + const float* keyboard_data_; + scoped_ptr<IFChannelBuffer> channels_; + scoped_ptr<SplitChannelBuffer> split_channels_; + scoped_ptr<SplitFilterStates[]> filter_states_; + scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_; + scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_; + scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; + scoped_ptr<ChannelBuffer<float> > input_buffer_; + scoped_ptr<ChannelBuffer<float> > process_buffer_; + ScopedVector<PushSincResampler> input_resamplers_; + ScopedVector<PushSincResampler> output_resamplers_; }; + } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_ +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi b/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi index 336b4eee75f..b1d18c5b06a 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi @@ -12,6 +12,7 @@ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', ], + 'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets', }, 'targets': [ { @@ -53,9 +54,9 @@ 'audio_buffer.h', 'audio_processing_impl.cc', 'audio_processing_impl.h', + 'common.h', 'echo_cancellation_impl.cc', 'echo_cancellation_impl.h', - 'echo_cancellation_impl_wrapper.h', 'echo_control_mobile_impl.cc', 'echo_control_mobile_impl.h', 'gain_control_impl.cc', @@ -67,10 +68,12 @@ 'level_estimator_impl.h', 'noise_suppression_impl.cc', 'noise_suppression_impl.h', - 'splitting_filter.cc', - 'splitting_filter.h', 'processing_component.cc', 'processing_component.h', + 'rms_level.cc', + 'rms_level.h', + 'typing_detection.cc', + 'typing_detection.h', 'utility/delay_estimator.c', 'utility/delay_estimator.h', 'utility/delay_estimator_internal.h', @@ -103,6 +106,17 @@ 'ns/nsx_core.h', 'ns/nsx_defines.h', ], + 'conditions': [ + ['target_arch=="mipsel"', { + 'sources': [ + 'ns/nsx_core_mips.c', + ], + }, { + 'sources': [ + 'ns/nsx_core_c.c', + ], + }], + ], }, { 'defines': ['WEBRTC_NS_FLOAT'], 'sources': [ @@ -124,6 +138,14 @@ 'sources': [ 'aecm/aecm_core_mips.c', ], + 'conditions': [ + ['mips_fpu==1', { + 'sources': [ + 'aec/aec_core_mips.c', + 'aec/aec_rdft_mips.c', + ], + }], + ], }, { 'sources': [ 'aecm/aecm_core_c.c', @@ -177,18 +199,22 @@ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', ], 'sources': [ + 'aec/aec_core_neon.c', 'aecm/aecm_core_neon.c', 'ns/nsx_core_neon.c', ], 'conditions': [ ['OS=="android" or OS=="ios"', { 'dependencies': [ - 'audio_processing_offsets', + '<(gen_core_neon_offsets_gyp):*', ], 'sources': [ 'aecm/aecm_core_neon.S', 'ns/nsx_core_neon.S', ], + 'include_dirs': [ + '<(shared_generated_dir)', + ], 'sources!': [ 'aecm/aecm_core_neon.c', 'ns/nsx_core_neon.c', @@ -197,22 +223,6 @@ }], ], }], - 'conditions': [ - ['OS=="android" or OS=="ios"', { - 'targets': [{ - 'target_name': 'audio_processing_offsets', - 'type': 'none', - 'sources': [ - 'aecm/aecm_core_neon_offsets.c', - 'ns/nsx_core_neon_offsets.c', - ], - 'variables': { - 'asm_header_dir': 'asm_offsets', - }, - 'includes': ['../../build/generate_asm_header.gypi',], - }], - }], - ], }], ], } diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc index 4d36ff7e7b4..de387edb2f5 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -12,17 +12,20 @@ #include <assert.h> +#include "webrtc/common_audio/include/audio_util.h" +#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/echo_cancellation_impl_wrapper.h" +#include "webrtc/modules/audio_processing/common.h" +#include "webrtc/modules/audio_processing/echo_cancellation_impl.h" #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h" #include "webrtc/modules/audio_processing/gain_control_impl.h" #include "webrtc/modules/audio_processing/high_pass_filter_impl.h" #include "webrtc/modules/audio_processing/level_estimator_impl.h" #include "webrtc/modules/audio_processing/noise_suppression_impl.h" #include "webrtc/modules/audio_processing/processing_component.h" -#include "webrtc/modules/audio_processing/splitting_filter.h" #include "webrtc/modules/audio_processing/voice_detection_impl.h" #include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/file_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" @@ -36,9 +39,30 @@ #endif #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP +#define RETURN_ON_ERR(expr) \ + do { \ + int err = expr; \ + if (err != kNoError) { \ + return err; \ + } \ + } while (0) + namespace webrtc { + +// Throughout webrtc, it's assumed that success is represented by zero. +COMPILE_ASSERT(AudioProcessing::kNoError == 0, no_error_must_be_zero); + AudioProcessing* AudioProcessing::Create(int id) { - AudioProcessingImpl* apm = new AudioProcessingImpl(id); + return Create(); +} + +AudioProcessing* AudioProcessing::Create() { + Config config; + return Create(config); +} + +AudioProcessing* AudioProcessing::Create(const Config& config) { + AudioProcessingImpl* apm = new AudioProcessingImpl(config); if (apm->Initialize() != kNoError) { delete apm; apm = NULL; @@ -47,12 +71,8 @@ AudioProcessing* AudioProcessing::Create(int id) { return apm; } -int32_t AudioProcessing::TimeUntilNextProcess() { return -1; } -int32_t AudioProcessing::Process() { return -1; } - -AudioProcessingImpl::AudioProcessingImpl(int id) - : id_(id), - echo_cancellation_(NULL), +AudioProcessingImpl::AudioProcessingImpl(const Config& config) + : echo_cancellation_(NULL), echo_control_mobile_(NULL), gain_control_(NULL), high_pass_filter_(NULL), @@ -60,41 +80,43 @@ AudioProcessingImpl::AudioProcessingImpl(int id) noise_suppression_(NULL), voice_detection_(NULL), crit_(CriticalSectionWrapper::CreateCriticalSection()), - render_audio_(NULL), - capture_audio_(NULL), #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP debug_file_(FileWrapper::Create()), event_msg_(new audioproc::Event()), #endif - sample_rate_hz_(kSampleRate16kHz), - split_sample_rate_hz_(kSampleRate16kHz), - samples_per_channel_(sample_rate_hz_ / 100), + fwd_in_format_(kSampleRate16kHz, 1), + fwd_proc_format_(kSampleRate16kHz, 1), + fwd_out_format_(kSampleRate16kHz), + rev_in_format_(kSampleRate16kHz, 1), + rev_proc_format_(kSampleRate16kHz, 1), + split_rate_(kSampleRate16kHz), stream_delay_ms_(0), delay_offset_ms_(0), was_stream_delay_set_(false), - num_reverse_channels_(1), - num_input_channels_(1), - num_output_channels_(1) { - echo_cancellation_ = EchoCancellationImplWrapper::Create(this); + output_will_be_muted_(false), + key_pressed_(false) { + echo_cancellation_ = new EchoCancellationImpl(this, crit_); component_list_.push_back(echo_cancellation_); - echo_control_mobile_ = new EchoControlMobileImpl(this); + echo_control_mobile_ = new EchoControlMobileImpl(this, crit_); component_list_.push_back(echo_control_mobile_); - gain_control_ = new GainControlImpl(this); + gain_control_ = new GainControlImpl(this, crit_); component_list_.push_back(gain_control_); - high_pass_filter_ = new HighPassFilterImpl(this); + high_pass_filter_ = new HighPassFilterImpl(this, crit_); component_list_.push_back(high_pass_filter_); - level_estimator_ = new LevelEstimatorImpl(this); + level_estimator_ = new LevelEstimatorImpl(this, crit_); component_list_.push_back(level_estimator_); - noise_suppression_ = new NoiseSuppressionImpl(this); + noise_suppression_ = new NoiseSuppressionImpl(this, crit_); component_list_.push_back(noise_suppression_); - voice_detection_ = new VoiceDetectionImpl(this); + voice_detection_ = new VoiceDetectionImpl(this, crit_); component_list_.push_back(voice_detection_); + + SetExtraOptions(config); } AudioProcessingImpl::~AudioProcessingImpl() { @@ -112,52 +134,52 @@ AudioProcessingImpl::~AudioProcessingImpl() { debug_file_->CloseFile(); } #endif - - if (render_audio_) { - delete render_audio_; - render_audio_ = NULL; - } - - if (capture_audio_) { - delete capture_audio_; - capture_audio_ = NULL; - } } - delete crit_; crit_ = NULL; } -CriticalSectionWrapper* AudioProcessingImpl::crit() const { - return crit_; -} - -int AudioProcessingImpl::split_sample_rate_hz() const { - return split_sample_rate_hz_; -} - int AudioProcessingImpl::Initialize() { CriticalSectionScoped crit_scoped(crit_); return InitializeLocked(); } -int AudioProcessingImpl::InitializeLocked() { - if (render_audio_ != NULL) { - delete render_audio_; - render_audio_ = NULL; - } - - if (capture_audio_ != NULL) { - delete capture_audio_; - capture_audio_ = NULL; - } - - render_audio_ = new AudioBuffer(num_reverse_channels_, - samples_per_channel_); - capture_audio_ = new AudioBuffer(num_input_channels_, - samples_per_channel_); +int AudioProcessingImpl::set_sample_rate_hz(int rate) { + CriticalSectionScoped crit_scoped(crit_); + return InitializeLocked(rate, + rate, + rev_in_format_.rate(), + fwd_in_format_.num_channels(), + fwd_proc_format_.num_channels(), + rev_in_format_.num_channels()); +} + +int AudioProcessingImpl::Initialize(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + ChannelLayout input_layout, + ChannelLayout output_layout, + ChannelLayout reverse_layout) { + CriticalSectionScoped crit_scoped(crit_); + return InitializeLocked(input_sample_rate_hz, + output_sample_rate_hz, + reverse_sample_rate_hz, + ChannelsFromLayout(input_layout), + ChannelsFromLayout(output_layout), + ChannelsFromLayout(reverse_layout)); +} - was_stream_delay_set_ = false; +int AudioProcessingImpl::InitializeLocked() { + render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(), + rev_in_format_.num_channels(), + rev_proc_format_.samples_per_channel(), + rev_proc_format_.num_channels(), + rev_proc_format_.samples_per_channel())); + capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(), + fwd_in_format_.num_channels(), + fwd_proc_format_.samples_per_channel(), + fwd_proc_format_.num_channels(), + fwd_out_format_.samples_per_channel())); // Initialize all components. std::list<ProcessingComponent*>::iterator it; @@ -180,115 +202,228 @@ int AudioProcessingImpl::InitializeLocked() { return kNoError; } -void AudioProcessingImpl::SetExtraOptions(const Config& config) { - std::list<ProcessingComponent*>::iterator it; - for (it = component_list_.begin(); it != component_list_.end(); ++it) - (*it)->SetExtraOptions(config); -} +int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + int num_input_channels, + int num_output_channels, + int num_reverse_channels) { + if (input_sample_rate_hz <= 0 || + output_sample_rate_hz <= 0 || + reverse_sample_rate_hz <= 0) { + return kBadSampleRateError; + } + if (num_output_channels > num_input_channels) { + return kBadNumberChannelsError; + } + // Only mono and stereo supported currently. + if (num_input_channels > 2 || num_input_channels < 1 || + num_output_channels > 2 || num_output_channels < 1 || + num_reverse_channels > 2 || num_reverse_channels < 1) { + return kBadNumberChannelsError; + } -int AudioProcessingImpl::EnableExperimentalNs(bool enable) { - return kNoError; -} + fwd_in_format_.set(input_sample_rate_hz, num_input_channels); + fwd_out_format_.set(output_sample_rate_hz); + rev_in_format_.set(reverse_sample_rate_hz, num_reverse_channels); -int AudioProcessingImpl::set_sample_rate_hz(int rate) { - CriticalSectionScoped crit_scoped(crit_); - if (rate == sample_rate_hz_) { - return kNoError; + // We process at the closest native rate >= min(input rate, output rate)... + int min_proc_rate = std::min(fwd_in_format_.rate(), fwd_out_format_.rate()); + int fwd_proc_rate; + if (min_proc_rate > kSampleRate16kHz) { + fwd_proc_rate = kSampleRate32kHz; + } else if (min_proc_rate > kSampleRate8kHz) { + fwd_proc_rate = kSampleRate16kHz; + } else { + fwd_proc_rate = kSampleRate8kHz; } - if (rate != kSampleRate8kHz && - rate != kSampleRate16kHz && - rate != kSampleRate32kHz) { - return kBadParameterError; + // ...with one exception. + if (echo_control_mobile_->is_enabled() && min_proc_rate > kSampleRate16kHz) { + fwd_proc_rate = kSampleRate16kHz; } - if (echo_control_mobile_->is_enabled() && rate > kSampleRate16kHz) { - LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates"; - return kUnsupportedComponentError; + + fwd_proc_format_.set(fwd_proc_rate, num_output_channels); + + // We normally process the reverse stream at 16 kHz. Unless... + int rev_proc_rate = kSampleRate16kHz; + if (fwd_proc_format_.rate() == kSampleRate8kHz) { + // ...the forward stream is at 8 kHz. + rev_proc_rate = kSampleRate8kHz; + } else { + if (rev_in_format_.rate() == kSampleRate32kHz) { + // ...or the input is at 32 kHz, in which case we use the splitting + // filter rather than the resampler. + rev_proc_rate = kSampleRate32kHz; + } } - sample_rate_hz_ = rate; - samples_per_channel_ = rate / 100; + // TODO(ajm): Enable this. + // Always downmix the reverse stream to mono for analysis. + //rev_proc_format_.set(rev_proc_rate, 1); + rev_proc_format_.set(rev_proc_rate, rev_in_format_.num_channels()); - if (sample_rate_hz_ == kSampleRate32kHz) { - split_sample_rate_hz_ = kSampleRate16kHz; + if (fwd_proc_format_.rate() == kSampleRate32kHz) { + split_rate_ = kSampleRate16kHz; } else { - split_sample_rate_hz_ = sample_rate_hz_; + split_rate_ = fwd_proc_format_.rate(); } return InitializeLocked(); } -int AudioProcessingImpl::sample_rate_hz() const { +// Calls InitializeLocked() if any of the audio parameters have changed from +// their current values. +int AudioProcessingImpl::MaybeInitializeLocked(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + int num_input_channels, + int num_output_channels, + int num_reverse_channels) { + if (input_sample_rate_hz == fwd_in_format_.rate() && + output_sample_rate_hz == fwd_out_format_.rate() && + reverse_sample_rate_hz == rev_in_format_.rate() && + num_input_channels == fwd_in_format_.num_channels() && + num_output_channels == fwd_proc_format_.num_channels() && + num_reverse_channels == rev_in_format_.num_channels()) { + return kNoError; + } + + return InitializeLocked(input_sample_rate_hz, + output_sample_rate_hz, + reverse_sample_rate_hz, + num_input_channels, + num_output_channels, + num_reverse_channels); +} + +void AudioProcessingImpl::SetExtraOptions(const Config& config) { CriticalSectionScoped crit_scoped(crit_); - return sample_rate_hz_; + std::list<ProcessingComponent*>::iterator it; + for (it = component_list_.begin(); it != component_list_.end(); ++it) + (*it)->SetExtraOptions(config); +} + +int AudioProcessingImpl::EnableExperimentalNs(bool enable) { + return kNoError; } -int AudioProcessingImpl::set_num_reverse_channels(int channels) { +int AudioProcessingImpl::input_sample_rate_hz() const { CriticalSectionScoped crit_scoped(crit_); - if (channels == num_reverse_channels_) { - return kNoError; - } - // Only stereo supported currently. - if (channels > 2 || channels < 1) { - return kBadParameterError; - } + return fwd_in_format_.rate(); +} - num_reverse_channels_ = channels; +int AudioProcessingImpl::sample_rate_hz() const { + CriticalSectionScoped crit_scoped(crit_); + return fwd_in_format_.rate(); +} - return InitializeLocked(); +int AudioProcessingImpl::proc_sample_rate_hz() const { + return fwd_proc_format_.rate(); +} + +int AudioProcessingImpl::proc_split_sample_rate_hz() const { + return split_rate_; } int AudioProcessingImpl::num_reverse_channels() const { - return num_reverse_channels_; + return rev_proc_format_.num_channels(); +} + +int AudioProcessingImpl::num_input_channels() const { + return fwd_in_format_.num_channels(); +} + +int AudioProcessingImpl::num_output_channels() const { + return fwd_proc_format_.num_channels(); +} + +void AudioProcessingImpl::set_output_will_be_muted(bool muted) { + output_will_be_muted_ = muted; } -int AudioProcessingImpl::set_num_channels( - int input_channels, - int output_channels) { +bool AudioProcessingImpl::output_will_be_muted() const { + return output_will_be_muted_; +} + +int AudioProcessingImpl::ProcessStream(const float* const* src, + int samples_per_channel, + int input_sample_rate_hz, + ChannelLayout input_layout, + int output_sample_rate_hz, + ChannelLayout output_layout, + float* const* dest) { CriticalSectionScoped crit_scoped(crit_); - if (input_channels == num_input_channels_ && - output_channels == num_output_channels_) { - return kNoError; - } - if (output_channels > input_channels) { - return kBadParameterError; + if (!src || !dest) { + return kNullPointerError; } - // Only stereo supported currently. - if (input_channels > 2 || input_channels < 1 || - output_channels > 2 || output_channels < 1) { - return kBadParameterError; + + RETURN_ON_ERR(MaybeInitializeLocked(input_sample_rate_hz, + output_sample_rate_hz, + rev_in_format_.rate(), + ChannelsFromLayout(input_layout), + ChannelsFromLayout(output_layout), + rev_in_format_.num_channels())); + if (samples_per_channel != fwd_in_format_.samples_per_channel()) { + return kBadDataLengthError; } - num_input_channels_ = input_channels; - num_output_channels_ = output_channels; +#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP + if (debug_file_->Open()) { + event_msg_->set_type(audioproc::Event::STREAM); + audioproc::Stream* msg = event_msg_->mutable_stream(); + const size_t channel_size = sizeof(float) * samples_per_channel; + for (int i = 0; i < fwd_in_format_.num_channels(); ++i) + msg->add_input_channel(src[i], channel_size); + } +#endif - return InitializeLocked(); -} + capture_audio_->CopyFrom(src, samples_per_channel, input_layout); + RETURN_ON_ERR(ProcessStreamLocked()); + if (output_copy_needed(is_data_processed())) { + capture_audio_->CopyTo(fwd_out_format_.samples_per_channel(), + output_layout, + dest); + } -int AudioProcessingImpl::num_input_channels() const { - return num_input_channels_; -} +#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP + if (debug_file_->Open()) { + audioproc::Stream* msg = event_msg_->mutable_stream(); + const size_t channel_size = sizeof(float) * samples_per_channel; + for (int i = 0; i < fwd_proc_format_.num_channels(); ++i) + msg->add_output_channel(dest[i], channel_size); + RETURN_ON_ERR(WriteMessageToDebugFile()); + } +#endif -int AudioProcessingImpl::num_output_channels() const { - return num_output_channels_; + return kNoError; } int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { CriticalSectionScoped crit_scoped(crit_); - int err = kNoError; - - if (frame == NULL) { + if (!frame) { return kNullPointerError; } - - if (frame->sample_rate_hz_ != sample_rate_hz_) { + // Must be a native rate. + if (frame->sample_rate_hz_ != kSampleRate8kHz && + frame->sample_rate_hz_ != kSampleRate16kHz && + frame->sample_rate_hz_ != kSampleRate32kHz) { return kBadSampleRateError; } - - if (frame->num_channels_ != num_input_channels_) { - return kBadNumberChannelsError; + if (echo_control_mobile_->is_enabled() && + frame->sample_rate_hz_ > kSampleRate16kHz) { + LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates"; + return kUnsupportedComponentError; } - if (frame->samples_per_channel_ != samples_per_channel_) { + // TODO(ajm): The input and output rates and channels are currently + // constrained to be identical in the int16 interface. + RETURN_ON_ERR(MaybeInitializeLocked(frame->sample_rate_hz_, + frame->sample_rate_hz_, + rev_in_format_.rate(), + frame->num_channels_, + frame->num_channels_, + rev_in_format_.num_channels())); + if (frame->samples_per_channel_ != fwd_in_format_.samples_per_channel()) { return kBadDataLengthError; } @@ -300,126 +435,142 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { frame->samples_per_channel_ * frame->num_channels_; msg->set_input_data(frame->data_, data_size); - msg->set_delay(stream_delay_ms_); - msg->set_drift(echo_cancellation_->stream_drift_samples()); - msg->set_level(gain_control_->stream_analog_level()); } #endif capture_audio_->DeinterleaveFrom(frame); + RETURN_ON_ERR(ProcessStreamLocked()); + capture_audio_->InterleaveTo(frame, output_copy_needed(is_data_processed())); - // TODO(ajm): experiment with mixing and AEC placement. - if (num_output_channels_ < num_input_channels_) { - capture_audio_->Mix(num_output_channels_); - frame->num_channels_ = num_output_channels_; - } - - bool data_processed = is_data_processed(); - if (analysis_needed(data_processed)) { - for (int i = 0; i < num_output_channels_; i++) { - // Split into a low and high band. - SplittingFilterAnalysis(capture_audio_->data(i), - capture_audio_->low_pass_split_data(i), - capture_audio_->high_pass_split_data(i), - capture_audio_->analysis_filter_state1(i), - capture_audio_->analysis_filter_state2(i)); - } - } - - err = high_pass_filter_->ProcessCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; - } - - err = gain_control_->AnalyzeCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; +#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP + if (debug_file_->Open()) { + audioproc::Stream* msg = event_msg_->mutable_stream(); + const size_t data_size = sizeof(int16_t) * + frame->samples_per_channel_ * + frame->num_channels_; + msg->set_output_data(frame->data_, data_size); + RETURN_ON_ERR(WriteMessageToDebugFile()); } +#endif - err = echo_cancellation_->ProcessCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; - } + return kNoError; +} - if (echo_control_mobile_->is_enabled() && - noise_suppression_->is_enabled()) { - capture_audio_->CopyLowPassToReference(); - } - err = noise_suppression_->ProcessCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; +int AudioProcessingImpl::ProcessStreamLocked() { +#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP + if (debug_file_->Open()) { + audioproc::Stream* msg = event_msg_->mutable_stream(); + msg->set_delay(stream_delay_ms_); + msg->set_drift(echo_cancellation_->stream_drift_samples()); + msg->set_level(gain_control_->stream_analog_level()); + msg->set_keypress(key_pressed_); } +#endif - err = echo_control_mobile_->ProcessCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; + AudioBuffer* ca = capture_audio_.get(); // For brevity. + bool data_processed = is_data_processed(); + if (analysis_needed(data_processed)) { + for (int i = 0; i < fwd_proc_format_.num_channels(); i++) { + // Split into a low and high band. + WebRtcSpl_AnalysisQMF(ca->data(i), + ca->samples_per_channel(), + ca->low_pass_split_data(i), + ca->high_pass_split_data(i), + ca->filter_states(i)->analysis_filter_state1, + ca->filter_states(i)->analysis_filter_state2); + } } - err = voice_detection_->ProcessCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; - } + RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca)); + RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca)); + RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca)); - err = gain_control_->ProcessCaptureAudio(capture_audio_); - if (err != kNoError) { - return err; + if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) { + ca->CopyLowPassToReference(); } + RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca)); + RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca)); + RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca)); + RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca)); if (synthesis_needed(data_processed)) { - for (int i = 0; i < num_output_channels_; i++) { + for (int i = 0; i < fwd_proc_format_.num_channels(); i++) { // Recombine low and high bands. - SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i), - capture_audio_->high_pass_split_data(i), - capture_audio_->data(i), - capture_audio_->synthesis_filter_state1(i), - capture_audio_->synthesis_filter_state2(i)); + WebRtcSpl_SynthesisQMF(ca->low_pass_split_data(i), + ca->high_pass_split_data(i), + ca->samples_per_split_channel(), + ca->data(i), + ca->filter_states(i)->synthesis_filter_state1, + ca->filter_states(i)->synthesis_filter_state2); } } // The level estimator operates on the recombined data. - err = level_estimator_->ProcessStream(capture_audio_); - if (err != kNoError) { - return err; + RETURN_ON_ERR(level_estimator_->ProcessStream(ca)); + + was_stream_delay_set_ = false; + return kNoError; +} + +int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data, + int samples_per_channel, + int sample_rate_hz, + ChannelLayout layout) { + CriticalSectionScoped crit_scoped(crit_); + if (data == NULL) { + return kNullPointerError; } - capture_audio_->InterleaveTo(frame, interleave_needed(data_processed)); + const int num_channels = ChannelsFromLayout(layout); + RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(), + fwd_out_format_.rate(), + sample_rate_hz, + fwd_in_format_.num_channels(), + fwd_proc_format_.num_channels(), + num_channels)); + if (samples_per_channel != rev_in_format_.samples_per_channel()) { + return kBadDataLengthError; + } #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP if (debug_file_->Open()) { - audioproc::Stream* msg = event_msg_->mutable_stream(); - const size_t data_size = sizeof(int16_t) * - frame->samples_per_channel_ * - frame->num_channels_; - msg->set_output_data(frame->data_, data_size); - err = WriteMessageToDebugFile(); - if (err != kNoError) { - return err; - } + event_msg_->set_type(audioproc::Event::REVERSE_STREAM); + audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream(); + const size_t channel_size = sizeof(float) * samples_per_channel; + for (int i = 0; i < num_channels; ++i) + msg->add_channel(data[i], channel_size); + RETURN_ON_ERR(WriteMessageToDebugFile()); } #endif - was_stream_delay_set_ = false; - return kNoError; + render_audio_->CopyFrom(data, samples_per_channel, layout); + return AnalyzeReverseStreamLocked(); } int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) { CriticalSectionScoped crit_scoped(crit_); - int err = kNoError; - if (frame == NULL) { return kNullPointerError; } - - if (frame->sample_rate_hz_ != sample_rate_hz_) { + // Must be a native rate. + if (frame->sample_rate_hz_ != kSampleRate8kHz && + frame->sample_rate_hz_ != kSampleRate16kHz && + frame->sample_rate_hz_ != kSampleRate32kHz) { return kBadSampleRateError; } - - if (frame->num_channels_ != num_reverse_channels_) { - return kBadNumberChannelsError; + // This interface does not tolerate different forward and reverse rates. + if (frame->sample_rate_hz_ != fwd_in_format_.rate()) { + return kBadSampleRateError; } - if (frame->samples_per_channel_ != samples_per_channel_) { + RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(), + fwd_out_format_.rate(), + frame->sample_rate_hz_, + fwd_in_format_.num_channels(), + fwd_in_format_.num_channels(), + frame->num_channels_)); + if (frame->samples_per_channel_ != rev_in_format_.samples_per_channel()) { return kBadDataLengthError; } @@ -431,44 +582,33 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) { frame->samples_per_channel_ * frame->num_channels_; msg->set_data(frame->data_, data_size); - err = WriteMessageToDebugFile(); - if (err != kNoError) { - return err; - } + RETURN_ON_ERR(WriteMessageToDebugFile()); } #endif render_audio_->DeinterleaveFrom(frame); + return AnalyzeReverseStreamLocked(); +} - // TODO(ajm): turn the splitting filter into a component? - if (sample_rate_hz_ == kSampleRate32kHz) { - for (int i = 0; i < num_reverse_channels_; i++) { +int AudioProcessingImpl::AnalyzeReverseStreamLocked() { + AudioBuffer* ra = render_audio_.get(); // For brevity. + if (rev_proc_format_.rate() == kSampleRate32kHz) { + for (int i = 0; i < rev_proc_format_.num_channels(); i++) { // Split into low and high band. - SplittingFilterAnalysis(render_audio_->data(i), - render_audio_->low_pass_split_data(i), - render_audio_->high_pass_split_data(i), - render_audio_->analysis_filter_state1(i), - render_audio_->analysis_filter_state2(i)); + WebRtcSpl_AnalysisQMF(ra->data(i), + ra->samples_per_channel(), + ra->low_pass_split_data(i), + ra->high_pass_split_data(i), + ra->filter_states(i)->analysis_filter_state1, + ra->filter_states(i)->analysis_filter_state2); } } - // TODO(ajm): warnings possible from components? - err = echo_cancellation_->ProcessRenderAudio(render_audio_); - if (err != kNoError) { - return err; - } - - err = echo_control_mobile_->ProcessRenderAudio(render_audio_); - if (err != kNoError) { - return err; - } + RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra)); + RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra)); + RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra)); - err = gain_control_->ProcessRenderAudio(render_audio_); - if (err != kNoError) { - return err; - } - - return err; // TODO(ajm): this is for returning warnings; necessary? + return kNoError; } int AudioProcessingImpl::set_stream_delay_ms(int delay) { @@ -499,6 +639,14 @@ bool AudioProcessingImpl::was_stream_delay_set() const { return was_stream_delay_set_; } +void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) { + key_pressed_ = key_pressed; +} + +bool AudioProcessingImpl::stream_key_pressed() const { + return key_pressed_; +} + void AudioProcessingImpl::set_delay_offset_ms(int offset) { CriticalSectionScoped crit_scoped(crit_); delay_offset_ms_ = offset; @@ -613,13 +761,6 @@ VoiceDetection* AudioProcessingImpl::voice_detection() const { return voice_detection_; } -int32_t AudioProcessingImpl::ChangeUniqueId(const int32_t id) { - CriticalSectionScoped crit_scoped(crit_); - id_ = id; - - return kNoError; -} - bool AudioProcessingImpl::is_data_processed() const { int enabled_count = 0; std::list<ProcessingComponent*>::const_iterator it; @@ -645,20 +786,21 @@ bool AudioProcessingImpl::is_data_processed() const { return true; } -bool AudioProcessingImpl::interleave_needed(bool is_data_processed) const { +bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const { // Check if we've upmixed or downmixed the audio. - return (num_output_channels_ != num_input_channels_ || is_data_processed); + return ((fwd_proc_format_.num_channels() != fwd_in_format_.num_channels()) || + is_data_processed); } bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const { - return (is_data_processed && sample_rate_hz_ == kSampleRate32kHz); + return (is_data_processed && fwd_proc_format_.rate() == kSampleRate32kHz); } bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const { if (!is_data_processed && !voice_detection_->is_enabled()) { // Only level_estimator_ is enabled. return false; - } else if (sample_rate_hz_ == kSampleRate32kHz) { + } else if (fwd_proc_format_.rate() == kSampleRate32kHz) { // Something besides level_estimator_ is enabled, and we have super-wb. return true; } @@ -690,17 +832,18 @@ int AudioProcessingImpl::WriteMessageToDebugFile() { event_msg_->Clear(); - return 0; + return kNoError; } int AudioProcessingImpl::WriteInitMessage() { event_msg_->set_type(audioproc::Event::INIT); audioproc::Init* msg = event_msg_->mutable_init(); - msg->set_sample_rate(sample_rate_hz_); - msg->set_device_sample_rate(echo_cancellation_->device_sample_rate_hz()); - msg->set_num_input_channels(num_input_channels_); - msg->set_num_output_channels(num_output_channels_); - msg->set_num_reverse_channels(num_reverse_channels_); + msg->set_sample_rate(fwd_in_format_.rate()); + msg->set_num_input_channels(fwd_in_format_.num_channels()); + msg->set_num_output_channels(fwd_proc_format_.num_channels()); + msg->set_num_reverse_channels(rev_in_format_.num_channels()); + msg->set_reverse_sample_rate(rev_in_format_.rate()); + msg->set_output_sample_rate(fwd_out_format_.rate()); int err = WriteMessageToDebugFile(); if (err != kNoError) { @@ -710,4 +853,5 @@ int AudioProcessingImpl::WriteInitMessage() { return kNoError; } #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h index e48a2c18a4f..d34f305a96b 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h @@ -19,9 +19,10 @@ #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { + class AudioBuffer; class CriticalSectionWrapper; -class EchoCancellationImplWrapper; +class EchoCancellationImpl; class EchoControlMobileImpl; class FileWrapper; class GainControlImpl; @@ -39,44 +40,92 @@ class Event; } // namespace audioproc #endif -class AudioProcessingImpl : public AudioProcessing { +class AudioRate { public: - enum { - kSampleRate8kHz = 8000, - kSampleRate16kHz = 16000, - kSampleRate32kHz = 32000 - }; + explicit AudioRate(int sample_rate_hz) + : rate_(sample_rate_hz), + samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {} + virtual ~AudioRate() {} + + void set(int rate) { + rate_ = rate; + samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000; + } - explicit AudioProcessingImpl(int id); - virtual ~AudioProcessingImpl(); + int rate() const { return rate_; } + int samples_per_channel() const { return samples_per_channel_; } - CriticalSectionWrapper* crit() const; + private: + int rate_; + int samples_per_channel_; +}; - int split_sample_rate_hz() const; - bool was_stream_delay_set() const; +class AudioFormat : public AudioRate { + public: + AudioFormat(int sample_rate_hz, int num_channels) + : AudioRate(sample_rate_hz), + num_channels_(num_channels) {} + virtual ~AudioFormat() {} + + void set(int rate, int num_channels) { + AudioRate::set(rate); + num_channels_ = num_channels; + } + + int num_channels() const { return num_channels_; } + + private: + int num_channels_; +}; + +class AudioProcessingImpl : public AudioProcessing { + public: + explicit AudioProcessingImpl(const Config& config); + virtual ~AudioProcessingImpl(); // AudioProcessing methods. virtual int Initialize() OVERRIDE; - virtual int InitializeLocked(); + virtual int Initialize(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + ChannelLayout input_layout, + ChannelLayout output_layout, + ChannelLayout reverse_layout) OVERRIDE; virtual void SetExtraOptions(const Config& config) OVERRIDE; virtual int EnableExperimentalNs(bool enable) OVERRIDE; virtual bool experimental_ns_enabled() const OVERRIDE { return false; } virtual int set_sample_rate_hz(int rate) OVERRIDE; + virtual int input_sample_rate_hz() const OVERRIDE; virtual int sample_rate_hz() const OVERRIDE; - virtual int set_num_channels(int input_channels, - int output_channels) OVERRIDE; + virtual int proc_sample_rate_hz() const OVERRIDE; + virtual int proc_split_sample_rate_hz() const OVERRIDE; virtual int num_input_channels() const OVERRIDE; virtual int num_output_channels() const OVERRIDE; - virtual int set_num_reverse_channels(int channels) OVERRIDE; virtual int num_reverse_channels() const OVERRIDE; + virtual void set_output_will_be_muted(bool muted) OVERRIDE; + virtual bool output_will_be_muted() const OVERRIDE; virtual int ProcessStream(AudioFrame* frame) OVERRIDE; + virtual int ProcessStream(const float* const* src, + int samples_per_channel, + int input_sample_rate_hz, + ChannelLayout input_layout, + int output_sample_rate_hz, + ChannelLayout output_layout, + float* const* dest) OVERRIDE; virtual int AnalyzeReverseStream(AudioFrame* frame) OVERRIDE; + virtual int AnalyzeReverseStream(const float* const* data, + int samples_per_channel, + int sample_rate_hz, + ChannelLayout layout) OVERRIDE; virtual int set_stream_delay_ms(int delay) OVERRIDE; virtual int stream_delay_ms() const OVERRIDE; + virtual bool was_stream_delay_set() const OVERRIDE; virtual void set_delay_offset_ms(int offset) OVERRIDE; virtual int delay_offset_ms() const OVERRIDE; + virtual void set_stream_key_pressed(bool key_pressed) OVERRIDE; + virtual bool stream_key_pressed() const OVERRIDE; virtual int StartDebugRecording( const char filename[kMaxFilenameSize]) OVERRIDE; virtual int StartDebugRecording(FILE* handle) OVERRIDE; @@ -89,18 +138,32 @@ class AudioProcessingImpl : public AudioProcessing { virtual NoiseSuppression* noise_suppression() const OVERRIDE; virtual VoiceDetection* voice_detection() const OVERRIDE; - // Module methods. - virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE; + protected: + // Overridden in a mock. + virtual int InitializeLocked(); private: + int InitializeLocked(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + int num_input_channels, + int num_output_channels, + int num_reverse_channels); + int MaybeInitializeLocked(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + int num_input_channels, + int num_output_channels, + int num_reverse_channels); + int ProcessStreamLocked(); + int AnalyzeReverseStreamLocked(); + bool is_data_processed() const; - bool interleave_needed(bool is_data_processed) const; + bool output_copy_needed(bool is_data_processed) const; bool synthesis_needed(bool is_data_processed) const; bool analysis_needed(bool is_data_processed) const; - int id_; - - EchoCancellationImplWrapper* echo_cancellation_; + EchoCancellationImpl* echo_cancellation_; EchoControlMobileImpl* echo_control_mobile_; GainControlImpl* gain_control_; HighPassFilterImpl* high_pass_filter_; @@ -110,29 +173,34 @@ class AudioProcessingImpl : public AudioProcessing { std::list<ProcessingComponent*> component_list_; CriticalSectionWrapper* crit_; - AudioBuffer* render_audio_; - AudioBuffer* capture_audio_; + scoped_ptr<AudioBuffer> render_audio_; + scoped_ptr<AudioBuffer> capture_audio_; #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP // TODO(andrew): make this more graceful. Ideally we would split this stuff // out into a separate class with an "enabled" and "disabled" implementation. int WriteMessageToDebugFile(); int WriteInitMessage(); scoped_ptr<FileWrapper> debug_file_; - scoped_ptr<audioproc::Event> event_msg_; // Protobuf message. - std::string event_str_; // Memory for protobuf serialization. + scoped_ptr<audioproc::Event> event_msg_; // Protobuf message. + std::string event_str_; // Memory for protobuf serialization. #endif - int sample_rate_hz_; - int split_sample_rate_hz_; - int samples_per_channel_; + AudioFormat fwd_in_format_; + AudioFormat fwd_proc_format_; + AudioRate fwd_out_format_; + AudioFormat rev_in_format_; + AudioFormat rev_proc_format_; + int split_rate_; + int stream_delay_ms_; int delay_offset_ms_; bool was_stream_delay_set_; - int num_reverse_channels_; - int num_input_channels_; - int num_output_channels_; + bool output_will_be_muted_; + + bool key_pressed_; }; + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc new file mode 100644 index 00000000000..09576175756 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc @@ -0,0 +1,74 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_processing/audio_processing_impl.h" + +#include "testing/gmock/include/gmock/gmock.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/config.h" +#include "webrtc/modules/audio_processing/test/test_utils.h" +#include "webrtc/modules/interface/module_common_types.h" + +using ::testing::Invoke; +using ::testing::Return; + +namespace webrtc { + +class MockInitialize : public AudioProcessingImpl { + public: + explicit MockInitialize(const Config& config) : AudioProcessingImpl(config) { + } + + MOCK_METHOD0(InitializeLocked, int()); + int RealInitializeLocked() { return AudioProcessingImpl::InitializeLocked(); } +}; + +TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { + Config config; + MockInitialize mock(config); + ON_CALL(mock, InitializeLocked()) + .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); + + EXPECT_CALL(mock, InitializeLocked()).Times(1); + mock.Initialize(); + + AudioFrame frame; + // Call with the default parameters; there should be no init. + frame.num_channels_ = 1; + SetFrameSampleRate(&frame, 16000); + EXPECT_CALL(mock, InitializeLocked()) + .Times(0); + EXPECT_NOERR(mock.ProcessStream(&frame)); + EXPECT_NOERR(mock.AnalyzeReverseStream(&frame)); + + // New sample rate. (Only impacts ProcessStream). + SetFrameSampleRate(&frame, 32000); + EXPECT_CALL(mock, InitializeLocked()) + .Times(1); + EXPECT_NOERR(mock.ProcessStream(&frame)); + + // New number of channels. + frame.num_channels_ = 2; + EXPECT_CALL(mock, InitializeLocked()) + .Times(2); + EXPECT_NOERR(mock.ProcessStream(&frame)); + // ProcessStream sets num_channels_ == num_output_channels. + frame.num_channels_ = 2; + EXPECT_NOERR(mock.AnalyzeReverseStream(&frame)); + + // A new sample rate passed to AnalyzeReverseStream should be an error and + // not cause an init. + SetFrameSampleRate(&frame, 16000); + EXPECT_CALL(mock, InitializeLocked()) + .Times(0); + EXPECT_EQ(mock.kBadSampleRateError, mock.AnalyzeReverseStream(&frame)); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_tests.gypi b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_tests.gypi index 05d7514bded..82aa7fd14fd 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_tests.gypi +++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_tests.gypi @@ -7,25 +7,23 @@ # be found in the AUTHORS file in the root of the source tree. { - 'targets': [ - { - 'target_name': 'audioproc_unittest_proto', - 'type': 'static_library', - 'sources': [ 'test/unittest.proto', ], - 'variables': { - 'proto_in_dir': 'test', - # Workaround to protect against gyp's pathname relativization when this - # file is included by modules.gyp. - 'proto_out_protected': 'webrtc/audio_processing', - 'proto_out_dir': '<(proto_out_protected)', - }, - 'includes': [ '../../build/protoc.gypi', ], - }, - ], 'conditions': [ ['enable_protobuf==1', { 'targets': [ { + 'target_name': 'audioproc_unittest_proto', + 'type': 'static_library', + 'sources': [ 'test/unittest.proto', ], + 'variables': { + 'proto_in_dir': 'test', + # Workaround to protect against gyp's pathname relativization when + # this file is included by modules.gyp. + 'proto_out_protected': 'webrtc/audio_processing', + 'proto_out_dir': '<(proto_out_protected)', + }, + 'includes': [ '../../build/protoc.gypi', ], + }, + { 'target_name': 'audioproc', 'type': 'executable', 'dependencies': [ diff --git a/chromium/third_party/webrtc/modules/audio_processing/common.h b/chromium/third_party/webrtc/modules/audio_processing/common.h new file mode 100644 index 00000000000..42454df299f --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/common.h @@ -0,0 +1,76 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_ + +#include <assert.h> +#include <string.h> + +#include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) { + switch (layout) { + case AudioProcessing::kMono: + case AudioProcessing::kMonoAndKeyboard: + return 1; + case AudioProcessing::kStereo: + case AudioProcessing::kStereoAndKeyboard: + return 2; + } + assert(false); + return -1; +} + +// Helper to encapsulate a contiguous data buffer with access to a pointer +// array of the deinterleaved channels. +template <typename T> +class ChannelBuffer { + public: + ChannelBuffer(int samples_per_channel, int num_channels) + : data_(new T[samples_per_channel * num_channels]), + channels_(new T*[num_channels]), + samples_per_channel_(samples_per_channel), + num_channels_(num_channels) { + memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels); + for (int i = 0; i < num_channels; ++i) + channels_[i] = &data_[i * samples_per_channel]; + } + ~ChannelBuffer() {} + + void CopyFrom(const void* channel_ptr, int i) { + assert(i < num_channels_); + memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T)); + } + + T* data() { return data_.get(); } + T* channel(int i) { + assert(i < num_channels_); + return channels_[i]; + } + T** channels() { return channels_.get(); } + + int samples_per_channel() { return samples_per_channel_; } + int num_channels() { return num_channels_; } + int length() { return samples_per_channel_ * num_channels_; } + + private: + scoped_ptr<T[]> data_; + scoped_ptr<T*[]> channels_; + int samples_per_channel_; + int num_channels_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/debug.proto b/chromium/third_party/webrtc/modules/audio_processing/debug.proto index 4b3a1638941..dce2f792093 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/debug.proto +++ b/chromium/third_party/webrtc/modules/audio_processing/debug.proto @@ -4,22 +4,39 @@ package webrtc.audioproc; message Init { optional int32 sample_rate = 1; - optional int32 device_sample_rate = 2; + optional int32 device_sample_rate = 2 [deprecated=true]; optional int32 num_input_channels = 3; optional int32 num_output_channels = 4; optional int32 num_reverse_channels = 5; + optional int32 reverse_sample_rate = 6; + optional int32 output_sample_rate = 7; } +// May contain interleaved or deinterleaved data, but don't store both formats. message ReverseStream { + // int16 interleaved data. optional bytes data = 1; + + // float deinterleaved data, where each repeated element points to a single + // channel buffer of data. + repeated bytes channel = 2; } +// May contain interleaved or deinterleaved data, but don't store both formats. message Stream { + // int16 interleaved data. optional bytes input_data = 1; optional bytes output_data = 2; + optional int32 delay = 3; optional sint32 drift = 4; optional int32 level = 5; + optional bool keypress = 6; + + // float deinterleaved data, where each repeated element points to a single + // channel buffer of data. + repeated bytes input_channel = 7; + repeated bytes output_channel = 8; } message Event { diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.cc index cd12363ec5e..e770f9fe377 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.cc @@ -18,7 +18,6 @@ extern "C" { } #include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h" #include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" namespace webrtc { @@ -56,23 +55,20 @@ AudioProcessing::Error MapError(int err) { } } // namespace -EchoCancellationImplWrapper* EchoCancellationImplWrapper::Create( - const AudioProcessingImpl* audioproc) { - return new EchoCancellationImpl(audioproc); -} - -EchoCancellationImpl::EchoCancellationImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), +EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), apm_(apm), + crit_(crit), drift_compensation_enabled_(false), metrics_enabled_(false), suppression_level_(kModerateSuppression), - device_sample_rate_hz_(48000), stream_drift_samples_(0), was_stream_drift_set_(false), stream_has_echo_(false), delay_logging_enabled_(false), - delay_correction_enabled_(false) {} + delay_correction_enabled_(false), + reported_delay_enabled_(true) {} EchoCancellationImpl::~EchoCancellationImpl() {} @@ -133,10 +129,10 @@ int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) { Handle* my_handle = handle(handle_index); err = WebRtcAec_Process( my_handle, - audio->low_pass_split_data(i), - audio->high_pass_split_data(i), - audio->low_pass_split_data(i), - audio->high_pass_split_data(i), + audio->low_pass_split_data_f(i), + audio->high_pass_split_data_f(i), + audio->low_pass_split_data_f(i), + audio->high_pass_split_data_f(i), static_cast<int16_t>(audio->samples_per_split_channel()), apm_->stream_delay_ms(), stream_drift_samples_); @@ -168,7 +164,7 @@ int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) { } int EchoCancellationImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); // Ensure AEC and AECM are not both enabled. if (enable && apm_->echo_control_mobile()->is_enabled()) { return apm_->kBadParameterError; @@ -182,7 +178,7 @@ bool EchoCancellationImpl::is_enabled() const { } int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (MapSetting(level) == -1) { return apm_->kBadParameterError; } @@ -197,7 +193,7 @@ EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level() } int EchoCancellationImpl::enable_drift_compensation(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); drift_compensation_enabled_ = enable; return Configure(); } @@ -206,20 +202,6 @@ bool EchoCancellationImpl::is_drift_compensation_enabled() const { return drift_compensation_enabled_; } -int EchoCancellationImpl::set_device_sample_rate_hz(int rate) { - CriticalSectionScoped crit_scoped(apm_->crit()); - if (rate < 8000 || rate > 96000) { - return apm_->kBadParameterError; - } - - device_sample_rate_hz_ = rate; - return Initialize(); -} - -int EchoCancellationImpl::device_sample_rate_hz() const { - return device_sample_rate_hz_; -} - void EchoCancellationImpl::set_stream_drift_samples(int drift) { was_stream_drift_set_ = true; stream_drift_samples_ = drift; @@ -230,7 +212,7 @@ int EchoCancellationImpl::stream_drift_samples() const { } int EchoCancellationImpl::enable_metrics(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); metrics_enabled_ = enable; return Configure(); } @@ -242,7 +224,7 @@ bool EchoCancellationImpl::are_metrics_enabled() const { // TODO(ajm): we currently just use the metrics from the first AEC. Think more // aboue the best way to extend this to multi-channel. int EchoCancellationImpl::GetMetrics(Metrics* metrics) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (metrics == NULL) { return apm_->kNullPointerError; } @@ -289,7 +271,7 @@ bool EchoCancellationImpl::stream_has_echo() const { } int EchoCancellationImpl::enable_delay_logging(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); delay_logging_enabled_ = enable; return Configure(); } @@ -300,7 +282,7 @@ bool EchoCancellationImpl::is_delay_logging_enabled() const { // TODO(bjornv): How should we handle the multi-channel case? int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (median == NULL) { return apm_->kNullPointerError; } @@ -322,7 +304,7 @@ int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) { } struct AecCore* EchoCancellationImpl::aec_core() const { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (!is_component_enabled()) { return NULL; } @@ -336,13 +318,12 @@ int EchoCancellationImpl::Initialize() { return err; } - was_stream_drift_set_ = false; - return apm_->kNoError; } void EchoCancellationImpl::SetExtraOptions(const Config& config) { delay_correction_enabled_ = config.Get<DelayCorrection>().enabled; + reported_delay_enabled_ = config.Get<ReportedDelay>().enabled; Configure(); } @@ -357,16 +338,19 @@ void* EchoCancellationImpl::CreateHandle() const { return handle; } -int EchoCancellationImpl::DestroyHandle(void* handle) const { +void EchoCancellationImpl::DestroyHandle(void* handle) const { assert(handle != NULL); - return WebRtcAec_Free(static_cast<Handle*>(handle)); + WebRtcAec_Free(static_cast<Handle*>(handle)); } int EchoCancellationImpl::InitializeHandle(void* handle) const { assert(handle != NULL); + // TODO(ajm): Drift compensation is disabled in practice. If restored, it + // should be managed internally and not depend on the hardware sample rate. + // For now, just hardcode a 48 kHz value. return WebRtcAec_Init(static_cast<Handle*>(handle), - apm_->sample_rate_hz(), - device_sample_rate_hz_); + apm_->proc_sample_rate_hz(), + 48000); } int EchoCancellationImpl::ConfigureHandle(void* handle) const { @@ -379,6 +363,8 @@ int EchoCancellationImpl::ConfigureHandle(void* handle) const { WebRtcAec_enable_delay_correction(WebRtcAec_aec_core( static_cast<Handle*>(handle)), delay_correction_enabled_ ? 1 : 0); + WebRtcAec_enable_reported_delay(WebRtcAec_aec_core( + static_cast<Handle*>(handle)), reported_delay_enabled_ ? 1 : 0); return WebRtcAec_set_config(static_cast<Handle*>(handle), config); } diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.h b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.h index 3ab0ce26689..b9c116a0650 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl.h @@ -11,25 +11,26 @@ #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_ #define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_ -#include "webrtc/modules/audio_processing/echo_cancellation_impl_wrapper.h" +#include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { -class AudioProcessingImpl; class AudioBuffer; +class CriticalSectionWrapper; -class EchoCancellationImpl : public EchoCancellationImplWrapper { +class EchoCancellationImpl : public EchoCancellation, + public ProcessingComponent { public: - explicit EchoCancellationImpl(const AudioProcessingImpl* apm); + EchoCancellationImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit); virtual ~EchoCancellationImpl(); - // EchoCancellationImplWrapper implementation. - virtual int ProcessRenderAudio(const AudioBuffer* audio) OVERRIDE; - virtual int ProcessCaptureAudio(AudioBuffer* audio) OVERRIDE; + int ProcessRenderAudio(const AudioBuffer* audio); + int ProcessCaptureAudio(AudioBuffer* audio); // EchoCancellation implementation. virtual bool is_enabled() const OVERRIDE; - virtual int device_sample_rate_hz() const OVERRIDE; virtual int stream_drift_samples() const OVERRIDE; // ProcessingComponent implementation. @@ -41,7 +42,6 @@ class EchoCancellationImpl : public EchoCancellationImplWrapper { virtual int Enable(bool enable) OVERRIDE; virtual int enable_drift_compensation(bool enable) OVERRIDE; virtual bool is_drift_compensation_enabled() const OVERRIDE; - virtual int set_device_sample_rate_hz(int rate) OVERRIDE; virtual void set_stream_drift_samples(int drift) OVERRIDE; virtual int set_suppression_level(SuppressionLevel level) OVERRIDE; virtual SuppressionLevel suppression_level() const OVERRIDE; @@ -58,20 +58,21 @@ class EchoCancellationImpl : public EchoCancellationImplWrapper { virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + const AudioProcessing* apm_; + CriticalSectionWrapper* crit_; bool drift_compensation_enabled_; bool metrics_enabled_; SuppressionLevel suppression_level_; - int device_sample_rate_hz_; int stream_drift_samples_; bool was_stream_drift_set_; bool stream_has_echo_; bool delay_logging_enabled_; bool delay_correction_enabled_; + bool reported_delay_enabled_; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc index f9bc3213ff1..49bcf9459b0 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc @@ -14,6 +14,7 @@ extern "C" { } #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace webrtc { @@ -47,4 +48,34 @@ TEST(EchoCancellationInternalTest, DelayCorrection) { EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core)); } +TEST(EchoCancellationInternalTest, ReportedDelay) { + scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(0)); + EXPECT_TRUE(ap->echo_cancellation()->aec_core() == NULL); + + EXPECT_EQ(ap->kNoError, ap->echo_cancellation()->Enable(true)); + EXPECT_TRUE(ap->echo_cancellation()->is_enabled()); + + AecCore* aec_core = ap->echo_cancellation()->aec_core(); + ASSERT_TRUE(aec_core != NULL); + // Enabled by default. + EXPECT_EQ(1, WebRtcAec_reported_delay_enabled(aec_core)); + + Config config; + config.Set<ReportedDelay>(new ReportedDelay(false)); + ap->SetExtraOptions(config); + EXPECT_EQ(0, WebRtcAec_reported_delay_enabled(aec_core)); + + // Retains setting after initialization. + EXPECT_EQ(ap->kNoError, ap->Initialize()); + EXPECT_EQ(0, WebRtcAec_reported_delay_enabled(aec_core)); + + config.Set<ReportedDelay>(new ReportedDelay(true)); + ap->SetExtraOptions(config); + EXPECT_EQ(1, WebRtcAec_reported_delay_enabled(aec_core)); + + // Retains setting after initialization. + EXPECT_EQ(ap->kNoError, ap->Initialize()); + EXPECT_EQ(1, WebRtcAec_reported_delay_enabled(aec_core)); +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_wrapper.h b/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_wrapper.h deleted file mode 100644 index f1c03f32d6f..00000000000 --- a/chromium/third_party/webrtc/modules/audio_processing/echo_cancellation_impl_wrapper.h +++ /dev/null @@ -1,35 +0,0 @@ -/* - * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_WRAPPER_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_WRAPPER_H_ - -#include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/audio_processing/processing_component.h" - -namespace webrtc { - -class AudioProcessingImpl; -class AudioBuffer; - -class EchoCancellationImplWrapper : public virtual EchoCancellation, - public virtual ProcessingComponent { - public: - static EchoCancellationImplWrapper* Create( - const AudioProcessingImpl* audioproc); - virtual ~EchoCancellationImplWrapper() {} - - virtual int ProcessRenderAudio(const AudioBuffer* audio) = 0; - virtual int ProcessCaptureAudio(AudioBuffer* audio) = 0; -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_WRAPPER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc index f7853814966..a03adc5300e 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc @@ -15,7 +15,6 @@ #include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h" #include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" @@ -63,9 +62,11 @@ size_t EchoControlMobile::echo_path_size_bytes() { return WebRtcAecm_echo_path_size_bytes(); } -EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), +EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), apm_(apm), + crit_(crit), routing_mode_(kSpeakerphone), comfort_noise_enabled_(true), external_echo_path_(NULL) {} @@ -127,7 +128,7 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) { for (int i = 0; i < audio->num_channels(); i++) { // TODO(ajm): improve how this works, possibly inside AECM. // This is kind of hacked up. - int16_t* noisy = audio->low_pass_reference(i); + const int16_t* noisy = audio->low_pass_reference(i); int16_t* clean = audio->low_pass_split_data(i); if (noisy == NULL) { noisy = clean; @@ -155,7 +156,7 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) { } int EchoControlMobileImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); // Ensure AEC and AECM are not both enabled. if (enable && apm_->echo_cancellation()->is_enabled()) { return apm_->kBadParameterError; @@ -169,7 +170,7 @@ bool EchoControlMobileImpl::is_enabled() const { } int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (MapSetting(mode) == -1) { return apm_->kBadParameterError; } @@ -184,7 +185,7 @@ EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode() } int EchoControlMobileImpl::enable_comfort_noise(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); comfort_noise_enabled_ = enable; return Configure(); } @@ -195,7 +196,7 @@ bool EchoControlMobileImpl::is_comfort_noise_enabled() const { int EchoControlMobileImpl::SetEchoPath(const void* echo_path, size_t size_bytes) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (echo_path == NULL) { return apm_->kNullPointerError; } @@ -214,7 +215,7 @@ int EchoControlMobileImpl::SetEchoPath(const void* echo_path, int EchoControlMobileImpl::GetEchoPath(void* echo_path, size_t size_bytes) const { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (echo_path == NULL) { return apm_->kNullPointerError; } @@ -240,7 +241,7 @@ int EchoControlMobileImpl::Initialize() { return apm_->kNoError; } - if (apm_->sample_rate_hz() == apm_->kSampleRate32kHz) { + if (apm_->proc_sample_rate_hz() > apm_->kSampleRate16kHz) { LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates"; return apm_->kBadSampleRateError; } @@ -259,14 +260,14 @@ void* EchoControlMobileImpl::CreateHandle() const { return handle; } -int EchoControlMobileImpl::DestroyHandle(void* handle) const { - return WebRtcAecm_Free(static_cast<Handle*>(handle)); +void EchoControlMobileImpl::DestroyHandle(void* handle) const { + WebRtcAecm_Free(static_cast<Handle*>(handle)); } int EchoControlMobileImpl::InitializeHandle(void* handle) const { assert(handle != NULL); Handle* my_handle = static_cast<Handle*>(handle); - if (WebRtcAecm_Init(my_handle, apm_->sample_rate_hz()) != 0) { + if (WebRtcAecm_Init(my_handle, apm_->proc_sample_rate_hz()) != 0) { return GetHandleError(my_handle); } if (external_echo_path_ != NULL) { diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.h index 5eefab0a3c2..4f5b5931a1c 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.h @@ -15,13 +15,15 @@ #include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { -class AudioProcessingImpl; + class AudioBuffer; +class CriticalSectionWrapper; class EchoControlMobileImpl : public EchoControlMobile, public ProcessingComponent { public: - explicit EchoControlMobileImpl(const AudioProcessingImpl* apm); + EchoControlMobileImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit); virtual ~EchoControlMobileImpl(); int ProcessRenderAudio(const AudioBuffer* audio); @@ -47,11 +49,12 @@ class EchoControlMobileImpl : public EchoControlMobile, virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + const AudioProcessing* apm_; + CriticalSectionWrapper* crit_; RoutingMode routing_mode_; bool comfort_noise_enabled_; unsigned char* external_echo_path_; diff --git a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc index 35547031e30..a67b67ecb16 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc @@ -12,12 +12,10 @@ #include <assert.h> +#include "webrtc/modules/audio_processing/audio_buffer.h" #include "webrtc/modules/audio_processing/agc/include/gain_control.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" - namespace webrtc { typedef void Handle; @@ -37,9 +35,11 @@ int16_t MapSetting(GainControl::Mode mode) { } } // namespace -GainControlImpl::GainControlImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), +GainControlImpl::GainControlImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), apm_(apm), + crit_(crit), mode_(kAdaptiveAnalog), minimum_capture_level_(0), maximum_capture_level_(255), @@ -59,7 +59,7 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { assert(audio->samples_per_split_channel() <= 160); - int16_t* mixed_data = audio->low_pass_split_data(0); + const int16_t* mixed_data = audio->low_pass_split_data(0); if (audio->num_channels() > 1) { audio->CopyAndMixLowPass(1); mixed_data = audio->mixed_low_pass_data(0); @@ -91,6 +91,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { int err = apm_->kNoError; if (mode_ == kAdaptiveAnalog) { + capture_levels_.assign(num_handles(), analog_capture_level_); for (int i = 0; i < num_handles(); i++) { Handle* my_handle = static_cast<Handle*>(handle(i)); err = WebRtcAgc_AddMic( @@ -114,7 +115,6 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { audio->low_pass_split_data(i), audio->high_pass_split_data(i), static_cast<int16_t>(audio->samples_per_split_channel()), - //capture_levels_[i], analog_capture_level_, &capture_level_out); @@ -190,13 +190,6 @@ int GainControlImpl::set_stream_analog_level(int level) { if (level < minimum_capture_level_ || level > maximum_capture_level_) { return apm_->kBadParameterError; } - - if (mode_ == kAdaptiveAnalog) { - if (level != analog_capture_level_) { - // The analog level has been changed; update our internal levels. - capture_levels_.assign(num_handles(), level); - } - } analog_capture_level_ = level; return apm_->kNoError; @@ -210,7 +203,7 @@ int GainControlImpl::stream_analog_level() { } int GainControlImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); return EnableComponent(enable); } @@ -219,7 +212,7 @@ bool GainControlImpl::is_enabled() const { } int GainControlImpl::set_mode(Mode mode) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (MapSetting(mode) == -1) { return apm_->kBadParameterError; } @@ -234,7 +227,7 @@ GainControl::Mode GainControlImpl::mode() const { int GainControlImpl::set_analog_level_limits(int minimum, int maximum) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (minimum < 0) { return apm_->kBadParameterError; } @@ -266,7 +259,7 @@ bool GainControlImpl::stream_is_saturated() const { } int GainControlImpl::set_target_level_dbfs(int level) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (level > 31 || level < 0) { return apm_->kBadParameterError; } @@ -280,7 +273,7 @@ int GainControlImpl::target_level_dbfs() const { } int GainControlImpl::set_compression_gain_db(int gain) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (gain < 0 || gain > 90) { return apm_->kBadParameterError; } @@ -294,7 +287,7 @@ int GainControlImpl::compression_gain_db() const { } int GainControlImpl::enable_limiter(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); limiter_enabled_ = enable; return Configure(); } @@ -309,11 +302,7 @@ int GainControlImpl::Initialize() { return err; } - analog_capture_level_ = - (maximum_capture_level_ - minimum_capture_level_) >> 1; capture_levels_.assign(num_handles(), analog_capture_level_); - was_analog_level_set_ = false; - return apm_->kNoError; } @@ -328,8 +317,8 @@ void* GainControlImpl::CreateHandle() const { return handle; } -int GainControlImpl::DestroyHandle(void* handle) const { - return WebRtcAgc_Free(static_cast<Handle*>(handle)); +void GainControlImpl::DestroyHandle(void* handle) const { + WebRtcAgc_Free(static_cast<Handle*>(handle)); } int GainControlImpl::InitializeHandle(void* handle) const { @@ -337,7 +326,7 @@ int GainControlImpl::InitializeHandle(void* handle) const { minimum_capture_level_, maximum_capture_level_, MapSetting(mode_), - apm_->sample_rate_hz()); + apm_->proc_sample_rate_hz()); } int GainControlImpl::ConfigureHandle(void* handle) const { diff --git a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h index 2de02f6e6b7..81159870009 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h @@ -17,13 +17,15 @@ #include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { -class AudioProcessingImpl; + class AudioBuffer; +class CriticalSectionWrapper; class GainControlImpl : public GainControl, public ProcessingComponent { public: - explicit GainControlImpl(const AudioProcessingImpl* apm); + GainControlImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit); virtual ~GainControlImpl(); int ProcessRenderAudio(AudioBuffer* audio); @@ -58,11 +60,12 @@ class GainControlImpl : public GainControl, virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + const AudioProcessing* apm_; + CriticalSectionWrapper* crit_; Mode mode_; int minimum_capture_level_; int maximum_capture_level_; diff --git a/chromium/third_party/webrtc/modules/audio_processing/gen_core_neon_offsets.gyp b/chromium/third_party/webrtc/modules/audio_processing/gen_core_neon_offsets.gyp new file mode 100644 index 00000000000..55c79689f7e --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/gen_core_neon_offsets.gyp @@ -0,0 +1,45 @@ +# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +{ + 'includes': ['lib_core_neon_offsets.gypi'], + 'targets' : [ + { + 'target_name': 'gen_nsx_core_neon_offsets_h', + 'type': 'none', + 'dependencies': [ + 'lib_core_neon_offsets', + '<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx_obj_int_extract#host', + ], + 'sources': ['<(shared_generated_dir)/nsx_core_neon_offsets.o',], + 'variables' : { + 'unpack_lib_name':'nsx_core_neon_offsets.o', + }, + 'includes': [ + '../../../third_party/libvpx/unpack_lib_posix.gypi', + '../../../third_party/libvpx/obj_int_extract.gypi', + ], + }, + { + 'target_name': 'gen_aecm_core_neon_offsets_h', + 'type': 'none', + 'dependencies': [ + 'lib_core_neon_offsets', + '<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx_obj_int_extract#host', + ], + 'variables': { + 'unpack_lib_name':'aecm_core_neon_offsets.o', + }, + 'sources': ['<(shared_generated_dir)/aecm_core_neon_offsets.o',], + 'includes': [ + '../../../third_party/libvpx/unpack_lib_posix.gypi', + '../../../third_party/libvpx/obj_int_extract.gypi', + ], + }, + ], +} diff --git a/chromium/third_party/webrtc/modules/audio_processing/gen_core_neon_offsets_chromium.gyp b/chromium/third_party/webrtc/modules/audio_processing/gen_core_neon_offsets_chromium.gyp new file mode 100644 index 00000000000..f4a9134fb21 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/gen_core_neon_offsets_chromium.gyp @@ -0,0 +1,45 @@ +# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +{ + 'includes': ['lib_core_neon_offsets.gypi'], + 'targets' : [ + { + 'target_name': 'gen_nsx_core_neon_offsets_h', + 'type': 'none', + 'dependencies': [ + 'lib_core_neon_offsets', + '<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx_obj_int_extract#host', + ], + 'sources': ['<(shared_generated_dir)/nsx_core_neon_offsets.o',], + 'variables' : { + 'unpack_lib_name':'nsx_core_neon_offsets.o', + }, + 'includes': [ + '../../../../third_party/libvpx/unpack_lib_posix.gypi', + '../../../../third_party/libvpx/obj_int_extract.gypi', + ], + }, + { + 'target_name': 'gen_aecm_core_neon_offsets_h', + 'type': 'none', + 'dependencies': [ + 'lib_core_neon_offsets', + '<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx_obj_int_extract#host', + ], + 'variables': { + 'unpack_lib_name':'aecm_core_neon_offsets.o', + }, + 'sources': ['<(shared_generated_dir)/aecm_core_neon_offsets.o',], + 'includes': [ + '../../../../third_party/libvpx/unpack_lib_posix.gypi', + '../../../../third_party/libvpx/obj_int_extract.gypi', + ], + }, + ], +} diff --git a/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.cc index da2170373b0..0a23ff23555 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.cc @@ -13,11 +13,10 @@ #include <assert.h> #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" +#include "webrtc/modules/audio_processing/audio_buffer.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/typedefs.h" -#include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" namespace webrtc { namespace { @@ -36,7 +35,7 @@ struct FilterState { int InitializeFilter(FilterState* hpf, int sample_rate_hz) { assert(hpf != NULL); - if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) { + if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) { hpf->ba = kFilterCoefficients8kHz; } else { hpf->ba = kFilterCoefficients; @@ -105,9 +104,11 @@ int Filter(FilterState* hpf, int16_t* data, int length) { typedef FilterState Handle; -HighPassFilterImpl::HighPassFilterImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), - apm_(apm) {} +HighPassFilterImpl::HighPassFilterImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), + apm_(apm), + crit_(crit) {} HighPassFilterImpl::~HighPassFilterImpl() {} @@ -135,7 +136,7 @@ int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { } int HighPassFilterImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); return EnableComponent(enable); } @@ -147,14 +148,13 @@ void* HighPassFilterImpl::CreateHandle() const { return new FilterState; } -int HighPassFilterImpl::DestroyHandle(void* handle) const { +void HighPassFilterImpl::DestroyHandle(void* handle) const { delete static_cast<Handle*>(handle); - return apm_->kNoError; } int HighPassFilterImpl::InitializeHandle(void* handle) const { return InitializeFilter(static_cast<Handle*>(handle), - apm_->sample_rate_hz()); + apm_->proc_sample_rate_hz()); } int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const { diff --git a/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.h b/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.h index 7e11ea9ceee..6f91f3bc049 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/high_pass_filter_impl.h @@ -15,13 +15,14 @@ #include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { -class AudioProcessingImpl; + class AudioBuffer; +class CriticalSectionWrapper; class HighPassFilterImpl : public HighPassFilter, public ProcessingComponent { public: - explicit HighPassFilterImpl(const AudioProcessingImpl* apm); + HighPassFilterImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit); virtual ~HighPassFilterImpl(); int ProcessCaptureAudio(AudioBuffer* audio); @@ -37,11 +38,12 @@ class HighPassFilterImpl : public HighPassFilter, virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + const AudioProcessing* apm_; + CriticalSectionWrapper* crit_; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h b/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h index b5c856de273..77c3f3add22 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h +++ b/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h @@ -15,7 +15,6 @@ #include <stdio.h> // FILE #include "webrtc/common.h" -#include "webrtc/modules/interface/module.h" #include "webrtc/typedefs.h" struct AecCore; @@ -50,11 +49,32 @@ class VoiceDetection; // except when really necessary. struct DelayCorrection { DelayCorrection() : enabled(false) {} - DelayCorrection(bool enabled) : enabled(enabled) {} + explicit DelayCorrection(bool enabled) : enabled(enabled) {} + bool enabled; +}; + +// Use to disable the reported system delays. By disabling the reported system +// delays the echo cancellation algorithm assumes the process and reverse +// streams to be aligned. This configuration only applies to EchoCancellation +// and not EchoControlMobile and is set with AudioProcessing::SetExtraOptions(). +// Note that by disabling reported system delays the EchoCancellation may +// regress in performance. +struct ReportedDelay { + ReportedDelay() : enabled(true) {} + explicit ReportedDelay(bool enabled) : enabled(enabled) {} + bool enabled; +}; +// Must be provided through AudioProcessing::Create(Confg&). It will have no +// impact if used with AudioProcessing::SetExtraOptions(). +struct ExperimentalAgc { + ExperimentalAgc() : enabled(true) {} + explicit ExperimentalAgc(bool enabled) : enabled(enabled) {} bool enabled; }; +static const int kAudioProcMaxNativeSampleRateHz = 32000; + // The Audio Processing Module (APM) provides a collection of voice processing // components designed for real-time communications software. // @@ -84,16 +104,12 @@ struct DelayCorrection { // 2. Parameter getters are never called concurrently with the corresponding // setter. // -// APM accepts only 16-bit linear PCM audio data in frames of 10 ms. Multiple -// channels should be interleaved. +// APM accepts only linear PCM audio data in chunks of 10 ms. The int16 +// interfaces use interleaved data, while the float interfaces use deinterleaved +// data. // // Usage example, omitting error checking: // AudioProcessing* apm = AudioProcessing::Create(0); -// apm->set_sample_rate_hz(32000); // Super-wideband processing. -// -// // Mono capture and stereo render. -// apm->set_num_channels(1, 1); -// apm->set_num_reverse_channels(2); // // apm->high_pass_filter()->Enable(true); // @@ -132,13 +148,27 @@ struct DelayCorrection { // // Close the application... // delete apm; // -class AudioProcessing : public Module { +class AudioProcessing { public: - // Creates a APM instance, with identifier |id|. Use one instance for every - // primary audio stream requiring processing. On the client-side, this would - // typically be one instance for the near-end stream, and additional instances - // for each far-end stream which requires processing. On the server-side, - // this would typically be one instance for every incoming stream. + enum ChannelLayout { + kMono, + // Left, right. + kStereo, + // Mono, keyboard mic. + kMonoAndKeyboard, + // Left, right, keyboard mic. + kStereoAndKeyboard + }; + + // Creates an APM instance. Use one instance for every primary audio stream + // requiring processing. On the client-side, this would typically be one + // instance for the near-end stream, and additional instances for each far-end + // stream which requires processing. On the server-side, this would typically + // be one instance for every incoming stream. + static AudioProcessing* Create(); + // Allows passing in an optional configuration at create-time. + static AudioProcessing* Create(const Config& config); + // TODO(ajm): Deprecated; remove all calls to it. static AudioProcessing* Create(int id); virtual ~AudioProcessing() {} @@ -147,11 +177,26 @@ class AudioProcessing : public Module { // it is not necessary to call before processing the first stream after // creation. // - // set_sample_rate_hz(), set_num_channels() and set_num_reverse_channels() - // will trigger a full initialization if the settings are changed from their - // existing values. Otherwise they are no-ops. + // It is also not necessary to call if the audio parameters (sample + // rate and number of channels) have changed. Passing updated parameters + // directly to |ProcessStream()| and |AnalyzeReverseStream()| is permissible. + // If the parameters are known at init-time though, they may be provided. virtual int Initialize() = 0; + // The int16 interfaces require: + // - only |NativeRate|s be used + // - that the input, output and reverse rates must match + // - that |output_layout| matches |input_layout| + // + // The float interfaces accept arbitrary rates and support differing input + // and output layouts, but the output may only remove channels, not add. + virtual int Initialize(int input_sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + ChannelLayout input_layout, + ChannelLayout output_layout, + ChannelLayout reverse_layout) = 0; + // Pass down additional options which don't have explicit setters. This // ensures the options are applied immediately. virtual void SetExtraOptions(const Config& config) = 0; @@ -159,23 +204,30 @@ class AudioProcessing : public Module { virtual int EnableExperimentalNs(bool enable) = 0; virtual bool experimental_ns_enabled() const = 0; - // Sets the sample |rate| in Hz for both the primary and reverse audio - // streams. 8000, 16000 or 32000 Hz are permitted. + // DEPRECATED. + // TODO(ajm): Remove after Chromium has upgraded to using Initialize(). virtual int set_sample_rate_hz(int rate) = 0; + // TODO(ajm): Remove after voice engine no longer requires it to resample + // the reverse stream to the forward rate. + virtual int input_sample_rate_hz() const = 0; + // TODO(ajm): Remove after Chromium no longer depends on it. virtual int sample_rate_hz() const = 0; - // Sets the number of channels for the primary audio stream. Input frames must - // contain a number of channels given by |input_channels|, while output frames - // will be returned with number of channels given by |output_channels|. - virtual int set_num_channels(int input_channels, int output_channels) = 0; + // TODO(ajm): Only intended for internal use. Make private and friend the + // necessary classes? + virtual int proc_sample_rate_hz() const = 0; + virtual int proc_split_sample_rate_hz() const = 0; virtual int num_input_channels() const = 0; virtual int num_output_channels() const = 0; - - // Sets the number of channels for the reverse audio stream. Input frames must - // contain a number of channels given by |channels|. - virtual int set_num_reverse_channels(int channels) = 0; virtual int num_reverse_channels() const = 0; + // Set to true when the output of AudioProcessing will be muted or in some + // other way not used. Ideally, the captured audio would still be processed, + // but some components may change behavior based on this information. + // Default false. + virtual void set_output_will_be_muted(bool muted) = 0; + virtual bool output_will_be_muted() const = 0; + // Processes a 10 ms |frame| of the primary audio stream. On the client-side, // this is the near-end (or captured) audio. // @@ -184,10 +236,25 @@ class AudioProcessing : public Module { // with the stream_ tag which is needed should be called after processing. // // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| - // members of |frame| must be valid, and correspond to settings supplied - // to APM. + // members of |frame| must be valid. If changed from the previous call to this + // method, it will trigger an initialization. virtual int ProcessStream(AudioFrame* frame) = 0; + // Accepts deinterleaved float audio with the range [-1, 1]. Each element + // of |src| points to a channel buffer, arranged according to + // |input_layout|. At output, the channels will be arranged according to + // |output_layout| at |output_sample_rate_hz| in |dest|. + // + // The output layout may only remove channels, not add. |src| and |dest| + // may use the same memory, if desired. + virtual int ProcessStream(const float* const* src, + int samples_per_channel, + int input_sample_rate_hz, + ChannelLayout input_layout, + int output_sample_rate_hz, + ChannelLayout output_layout, + float* const* dest) = 0; + // Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame // will not be modified. On the client-side, this is the far-end (or to be // rendered) audio. @@ -199,11 +266,19 @@ class AudioProcessing : public Module { // chances are you don't need to use it. // // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| - // members of |frame| must be valid. + // members of |frame| must be valid. |sample_rate_hz_| must correspond to + // |input_sample_rate_hz()| // // TODO(ajm): add const to input; requires an implementation fix. virtual int AnalyzeReverseStream(AudioFrame* frame) = 0; + // Accepts deinterleaved float audio with the range [-1, 1]. Each element + // of |data| points to a channel buffer, arranged according to |layout|. + virtual int AnalyzeReverseStream(const float* const* data, + int samples_per_channel, + int sample_rate_hz, + ChannelLayout layout) = 0; + // This must be called if and only if echo processing is enabled. // // Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end @@ -219,6 +294,12 @@ class AudioProcessing : public Module { // ProcessStream(). virtual int set_stream_delay_ms(int delay) = 0; virtual int stream_delay_ms() const = 0; + virtual bool was_stream_delay_set() const = 0; + + // Call to signal that a key press occurred (true) or did not occur (false) + // with this chunk of audio. + virtual void set_stream_key_pressed(bool key_pressed) = 0; + virtual bool stream_key_pressed() const = 0; // Sets a delay |offset| in ms to add to the values passed in through // set_stream_delay_ms(). May be positive or negative. @@ -283,9 +364,13 @@ class AudioProcessing : public Module { kBadStreamParameterWarning = -13 }; - // Inherited from Module. - virtual int32_t TimeUntilNextProcess() OVERRIDE; - virtual int32_t Process() OVERRIDE; + enum NativeRate { + kSampleRate8kHz = 8000, + kSampleRate16kHz = 16000, + kSampleRate32kHz = 32000 + }; + + static const int kChunkSizeMs = 10; }; // The acoustic echo cancellation (AEC) component provides better performance @@ -306,16 +391,10 @@ class EchoCancellation { // render and capture devices are used, particularly with webcams. // // This enables a compensation mechanism, and requires that - // |set_device_sample_rate_hz()| and |set_stream_drift_samples()| be called. + // set_stream_drift_samples() be called. virtual int enable_drift_compensation(bool enable) = 0; virtual bool is_drift_compensation_enabled() const = 0; - // Provides the sampling rate of the audio devices. It is assumed the render - // and capture devices use the same nominal sample rate. Required if and only - // if drift compensation is enabled. - virtual int set_device_sample_rate_hz(int rate) = 0; - virtual int device_sample_rate_hz() const = 0; - // Sets the difference between the number of samples rendered and captured by // the audio devices since the last call to |ProcessStream()|. Must be called // if drift compensation is enabled, prior to |ProcessStream()|. @@ -555,8 +634,7 @@ class LevelEstimator { // frames since the last call to RMS(). The returned value is positive but // should be interpreted as negative. It is constrained to [0, 127]. // - // The computation follows: - // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-05 + // The computation follows: https://tools.ietf.org/html/rfc6465 // with the intent that it can provide the RTP audio level indication. // // Frames passed to ProcessStream() with an |_energy| of zero are considered diff --git a/chromium/third_party/webrtc/modules/audio_processing/include/mock_audio_processing.h b/chromium/third_party/webrtc/modules/audio_processing/include/mock_audio_processing.h index 46520ab494e..c1ac23adf76 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/include/mock_audio_processing.h +++ b/chromium/third_party/webrtc/modules/audio_processing/include/mock_audio_processing.h @@ -26,10 +26,6 @@ class MockEchoCancellation : public EchoCancellation { int(bool enable)); MOCK_CONST_METHOD0(is_drift_compensation_enabled, bool()); - MOCK_METHOD1(set_device_sample_rate_hz, - int(int rate)); - MOCK_CONST_METHOD0(device_sample_rate_hz, - int()); MOCK_METHOD1(set_stream_drift_samples, void(int drift)); MOCK_CONST_METHOD0(stream_drift_samples, @@ -181,6 +177,13 @@ class MockAudioProcessing : public AudioProcessing { MOCK_METHOD0(Initialize, int()); + MOCK_METHOD6(Initialize, + int(int sample_rate_hz, + int output_sample_rate_hz, + int reverse_sample_rate_hz, + ChannelLayout input_layout, + ChannelLayout output_layout, + ChannelLayout reverse_layout)); MOCK_METHOD1(SetExtraOptions, void(const Config& config)); MOCK_METHOD1(EnableExperimentalNs, @@ -189,26 +192,49 @@ class MockAudioProcessing : public AudioProcessing { bool()); MOCK_METHOD1(set_sample_rate_hz, int(int rate)); + MOCK_CONST_METHOD0(input_sample_rate_hz, + int()); MOCK_CONST_METHOD0(sample_rate_hz, int()); - MOCK_METHOD2(set_num_channels, - int(int input_channels, int output_channels)); + MOCK_CONST_METHOD0(proc_sample_rate_hz, + int()); + MOCK_CONST_METHOD0(proc_split_sample_rate_hz, + int()); MOCK_CONST_METHOD0(num_input_channels, int()); MOCK_CONST_METHOD0(num_output_channels, int()); - MOCK_METHOD1(set_num_reverse_channels, - int(int channels)); MOCK_CONST_METHOD0(num_reverse_channels, int()); + MOCK_METHOD1(set_output_will_be_muted, + void(bool muted)); + MOCK_CONST_METHOD0(output_will_be_muted, + bool()); MOCK_METHOD1(ProcessStream, int(AudioFrame* frame)); + MOCK_METHOD7(ProcessStream, + int(const float* const* src, + int samples_per_channel, + int input_sample_rate_hz, + ChannelLayout input_layout, + int output_sample_rate_hz, + ChannelLayout output_layout, + float* const* dest)); MOCK_METHOD1(AnalyzeReverseStream, int(AudioFrame* frame)); + MOCK_METHOD4(AnalyzeReverseStream, + int(const float* const* data, int frames, int sample_rate_hz, + ChannelLayout input_layout)); MOCK_METHOD1(set_stream_delay_ms, int(int delay)); MOCK_CONST_METHOD0(stream_delay_ms, int()); + MOCK_CONST_METHOD0(was_stream_delay_set, + bool()); + MOCK_METHOD1(set_stream_key_pressed, + void(bool key_pressed)); + MOCK_CONST_METHOD0(stream_key_pressed, + bool()); MOCK_METHOD1(set_delay_offset_ms, void(int offset)); MOCK_CONST_METHOD0(delay_offset_ms, @@ -230,20 +256,16 @@ class MockAudioProcessing : public AudioProcessing { } virtual MockHighPassFilter* high_pass_filter() const { return high_pass_filter_.get(); - }; + } virtual MockLevelEstimator* level_estimator() const { return level_estimator_.get(); - }; + } virtual MockNoiseSuppression* noise_suppression() const { return noise_suppression_.get(); - }; + } virtual MockVoiceDetection* voice_detection() const { return voice_detection_.get(); - }; - MOCK_METHOD0(TimeUntilNextProcess, - int32_t()); - MOCK_METHOD0(Process, - int32_t()); + } private: scoped_ptr<MockEchoCancellation> echo_cancellation_; diff --git a/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.cc index 29dbdfc78e0..cfe295a6a0b 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.cc @@ -10,110 +10,35 @@ #include "webrtc/modules/audio_processing/level_estimator_impl.h" -#include <assert.h> -#include <math.h> -#include <string.h> - #include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" +#include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/modules/audio_processing/rms_level.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" namespace webrtc { -namespace { - -const double kMaxSquaredLevel = 32768.0 * 32768.0; - -class Level { - public: - static const int kMinLevel = 127; - - Level() - : sum_square_(0.0), - sample_count_(0) {} - ~Level() {} - - void Init() { - sum_square_ = 0.0; - sample_count_ = 0; - } - - void Process(int16_t* data, int length) { - assert(data != NULL); - assert(length > 0); - sum_square_ += SumSquare(data, length); - sample_count_ += length; - } - - void ProcessMuted(int length) { - assert(length > 0); - sample_count_ += length; - } - int RMS() { - if (sample_count_ == 0 || sum_square_ == 0.0) { - Init(); - return kMinLevel; - } - - // Normalize by the max level. - double rms = sum_square_ / (sample_count_ * kMaxSquaredLevel); - // 20log_10(x^0.5) = 10log_10(x) - rms = 10 * log10(rms); - if (rms > 0) - rms = 0; - else if (rms < -kMinLevel) - rms = -kMinLevel; - - rms = -rms; - Init(); - return static_cast<int>(rms + 0.5); - } - - private: - static double SumSquare(int16_t* data, int length) { - double sum_square = 0.0; - for (int i = 0; i < length; ++i) { - double data_d = static_cast<double>(data[i]); - sum_square += data_d * data_d; - } - return sum_square; - } - - double sum_square_; - int sample_count_; -}; -} // namespace - -LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), - apm_(apm) {} +LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), + crit_(crit) {} LevelEstimatorImpl::~LevelEstimatorImpl() {} int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { if (!is_component_enabled()) { - return apm_->kNoError; + return AudioProcessing::kNoError; } - Level* level = static_cast<Level*>(handle(0)); - if (audio->is_muted()) { - level->ProcessMuted(audio->samples_per_channel()); - return apm_->kNoError; + RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0)); + for (int i = 0; i < audio->num_channels(); ++i) { + rms_level->Process(audio->data(i), audio->samples_per_channel()); } - int16_t* mixed_data = audio->data(0); - if (audio->num_channels() > 1) { - audio->CopyAndMix(1); - mixed_data = audio->mixed_data(0); - } - - level->Process(mixed_data, audio->samples_per_channel()); - - return apm_->kNoError; + return AudioProcessing::kNoError; } int LevelEstimatorImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); return EnableComponent(enable); } @@ -123,43 +48,38 @@ bool LevelEstimatorImpl::is_enabled() const { int LevelEstimatorImpl::RMS() { if (!is_component_enabled()) { - return apm_->kNotEnabledError; + return AudioProcessing::kNotEnabledError; } - Level* level = static_cast<Level*>(handle(0)); - return level->RMS(); + RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0)); + return rms_level->RMS(); } +// The ProcessingComponent implementation is pretty weird in this class since +// we have only a single instance of the trivial underlying component. void* LevelEstimatorImpl::CreateHandle() const { - return new Level; + return new RMSLevel; } -int LevelEstimatorImpl::DestroyHandle(void* handle) const { - assert(handle != NULL); - Level* level = static_cast<Level*>(handle); - delete level; - return apm_->kNoError; +void LevelEstimatorImpl::DestroyHandle(void* handle) const { + delete static_cast<RMSLevel*>(handle); } int LevelEstimatorImpl::InitializeHandle(void* handle) const { - assert(handle != NULL); - Level* level = static_cast<Level*>(handle); - level->Init(); - - return apm_->kNoError; + static_cast<RMSLevel*>(handle)->Reset(); + return AudioProcessing::kNoError; } int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const { - return apm_->kNoError; + return AudioProcessing::kNoError; } int LevelEstimatorImpl::num_handles_required() const { return 1; } -int LevelEstimatorImpl::GetHandleError(void* handle) const { - // The component has no detailed errors. - assert(handle != NULL); - return apm_->kUnspecifiedError; +int LevelEstimatorImpl::GetHandleError(void* /*handle*/) const { + return AudioProcessing::kUnspecifiedError; } + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.h b/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.h index 20dc18dc425..b38337d4d41 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/level_estimator_impl.h @@ -13,15 +13,18 @@ #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/audio_processing/processing_component.h" +#include "webrtc/modules/audio_processing/rms_level.h" namespace webrtc { -class AudioProcessingImpl; + class AudioBuffer; +class CriticalSectionWrapper; class LevelEstimatorImpl : public LevelEstimator, public ProcessingComponent { public: - explicit LevelEstimatorImpl(const AudioProcessingImpl* apm); + LevelEstimatorImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit); virtual ~LevelEstimatorImpl(); int ProcessStream(AudioBuffer* audio); @@ -38,12 +41,13 @@ class LevelEstimatorImpl : public LevelEstimator, virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + CriticalSectionWrapper* crit_; }; + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.gypi b/chromium/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.gypi new file mode 100644 index 00000000000..f32ddd47f78 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.gypi @@ -0,0 +1,51 @@ +# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +# This file has common information for gen_core_neon_offsets.gyp +# and gen_core_neon_offsets_chromium.gyp +{ + 'variables': { + 'variables' : { + 'lib_intermediate_name': '', + 'conditions' : [ + ['android_webview_build==1', { + 'lib_intermediate_name' : '$(abspath $(call intermediates-dir-for,STATIC_LIBRARIES,lib_core_neon_offsets,,,$(gyp_var_prefix)))/lib_core_neon_offsets.a', + }], + ], + }, + 'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets', + 'output_dir': '<(shared_generated_dir)', + 'output_format': 'cheader', + 'unpack_lib_search_path_list': [ + '-a', '<(PRODUCT_DIR)/lib_core_neon_offsets.a', + '-a', '<(LIB_DIR)/webrtc/modules/audio_processing/lib_core_neon_offsets.a', + '-a', '<(LIB_DIR)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a', + '-a', '<(lib_intermediate_name)', + ], + 'unpack_lib_output_dir':'<(shared_generated_dir)', + }, + 'includes': [ + '../../build/common.gypi', + ], + 'conditions': [ + ['((target_arch=="arm" and arm_version==7) or target_arch=="armv7") and (OS=="android" or OS=="ios")', { + 'targets' : [ + { + 'target_name': 'lib_core_neon_offsets', + 'type': 'static_library', + 'android_unmangled_name': 1, + 'hard_dependency': 1, + 'sources': [ + 'ns/nsx_core_neon_offsets.c', + 'aecm/aecm_core_neon_offsets.c', + ], + }, + ], + }], + ], +} diff --git a/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc index 41c11b1cd72..eea0a04a2a6 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc @@ -12,15 +12,14 @@ #include <assert.h> -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/modules/audio_processing/audio_buffer.h" #if defined(WEBRTC_NS_FLOAT) #include "webrtc/modules/audio_processing/ns/include/noise_suppression.h" #elif defined(WEBRTC_NS_FIXED) #include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h" #endif +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" namespace webrtc { @@ -47,9 +46,11 @@ int MapSetting(NoiseSuppression::Level level) { } } // namespace -NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), +NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), apm_(apm), + crit_(crit), level_(kModerate) {} NoiseSuppressionImpl::~NoiseSuppressionImpl() {} @@ -67,10 +68,10 @@ int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { Handle* my_handle = static_cast<Handle*>(handle(i)); #if defined(WEBRTC_NS_FLOAT) err = WebRtcNs_Process(static_cast<Handle*>(handle(i)), - audio->low_pass_split_data(i), - audio->high_pass_split_data(i), - audio->low_pass_split_data(i), - audio->high_pass_split_data(i)); + audio->low_pass_split_data_f(i), + audio->high_pass_split_data_f(i), + audio->low_pass_split_data_f(i), + audio->high_pass_split_data_f(i)); #elif defined(WEBRTC_NS_FIXED) err = WebRtcNsx_Process(static_cast<Handle*>(handle(i)), audio->low_pass_split_data(i), @@ -88,7 +89,7 @@ int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { } int NoiseSuppressionImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); return EnableComponent(enable); } @@ -97,7 +98,7 @@ bool NoiseSuppressionImpl::is_enabled() const { } int NoiseSuppressionImpl::set_level(Level level) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (MapSetting(level) == -1) { return apm_->kBadParameterError; } @@ -140,19 +141,21 @@ void* NoiseSuppressionImpl::CreateHandle() const { return handle; } -int NoiseSuppressionImpl::DestroyHandle(void* handle) const { +void NoiseSuppressionImpl::DestroyHandle(void* handle) const { #if defined(WEBRTC_NS_FLOAT) - return WebRtcNs_Free(static_cast<Handle*>(handle)); + WebRtcNs_Free(static_cast<Handle*>(handle)); #elif defined(WEBRTC_NS_FIXED) - return WebRtcNsx_Free(static_cast<Handle*>(handle)); + WebRtcNsx_Free(static_cast<Handle*>(handle)); #endif } int NoiseSuppressionImpl::InitializeHandle(void* handle) const { #if defined(WEBRTC_NS_FLOAT) - return WebRtcNs_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz()); + return WebRtcNs_Init(static_cast<Handle*>(handle), + apm_->proc_sample_rate_hz()); #elif defined(WEBRTC_NS_FIXED) - return WebRtcNsx_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz()); + return WebRtcNsx_Init(static_cast<Handle*>(handle), + apm_->proc_sample_rate_hz()); #endif } diff --git a/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.h b/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.h index f6dd8cbd78a..cadbbd9cd4c 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.h @@ -15,13 +15,15 @@ #include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { -class AudioProcessingImpl; + class AudioBuffer; +class CriticalSectionWrapper; class NoiseSuppressionImpl : public NoiseSuppression, public ProcessingComponent { public: - explicit NoiseSuppressionImpl(const AudioProcessingImpl* apm); + NoiseSuppressionImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit); virtual ~NoiseSuppressionImpl(); int ProcessCaptureAudio(AudioBuffer* audio); @@ -40,13 +42,15 @@ class NoiseSuppressionImpl : public NoiseSuppression, virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + const AudioProcessing* apm_; + CriticalSectionWrapper* crit_; Level level_; }; + } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/include/noise_suppression.h b/chromium/third_party/webrtc/modules/audio_processing/ns/include/noise_suppression.h index 32b18038089..3cf889e2d07 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/ns/include/noise_suppression.h +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/include/noise_suppression.h @@ -99,10 +99,10 @@ int WebRtcNs_set_policy(NsHandle* NS_inst, int mode); * -1 - Error */ int WebRtcNs_Process(NsHandle* NS_inst, - short* spframe, - short* spframe_H, - short* outframe, - short* outframe_H); + float* spframe, + float* spframe_H, + float* outframe, + float* outframe_H); /* Returns the internally used prior speech probability of the current frame. * There is a frequency bin based one as well, with which this should not be diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/noise_suppression.c b/chromium/third_party/webrtc/modules/audio_processing/ns/noise_suppression.c index 848467f080c..075ab88c1c6 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/ns/noise_suppression.c +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/noise_suppression.c @@ -43,8 +43,8 @@ int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) { } -int WebRtcNs_Process(NsHandle* NS_inst, short* spframe, short* spframe_H, - short* outframe, short* outframe_H) { +int WebRtcNs_Process(NsHandle* NS_inst, float* spframe, float* spframe_H, + float* outframe, float* outframe_H) { return WebRtcNs_ProcessCore( (NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H); } diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.c b/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.c index 124a66d8df5..ec267ae0f69 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.c +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.c @@ -715,10 +715,10 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr } int WebRtcNs_ProcessCore(NSinst_t* inst, - short* speechFrame, - short* speechFrameHB, - short* outFrame, - short* outFrameHB) { + float* speechFrame, + float* speechFrameHB, + float* outFrame, + float* outFrameHB) { // main routine for noise reduction int flagHB = 0; @@ -731,8 +731,8 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, float snrPrior, currentEstimateStsa; float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech; float gammaNoiseTmp, gammaNoiseOld; - float noiseUpdateTmp, fTmp, dTmp; - float fin[BLOCKL_MAX], fout[BLOCKL_MAX]; + float noiseUpdateTmp, fTmp; + float fout[BLOCKL_MAX]; float winData[ANAL_BLOCKL_MAX]; float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL]; float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL]; @@ -775,26 +775,17 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, updateParsFlag = inst->modelUpdatePars[0]; // - //for LB do all processing - // convert to float - for (i = 0; i < inst->blockLen10ms; i++) { - fin[i] = (float)speechFrame[i]; - } // update analysis buffer for L band memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms, sizeof(float) * (inst->anaLen - inst->blockLen10ms)); - memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin, + memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, speechFrame, sizeof(float) * inst->blockLen10ms); if (flagHB == 1) { - // convert to float - for (i = 0; i < inst->blockLen10ms; i++) { - fin[i] = (float)speechFrameHB[i]; - } // update analysis buffer for H band memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms, sizeof(float) * (inst->anaLen - inst->blockLen10ms)); - memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin, + memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, speechFrameHB, sizeof(float) * inst->blockLen10ms); } @@ -833,30 +824,16 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, inst->outBuf[i] = fout[i + inst->blockLen10ms]; } } - // convert to short - for (i = 0; i < inst->blockLen10ms; i++) { - dTmp = fout[i]; - if (dTmp < WEBRTC_SPL_WORD16_MIN) { - dTmp = WEBRTC_SPL_WORD16_MIN; - } else if (dTmp > WEBRTC_SPL_WORD16_MAX) { - dTmp = WEBRTC_SPL_WORD16_MAX; - } - outFrame[i] = (short)dTmp; - } + for (i = 0; i < inst->blockLen10ms; ++i) + outFrame[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN); // for time-domain gain of HB - if (flagHB == 1) { - for (i = 0; i < inst->blockLen10ms; i++) { - dTmp = inst->dataBufHB[i]; - if (dTmp < WEBRTC_SPL_WORD16_MIN) { - dTmp = WEBRTC_SPL_WORD16_MIN; - } else if (dTmp > WEBRTC_SPL_WORD16_MAX) { - dTmp = WEBRTC_SPL_WORD16_MAX; - } - outFrameHB[i] = (short)dTmp; - } - } // end of H band gain computation - // + if (flagHB == 1) + for (i = 0; i < inst->blockLen10ms; ++i) + outFrameHB[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN); + return 0; } @@ -1239,16 +1216,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, inst->outLen -= inst->blockLen10ms; } - // convert to short - for (i = 0; i < inst->blockLen10ms; i++) { - dTmp = fout[i]; - if (dTmp < WEBRTC_SPL_WORD16_MIN) { - dTmp = WEBRTC_SPL_WORD16_MIN; - } else if (dTmp > WEBRTC_SPL_WORD16_MAX) { - dTmp = WEBRTC_SPL_WORD16_MAX; - } - outFrame[i] = (short)dTmp; - } + for (i = 0; i < inst->blockLen10ms; ++i) + outFrame[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN); // for time-domain gain of HB if (flagHB == 1) { @@ -1289,13 +1259,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, } //apply gain for (i = 0; i < inst->blockLen10ms; i++) { - dTmp = gainTimeDomainHB * inst->dataBufHB[i]; - if (dTmp < WEBRTC_SPL_WORD16_MIN) { - dTmp = WEBRTC_SPL_WORD16_MIN; - } else if (dTmp > WEBRTC_SPL_WORD16_MAX) { - dTmp = WEBRTC_SPL_WORD16_MAX; - } - outFrameHB[i] = (short)dTmp; + float o = gainTimeDomainHB * inst->dataBufHB[i]; + outFrameHB[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, o, WEBRTC_SPL_WORD16_MIN); } } // end of H band gain computation // diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.h b/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.h index 50daa137cf8..785239ebdac 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.h +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/ns_core.h @@ -167,10 +167,10 @@ int WebRtcNs_set_policy_core(NSinst_t* inst, int mode); int WebRtcNs_ProcessCore(NSinst_t* inst, - short* inFrameLow, - short* inFrameHigh, - short* outFrameLow, - short* outFrameHigh); + float* inFrameLow, + float* inFrameHigh, + float* outFrameLow, + float* outFrameHigh); #ifdef __cplusplus diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c index c7229579f4c..2c8270f568c 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c @@ -70,11 +70,6 @@ static const int16_t WebRtcNsx_kLogTableFrac[256] = { // Skip first frequency bins during estimation. (0 <= value < 64) static const int kStartBand = 5; -static const int16_t kIndicatorTable[17] = { - 0, 2017, 3809, 5227, 6258, 6963, 7424, 7718, - 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187 -}; - // hybrib Hanning & flat window static const int16_t kBlocks80w128x[128] = { 0, 536, 1072, 1606, 2139, 2669, 3196, 3720, 4240, 4756, 5266, @@ -481,7 +476,7 @@ static void PrepareSpectrumC(NsxInst_t* inst, int16_t* freq_buf) { } // Denormalize the real-valued signal |in|, the output from inverse FFT. -static __inline void Denormalize(NsxInst_t* inst, int16_t* in, int factor) { +static void DenormalizeC(NsxInst_t* inst, int16_t* in, int factor) { int i = 0; int32_t tmp32 = 0; for (i = 0; i < inst->anaLen; i += 1) { @@ -546,9 +541,9 @@ static void AnalysisUpdateC(NsxInst_t* inst, } // Normalize the real-valued signal |in|, the input to forward FFT. -static __inline void NormalizeRealBuffer(NsxInst_t* inst, - const int16_t* in, - int16_t* out) { +static void NormalizeRealBufferC(NsxInst_t* inst, + const int16_t* in, + int16_t* out) { int i = 0; for (i = 0; i < inst->anaLen; ++i) { out[i] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData) @@ -560,6 +555,8 @@ NoiseEstimation WebRtcNsx_NoiseEstimation; PrepareSpectrum WebRtcNsx_PrepareSpectrum; SynthesisUpdate WebRtcNsx_SynthesisUpdate; AnalysisUpdate WebRtcNsx_AnalysisUpdate; +Denormalize WebRtcNsx_Denormalize; +NormalizeRealBuffer WebRtcNsx_NormalizeRealBuffer; #if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON) // Initialize function pointers for ARM Neon platform. @@ -571,6 +568,19 @@ static void WebRtcNsx_InitNeon(void) { } #endif +#if defined(MIPS32_LE) +// Initialize function pointers for MIPS platform. +static void WebRtcNsx_InitMips(void) { + WebRtcNsx_PrepareSpectrum = WebRtcNsx_PrepareSpectrum_mips; + WebRtcNsx_SynthesisUpdate = WebRtcNsx_SynthesisUpdate_mips; + WebRtcNsx_AnalysisUpdate = WebRtcNsx_AnalysisUpdate_mips; + WebRtcNsx_NormalizeRealBuffer = WebRtcNsx_NormalizeRealBuffer_mips; +#if defined(MIPS_DSP_R1_LE) + WebRtcNsx_Denormalize = WebRtcNsx_Denormalize_mips; +#endif +} +#endif + void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst, int16_t pink_noise_exp_avg, int32_t pink_noise_num_avg, @@ -758,6 +768,8 @@ int32_t WebRtcNsx_InitCore(NsxInst_t* inst, uint32_t fs) { WebRtcNsx_PrepareSpectrum = PrepareSpectrumC; WebRtcNsx_SynthesisUpdate = SynthesisUpdateC; WebRtcNsx_AnalysisUpdate = AnalysisUpdateC; + WebRtcNsx_Denormalize = DenormalizeC; + WebRtcNsx_NormalizeRealBuffer = NormalizeRealBufferC; #ifdef WEBRTC_DETECT_ARM_NEON uint64_t features = WebRtc_GetCPUFeaturesARM(); @@ -768,6 +780,10 @@ int32_t WebRtcNsx_InitCore(NsxInst_t* inst, uint32_t fs) { WebRtcNsx_InitNeon(); #endif +#if defined(MIPS32_LE) + WebRtcNsx_InitMips(); +#endif + inst->initFlag = 1; return 0; @@ -1169,239 +1185,6 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) { } } -// Compute speech/noise probability -// speech/noise probability is returned in: probSpeechFinal -//snrLocPrior is the prior SNR for each frequency (in Q11) -//snrLocPost is the post SNR for each frequency (in Q11) -void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, uint16_t* nonSpeechProbFinal, - uint32_t* priorLocSnr, uint32_t* postLocSnr) { - uint32_t zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3; - - int32_t invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32; - int32_t frac32, logTmp; - int32_t logLrtTimeAvgKsumFX; - - int16_t indPriorFX16; - int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart; - - int i, normTmp, normTmp2, nShifts; - - // compute feature based on average LR factor - // this is the average over all frequencies of the smooth log LRT - logLrtTimeAvgKsumFX = 0; - for (i = 0; i < inst->magnLen; i++) { - besselTmpFX32 = (int32_t)postLocSnr[i]; // Q11 - normTmp = WebRtcSpl_NormU32(postLocSnr[i]); - num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp) - if (normTmp > 10) { - den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp) - } else { - den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp) - } - if (den > 0) { - besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11 - } else { - besselTmpFX32 -= num; // Q11 - } - - // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) - inst->logLrtTimeAvg[i]); - // Here, LRT_TAVG = 0.5 - zeros = WebRtcSpl_NormU32(priorLocSnr[i]); - frac32 = (int32_t)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19); - tmp32 = WEBRTC_SPL_MUL(frac32, frac32); - tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19); - tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)frac32, 5412, 12); - frac32 = tmp32 + 37; - // tmp32 = log2(priorLocSnr[i]) - tmp32 = (int32_t)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12 - logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); // log2(priorLocSnr[i])*log(2) - tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); // Q12 - inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12 - - logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12 - } - inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, inst->stages + 10); // 5 = BIN_SIZE_LRT / 2 - // done with computation of LR factor - - // - //compute the indicator functions - // - - // average LRT feature - // FLOAT code - // indicator0 = 0.5 * (tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.0); - tmpIndFX = 16384; // Q14(1.0) - tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12 - nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5; - //use larger width in tanh map for pause regions - if (tmp32no1 < 0) { - tmpIndFX = 0; - tmp32no1 = -tmp32no1; - //widthPrior = widthPrior * 2.0; - nShifts++; - } - tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14 - // compute indicator function: sigmoid map - tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14); - if ((tableIndex < 16) && (tableIndex >= 0)) { - tmp16no2 = kIndicatorTable[tableIndex]; - tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; - frac = (int16_t)(tmp32no1 & 0x00003fff); // Q14 - tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14); - if (tmpIndFX == 0) { - tmpIndFX = 8192 - tmp16no2; // Q14 - } else { - tmpIndFX = 8192 + tmp16no2; // Q14 - } - } - indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14 - - //spectral flatness feature - if (inst->weightSpecFlat) { - tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10 - tmpIndFX = 16384; // Q14(1.0) - //use larger width in tanh map for pause regions - tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10 - nShifts = 4; - if (inst->thresholdSpecFlat < tmpU32no1) { - tmpIndFX = 0; - tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat; - //widthPrior = widthPrior * 2.0; - nShifts++; - } - tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, - nShifts), 25); //Q14 - tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), 25); //Q14 - // compute indicator function: sigmoid map - // FLOAT code - // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + 1.0); - tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14); - if (tableIndex < 16) { - tmp16no2 = kIndicatorTable[tableIndex]; - tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; - frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14 - tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14); - if (tmpIndFX) { - tmpIndFX = 8192 + tmp16no2; // Q14 - } else { - tmpIndFX = 8192 - tmp16no2; // Q14 - } - } - indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14 - } - - //for template spectral-difference - if (inst->weightSpecDiff) { - tmpU32no1 = 0; - if (inst->featureSpecDiff) { - normTmp = WEBRTC_SPL_MIN(20 - inst->stages, - WebRtcSpl_NormU32(inst->featureSpecDiff)); - tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); // Q(normTmp-2*stages) - tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, 20 - inst->stages - - normTmp); - if (tmpU32no2 > 0) { - // Q(20 - inst->stages) - tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); - } else { - tmpU32no1 = (uint32_t)(0x7fffffff); - } - } - tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, 17), 25); - tmpU32no2 = tmpU32no1 - tmpU32no3; - nShifts = 1; - tmpIndFX = 16384; // Q14(1.0) - //use larger width in tanh map for pause regions - if (tmpU32no2 & 0x80000000) { - tmpIndFX = 0; - tmpU32no2 = tmpU32no3 - tmpU32no1; - //widthPrior = widthPrior * 2.0; - nShifts--; - } - tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts); - // compute indicator function: sigmoid map - /* FLOAT code - indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0); - */ - tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14); - if (tableIndex < 16) { - tmp16no2 = kIndicatorTable[tableIndex]; - tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; - frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14 - tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( - tmp16no1, frac, 14); - if (tmpIndFX) { - tmpIndFX = 8192 + tmp16no2; - } else { - tmpIndFX = 8192 - tmp16no2; - } - } - indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14 - } - - //combine the indicator function with the feature weights - // FLOAT code - // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 * indicator2); - indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14 - // done with computing indicator function - - //compute the prior probability - // FLOAT code - // inst->priorNonSpeechProb += PRIOR_UPDATE * (indPriorNonSpeech - inst->priorNonSpeechProb); - tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14 - inst->priorNonSpeechProb += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( - PRIOR_UPDATE_Q14, tmp16, 14); // Q14 - - //final speech probability: combine prior model with LR factor: - - memset(nonSpeechProbFinal, 0, sizeof(uint16_t) * inst->magnLen); - - if (inst->priorNonSpeechProb > 0) { - for (i = 0; i < inst->magnLen; i++) { - // FLOAT code - // invLrt = exp(inst->logLrtTimeAvg[i]); - // invLrt = inst->priorSpeechProb * invLrt; - // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / (1.0 - inst->priorSpeechProb + invLrt); - // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt; - // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / (inst->priorNonSpeechProb + invLrt); - if (inst->logLrtTimeAvgW32[i] < 65300) { - tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(inst->logLrtTimeAvgW32[i], 23637), - 14); // Q12 - intPart = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12); - if (intPart < -8) { - intPart = -8; - } - frac = (int16_t)(tmp32no1 & 0x00000fff); // Q12 - - // Quadratic approximation of 2^frac - tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12 - tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12 - invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart) - + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8 - - normTmp = WebRtcSpl_NormW32(invLrtFX); - normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb)); - if (normTmp + normTmp2 >= 7) { - if (normTmp + normTmp2 < 15) { - invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp); - // Q(normTmp+normTmp2-7) - tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); - // Q(normTmp+normTmp2+7) - invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); // Q14 - } else { - tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q22 - invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14 - } - - tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)inst->priorNonSpeechProb, 8); // Q22 - - nonSpeechProbFinal[i] = (uint16_t)WEBRTC_SPL_DIV(tmp32no1, - (int32_t)inst->priorNonSpeechProb + invLrtFX); // Q8 - } - } - } - } -} - // Transform input (speechFrame) to frequency domain magnitude (magnU16) void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU16) { @@ -1461,7 +1244,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU right_shifts_in_magnU16 = WEBRTC_SPL_MAX(right_shifts_in_magnU16, 0); // create realImag as winData interleaved with zeros (= imag. part), normalize it - NormalizeRealBuffer(inst, winData, realImag); + WebRtcNsx_NormalizeRealBuffer(inst, winData, realImag); // FFT output will be in winData[]. WebRtcSpl_RealForwardFFT(inst->real_fft, realImag, winData); @@ -1624,9 +1407,9 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((uint32_t)sum_log_i_log_magn, 12); // Q5 // Shift the largest value of sum_log_i and tmp32no3 before multiplication - tmp_u16 = WEBRTC_SPL_LSHIFT_U16((uint16_t)sum_log_i, 1); // Q6 + tmp_u16 = ((uint16_t)sum_log_i << 1); // Q6 if ((uint32_t)sum_log_i > tmpU32no1) { - tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros); + tmp_u16 >>= zeros; } else { tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros); } @@ -1693,7 +1476,7 @@ void WebRtcNsx_DataSynthesis(NsxInst_t* inst, short* outFrame) { // Inverse FFT output will be in rfft_out[]. outCIFFT = WebRtcSpl_RealInverseFFT(inst->real_fft, realImag, rfft_out); - Denormalize(inst, rfft_out, outCIFFT); + WebRtcNsx_Denormalize(inst, rfft_out, outCIFFT); //scale factor: only do it after END_STARTUP_LONG time gainFactor = 8192; // 8192 = Q13(1.0) @@ -2288,8 +2071,8 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram tmpU16no1 += nonSpeechProbFinal[i]; // Q8 tmpU32no1 += (uint32_t)(inst->noiseSupFilter[i]); // Q14 } - avgProbSpeechHB = (int16_t)(4096 - - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12 + assert(inst->stages >= 7); + avgProbSpeechHB = (4096 - (tmpU16no1 >> (inst->stages - 7))); // Q12 avgFilterGainHB = (int16_t)WEBRTC_SPL_RSHIFT_U32( tmpU32no1, inst->stages - 3); // Q14 diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h index 1ad369ffbeb..5b3c5e78f4e 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h @@ -201,6 +201,23 @@ typedef void (*AnalysisUpdate)(NsxInst_t* inst, int16_t* new_speech); extern AnalysisUpdate WebRtcNsx_AnalysisUpdate; +// Denormalize the real-valued signal |in|, the output from inverse FFT. +typedef void (*Denormalize) (NsxInst_t* inst, int16_t* in, int factor); +extern Denormalize WebRtcNsx_Denormalize; + +// Normalize the real-valued signal |in|, the input to forward FFT. +typedef void (*NormalizeRealBuffer) (NsxInst_t* inst, + const int16_t* in, + int16_t* out); +extern NormalizeRealBuffer WebRtcNsx_NormalizeRealBuffer; + +// Compute speech/noise probability. +// Intended to be private. +void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, + uint16_t* nonSpeechProbFinal, + uint32_t* priorLocSnr, + uint32_t* postLocSnr); + #if (defined WEBRTC_DETECT_ARM_NEON) || defined (WEBRTC_ARCH_ARM_NEON) // For the above function pointers, functions for generic platforms are declared // and defined as static in file nsx_core.c, while those for ARM Neon platforms @@ -218,6 +235,26 @@ void WebRtcNsx_AnalysisUpdateNeon(NsxInst_t* inst, void WebRtcNsx_PrepareSpectrumNeon(NsxInst_t* inst, int16_t* freq_buff); #endif +#if defined(MIPS32_LE) +// For the above function pointers, functions for generic platforms are declared +// and defined as static in file nsx_core.c, while those for MIPS platforms +// are declared below and defined in file nsx_core_mips.c. +void WebRtcNsx_SynthesisUpdate_mips(NsxInst_t* inst, + int16_t* out_frame, + int16_t gain_factor); +void WebRtcNsx_AnalysisUpdate_mips(NsxInst_t* inst, + int16_t* out, + int16_t* new_speech); +void WebRtcNsx_PrepareSpectrum_mips(NsxInst_t* inst, int16_t* freq_buff); +void WebRtcNsx_NormalizeRealBuffer_mips(NsxInst_t* inst, + const int16_t* in, + int16_t* out); +#if defined(MIPS_DSP_R1_LE) +void WebRtcNsx_Denormalize_mips(NsxInst_t* inst, int16_t* in, int factor); +#endif + +#endif + #ifdef __cplusplus } #endif diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c new file mode 100644 index 00000000000..452b96e77b0 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c @@ -0,0 +1,273 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h" +#include "webrtc/modules/audio_processing/ns/nsx_core.h" + +static const int16_t kIndicatorTable[17] = { + 0, 2017, 3809, 5227, 6258, 6963, 7424, 7718, + 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187 +}; + +// Compute speech/noise probability +// speech/noise probability is returned in: probSpeechFinal +//snrLocPrior is the prior SNR for each frequency (in Q11) +//snrLocPost is the post SNR for each frequency (in Q11) +void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, + uint16_t* nonSpeechProbFinal, + uint32_t* priorLocSnr, + uint32_t* postLocSnr) { + + uint32_t zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3; + int32_t invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32; + int32_t frac32, logTmp; + int32_t logLrtTimeAvgKsumFX; + int16_t indPriorFX16; + int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart; + int i, normTmp, normTmp2, nShifts; + + // compute feature based on average LR factor + // this is the average over all frequencies of the smooth log LRT + logLrtTimeAvgKsumFX = 0; + for (i = 0; i < inst->magnLen; i++) { + besselTmpFX32 = (int32_t)postLocSnr[i]; // Q11 + normTmp = WebRtcSpl_NormU32(postLocSnr[i]); + num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp) + if (normTmp > 10) { + den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp) + } else { + den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp) + } + if (den > 0) { + besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11 + } else { + besselTmpFX32 -= num; // Q11 + } + + // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) + // - inst->logLrtTimeAvg[i]); + // Here, LRT_TAVG = 0.5 + zeros = WebRtcSpl_NormU32(priorLocSnr[i]); + frac32 = (int32_t)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19); + tmp32 = WEBRTC_SPL_MUL(frac32, frac32); + tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19); + tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)frac32, 5412, 12); + frac32 = tmp32 + 37; + // tmp32 = log2(priorLocSnr[i]) + tmp32 = (int32_t)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12 + logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); + // log2(priorLocSnr[i])*log(2) + tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); + // Q12 + inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12 + + logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12 + } + inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, + inst->stages + 10); + // 5 = BIN_SIZE_LRT / 2 + // done with computation of LR factor + + // + //compute the indicator functions + // + + // average LRT feature + // FLOAT code + // indicator0 = 0.5 * (tanh(widthPrior * + // (logLrtTimeAvgKsum - threshPrior0)) + 1.0); + tmpIndFX = 16384; // Q14(1.0) + tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12 + nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5; + //use larger width in tanh map for pause regions + if (tmp32no1 < 0) { + tmpIndFX = 0; + tmp32no1 = -tmp32no1; + //widthPrior = widthPrior * 2.0; + nShifts++; + } + tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14 + // compute indicator function: sigmoid map + tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14); + if ((tableIndex < 16) && (tableIndex >= 0)) { + tmp16no2 = kIndicatorTable[tableIndex]; + tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; + frac = (int16_t)(tmp32no1 & 0x00003fff); // Q14 + tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14); + if (tmpIndFX == 0) { + tmpIndFX = 8192 - tmp16no2; // Q14 + } else { + tmpIndFX = 8192 + tmp16no2; // Q14 + } + } + indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14 + + //spectral flatness feature + if (inst->weightSpecFlat) { + tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10 + tmpIndFX = 16384; // Q14(1.0) + //use larger width in tanh map for pause regions + tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10 + nShifts = 4; + if (inst->thresholdSpecFlat < tmpU32no1) { + tmpIndFX = 0; + tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat; + //widthPrior = widthPrior * 2.0; + nShifts++; + } + tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, + nShifts), 25); + //Q14 + tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), + 25); //Q14 + // compute indicator function: sigmoid map + // FLOAT code + // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * + // (threshPrior1 - tmpFloat1)) + 1.0); + tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14); + if (tableIndex < 16) { + tmp16no2 = kIndicatorTable[tableIndex]; + tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; + frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14 + tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14); + if (tmpIndFX) { + tmpIndFX = 8192 + tmp16no2; // Q14 + } else { + tmpIndFX = 8192 - tmp16no2; // Q14 + } + } + indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14 + } + + //for template spectral-difference + if (inst->weightSpecDiff) { + tmpU32no1 = 0; + if (inst->featureSpecDiff) { + normTmp = WEBRTC_SPL_MIN(20 - inst->stages, + WebRtcSpl_NormU32(inst->featureSpecDiff)); + tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); + // Q(normTmp-2*stages) + tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, + 20 - inst->stages - normTmp); + if (tmpU32no2 > 0) { + // Q(20 - inst->stages) + tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); + } else { + tmpU32no1 = (uint32_t)(0x7fffffff); + } + } + tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, + 17), + 25); + tmpU32no2 = tmpU32no1 - tmpU32no3; + nShifts = 1; + tmpIndFX = 16384; // Q14(1.0) + //use larger width in tanh map for pause regions + if (tmpU32no2 & 0x80000000) { + tmpIndFX = 0; + tmpU32no2 = tmpU32no3 - tmpU32no1; + //widthPrior = widthPrior * 2.0; + nShifts--; + } + tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts); + // compute indicator function: sigmoid map + /* FLOAT code + indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0); + */ + tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14); + if (tableIndex < 16) { + tmp16no2 = kIndicatorTable[tableIndex]; + tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; + frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14 + tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( + tmp16no1, frac, 14); + if (tmpIndFX) { + tmpIndFX = 8192 + tmp16no2; + } else { + tmpIndFX = 8192 - tmp16no2; + } + } + indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14 + } + + //combine the indicator function with the feature weights + // FLOAT code + // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * + // indicator1 + weightIndPrior2 * indicator2); + indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14 + // done with computing indicator function + + //compute the prior probability + // FLOAT code + // inst->priorNonSpeechProb += PRIOR_UPDATE * + // (indPriorNonSpeech - inst->priorNonSpeechProb); + tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14 + inst->priorNonSpeechProb += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( + PRIOR_UPDATE_Q14, tmp16, 14); // Q14 + + //final speech probability: combine prior model with LR factor: + + memset(nonSpeechProbFinal, 0, sizeof(uint16_t) * inst->magnLen); + + if (inst->priorNonSpeechProb > 0) { + for (i = 0; i < inst->magnLen; i++) { + // FLOAT code + // invLrt = exp(inst->logLrtTimeAvg[i]); + // invLrt = inst->priorSpeechProb * invLrt; + // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / + // (1.0 - inst->priorSpeechProb + invLrt); + // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt; + // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / + // (inst->priorNonSpeechProb + invLrt); + if (inst->logLrtTimeAvgW32[i] < 65300) { + tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL( + inst->logLrtTimeAvgW32[i], 23637), + 14); // Q12 + intPart = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12); + if (intPart < -8) { + intPart = -8; + } + frac = (int16_t)(tmp32no1 & 0x00000fff); // Q12 + + // Quadratic approximation of 2^frac + tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12 + tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12 + invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart) + + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8 + + normTmp = WebRtcSpl_NormW32(invLrtFX); + normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb)); + if (normTmp + normTmp2 >= 7) { + if (normTmp + normTmp2 < 15) { + invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp); + // Q(normTmp+normTmp2-7) + tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, + (16384 - inst->priorNonSpeechProb)); + // Q(normTmp+normTmp2+7) + invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); + // Q14 + } else { + tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, + (16384 - inst->priorNonSpeechProb)); + // Q22 + invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14 + } + + tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)inst->priorNonSpeechProb, + 8); // Q22 + + nonSpeechProbFinal[i] = (uint16_t)WEBRTC_SPL_DIV(tmp32no1, + (int32_t)inst->priorNonSpeechProb + invLrtFX); // Q8 + } + } + } + } +} + diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c new file mode 100644 index 00000000000..ccb0c376324 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c @@ -0,0 +1,1008 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h" +#include "webrtc/modules/audio_processing/ns/nsx_core.h" + +static const int16_t kIndicatorTable[17] = { + 0, 2017, 3809, 5227, 6258, 6963, 7424, 7718, + 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187 +}; + +// Compute speech/noise probability +// speech/noise probability is returned in: probSpeechFinal +//snrLocPrior is the prior SNR for each frequency (in Q11) +//snrLocPost is the post SNR for each frequency (in Q11) +void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, + uint16_t* nonSpeechProbFinal, + uint32_t* priorLocSnr, + uint32_t* postLocSnr) { + + uint32_t tmpU32no1, tmpU32no2, tmpU32no3; + int32_t indPriorFX, tmp32no1; + int32_t logLrtTimeAvgKsumFX; + int16_t indPriorFX16; + int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac; + int i, normTmp, nShifts; + + int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9; + int32_t const_max = 0x7fffffff; + int32_t const_neg43 = -43; + int32_t const_5412 = 5412; + int32_t const_11rsh12 = (11 << 12); + int32_t const_178 = 178; + + + // compute feature based on average LR factor + // this is the average over all frequencies of the smooth log LRT + logLrtTimeAvgKsumFX = 0; + for (i = 0; i < inst->magnLen; i++) { + r0 = postLocSnr[i]; // Q11 + r1 = priorLocSnr[i]; + r2 = inst->logLrtTimeAvgW32[i]; + + __asm __volatile( + ".set push \n\t" + ".set noreorder \n\t" + "clz %[r3], %[r0] \n\t" + "clz %[r5], %[r1] \n\t" + "slti %[r4], %[r3], 32 \n\t" + "slti %[r6], %[r5], 32 \n\t" + "movz %[r3], $0, %[r4] \n\t" + "movz %[r5], $0, %[r6] \n\t" + "slti %[r4], %[r3], 11 \n\t" + "addiu %[r6], %[r3], -11 \n\t" + "neg %[r7], %[r6] \n\t" + "sllv %[r6], %[r1], %[r6] \n\t" + "srav %[r7], %[r1], %[r7] \n\t" + "movn %[r6], %[r7], %[r4] \n\t" + "sllv %[r1], %[r1], %[r5] \n\t" + "and %[r1], %[r1], %[const_max] \n\t" + "sra %[r1], %[r1], 19 \n\t" + "mul %[r7], %[r1], %[r1] \n\t" + "sllv %[r3], %[r0], %[r3] \n\t" + "divu %[r8], %[r3], %[r6] \n\t" + "slti %[r6], %[r6], 1 \n\t" + "mul %[r7], %[r7], %[const_neg43] \n\t" + "sra %[r7], %[r7], 19 \n\t" + "movz %[r3], %[r8], %[r6] \n\t" + "subu %[r0], %[r0], %[r3] \n\t" + "mul %[r1], %[r1], %[const_5412] \n\t" + "sra %[r1], %[r1], 12 \n\t" + "addu %[r7], %[r7], %[r1] \n\t" + "addiu %[r1], %[r7], 37 \n\t" + "addiu %[r5], %[r5], -31 \n\t" + "neg %[r5], %[r5] \n\t" + "sll %[r5], %[r5], 12 \n\t" + "addu %[r5], %[r5], %[r1] \n\t" + "subu %[r7], %[r5], %[const_11rsh12] \n\t" + "mul %[r7], %[r7], %[const_178] \n\t" + "sra %[r7], %[r7], 8 \n\t" + "addu %[r7], %[r7], %[r2] \n\t" + "sra %[r7], %[r7], 1 \n\t" + "subu %[r2], %[r2], %[r7] \n\t" + "addu %[r2], %[r2], %[r0] \n\t" + ".set pop \n\t" + : [r0] "+r" (r0), [r1] "+r" (r1), [r2] "+r" (r2), + [r3] "=&r" (r3), [r4] "=&r" (r4), [r5] "=&r" (r5), + [r6] "=&r" (r6), [r7] "=&r" (r7), [r8] "=&r" (r8) + : [const_max] "r" (const_max), [const_neg43] "r" (const_neg43), + [const_5412] "r" (const_5412), [const_11rsh12] "r" (const_11rsh12), + [const_178] "r" (const_178) + : "hi", "lo" + ); + inst->logLrtTimeAvgW32[i] = r2; + logLrtTimeAvgKsumFX += r2; + } + + inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, + inst->stages + 10); + // 5 = BIN_SIZE_LRT / 2 + // done with computation of LR factor + + // + // compute the indicator functions + // + + // average LRT feature + // FLOAT code + // indicator0 = 0.5 * (tanh(widthPrior * + // (logLrtTimeAvgKsum - threshPrior0)) + 1.0); + tmpIndFX = 16384; // Q14(1.0) + tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12 + nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5; + //use larger width in tanh map for pause regions + if (tmp32no1 < 0) { + tmpIndFX = 0; + tmp32no1 = -tmp32no1; + //widthPrior = widthPrior * 2.0; + nShifts++; + } + tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14 + // compute indicator function: sigmoid map + tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14); + if ((tableIndex < 16) && (tableIndex >= 0)) { + tmp16no2 = kIndicatorTable[tableIndex]; + tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; + frac = (int16_t)(tmp32no1 & 0x00003fff); // Q14 + tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14); + if (tmpIndFX == 0) { + tmpIndFX = 8192 - tmp16no2; // Q14 + } else { + tmpIndFX = 8192 + tmp16no2; // Q14 + } + } + indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14 + + //spectral flatness feature + if (inst->weightSpecFlat) { + tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10 + tmpIndFX = 16384; // Q14(1.0) + //use larger width in tanh map for pause regions + tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10 + nShifts = 4; + if (inst->thresholdSpecFlat < tmpU32no1) { + tmpIndFX = 0; + tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat; + //widthPrior = widthPrior * 2.0; + nShifts++; + } + tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, + nShifts), 25); + //Q14 + tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), + 25); //Q14 + // compute indicator function: sigmoid map + // FLOAT code + // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * + // (threshPrior1 - tmpFloat1)) + 1.0); + tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14); + if (tableIndex < 16) { + tmp16no2 = kIndicatorTable[tableIndex]; + tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; + frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14 + tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14); + if (tmpIndFX) { + tmpIndFX = 8192 + tmp16no2; // Q14 + } else { + tmpIndFX = 8192 - tmp16no2; // Q14 + } + } + indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14 + } + + //for template spectral-difference + if (inst->weightSpecDiff) { + tmpU32no1 = 0; + if (inst->featureSpecDiff) { + normTmp = WEBRTC_SPL_MIN(20 - inst->stages, + WebRtcSpl_NormU32(inst->featureSpecDiff)); + tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); + // Q(normTmp-2*stages) + tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, + 20 - inst->stages - normTmp); + if (tmpU32no2 > 0) { + // Q(20 - inst->stages) + tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); + } else { + tmpU32no1 = (uint32_t)(0x7fffffff); + } + } + tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, + 17), + 25); + tmpU32no2 = tmpU32no1 - tmpU32no3; + nShifts = 1; + tmpIndFX = 16384; // Q14(1.0) + //use larger width in tanh map for pause regions + if (tmpU32no2 & 0x80000000) { + tmpIndFX = 0; + tmpU32no2 = tmpU32no3 - tmpU32no1; + //widthPrior = widthPrior * 2.0; + nShifts--; + } + tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts); + // compute indicator function: sigmoid map + /* FLOAT code + indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0); + */ + tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14); + if (tableIndex < 16) { + tmp16no2 = kIndicatorTable[tableIndex]; + tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex]; + frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14 + tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( + tmp16no1, frac, 14); + if (tmpIndFX) { + tmpIndFX = 8192 + tmp16no2; + } else { + tmpIndFX = 8192 - tmp16no2; + } + } + indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14 + } + + //combine the indicator function with the feature weights + // FLOAT code + // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * + // indicator1 + weightIndPrior2 * indicator2); + indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14 + // done with computing indicator function + + //compute the prior probability + // FLOAT code + // inst->priorNonSpeechProb += PRIOR_UPDATE * + // (indPriorNonSpeech - inst->priorNonSpeechProb); + tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14 + inst->priorNonSpeechProb += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( + PRIOR_UPDATE_Q14, tmp16, 14); // Q14 + + //final speech probability: combine prior model with LR factor: + + memset(nonSpeechProbFinal, 0, sizeof(uint16_t) * inst->magnLen); + + if (inst->priorNonSpeechProb > 0) { + r0 = inst->priorNonSpeechProb; + r1 = 16384 - r0; + int32_t const_23637 = 23637; + int32_t const_44 = 44; + int32_t const_84 = 84; + int32_t const_1 = 1; + int32_t const_neg8 = -8; + for (i = 0; i < inst->magnLen; i++) { + r2 = inst->logLrtTimeAvgW32[i]; + if (r2 < 65300) { + __asm __volatile( + ".set push \n\t" + ".set noreorder \n\t" + "mul %[r2], %[r2], %[const_23637] \n\t" + "sll %[r6], %[r1], 16 \n\t" + "clz %[r7], %[r6] \n\t" + "clo %[r8], %[r6] \n\t" + "slt %[r9], %[r6], $0 \n\t" + "movn %[r7], %[r8], %[r9] \n\t" + "sra %[r2], %[r2], 14 \n\t" + "andi %[r3], %[r2], 0xfff \n\t" + "mul %[r4], %[r3], %[r3] \n\t" + "mul %[r3], %[r3], %[const_84] \n\t" + "sra %[r2], %[r2], 12 \n\t" + "slt %[r5], %[r2], %[const_neg8] \n\t" + "movn %[r2], %[const_neg8], %[r5] \n\t" + "mul %[r4], %[r4], %[const_44] \n\t" + "sra %[r3], %[r3], 7 \n\t" + "addiu %[r7], %[r7], -1 \n\t" + "slti %[r9], %[r7], 31 \n\t" + "movz %[r7], $0, %[r9] \n\t" + "sra %[r4], %[r4], 19 \n\t" + "addu %[r4], %[r4], %[r3] \n\t" + "addiu %[r3], %[r2], 8 \n\t" + "addiu %[r2], %[r2], -4 \n\t" + "neg %[r5], %[r2] \n\t" + "sllv %[r6], %[r4], %[r2] \n\t" + "srav %[r5], %[r4], %[r5] \n\t" + "slt %[r2], %[r2], $0 \n\t" + "movn %[r6], %[r5], %[r2] \n\t" + "sllv %[r3], %[const_1], %[r3] \n\t" + "addu %[r2], %[r3], %[r6] \n\t" + "clz %[r4], %[r2] \n\t" + "clo %[r5], %[r2] \n\t" + "slt %[r8], %[r2], $0 \n\t" + "movn %[r4], %[r5], %[r8] \n\t" + "addiu %[r4], %[r4], -1 \n\t" + "slt %[r5], $0, %[r2] \n\t" + "or %[r5], %[r5], %[r7] \n\t" + "movz %[r4], $0, %[r5] \n\t" + "addiu %[r6], %[r7], -7 \n\t" + "addu %[r6], %[r6], %[r4] \n\t" + "bltz %[r6], 1f \n\t" + " nop \n\t" + "addiu %[r4], %[r6], -8 \n\t" + "neg %[r3], %[r4] \n\t" + "srav %[r5], %[r2], %[r3] \n\t" + "mul %[r5], %[r5], %[r1] \n\t" + "mul %[r2], %[r2], %[r1] \n\t" + "slt %[r4], %[r4], $0 \n\t" + "srav %[r5], %[r5], %[r6] \n\t" + "sra %[r2], %[r2], 8 \n\t" + "movn %[r2], %[r5], %[r4] \n\t" + "sll %[r3], %[r0], 8 \n\t" + "addu %[r2], %[r0], %[r2] \n\t" + "divu %[r3], %[r3], %[r2] \n\t" + "1: \n\t" + ".set pop \n\t" + : [r2] "+r" (r2), [r3] "=&r" (r3), [r4] "=&r" (r4), + [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), + [r8] "=&r" (r8), [r9] "=&r" (r9) + : [r0] "r" (r0), [r1] "r" (r1), [const_23637] "r" (const_23637), + [const_neg8] "r" (const_neg8), [const_84] "r" (const_84), + [const_1] "r" (const_1), [const_44] "r" (const_44) + : "hi", "lo" + ); + nonSpeechProbFinal[i] = r3; + } + } + } +} + +// Update analysis buffer for lower band, and window data before FFT. +void WebRtcNsx_AnalysisUpdate_mips(NsxInst_t* inst, + int16_t* out, + int16_t* new_speech) { + + int iters, after; + int anaLen = inst->anaLen; + int *window = (int*)inst->window; + int *anaBuf = (int*)inst->analysisBuffer; + int *outBuf = (int*)out; + int r0, r1, r2, r3, r4, r5, r6, r7; +#if defined(MIPS_DSP_R1_LE) + int r8; +#endif + + // For lower band update analysis buffer. + WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer, + inst->analysisBuffer + inst->blockLen10ms, + inst->anaLen - inst->blockLen10ms); + WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer + + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms); + + // Window data before FFT. +#if defined(MIPS_DSP_R1_LE) + __asm __volatile( + ".set push \n\t" + ".set noreorder \n\t" + "sra %[iters], %[anaLen], 3 \n\t" + "1: \n\t" + "blez %[iters], 2f \n\t" + " nop \n\t" + "lw %[r0], 0(%[window]) \n\t" + "lw %[r1], 0(%[anaBuf]) \n\t" + "lw %[r2], 4(%[window]) \n\t" + "lw %[r3], 4(%[anaBuf]) \n\t" + "lw %[r4], 8(%[window]) \n\t" + "lw %[r5], 8(%[anaBuf]) \n\t" + "lw %[r6], 12(%[window]) \n\t" + "lw %[r7], 12(%[anaBuf]) \n\t" + "muleq_s.w.phl %[r8], %[r0], %[r1] \n\t" + "muleq_s.w.phr %[r0], %[r0], %[r1] \n\t" + "muleq_s.w.phl %[r1], %[r2], %[r3] \n\t" + "muleq_s.w.phr %[r2], %[r2], %[r3] \n\t" + "muleq_s.w.phl %[r3], %[r4], %[r5] \n\t" + "muleq_s.w.phr %[r4], %[r4], %[r5] \n\t" + "muleq_s.w.phl %[r5], %[r6], %[r7] \n\t" + "muleq_s.w.phr %[r6], %[r6], %[r7] \n\t" +#if defined(MIPS_DSP_R2_LE) + "precr_sra_r.ph.w %[r8], %[r0], 15 \n\t" + "precr_sra_r.ph.w %[r1], %[r2], 15 \n\t" + "precr_sra_r.ph.w %[r3], %[r4], 15 \n\t" + "precr_sra_r.ph.w %[r5], %[r6], 15 \n\t" + "sw %[r8], 0(%[outBuf]) \n\t" + "sw %[r1], 4(%[outBuf]) \n\t" + "sw %[r3], 8(%[outBuf]) \n\t" + "sw %[r5], 12(%[outBuf]) \n\t" +#else + "shra_r.w %[r8], %[r8], 15 \n\t" + "shra_r.w %[r0], %[r0], 15 \n\t" + "shra_r.w %[r1], %[r1], 15 \n\t" + "shra_r.w %[r2], %[r2], 15 \n\t" + "shra_r.w %[r3], %[r3], 15 \n\t" + "shra_r.w %[r4], %[r4], 15 \n\t" + "shra_r.w %[r5], %[r5], 15 \n\t" + "shra_r.w %[r6], %[r6], 15 \n\t" + "sll %[r0], %[r0], 16 \n\t" + "sll %[r2], %[r2], 16 \n\t" + "sll %[r4], %[r4], 16 \n\t" + "sll %[r6], %[r6], 16 \n\t" + "packrl.ph %[r0], %[r8], %[r0] \n\t" + "packrl.ph %[r2], %[r1], %[r2] \n\t" + "packrl.ph %[r4], %[r3], %[r4] \n\t" + "packrl.ph %[r6], %[r5], %[r6] \n\t" + "sw %[r0], 0(%[outBuf]) \n\t" + "sw %[r2], 4(%[outBuf]) \n\t" + "sw %[r4], 8(%[outBuf]) \n\t" + "sw %[r6], 12(%[outBuf]) \n\t" +#endif + "addiu %[window], %[window], 16 \n\t" + "addiu %[anaBuf], %[anaBuf], 16 \n\t" + "addiu %[outBuf], %[outBuf], 16 \n\t" + "b 1b \n\t" + " addiu %[iters], %[iters], -1 \n\t" + "2: \n\t" + "andi %[after], %[anaLen], 7 \n\t" + "3: \n\t" + "blez %[after], 4f \n\t" + " nop \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[anaBuf]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "addiu %[window], %[window], 2 \n\t" + "addiu %[anaBuf], %[anaBuf], 2 \n\t" + "addiu %[outBuf], %[outBuf], 2 \n\t" + "shra_r.w %[r0], %[r0], 14 \n\t" + "sh %[r0], -2(%[outBuf]) \n\t" + "b 3b \n\t" + " addiu %[after], %[after], -1 \n\t" + "4: \n\t" + ".set pop \n\t" + : [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), + [r3] "=&r" (r3), [r4] "=&r" (r4), [r5] "=&r" (r5), + [r6] "=&r" (r6), [r7] "=&r" (r7), [r8] "=&r" (r8), + [iters] "=&r" (iters), [after] "=&r" (after), + [window] "+r" (window),[anaBuf] "+r" (anaBuf), + [outBuf] "+r" (outBuf) + : [anaLen] "r" (anaLen) + : "memory", "hi", "lo" + ); +#else + __asm __volatile( + ".set push \n\t" + ".set noreorder \n\t" + "sra %[iters], %[anaLen], 2 \n\t" + "1: \n\t" + "blez %[iters], 2f \n\t" + " nop \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[anaBuf]) \n\t" + "lh %[r2], 2(%[window]) \n\t" + "lh %[r3], 2(%[anaBuf]) \n\t" + "lh %[r4], 4(%[window]) \n\t" + "lh %[r5], 4(%[anaBuf]) \n\t" + "lh %[r6], 6(%[window]) \n\t" + "lh %[r7], 6(%[anaBuf]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "mul %[r2], %[r2], %[r3] \n\t" + "mul %[r4], %[r4], %[r5] \n\t" + "mul %[r6], %[r6], %[r7] \n\t" + "addiu %[window], %[window], 8 \n\t" + "addiu %[anaBuf], %[anaBuf], 8 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "addiu %[r2], %[r2], 0x2000 \n\t" + "addiu %[r4], %[r4], 0x2000 \n\t" + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "sra %[r2], %[r2], 14 \n\t" + "sra %[r4], %[r4], 14 \n\t" + "sra %[r6], %[r6], 14 \n\t" + "sh %[r0], 0(%[outBuf]) \n\t" + "sh %[r2], 2(%[outBuf]) \n\t" + "sh %[r4], 4(%[outBuf]) \n\t" + "sh %[r6], 6(%[outBuf]) \n\t" + "addiu %[outBuf], %[outBuf], 8 \n\t" + "b 1b \n\t" + " addiu %[iters], %[iters], -1 \n\t" + "2: \n\t" + "andi %[after], %[anaLen], 3 \n\t" + "3: \n\t" + "blez %[after], 4f \n\t" + " nop \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[anaBuf]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "addiu %[window], %[window], 2 \n\t" + "addiu %[anaBuf], %[anaBuf], 2 \n\t" + "addiu %[outBuf], %[outBuf], 2 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "sh %[r0], -2(%[outBuf]) \n\t" + "b 3b \n\t" + " addiu %[after], %[after], -1 \n\t" + "4: \n\t" + ".set pop \n\t" + : [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), + [r3] "=&r" (r3), [r4] "=&r" (r4), [r5] "=&r" (r5), + [r6] "=&r" (r6), [r7] "=&r" (r7), [iters] "=&r" (iters), + [after] "=&r" (after), [window] "+r" (window), + [anaBuf] "+r" (anaBuf), [outBuf] "+r" (outBuf) + : [anaLen] "r" (anaLen) + : "memory", "hi", "lo" + ); +#endif +} + +// For the noise supression process, synthesis, read out fully processed +// segment, and update synthesis buffer. +void WebRtcNsx_SynthesisUpdate_mips(NsxInst_t* inst, + int16_t* out_frame, + int16_t gain_factor) { + + int iters = inst->blockLen10ms >> 2; + int after = inst->blockLen10ms & 3; + int r0, r1, r2, r3, r4, r5, r6, r7; + int16_t *window = (int16_t*)inst->window; + int16_t *real = inst->real; + int16_t *synthBuf = inst->synthesisBuffer; + int16_t *out = out_frame; + int sat_pos = 0x7fff; + int sat_neg = 0xffff8000; + int block10 = (int)inst->blockLen10ms; + int anaLen = (int)inst->anaLen; + + __asm __volatile( + ".set push \n\t" + ".set noreorder \n\t" + "1: \n\t" + "blez %[iters], 2f \n\t" + " nop \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[real]) \n\t" + "lh %[r2], 2(%[window]) \n\t" + "lh %[r3], 2(%[real]) \n\t" + "lh %[r4], 4(%[window]) \n\t" + "lh %[r5], 4(%[real]) \n\t" + "lh %[r6], 6(%[window]) \n\t" + "lh %[r7], 6(%[real]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "mul %[r2], %[r2], %[r3] \n\t" + "mul %[r4], %[r4], %[r5] \n\t" + "mul %[r6], %[r6], %[r7] \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "addiu %[r2], %[r2], 0x2000 \n\t" + "addiu %[r4], %[r4], 0x2000 \n\t" + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "sra %[r2], %[r2], 14 \n\t" + "sra %[r4], %[r4], 14 \n\t" + "sra %[r6], %[r6], 14 \n\t" + "mul %[r0], %[r0], %[gain_factor] \n\t" + "mul %[r2], %[r2], %[gain_factor] \n\t" + "mul %[r4], %[r4], %[gain_factor] \n\t" + "mul %[r6], %[r6], %[gain_factor] \n\t" + "addiu %[r0], %[r0], 0x1000 \n\t" + "addiu %[r2], %[r2], 0x1000 \n\t" + "addiu %[r4], %[r4], 0x1000 \n\t" + "addiu %[r6], %[r6], 0x1000 \n\t" + "sra %[r0], %[r0], 13 \n\t" + "sra %[r2], %[r2], 13 \n\t" + "sra %[r4], %[r4], 13 \n\t" + "sra %[r6], %[r6], 13 \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "slt %[r3], %[r2], %[sat_pos] \n\t" + "slt %[r5], %[r4], %[sat_pos] \n\t" + "slt %[r7], %[r6], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "movz %[r2], %[sat_pos], %[r3] \n\t" + "movz %[r4], %[sat_pos], %[r5] \n\t" + "movz %[r6], %[sat_pos], %[r7] \n\t" + "lh %[r1], 0(%[synthBuf]) \n\t" + "lh %[r3], 2(%[synthBuf]) \n\t" + "lh %[r5], 4(%[synthBuf]) \n\t" + "lh %[r7], 6(%[synthBuf]) \n\t" + "addu %[r0], %[r0], %[r1] \n\t" + "addu %[r2], %[r2], %[r3] \n\t" + "addu %[r4], %[r4], %[r5] \n\t" + "addu %[r6], %[r6], %[r7] \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "slt %[r3], %[r2], %[sat_pos] \n\t" + "slt %[r5], %[r4], %[sat_pos] \n\t" + "slt %[r7], %[r6], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "movz %[r2], %[sat_pos], %[r3] \n\t" + "movz %[r4], %[sat_pos], %[r5] \n\t" + "movz %[r6], %[sat_pos], %[r7] \n\t" + "slt %[r1], %[r0], %[sat_neg] \n\t" + "slt %[r3], %[r2], %[sat_neg] \n\t" + "slt %[r5], %[r4], %[sat_neg] \n\t" + "slt %[r7], %[r6], %[sat_neg] \n\t" + "movn %[r0], %[sat_neg], %[r1] \n\t" + "movn %[r2], %[sat_neg], %[r3] \n\t" + "movn %[r4], %[sat_neg], %[r5] \n\t" + "movn %[r6], %[sat_neg], %[r7] \n\t" + "sh %[r0], 0(%[synthBuf]) \n\t" + "sh %[r2], 2(%[synthBuf]) \n\t" + "sh %[r4], 4(%[synthBuf]) \n\t" + "sh %[r6], 6(%[synthBuf]) \n\t" + "sh %[r0], 0(%[out]) \n\t" + "sh %[r2], 2(%[out]) \n\t" + "sh %[r4], 4(%[out]) \n\t" + "sh %[r6], 6(%[out]) \n\t" + "addiu %[window], %[window], 8 \n\t" + "addiu %[real], %[real], 8 \n\t" + "addiu %[synthBuf],%[synthBuf], 8 \n\t" + "addiu %[out], %[out], 8 \n\t" + "b 1b \n\t" + " addiu %[iters], %[iters], -1 \n\t" + "2: \n\t" + "blez %[after], 3f \n\t" + " subu %[block10], %[anaLen], %[block10] \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[real]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "addiu %[window], %[window], 2 \n\t" + "addiu %[real], %[real], 2 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "mul %[r0], %[r0], %[gain_factor] \n\t" + "addiu %[r0], %[r0], 0x1000 \n\t" + "sra %[r0], %[r0], 13 \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "lh %[r1], 0(%[synthBuf]) \n\t" + "addu %[r0], %[r0], %[r1] \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "slt %[r1], %[r0], %[sat_neg] \n\t" + "movn %[r0], %[sat_neg], %[r1] \n\t" + "sh %[r0], 0(%[synthBuf]) \n\t" + "sh %[r0], 0(%[out]) \n\t" + "addiu %[synthBuf],%[synthBuf], 2 \n\t" + "addiu %[out], %[out], 2 \n\t" + "b 2b \n\t" + " addiu %[after], %[after], -1 \n\t" + "3: \n\t" + "sra %[iters], %[block10], 2 \n\t" + "4: \n\t" + "blez %[iters], 5f \n\t" + " andi %[after], %[block10], 3 \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[real]) \n\t" + "lh %[r2], 2(%[window]) \n\t" + "lh %[r3], 2(%[real]) \n\t" + "lh %[r4], 4(%[window]) \n\t" + "lh %[r5], 4(%[real]) \n\t" + "lh %[r6], 6(%[window]) \n\t" + "lh %[r7], 6(%[real]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "mul %[r2], %[r2], %[r3] \n\t" + "mul %[r4], %[r4], %[r5] \n\t" + "mul %[r6], %[r6], %[r7] \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "addiu %[r2], %[r2], 0x2000 \n\t" + "addiu %[r4], %[r4], 0x2000 \n\t" + "addiu %[r6], %[r6], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "sra %[r2], %[r2], 14 \n\t" + "sra %[r4], %[r4], 14 \n\t" + "sra %[r6], %[r6], 14 \n\t" + "mul %[r0], %[r0], %[gain_factor] \n\t" + "mul %[r2], %[r2], %[gain_factor] \n\t" + "mul %[r4], %[r4], %[gain_factor] \n\t" + "mul %[r6], %[r6], %[gain_factor] \n\t" + "addiu %[r0], %[r0], 0x1000 \n\t" + "addiu %[r2], %[r2], 0x1000 \n\t" + "addiu %[r4], %[r4], 0x1000 \n\t" + "addiu %[r6], %[r6], 0x1000 \n\t" + "sra %[r0], %[r0], 13 \n\t" + "sra %[r2], %[r2], 13 \n\t" + "sra %[r4], %[r4], 13 \n\t" + "sra %[r6], %[r6], 13 \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "slt %[r3], %[r2], %[sat_pos] \n\t" + "slt %[r5], %[r4], %[sat_pos] \n\t" + "slt %[r7], %[r6], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "movz %[r2], %[sat_pos], %[r3] \n\t" + "movz %[r4], %[sat_pos], %[r5] \n\t" + "movz %[r6], %[sat_pos], %[r7] \n\t" + "lh %[r1], 0(%[synthBuf]) \n\t" + "lh %[r3], 2(%[synthBuf]) \n\t" + "lh %[r5], 4(%[synthBuf]) \n\t" + "lh %[r7], 6(%[synthBuf]) \n\t" + "addu %[r0], %[r0], %[r1] \n\t" + "addu %[r2], %[r2], %[r3] \n\t" + "addu %[r4], %[r4], %[r5] \n\t" + "addu %[r6], %[r6], %[r7] \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "slt %[r3], %[r2], %[sat_pos] \n\t" + "slt %[r5], %[r4], %[sat_pos] \n\t" + "slt %[r7], %[r6], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "movz %[r2], %[sat_pos], %[r3] \n\t" + "movz %[r4], %[sat_pos], %[r5] \n\t" + "movz %[r6], %[sat_pos], %[r7] \n\t" + "slt %[r1], %[r0], %[sat_neg] \n\t" + "slt %[r3], %[r2], %[sat_neg] \n\t" + "slt %[r5], %[r4], %[sat_neg] \n\t" + "slt %[r7], %[r6], %[sat_neg] \n\t" + "movn %[r0], %[sat_neg], %[r1] \n\t" + "movn %[r2], %[sat_neg], %[r3] \n\t" + "movn %[r4], %[sat_neg], %[r5] \n\t" + "movn %[r6], %[sat_neg], %[r7] \n\t" + "sh %[r0], 0(%[synthBuf]) \n\t" + "sh %[r2], 2(%[synthBuf]) \n\t" + "sh %[r4], 4(%[synthBuf]) \n\t" + "sh %[r6], 6(%[synthBuf]) \n\t" + "addiu %[window], %[window], 8 \n\t" + "addiu %[real], %[real], 8 \n\t" + "addiu %[synthBuf],%[synthBuf], 8 \n\t" + "b 4b \n\t" + " addiu %[iters], %[iters], -1 \n\t" + "5: \n\t" + "blez %[after], 6f \n\t" + " nop \n\t" + "lh %[r0], 0(%[window]) \n\t" + "lh %[r1], 0(%[real]) \n\t" + "mul %[r0], %[r0], %[r1] \n\t" + "addiu %[window], %[window], 2 \n\t" + "addiu %[real], %[real], 2 \n\t" + "addiu %[r0], %[r0], 0x2000 \n\t" + "sra %[r0], %[r0], 14 \n\t" + "mul %[r0], %[r0], %[gain_factor] \n\t" + "addiu %[r0], %[r0], 0x1000 \n\t" + "sra %[r0], %[r0], 13 \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "lh %[r1], 0(%[synthBuf]) \n\t" + "addu %[r0], %[r0], %[r1] \n\t" + "slt %[r1], %[r0], %[sat_pos] \n\t" + "movz %[r0], %[sat_pos], %[r1] \n\t" + "slt %[r1], %[r0], %[sat_neg] \n\t" + "movn %[r0], %[sat_neg], %[r1] \n\t" + "sh %[r0], 0(%[synthBuf]) \n\t" + "addiu %[synthBuf],%[synthBuf], 2 \n\t" + "b 2b \n\t" + " addiu %[after], %[after], -1 \n\t" + "6: \n\t" + ".set pop \n\t" + : [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), + [r3] "=&r" (r3), [r4] "=&r" (r4), [r5] "=&r" (r5), + [r6] "=&r" (r6), [r7] "=&r" (r7), [iters] "+r" (iters), + [after] "+r" (after), [block10] "+r" (block10), + [window] "+r" (window), [real] "+r" (real), + [synthBuf] "+r" (synthBuf), [out] "+r" (out) + : [gain_factor] "r" (gain_factor), [sat_pos] "r" (sat_pos), + [sat_neg] "r" (sat_neg), [anaLen] "r" (anaLen) + : "memory", "hi", "lo" + ); + + // update synthesis buffer + WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer, + inst->synthesisBuffer + inst->blockLen10ms, + inst->anaLen - inst->blockLen10ms); + WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms); +} + +// Filter the data in the frequency domain, and create spectrum. +void WebRtcNsx_PrepareSpectrum_mips(NsxInst_t* inst, int16_t* freq_buf) { + + uint16_t *noiseSupFilter = inst->noiseSupFilter; + int16_t *real = inst->real; + int16_t *imag = inst->imag; + int32_t loop_count = 2; + int16_t tmp_1, tmp_2, tmp_3, tmp_4, tmp_5, tmp_6; + int16_t tmp16 = (inst->anaLen << 1) - 4; + int16_t* freq_buf_f = freq_buf; + int16_t* freq_buf_s = &freq_buf[tmp16]; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + //first sample + "lh %[tmp_1], 0(%[noiseSupFilter]) \n\t" + "lh %[tmp_2], 0(%[real]) \n\t" + "lh %[tmp_3], 0(%[imag]) \n\t" + "mul %[tmp_2], %[tmp_2], %[tmp_1] \n\t" + "mul %[tmp_3], %[tmp_3], %[tmp_1] \n\t" + "sra %[tmp_2], %[tmp_2], 14 \n\t" + "sra %[tmp_3], %[tmp_3], 14 \n\t" + "sh %[tmp_2], 0(%[real]) \n\t" + "sh %[tmp_3], 0(%[imag]) \n\t" + "negu %[tmp_3], %[tmp_3] \n\t" + "sh %[tmp_2], 0(%[freq_buf_f]) \n\t" + "sh %[tmp_3], 2(%[freq_buf_f]) \n\t" + "addiu %[real], %[real], 2 \n\t" + "addiu %[imag], %[imag], 2 \n\t" + "addiu %[noiseSupFilter], %[noiseSupFilter], 2 \n\t" + "addiu %[freq_buf_f], %[freq_buf_f], 4 \n\t" + "1: \n\t" + "lh %[tmp_1], 0(%[noiseSupFilter]) \n\t" + "lh %[tmp_2], 0(%[real]) \n\t" + "lh %[tmp_3], 0(%[imag]) \n\t" + "lh %[tmp_4], 2(%[noiseSupFilter]) \n\t" + "lh %[tmp_5], 2(%[real]) \n\t" + "lh %[tmp_6], 2(%[imag]) \n\t" + "mul %[tmp_2], %[tmp_2], %[tmp_1] \n\t" + "mul %[tmp_3], %[tmp_3], %[tmp_1] \n\t" + "mul %[tmp_5], %[tmp_5], %[tmp_4] \n\t" + "mul %[tmp_6], %[tmp_6], %[tmp_4] \n\t" + "addiu %[loop_count], %[loop_count], 2 \n\t" + "sra %[tmp_2], %[tmp_2], 14 \n\t" + "sra %[tmp_3], %[tmp_3], 14 \n\t" + "sra %[tmp_5], %[tmp_5], 14 \n\t" + "sra %[tmp_6], %[tmp_6], 14 \n\t" + "addiu %[noiseSupFilter], %[noiseSupFilter], 4 \n\t" + "sh %[tmp_2], 0(%[real]) \n\t" + "sh %[tmp_2], 4(%[freq_buf_s]) \n\t" + "sh %[tmp_3], 0(%[imag]) \n\t" + "sh %[tmp_3], 6(%[freq_buf_s]) \n\t" + "negu %[tmp_3], %[tmp_3] \n\t" + "sh %[tmp_5], 2(%[real]) \n\t" + "sh %[tmp_5], 0(%[freq_buf_s]) \n\t" + "sh %[tmp_6], 2(%[imag]) \n\t" + "sh %[tmp_6], 2(%[freq_buf_s]) \n\t" + "negu %[tmp_6], %[tmp_6] \n\t" + "addiu %[freq_buf_s], %[freq_buf_s], -8 \n\t" + "addiu %[real], %[real], 4 \n\t" + "addiu %[imag], %[imag], 4 \n\t" + "sh %[tmp_2], 0(%[freq_buf_f]) \n\t" + "sh %[tmp_3], 2(%[freq_buf_f]) \n\t" + "sh %[tmp_5], 4(%[freq_buf_f]) \n\t" + "sh %[tmp_6], 6(%[freq_buf_f]) \n\t" + "blt %[loop_count], %[loop_size], 1b \n\t" + " addiu %[freq_buf_f], %[freq_buf_f], 8 \n\t" + //last two samples: + "lh %[tmp_1], 0(%[noiseSupFilter]) \n\t" + "lh %[tmp_2], 0(%[real]) \n\t" + "lh %[tmp_3], 0(%[imag]) \n\t" + "lh %[tmp_4], 2(%[noiseSupFilter]) \n\t" + "lh %[tmp_5], 2(%[real]) \n\t" + "lh %[tmp_6], 2(%[imag]) \n\t" + "mul %[tmp_2], %[tmp_2], %[tmp_1] \n\t" + "mul %[tmp_3], %[tmp_3], %[tmp_1] \n\t" + "mul %[tmp_5], %[tmp_5], %[tmp_4] \n\t" + "mul %[tmp_6], %[tmp_6], %[tmp_4] \n\t" + "sra %[tmp_2], %[tmp_2], 14 \n\t" + "sra %[tmp_3], %[tmp_3], 14 \n\t" + "sra %[tmp_5], %[tmp_5], 14 \n\t" + "sra %[tmp_6], %[tmp_6], 14 \n\t" + "sh %[tmp_2], 0(%[real]) \n\t" + "sh %[tmp_2], 4(%[freq_buf_s]) \n\t" + "sh %[tmp_3], 0(%[imag]) \n\t" + "sh %[tmp_3], 6(%[freq_buf_s]) \n\t" + "negu %[tmp_3], %[tmp_3] \n\t" + "sh %[tmp_2], 0(%[freq_buf_f]) \n\t" + "sh %[tmp_3], 2(%[freq_buf_f]) \n\t" + "sh %[tmp_5], 4(%[freq_buf_f]) \n\t" + "sh %[tmp_6], 6(%[freq_buf_f]) \n\t" + "sh %[tmp_5], 2(%[real]) \n\t" + "sh %[tmp_6], 2(%[imag]) \n\t" + ".set pop \n\t" + : [real] "+r" (real), [imag] "+r" (imag), + [freq_buf_f] "+r" (freq_buf_f), [freq_buf_s] "+r" (freq_buf_s), + [loop_count] "+r" (loop_count), [noiseSupFilter] "+r" (noiseSupFilter), + [tmp_1] "=&r" (tmp_1), [tmp_2] "=&r" (tmp_2), [tmp_3] "=&r" (tmp_3), + [tmp_4] "=&r" (tmp_4), [tmp_5] "=&r" (tmp_5), [tmp_6] "=&r" (tmp_6) + : [loop_size] "r" (inst->anaLen2) + : "memory", "hi", "lo" + ); +} + +#if defined(MIPS_DSP_R1_LE) +// Denormalize the real-valued signal |in|, the output from inverse FFT. +void WebRtcNsx_Denormalize_mips(NsxInst_t* inst, int16_t* in, int factor) { + int32_t r0, r1, r2, r3, t0; + int len = inst->anaLen; + int16_t *out = &inst->real[0]; + int shift = factor - inst->normData; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "beqz %[len], 8f \n\t" + " nop \n\t" + "bltz %[shift], 4f \n\t" + " sra %[t0], %[len], 2 \n\t" + "beqz %[t0], 2f \n\t" + " andi %[len], %[len], 3 \n\t" + "1: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 2(%[in]) \n\t" + "lh %[r2], 4(%[in]) \n\t" + "lh %[r3], 6(%[in]) \n\t" + "shllv_s.ph %[r0], %[r0], %[shift] \n\t" + "shllv_s.ph %[r1], %[r1], %[shift] \n\t" + "shllv_s.ph %[r2], %[r2], %[shift] \n\t" + "shllv_s.ph %[r3], %[r3], %[shift] \n\t" + "addiu %[in], %[in], 8 \n\t" + "addiu %[t0], %[t0], -1 \n\t" + "sh %[r0], 0(%[out]) \n\t" + "sh %[r1], 2(%[out]) \n\t" + "sh %[r2], 4(%[out]) \n\t" + "sh %[r3], 6(%[out]) \n\t" + "bgtz %[t0], 1b \n\t" + " addiu %[out], %[out], 8 \n\t" + "2: \n\t" + "beqz %[len], 8f \n\t" + " nop \n\t" + "3: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "addiu %[in], %[in], 2 \n\t" + "addiu %[len], %[len], -1 \n\t" + "shllv_s.ph %[r0], %[r0], %[shift] \n\t" + "addiu %[out], %[out], 2 \n\t" + "bgtz %[len], 3b \n\t" + " sh %[r0], -2(%[out]) \n\t" + "b 8f \n\t" + "4: \n\t" + "negu %[shift], %[shift] \n\t" + "beqz %[t0], 6f \n\t" + " andi %[len], %[len], 3 \n\t" + "5: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 2(%[in]) \n\t" + "lh %[r2], 4(%[in]) \n\t" + "lh %[r3], 6(%[in]) \n\t" + "srav %[r0], %[r0], %[shift] \n\t" + "srav %[r1], %[r1], %[shift] \n\t" + "srav %[r2], %[r2], %[shift] \n\t" + "srav %[r3], %[r3], %[shift] \n\t" + "addiu %[in], %[in], 8 \n\t" + "addiu %[t0], %[t0], -1 \n\t" + "sh %[r0], 0(%[out]) \n\t" + "sh %[r1], 2(%[out]) \n\t" + "sh %[r2], 4(%[out]) \n\t" + "sh %[r3], 6(%[out]) \n\t" + "bgtz %[t0], 5b \n\t" + " addiu %[out], %[out], 8 \n\t" + "6: \n\t" + "beqz %[len], 8f \n\t" + " nop \n\t" + "7: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "addiu %[in], %[in], 2 \n\t" + "addiu %[len], %[len], -1 \n\t" + "srav %[r0], %[r0], %[shift] \n\t" + "addiu %[out], %[out], 2 \n\t" + "bgtz %[len], 7b \n\t" + " sh %[r0], -2(%[out]) \n\t" + "8: \n\t" + ".set pop \n\t" + : [t0] "=&r" (t0), [r0] "=&r" (r0), [r1] "=&r" (r1), + [r2] "=&r" (r2), [r3] "=&r" (r3) + : [len] "r" (len), [shift] "r" (shift), [in] "r" (in), + [out] "r" (out) + : "memory" + ); +} +#endif + +// Normalize the real-valued signal |in|, the input to forward FFT. +void WebRtcNsx_NormalizeRealBuffer_mips(NsxInst_t* inst, + const int16_t* in, + int16_t* out) { + int32_t r0, r1, r2, r3, t0; + int len = inst->anaLen; + int shift = inst->normData; + + __asm __volatile ( + ".set push \n\t" + ".set noreorder \n\t" + "beqz %[len], 4f \n\t" + " sra %[t0], %[len], 2 \n\t" + "beqz %[t0], 2f \n\t" + " andi %[len], %[len], 3 \n\t" + "1: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "lh %[r1], 2(%[in]) \n\t" + "lh %[r2], 4(%[in]) \n\t" + "lh %[r3], 6(%[in]) \n\t" + "sllv %[r0], %[r0], %[shift] \n\t" + "sllv %[r1], %[r1], %[shift] \n\t" + "sllv %[r2], %[r2], %[shift] \n\t" + "sllv %[r3], %[r3], %[shift] \n\t" + "addiu %[in], %[in], 8 \n\t" + "addiu %[t0], %[t0], -1 \n\t" + "sh %[r0], 0(%[out]) \n\t" + "sh %[r1], 2(%[out]) \n\t" + "sh %[r2], 4(%[out]) \n\t" + "sh %[r3], 6(%[out]) \n\t" + "bgtz %[t0], 1b \n\t" + " addiu %[out], %[out], 8 \n\t" + "2: \n\t" + "beqz %[len], 4f \n\t" + " nop \n\t" + "3: \n\t" + "lh %[r0], 0(%[in]) \n\t" + "addiu %[in], %[in], 2 \n\t" + "addiu %[len], %[len], -1 \n\t" + "sllv %[r0], %[r0], %[shift] \n\t" + "addiu %[out], %[out], 2 \n\t" + "bgtz %[len], 3b \n\t" + " sh %[r0], -2(%[out]) \n\t" + "4: \n\t" + ".set pop \n\t" + : [t0] "=&r" (t0), [r0] "=&r" (r0), [r1] "=&r" (r1), + [r2] "=&r" (r2), [r3] "=&r" (r3) + : [len] "r" (len), [shift] "r" (shift), [in] "r" (in), + [out] "r" (out) + : "memory" + ); +} + diff --git a/chromium/third_party/webrtc/modules/audio_processing/processing_component.cc b/chromium/third_party/webrtc/modules/audio_processing/processing_component.cc index 23bf22570b1..9e16d7c4eea 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/processing_component.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/processing_component.cc @@ -12,15 +12,12 @@ #include <assert.h> -#include "webrtc/modules/audio_processing/audio_processing_impl.h" +#include "webrtc/modules/audio_processing/include/audio_processing.h" namespace webrtc { -ProcessingComponent::ProcessingComponent() {} - -ProcessingComponent::ProcessingComponent(const AudioProcessingImpl* apm) - : apm_(apm), - initialized_(false), +ProcessingComponent::ProcessingComponent() + : initialized_(false), enabled_(false), num_handles_(0) {} @@ -35,7 +32,7 @@ int ProcessingComponent::Destroy() { } initialized_ = false; - return apm_->kNoError; + return AudioProcessing::kNoError; } int ProcessingComponent::EnableComponent(bool enable) { @@ -43,7 +40,7 @@ int ProcessingComponent::EnableComponent(bool enable) { enabled_ = enable; // Must be set before Initialize() is called. int err = Initialize(); - if (err != apm_->kNoError) { + if (err != AudioProcessing::kNoError) { enabled_ = false; return err; } @@ -51,7 +48,7 @@ int ProcessingComponent::EnableComponent(bool enable) { enabled_ = enable; } - return apm_->kNoError; + return AudioProcessing::kNoError; } bool ProcessingComponent::is_component_enabled() const { @@ -69,7 +66,7 @@ int ProcessingComponent::num_handles() const { int ProcessingComponent::Initialize() { if (!enabled_) { - return apm_->kNoError; + return AudioProcessing::kNoError; } num_handles_ = num_handles_required(); @@ -82,12 +79,12 @@ int ProcessingComponent::Initialize() { if (handles_[i] == NULL) { handles_[i] = CreateHandle(); if (handles_[i] == NULL) { - return apm_->kCreationFailedError; + return AudioProcessing::kCreationFailedError; } } int err = InitializeHandle(handles_[i]); - if (err != apm_->kNoError) { + if (err != AudioProcessing::kNoError) { return GetHandleError(handles_[i]); } } @@ -98,17 +95,17 @@ int ProcessingComponent::Initialize() { int ProcessingComponent::Configure() { if (!initialized_) { - return apm_->kNoError; + return AudioProcessing::kNoError; } assert(static_cast<int>(handles_.size()) >= num_handles_); for (int i = 0; i < num_handles_; i++) { int err = ConfigureHandle(handles_[i]); - if (err != apm_->kNoError) { + if (err != AudioProcessing::kNoError) { return GetHandleError(handles_[i]); } } - return apm_->kNoError; + return AudioProcessing::kNoError; } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/processing_component.h b/chromium/third_party/webrtc/modules/audio_processing/processing_component.h index c090d222456..8ee3ac6c7db 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/processing_component.h +++ b/chromium/third_party/webrtc/modules/audio_processing/processing_component.h @@ -13,16 +13,13 @@ #include <vector> -#include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/common.h" namespace webrtc { -class AudioProcessingImpl; - class ProcessingComponent { public: ProcessingComponent(); - explicit ProcessingComponent(const AudioProcessingImpl* apm); virtual ~ProcessingComponent(); virtual int Initialize(); @@ -41,11 +38,10 @@ class ProcessingComponent { virtual void* CreateHandle() const = 0; virtual int InitializeHandle(void* handle) const = 0; virtual int ConfigureHandle(void* handle) const = 0; - virtual int DestroyHandle(void* handle) const = 0; + virtual void DestroyHandle(void* handle) const = 0; virtual int num_handles_required() const = 0; virtual int GetHandleError(void* handle) const = 0; - const AudioProcessingImpl* apm_; std::vector<void*> handles_; bool initialized_; bool enabled_; diff --git a/chromium/third_party/webrtc/modules/audio_processing/rms_level.cc b/chromium/third_party/webrtc/modules/audio_processing/rms_level.cc new file mode 100644 index 00000000000..14136bf3049 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/rms_level.cc @@ -0,0 +1,61 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_processing/rms_level.h" + +#include <assert.h> +#include <math.h> + +namespace webrtc { + +static const float kMaxSquaredLevel = 32768 * 32768; + +RMSLevel::RMSLevel() + : sum_square_(0), + sample_count_(0) {} + +RMSLevel::~RMSLevel() {} + +void RMSLevel::Reset() { + sum_square_ = 0; + sample_count_ = 0; +} + +void RMSLevel::Process(const int16_t* data, int length) { + for (int i = 0; i < length; ++i) { + sum_square_ += data[i] * data[i]; + } + sample_count_ += length; +} + +void RMSLevel::ProcessMuted(int length) { + sample_count_ += length; +} + +int RMSLevel::RMS() { + if (sample_count_ == 0 || sum_square_ == 0) { + Reset(); + return kMinLevel; + } + + // Normalize by the max level. + float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel); + // 20log_10(x^0.5) = 10log_10(x) + rms = 10 * log10(rms); + assert(rms <= 0); + if (rms < -kMinLevel) + rms = -kMinLevel; + + rms = -rms; + Reset(); + return static_cast<int>(rms + 0.5); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/rms_level.h b/chromium/third_party/webrtc/modules/audio_processing/rms_level.h new file mode 100644 index 00000000000..055d271bb19 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/rms_level.h @@ -0,0 +1,57 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ + +#include "webrtc/typedefs.h" + +namespace webrtc { + +// Computes the root mean square (RMS) level in dBFs (decibels from digital +// full-scale) of audio data. The computation follows RFC 6465: +// https://tools.ietf.org/html/rfc6465 +// with the intent that it can provide the RTP audio level indication. +// +// The expected approach is to provide constant-sized chunks of audio to +// Process(). When enough chunks have been accumulated to form a packet, call +// RMS() to get the audio level indicator for the RTP header. +class RMSLevel { + public: + static const int kMinLevel = 127; + + RMSLevel(); + ~RMSLevel(); + + // Can be called to reset internal states, but is not required during normal + // operation. + void Reset(); + + // Pass each chunk of audio to Process() to accumulate the level. + void Process(const int16_t* data, int length); + + // If all samples with the given |length| have a magnitude of zero, this is + // a shortcut to avoid some computation. + void ProcessMuted(int length); + + // Computes the RMS level over all data passed to Process() since the last + // call to RMS(). The returned value is positive but should be interpreted as + // negative as per the RFC. It is constrained to [0, 127]. + int RMS(); + + private: + float sum_square_; + int sample_count_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ + diff --git a/chromium/third_party/webrtc/modules/audio_processing/splitting_filter.cc b/chromium/third_party/webrtc/modules/audio_processing/splitting_filter.cc deleted file mode 100644 index 372c8dc426e..00000000000 --- a/chromium/third_party/webrtc/modules/audio_processing/splitting_filter.cc +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_processing/splitting_filter.h" - -namespace webrtc { - -void SplittingFilterAnalysis(const int16_t* in_data, - int16_t* low_band, - int16_t* high_band, - int32_t* filter_state1, - int32_t* filter_state2) -{ - WebRtcSpl_AnalysisQMF(in_data, low_band, high_band, filter_state1, filter_state2); -} - -void SplittingFilterSynthesis(const int16_t* low_band, - const int16_t* high_band, - int16_t* out_data, - int32_t* filt_state1, - int32_t* filt_state2) -{ - WebRtcSpl_SynthesisQMF(low_band, high_band, out_data, filt_state1, filt_state2); -} -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/splitting_filter.h b/chromium/third_party/webrtc/modules/audio_processing/splitting_filter.h deleted file mode 100644 index b6c851273a6..00000000000 --- a/chromium/third_party/webrtc/modules/audio_processing/splitting_filter.h +++ /dev/null @@ -1,63 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_ - -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/typedefs.h" - -namespace webrtc { -/* - * SplittingFilterbank_analysisQMF(...) - * - * Splits a super-wb signal into two subbands: 0-8 kHz and 8-16 kHz. - * - * Input: - * - in_data : super-wb audio signal - * - * Input & Output: - * - filt_state1: Filter state for first all-pass filter - * - filt_state2: Filter state for second all-pass filter - * - * Output: - * - low_band : The signal from the 0-4 kHz band - * - high_band : The signal from the 4-8 kHz band - */ -void SplittingFilterAnalysis(const int16_t* in_data, - int16_t* low_band, - int16_t* high_band, - int32_t* filt_state1, - int32_t* filt_state2); - -/* - * SplittingFilterbank_synthesisQMF(...) - * - * Combines the two subbands (0-8 and 8-16 kHz) into a super-wb signal. - * - * Input: - * - low_band : The signal with the 0-8 kHz band - * - high_band : The signal with the 8-16 kHz band - * - * Input & Output: - * - filt_state1: Filter state for first all-pass filter - * - filt_state2: Filter state for second all-pass filter - * - * Output: - * - out_data : super-wb speech signal - */ -void SplittingFilterSynthesis(const int16_t* low_band, - const int16_t* high_band, - int16_t* out_data, - int32_t* filt_state1, - int32_t* filt_state2); -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/typing_detection.cc b/chromium/third_party/webrtc/modules/audio_processing/typing_detection.cc new file mode 100644 index 00000000000..5f5ce0abafd --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/typing_detection.cc @@ -0,0 +1,90 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_processing/typing_detection.h" + +namespace webrtc { + +TypingDetection::TypingDetection() + : time_active_(0), + time_since_last_typing_(0), + penalty_counter_(0), + counter_since_last_detection_update_(0), + detection_to_report_(false), + new_detection_to_report_(false), + time_window_(10), + cost_per_typing_(100), + reporting_threshold_(300), + penalty_decay_(1), + type_event_delay_(2), + report_detection_update_period_(1) { +} + +TypingDetection::~TypingDetection() {} + +bool TypingDetection::Process(bool key_pressed, bool vad_activity) { + if (vad_activity) + time_active_++; + else + time_active_ = 0; + + // Keep track if time since last typing event + if (key_pressed) + time_since_last_typing_ = 0; + else + ++time_since_last_typing_; + + if (time_since_last_typing_ < type_event_delay_ && + vad_activity && + time_active_ < time_window_) { + penalty_counter_ += cost_per_typing_; + if (penalty_counter_ > reporting_threshold_) + new_detection_to_report_ = true; + } + + if (penalty_counter_ > 0) + penalty_counter_ -= penalty_decay_; + + if (++counter_since_last_detection_update_ == + report_detection_update_period_) { + detection_to_report_ = new_detection_to_report_; + new_detection_to_report_ = false; + counter_since_last_detection_update_ = 0; + } + + return detection_to_report_; +} + +int TypingDetection::TimeSinceLastDetectionInSeconds() { + // Round to whole seconds. + return (time_since_last_typing_ + 50) / 100; +} + +void TypingDetection::SetParameters(int time_window, + int cost_per_typing, + int reporting_threshold, + int penalty_decay, + int type_event_delay, + int report_detection_update_period) { + if (time_window) time_window_ = time_window; + + if (cost_per_typing) cost_per_typing_ = cost_per_typing; + + if (reporting_threshold) reporting_threshold_ = reporting_threshold; + + if (penalty_decay) penalty_decay_ = penalty_decay; + + if (type_event_delay) type_event_delay_ = type_event_delay; + + if (report_detection_update_period) + report_detection_update_period_ = report_detection_update_period; +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/audio_processing/typing_detection.h b/chromium/third_party/webrtc/modules/audio_processing/typing_detection.h new file mode 100644 index 00000000000..5fa6456e9e9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_processing/typing_detection.h @@ -0,0 +1,93 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_ + +#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class TypingDetection { + public: + TypingDetection(); + virtual ~TypingDetection(); + + // Run the detection algortihm. Shall be called every 10 ms. Returns true if + // typing is detected, or false if not, based on the update period as set with + // SetParameters(). See |report_detection_update_period_| description below. + bool Process(bool key_pressed, bool vad_activity); + + // Gets the time in seconds since the last detection. + int TimeSinceLastDetectionInSeconds(); + + // Sets the algorithm parameters. A parameter value of 0 leaves it unchanged. + // See the correspondning member variables below for descriptions. + void SetParameters(int time_window, + int cost_per_typing, + int reporting_threshold, + int penalty_decay, + int type_event_delay, + int report_detection_update_period); + + private: + int time_active_; + int time_since_last_typing_; + int penalty_counter_; + + // Counter since last time the detection status reported by Process() was + // updated. See also |report_detection_update_period_|. + int counter_since_last_detection_update_; + + // The detection status to report. Updated every + // |report_detection_update_period_| call to Process(). + bool detection_to_report_; + + // What |detection_to_report_| should be set to next time it is updated. + bool new_detection_to_report_; + + // Settable threshold values. + + // Number of 10 ms slots accepted to count as a hit. + int time_window_; + + // Penalty added for a typing + activity coincide. + int cost_per_typing_; + + // Threshold for |penalty_counter_|. + int reporting_threshold_; + + // How much we reduce |penalty_counter_| every 10 ms. + int penalty_decay_; + + // How old typing events we allow. + int type_event_delay_; + + // Settable update period. + + // Number of 10 ms slots between each update of the detection status returned + // by Process(). This inertia added to the algorithm is usually desirable and + // provided so that consumers of the class don't have to implement that + // themselves if they don't wish. + // If set to 1, each call to Process() will return the detection status for + // that 10 ms slot. + // If set to N (where N > 1), the detection status returned from Process() + // will remain the same until Process() has been called N times. Then, if none + // of the last N calls to Process() has detected typing for each respective + // 10 ms slot, Process() will return false. If at least one of the last N + // calls has detected typing, Process() will return true. And that returned + // status will then remain the same until the next N calls have been done. + int report_detection_update_period_; +}; + +} // namespace webrtc + +#endif // #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.c b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.c index 6d6e9bc97bb..3b2043267ff 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.c +++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.c @@ -30,10 +30,6 @@ static const float kMinHistogramThreshold = 1.5f; static const int kMinRequiredHits = 10; static const int kMaxHitsWhenPossiblyNonCausal = 10; static const int kMaxHitsWhenPossiblyCausal = 1000; -// TODO(bjornv): Make kMaxDelayDifference a configurable parameter, since it -// corresponds to the filter length if the delay estimation is used in echo -// control. -static const int kMaxDelayDifference = 32; static const float kQ14Scaling = 1.f / (1 << 14); // Scaling by 2^14 to get Q0. static const float kFractionSlope = 0.05f; static const float kMinFractionWhenPossiblyCausal = 0.5f; @@ -195,8 +191,8 @@ static int HistogramBasedValidation(const BinaryDelayEstimator* self, // depending on the distance between the |candidate_delay| and |last_delay|. // TODO(bjornv): How much can we gain by turning the fraction calculation // into tables? - if (delay_difference >= kMaxDelayDifference) { - fraction = 1.f - kFractionSlope * (delay_difference - kMaxDelayDifference); + if (delay_difference > self->allowed_offset) { + fraction = 1.f - kFractionSlope * (delay_difference - self->allowed_offset); fraction = (fraction > kMinFractionWhenPossiblyCausal ? fraction : kMinFractionWhenPossiblyCausal); } else if (delay_difference < 0) { @@ -308,6 +304,39 @@ void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) { memset(self->far_bit_counts, 0, sizeof(int) * self->history_size); } +void WebRtc_SoftResetBinaryDelayEstimatorFarend( + BinaryDelayEstimatorFarend* self, int delay_shift) { + int abs_shift = abs(delay_shift); + int shift_size = 0; + int dest_index = 0; + int src_index = 0; + int padding_index = 0; + + assert(self != NULL); + shift_size = self->history_size - abs_shift; + assert(shift_size > 0); + if (delay_shift == 0) { + return; + } else if (delay_shift > 0) { + dest_index = abs_shift; + } else if (delay_shift < 0) { + src_index = abs_shift; + padding_index = shift_size; + } + + // Shift and zero pad buffers. + memmove(&self->binary_far_history[dest_index], + &self->binary_far_history[src_index], + sizeof(*self->binary_far_history) * shift_size); + memset(&self->binary_far_history[padding_index], 0, + sizeof(*self->binary_far_history) * abs_shift); + memmove(&self->far_bit_counts[dest_index], + &self->far_bit_counts[src_index], + sizeof(*self->far_bit_counts) * shift_size); + memset(&self->far_bit_counts[padding_index], 0, + sizeof(*self->far_bit_counts) * abs_shift); +} + void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* handle, uint32_t binary_far_spectrum) { assert(handle != NULL); @@ -349,10 +378,10 @@ void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) { } BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( - BinaryDelayEstimatorFarend* farend, int lookahead) { + BinaryDelayEstimatorFarend* farend, int max_lookahead) { BinaryDelayEstimator* self = NULL; - if ((farend != NULL) && (lookahead >= 0)) { + if ((farend != NULL) && (max_lookahead >= 0)) { // Sanity conditions fulfilled. self = malloc(sizeof(BinaryDelayEstimator)); } @@ -361,7 +390,11 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( int malloc_fail = 0; self->farend = farend; - self->near_history_size = lookahead + 1; + self->near_history_size = max_lookahead + 1; + self->robust_validation_enabled = 0; // Disabled by default. + self->allowed_offset = 0; + + self->lookahead = max_lookahead; // Allocate memory for spectrum buffers. The extra array element in // |mean_bit_counts| and |histogram| is a dummy element only used while @@ -374,7 +407,7 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( malloc_fail |= (self->bit_counts == NULL); // Allocate memory for history buffers. - self->binary_near_history = malloc((lookahead + 1) * sizeof(uint32_t)); + self->binary_near_history = malloc((max_lookahead + 1) * sizeof(uint32_t)); malloc_fail |= (self->binary_near_history == NULL); self->histogram = malloc((farend->history_size + 1) * sizeof(float)); @@ -400,26 +433,40 @@ void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self) { self->mean_bit_counts[i] = (20 << 9); // 20 in Q9. self->histogram[i] = 0.f; } - self->minimum_probability = (32 << 9); // 32 in Q9. - self->last_delay_probability = (32 << 9); // 32 in Q9. + self->minimum_probability = kMaxBitCountsQ9; // 32 in Q9. + self->last_delay_probability = (int) kMaxBitCountsQ9; // 32 in Q9. // Default return value if we're unable to estimate. -1 is used for errors. self->last_delay = -2; - self->robust_validation_enabled = 0; // Disabled by default. self->last_candidate_delay = -2; self->compare_delay = self->farend->history_size; self->candidate_hits = 0; self->last_delay_histogram = 0.f; } +int WebRtc_SoftResetBinaryDelayEstimator(BinaryDelayEstimator* self, + int delay_shift) { + int lookahead = 0; + assert(self != NULL); + lookahead = self->lookahead; + self->lookahead -= delay_shift; + if (self->lookahead < 0) { + self->lookahead = 0; + } + if (self->lookahead > self->near_history_size - 1) { + self->lookahead = self->near_history_size - 1; + } + return lookahead - self->lookahead; +} + int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self, uint32_t binary_near_spectrum) { int i = 0; int candidate_delay = -1; int valid_candidate = 0; - int32_t value_best_candidate = 32 << 9; // 32 in Q9, (max |mean_bit_counts|). + int32_t value_best_candidate = kMaxBitCountsQ9; int32_t value_worst_candidate = 0; int32_t valley_depth = 0; @@ -430,8 +477,7 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self, memmove(&(self->binary_near_history[1]), &(self->binary_near_history[0]), (self->near_history_size - 1) * sizeof(uint32_t)); self->binary_near_history[0] = binary_near_spectrum; - binary_near_spectrum = - self->binary_near_history[self->near_history_size - 1]; + binary_near_spectrum = self->binary_near_history[self->lookahead]; } // Compare with delayed spectra and store the |bit_counts| for each delay. @@ -547,21 +593,23 @@ int WebRtc_binary_last_delay(BinaryDelayEstimator* self) { return self->last_delay; } -int WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self) { - int delay_quality = 0; +float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self) { + float quality = 0; assert(self != NULL); - // |last_delay_probability| is the opposite of quality and states how deep the - // minimum of the cost function is. The value states how many non-matching - // bits we have between the binary spectra for the corresponding delay - // estimate. The range is thus from 0 to 32, since we use 32 bits in the - // binary spectra. - - // Return the |delay_quality| = 1 - |last_delay_probability| / 32 (in Q14). - delay_quality = (32 << 9) - self->last_delay_probability; - if (delay_quality < 0) { - delay_quality = 0; + + if (self->robust_validation_enabled) { + // Simply a linear function of the histogram height at delay estimate. + quality = self->histogram[self->compare_delay] / kHistogramMax; + } else { + // Note that |last_delay_probability| states how deep the minimum of the + // cost function is, so it is rather an error probability. + quality = (float) (kMaxBitCountsQ9 - self->last_delay_probability) / + kMaxBitCountsQ9; + if (quality < 0) { + quality = 0; + } } - return delay_quality; + return quality; } void WebRtc_MeanEstimatorFix(int32_t new_value, diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.h b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.h index 7ffb81b8b18..3d5ffce20e9 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.h +++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.h @@ -16,6 +16,8 @@ #include "webrtc/typedefs.h" +static const int32_t kMaxBitCountsQ9 = (32 << 9); // 32 matching bits in Q9. + typedef struct { // Pointer to bit counts. int* far_bit_counts; @@ -44,12 +46,16 @@ typedef struct { // Robust validation int robust_validation_enabled; + int allowed_offset; int last_candidate_delay; int compare_delay; int candidate_hits; float* histogram; float last_delay_histogram; + // For dynamically changing the lookahead when using SoftReset...(). + int lookahead; + // Far-end binary spectrum history buffer etc. BinaryDelayEstimatorFarend* farend; } BinaryDelayEstimator; @@ -90,6 +96,15 @@ BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend( // void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self); +// Soft resets the delay estimation far-end instance created with +// WebRtc_CreateBinaryDelayEstimatorFarend(...). +// +// Input: +// - delay_shift : The amount of blocks to shift history buffers. +// +void WebRtc_SoftResetBinaryDelayEstimatorFarend( + BinaryDelayEstimatorFarend* self, int delay_shift); + // Adds the binary far-end spectrum to the internal far-end history buffer. This // spectrum is used as reference when calculating the delay using // WebRtc_ProcessBinarySpectrum(). @@ -121,38 +136,10 @@ void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self); // Allocates the memory needed by the binary delay estimation. The memory needs // to be initialized separately through WebRtc_InitBinaryDelayEstimator(...). // -// Inputs: -// - farend : Pointer to the far-end part of the Binary Delay -// Estimator. This memory has to be created separately -// prior to this call using -// WebRtc_CreateBinaryDelayEstimatorFarend(). -// -// Note that BinaryDelayEstimator does not take -// ownership of |farend|. -// -// - lookahead : Amount of non-causal lookahead to use. This can -// detect cases in which a near-end signal occurs before -// the corresponding far-end signal. It will delay the -// estimate for the current block by an equal amount, -// and the returned values will be offset by it. -// -// A value of zero is the typical no-lookahead case. -// This also represents the minimum delay which can be -// estimated. -// -// Note that the effective range of delay estimates is -// [-|lookahead|,... ,|history_size|-|lookahead|) -// where |history_size| was set upon creating the far-end -// history buffer size. -// -// Return value: -// - BinaryDelayEstimator* -// : Created |handle|. If the memory can't be allocated -// or if any of the input parameters are invalid NULL -// is returned. -// +// See WebRtc_CreateDelayEstimator(..) in delay_estimator_wrapper.c for detailed +// description. BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( - BinaryDelayEstimatorFarend* farend, int lookahead); + BinaryDelayEstimatorFarend* farend, int max_lookahead); // Initializes the delay estimation instance created with // WebRtc_CreateBinaryDelayEstimator(...). @@ -165,6 +152,18 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( // void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self); +// Soft resets the delay estimation instance created with +// WebRtc_CreateBinaryDelayEstimator(...). +// +// Input: +// - delay_shift : The amount of blocks to shift history buffers. +// +// Return value: +// - actual_shifts : The actual number of shifts performed. +// +int WebRtc_SoftResetBinaryDelayEstimator(BinaryDelayEstimator* self, + int delay_shift); + // Estimates and returns the delay between the binary far-end and binary near- // end spectra. It is assumed the binary far-end spectrum has been added using // WebRtc_AddBinaryFarSpectrum() prior to this call. The value will be offset by @@ -199,17 +198,12 @@ int WebRtc_binary_last_delay(BinaryDelayEstimator* self); // Returns the estimation quality of the last calculated delay updated by the // function WebRtc_ProcessBinarySpectrum(...). The estimation quality is a value -// in the interval [0, 1] in Q14. The higher the value, the better quality. -// -// Input: -// - self : Pointer to the delay estimation instance. +// in the interval [0, 1]. The higher the value, the better the quality. // // Return value: -// - delay_quality : >= 0 - Estimation quality (in Q14) of last -// calculated delay value. -// -2 - Insufficient data for estimation. -// -int WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self); +// - delay_quality : >= 0 - Estimation quality of last calculated +// delay value. +float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self); // Updates the |mean_value| recursively with a step size of 2^-|factor|. This // function is used internally in the Binary Delay Estimator as well as the diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc index bdc199cafbb..ca0901d6db2 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc @@ -26,6 +26,9 @@ enum { kLookahead = 10 }; // Length of binary spectrum sequence. enum { kSequenceLength = 400 }; +const int kEnable[] = { 0, 1 }; +const size_t kSizeEnable = sizeof(kEnable) / sizeof(*kEnable); + class DelayEstimatorTest : public ::testing::Test { protected: DelayEstimatorTest(); @@ -38,7 +41,8 @@ class DelayEstimatorTest : public ::testing::Test { void RunBinarySpectra(BinaryDelayEstimator* binary1, BinaryDelayEstimator* binary2, int near_offset, int lookahead_offset, int far_offset); - void RunBinarySpectraTest(int near_offset, int lookahead_offset); + void RunBinarySpectraTest(int near_offset, int lookahead_offset, + int ref_robust_validation, int robust_validation); void* handle_; DelayEstimator* self_; @@ -113,7 +117,7 @@ void DelayEstimatorTest::Init() { EXPECT_EQ(0, farend_self_->far_spectrum_initialized); EXPECT_EQ(0, self_->near_spectrum_initialized); EXPECT_EQ(-2, WebRtc_last_delay(handle_)); // Delay in initial state. - EXPECT_EQ(0, WebRtc_last_delay_quality(handle_)); // Zero quality. + EXPECT_FLOAT_EQ(0, WebRtc_last_delay_quality(handle_)); // Zero quality. } void DelayEstimatorTest::InitBinary() { @@ -143,6 +147,8 @@ void DelayEstimatorTest::RunBinarySpectra(BinaryDelayEstimator* binary1, int near_offset, int lookahead_offset, int far_offset) { + int different_validations = binary1->robust_validation_enabled ^ + binary2->robust_validation_enabled; WebRtc_InitBinaryDelayEstimatorFarend(binary_farend_); WebRtc_InitBinaryDelayEstimator(binary1); WebRtc_InitBinaryDelayEstimator(binary2); @@ -167,19 +173,32 @@ void DelayEstimatorTest::RunBinarySpectra(BinaryDelayEstimator* binary1, if ((delay_1 != -2) && (delay_2 != -2)) { EXPECT_EQ(delay_1, delay_2 - lookahead_offset - near_offset); } + // For the case of identical signals |delay_1| and |delay_2| should match + // all the time, unless one of them has robust validation turned on. In + // that case the robust validation leaves the initial state faster. if ((near_offset == 0) && (lookahead_offset == 0)) { - EXPECT_EQ(delay_1, delay_2); + if (!different_validations) { + EXPECT_EQ(delay_1, delay_2); + } else { + if (binary1->robust_validation_enabled) { + EXPECT_GE(delay_1, delay_2); + } else { + EXPECT_GE(delay_2, delay_1); + } + } } } // Verify that we have left the initialized state. EXPECT_NE(-2, WebRtc_binary_last_delay(binary1)); - EXPECT_NE(0, WebRtc_binary_last_delay_quality(binary1)); + EXPECT_LT(0, WebRtc_binary_last_delay_quality(binary1)); EXPECT_NE(-2, WebRtc_binary_last_delay(binary2)); - EXPECT_NE(0, WebRtc_binary_last_delay_quality(binary2)); + EXPECT_LT(0, WebRtc_binary_last_delay_quality(binary2)); } void DelayEstimatorTest::RunBinarySpectraTest(int near_offset, - int lookahead_offset) { + int lookahead_offset, + int ref_robust_validation, + int robust_validation) { BinaryDelayEstimator* binary2 = WebRtc_CreateBinaryDelayEstimator(binary_farend_, kLookahead + lookahead_offset); @@ -187,6 +206,8 @@ void DelayEstimatorTest::RunBinarySpectraTest(int near_offset, // the delay is equivalent with a positive |offset| of the far-end sequence. // For non-causal systems the delay is equivalent with a negative |offset| of // the far-end sequence. + binary_->robust_validation_enabled = ref_robust_validation; + binary2->robust_validation_enabled = robust_validation; for (int offset = -kLookahead; offset < kMaxDelay - lookahead_offset - near_offset; offset++) { @@ -194,6 +215,7 @@ void DelayEstimatorTest::RunBinarySpectraTest(int near_offset, } WebRtc_FreeBinaryDelayEstimator(binary2); binary2 = NULL; + binary_->robust_validation_enabled = 0; // Reset reference. } TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { @@ -206,14 +228,12 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { void* handle = farend_handle_; handle = WebRtc_CreateDelayEstimatorFarend(33, kMaxDelay + kLookahead); EXPECT_TRUE(handle == NULL); - handle = farend_handle_; handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, 1); EXPECT_TRUE(handle == NULL); handle = handle_; handle = WebRtc_CreateDelayEstimator(NULL, kLookahead); EXPECT_TRUE(handle == NULL); - handle = handle_; handle = WebRtc_CreateDelayEstimator(farend_handle_, -1); EXPECT_TRUE(handle == NULL); @@ -246,10 +266,18 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, spectrum_size_, 16)); + // WebRtc_set_allowed_offset() should return -1 if we have: + // 1) NULL pointer as |handle|. + // 2) |allowed_offset| < 0. + EXPECT_EQ(-1, WebRtc_set_allowed_offset(NULL, 0)); + EXPECT_EQ(-1, WebRtc_set_allowed_offset(handle_, -1)); + + EXPECT_EQ(-1, WebRtc_get_allowed_offset(NULL)); + // WebRtc_enable_robust_validation() should return -1 if we have: // 1) NULL pointer as |handle|. // 2) Incorrect |enable| value (not 0 or 1). - EXPECT_EQ(-1, WebRtc_enable_robust_validation(NULL, 0)); + EXPECT_EQ(-1, WebRtc_enable_robust_validation(NULL, kEnable[0])); EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, -1)); EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, 2)); @@ -286,21 +314,31 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // WebRtc_last_delay() should return -1 if we have a NULL pointer as |handle|. EXPECT_EQ(-1, WebRtc_last_delay(NULL)); - // WebRtc_last_delay_quality() should return -1 if we have a NULL pointer as - // |handle|. - EXPECT_EQ(-1, WebRtc_last_delay_quality(NULL)); - // Free any local memory if needed. WebRtc_FreeDelayEstimator(handle); } +TEST_F(DelayEstimatorTest, VerifyAllowedOffset) { + // Is set to zero by default. + EXPECT_EQ(0, WebRtc_get_allowed_offset(handle_)); + for (int i = 1; i >= 0; i--) { + EXPECT_EQ(0, WebRtc_set_allowed_offset(handle_, i)); + EXPECT_EQ(i, WebRtc_get_allowed_offset(handle_)); + Init(); + // Unaffected over a reset. + EXPECT_EQ(i, WebRtc_get_allowed_offset(handle_)); + } +} + TEST_F(DelayEstimatorTest, VerifyEnableRobustValidation) { - Init(); // Disabled by default. EXPECT_EQ(0, WebRtc_is_robust_validation_enabled(handle_)); - for (int i = 1; i >= 0; i--) { - EXPECT_EQ(0, WebRtc_enable_robust_validation(handle_, i)); - EXPECT_EQ(i, WebRtc_is_robust_validation_enabled(handle_)); + for (size_t i = 0; i < kSizeEnable; ++i) { + EXPECT_EQ(0, WebRtc_enable_robust_validation(handle_, kEnable[i])); + EXPECT_EQ(kEnable[i], WebRtc_is_robust_validation_enabled(handle_)); + Init(); + // Unaffected over a reset. + EXPECT_EQ(kEnable[i], WebRtc_is_robust_validation_enabled(handle_)); } } @@ -335,6 +373,7 @@ TEST_F(DelayEstimatorTest, CorrectLastDelay) { // (|last_delay| = -2). Then we compare the Process() output with the // last_delay() call. + // TODO(bjornv): Update quality values for robust validation. int last_delay = 0; // Floating point operations. Init(); @@ -345,13 +384,16 @@ TEST_F(DelayEstimatorTest, CorrectLastDelay) { spectrum_size_); if (last_delay != -2) { EXPECT_EQ(last_delay, WebRtc_last_delay(handle_)); - EXPECT_EQ(7203, WebRtc_last_delay_quality(handle_)); + if (!WebRtc_is_robust_validation_enabled(handle_)) { + EXPECT_FLOAT_EQ(7203.f / kMaxBitCountsQ9, + WebRtc_last_delay_quality(handle_)); + } break; } } // Verify that we have left the initialized state. EXPECT_NE(-2, WebRtc_last_delay(handle_)); - EXPECT_NE(0, WebRtc_last_delay_quality(handle_)); + EXPECT_LT(0, WebRtc_last_delay_quality(handle_)); // Fixed point operations. Init(); @@ -362,13 +404,16 @@ TEST_F(DelayEstimatorTest, CorrectLastDelay) { spectrum_size_, 0); if (last_delay != -2) { EXPECT_EQ(last_delay, WebRtc_last_delay(handle_)); - EXPECT_EQ(7203, WebRtc_last_delay_quality(handle_)); + if (!WebRtc_is_robust_validation_enabled(handle_)) { + EXPECT_FLOAT_EQ(7203.f / kMaxBitCountsQ9, + WebRtc_last_delay_quality(handle_)); + } break; } } // Verify that we have left the initialized state. EXPECT_NE(-2, WebRtc_last_delay(handle_)); - EXPECT_NE(0, WebRtc_last_delay_quality(handle_)); + EXPECT_LT(0, WebRtc_last_delay_quality(handle_)); } TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimatorFarend) { @@ -391,18 +436,14 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimator) { BinaryDelayEstimator* binary_handle = binary_; // WebRtc_CreateBinaryDelayEstimator() should return -1 if we have a NULL - // pointer as |binary_handle| or invalid input values. Upon failure, the + // pointer as |binary_farend| or invalid input values. Upon failure, the // |binary_handle| should be NULL. // Make sure we have a non-NULL value at start, so we can detect NULL after // create failure. binary_handle = WebRtc_CreateBinaryDelayEstimator(NULL, kLookahead); EXPECT_TRUE(binary_handle == NULL); - binary_handle = binary_; binary_handle = WebRtc_CreateBinaryDelayEstimator(binary_farend_, -1); EXPECT_TRUE(binary_handle == NULL); - binary_handle = binary_; - binary_handle = WebRtc_CreateBinaryDelayEstimator(0, 0); - EXPECT_TRUE(binary_handle == NULL); } TEST_F(DelayEstimatorTest, MeanEstimatorFix) { @@ -431,26 +472,70 @@ TEST_F(DelayEstimatorTest, ExactDelayEstimateMultipleNearSameSpectrum) { // the signal accordingly. We create two Binary Delay Estimators and feed them // with the same signals, so they should output the same results. // We verify both causal and non-causal delays. + // For these noise free signals, the robust validation should not have an + // impact, hence we turn robust validation on/off for both reference and + // delayed near end. - RunBinarySpectraTest(0, 0); + for (size_t i = 0; i < kSizeEnable; ++i) { + for (size_t j = 0; j < kSizeEnable; ++j) { + RunBinarySpectraTest(0, 0, kEnable[i], kEnable[j]); + } + } } TEST_F(DelayEstimatorTest, ExactDelayEstimateMultipleNearDifferentSpectrum) { // In this test we use the same setup as above, but we now feed the two Binary // Delay Estimators with different signals, so they should output different // results. + // For these noise free signals, the robust validation should not have an + // impact, hence we turn robust validation on/off for both reference and + // delayed near end. const int kNearOffset = 1; - RunBinarySpectraTest(kNearOffset, 0); + for (size_t i = 0; i < kSizeEnable; ++i) { + for (size_t j = 0; j < kSizeEnable; ++j) { + RunBinarySpectraTest(kNearOffset, 0, kEnable[i], kEnable[j]); + } + } } TEST_F(DelayEstimatorTest, ExactDelayEstimateMultipleNearDifferentLookahead) { // In this test we use the same setup as above, feeding the two Binary // Delay Estimators with the same signals. The difference is that we create // them with different lookahead. + // For these noise free signals, the robust validation should not have an + // impact, hence we turn robust validation on/off for both reference and + // delayed near end. const int kLookaheadOffset = 1; - RunBinarySpectraTest(0, kLookaheadOffset); + for (size_t i = 0; i < kSizeEnable; ++i) { + for (size_t j = 0; j < kSizeEnable; ++j) { + RunBinarySpectraTest(0, kLookaheadOffset, kEnable[i], kEnable[j]); + } + } } +TEST_F(DelayEstimatorTest, AllowedOffsetNoImpactWhenRobustValidationDisabled) { + // The same setup as in ExactDelayEstimateMultipleNearSameSpectrum with the + // difference that |allowed_offset| is set for the reference binary delay + // estimator. + + binary_->allowed_offset = 10; + RunBinarySpectraTest(0, 0, 0, 0); + binary_->allowed_offset = 0; // Reset reference. +} + +TEST_F(DelayEstimatorTest, VerifyLookaheadAtCreate) { + void* farend_handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, + kMaxDelay); + ASSERT_TRUE(farend_handle != NULL); + void* handle = WebRtc_CreateDelayEstimator(farend_handle, kLookahead); + ASSERT_TRUE(handle != NULL); + EXPECT_EQ(kLookahead, WebRtc_lookahead(handle)); + WebRtc_FreeDelayEstimator(handle); + WebRtc_FreeDelayEstimatorFarend(farend_handle); +} + +// TODO(bjornv): Add tests for SoftReset...(...). + } // namespace diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c index ce4431844ee..6ec894e65e9 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c +++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c @@ -191,6 +191,12 @@ int WebRtc_InitDelayEstimatorFarend(void* handle) { return 0; } +void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift) { + DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle; + assert(self != NULL); + WebRtc_SoftResetBinaryDelayEstimatorFarend(self->binary_farend, delay_shift); +} + int WebRtc_AddFarSpectrumFix(void* handle, uint16_t* far_spectrum, int spectrum_size, int far_q) { DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle; @@ -261,7 +267,7 @@ void WebRtc_FreeDelayEstimator(void* handle) { free(self); } -void* WebRtc_CreateDelayEstimator(void* farend_handle, int lookahead) { +void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) { DelayEstimator* self = NULL; DelayEstimatorFarend* farend = (DelayEstimatorFarend*) farend_handle; @@ -274,7 +280,7 @@ void* WebRtc_CreateDelayEstimator(void* farend_handle, int lookahead) { // Allocate memory for the farend spectrum handling. self->binary_handle = - WebRtc_CreateBinaryDelayEstimator(farend->binary_farend, lookahead); + WebRtc_CreateBinaryDelayEstimator(farend->binary_farend, max_lookahead); memory_fail |= (self->binary_handle == NULL); // Allocate memory for spectrum buffers. @@ -312,6 +318,50 @@ int WebRtc_InitDelayEstimator(void* handle) { return 0; } +int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) { + DelayEstimator* self = (DelayEstimator*) handle; + assert(self != NULL); + return WebRtc_SoftResetBinaryDelayEstimator(self->binary_handle, delay_shift); +} + +int WebRtc_set_lookahead(void* handle, int lookahead) { + DelayEstimator* self = (DelayEstimator*) handle; + assert(self != NULL); + assert(self->binary_handle != NULL); + if ((lookahead > self->binary_handle->near_history_size - 1) || + (lookahead < 0)) { + return -1; + } + self->binary_handle->lookahead = lookahead; + return self->binary_handle->lookahead; +} + +int WebRtc_lookahead(void* handle) { + DelayEstimator* self = (DelayEstimator*) handle; + assert(self != NULL); + assert(self->binary_handle != NULL); + return self->binary_handle->lookahead; +} + +int WebRtc_set_allowed_offset(void* handle, int allowed_offset) { + DelayEstimator* self = (DelayEstimator*) handle; + + if ((self == NULL) || (allowed_offset < 0)) { + return -1; + } + self->binary_handle->allowed_offset = allowed_offset; + return 0; +} + +int WebRtc_get_allowed_offset(const void* handle) { + const DelayEstimator* self = (const DelayEstimator*) handle; + + if (self == NULL) { + return -1; + } + return self->binary_handle->allowed_offset; +} + int WebRtc_enable_robust_validation(void* handle, int enable) { DelayEstimator* self = (DelayEstimator*) handle; @@ -326,13 +376,12 @@ int WebRtc_enable_robust_validation(void* handle, int enable) { return 0; } -int WebRtc_is_robust_validation_enabled(void* handle) { - DelayEstimator* self = (DelayEstimator*) handle; +int WebRtc_is_robust_validation_enabled(const void* handle) { + const DelayEstimator* self = (const DelayEstimator*) handle; if (self == NULL) { return -1; } - assert(self->binary_handle != NULL); return self->binary_handle->robust_validation_enabled; } @@ -403,12 +452,8 @@ int WebRtc_last_delay(void* handle) { return WebRtc_binary_last_delay(self->binary_handle); } -int WebRtc_last_delay_quality(void* handle) { +float WebRtc_last_delay_quality(void* handle) { DelayEstimator* self = (DelayEstimator*) handle; - - if (self == NULL) { - return -1; - } - + assert(self != NULL); return WebRtc_binary_last_delay_quality(self->binary_handle); } diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h index 50bcddeddc2..13e86bdd438 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h +++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h @@ -52,6 +52,13 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size); // int WebRtc_InitDelayEstimatorFarend(void* handle); +// Soft resets the far-end part of the delay estimation instance returned by +// WebRtc_CreateDelayEstimatorFarend(...). +// Input: +// - delay_shift : The amount of blocks to shift history buffers. +// +void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift); + // Adds the far-end spectrum to the far-end history buffer. This spectrum is // used as reference when calculating the delay using // WebRtc_ProcessSpectrum(). @@ -91,11 +98,18 @@ void WebRtc_FreeDelayEstimator(void* handle); // ownership of |farend_handle|, which has to be torn // down properly after this instance. // -// - lookahead : Amount of non-causal lookahead to use. This can -// detect cases in which a near-end signal occurs before -// the corresponding far-end signal. It will delay the -// estimate for the current block by an equal amount, -// and the returned values will be offset by it. +// - max_lookahead : Maximum amount of non-causal lookahead allowed. The +// actual amount of lookahead used can be controlled by +// WebRtc_set_lookahead(...). The default |lookahead| is +// set to |max_lookahead| at create time. Use +// WebRtc_set_lookahead(...) before start if a different +// value is desired. +// +// Using lookahead can detect cases in which a near-end +// signal occurs before the corresponding far-end signal. +// It will delay the estimate for the current block by an +// equal amount, and the returned values will be offset +// by it. // // A value of zero is the typical no-lookahead case. // This also represents the minimum delay which can be @@ -111,7 +125,7 @@ void WebRtc_FreeDelayEstimator(void* handle); // if any of the input parameters are invalid NULL is // returned. // -void* WebRtc_CreateDelayEstimator(void* farend_handle, int lookahead); +void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead); // Initializes the delay estimation instance returned by // WebRtc_CreateDelayEstimator(...) @@ -123,17 +137,59 @@ void* WebRtc_CreateDelayEstimator(void* farend_handle, int lookahead); // int WebRtc_InitDelayEstimator(void* handle); +// Soft resets the delay estimation instance returned by +// WebRtc_CreateDelayEstimator(...) +// Input: +// - delay_shift : The amount of blocks to shift history buffers. +// +// Return value: +// - actual_shifts : The actual number of shifts performed. +// +int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift); + +// Sets the amount of |lookahead| to use. Valid values are [0, max_lookahead] +// where |max_lookahead| was set at create time through +// WebRtc_CreateDelayEstimator(...). +// +// Input: +// - lookahead : The amount of blocks to shift history buffers. +// +// Return value: +// - new_lookahead : The actual number of shifts performed. +// +int WebRtc_set_lookahead(void* handle, int lookahead); + +// Returns the amount of lookahead we currently use. +int WebRtc_lookahead(void* handle); + +// Sets the |allowed_offset| used in the robust validation scheme. If the +// delay estimator is used in an echo control component, this parameter is +// related to the filter length. In principle |allowed_offset| should be set to +// the echo control filter length minus the expected echo duration, i.e., the +// delay offset the echo control can handle without quality regression. The +// default value, used if not set manually, is zero. Note that |allowed_offset| +// has to be non-negative. +// Inputs: +// - handle : Pointer to the delay estimation instance. +// - allowed_offset : The amount of delay offset, measured in partitions, +// the echo control filter can handle. +int WebRtc_set_allowed_offset(void* handle, int allowed_offset); + +// Returns the |allowed_offset| in number of partitions. +int WebRtc_get_allowed_offset(const void* handle); + // TODO(bjornv): Implement this functionality. Currently, enabling it has no // impact, hence this is an empty API. // Enables/Disables a robust validation functionality in the delay estimation. -// This is by default disabled upon initialization. +// This is by default set to disabled at create time. The state is preserved +// over a reset. // Inputs: // - handle : Pointer to the delay estimation instance. // - enable : Enable (1) or disable (0) this feature. int WebRtc_enable_robust_validation(void* handle, int enable); // Returns 1 if robust validation is enabled and 0 if disabled. -int WebRtc_is_robust_validation_enabled(void* handle); +int WebRtc_is_robust_validation_enabled(const void* handle); // Estimates and returns the delay between the far-end and near-end blocks. The // value will be offset by the lookahead (i.e. the lookahead should be @@ -179,18 +235,11 @@ int WebRtc_last_delay(void* handle); // Returns the estimation quality/probability of the last calculated delay // updated by the function WebRtc_DelayEstimatorProcess(...). The estimation -// quality is a value in the interval [0, 1] in Q9. The higher the value, the -// better quality. -// -// Input: -// - handle : Pointer to the delay estimation instance. +// quality is a value in the interval [0, 1]. The higher the value, the better +// the quality. // // Return value: -// - delay_quality : >= 0 - Estimation quality (in Q9) of last calculated -// delay value. -// -1 - Error. -// -2 - Insufficient data for estimation. -// -int WebRtc_last_delay_quality(void* handle); +// - delay_quality : >= 0 - Estimation quality of last calculated delay. +float WebRtc_last_delay_quality(void* handle); #endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_ diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc index 2b7634dd073..5dacf0b804c 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc @@ -22,7 +22,12 @@ extern "C" { namespace webrtc { -typedef scoped_ptr_malloc<RingBuffer, WebRtc_FreeBuffer> scoped_ring_buffer; +struct FreeBufferDeleter { + inline void operator()(void* ptr) const { + WebRtc_FreeBuffer(ptr); + } +}; +typedef scoped_ptr<RingBuffer, FreeBufferDeleter> scoped_ring_buffer; static void AssertElementEq(int expected, int actual) { ASSERT_EQ(expected, actual); @@ -56,8 +61,8 @@ static void RandomStressTest(int** data_ptr) { srand(seed); for (int i = 0; i < kNumTests; i++) { const int buffer_size = std::max(rand() % kMaxBufferSize, 1); - scoped_array<int> write_data(new int[buffer_size]); - scoped_array<int> read_data(new int[buffer_size]); + scoped_ptr<int[]> write_data(new int[buffer_size]); + scoped_ptr<int[]> read_data(new int[buffer_size]); scoped_ring_buffer buffer(WebRtc_CreateBuffer(buffer_size, sizeof(int))); ASSERT_TRUE(buffer.get() != NULL); ASSERT_EQ(0, WebRtc_InitBuffer(buffer.get())); diff --git a/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc index d41547c8022..c6e497ffa3e 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc +++ b/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc @@ -13,10 +13,8 @@ #include <assert.h> #include "webrtc/common_audio/vad/include/webrtc_vad.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" - #include "webrtc/modules/audio_processing/audio_buffer.h" -#include "webrtc/modules/audio_processing/audio_processing_impl.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" namespace webrtc { @@ -39,9 +37,11 @@ int MapSetting(VoiceDetection::Likelihood likelihood) { } } // namespace -VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessingImpl* apm) - : ProcessingComponent(apm), +VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessing* apm, + CriticalSectionWrapper* crit) + : ProcessingComponent(), apm_(apm), + crit_(crit), stream_has_voice_(false), using_external_vad_(false), likelihood_(kLowLikelihood), @@ -61,7 +61,7 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { } assert(audio->samples_per_split_channel() <= 160); - int16_t* mixed_data = audio->low_pass_split_data(0); + const int16_t* mixed_data = audio->low_pass_split_data(0); if (audio->num_channels() > 1) { audio->CopyAndMixLowPass(1); mixed_data = audio->mixed_low_pass_data(0); @@ -70,7 +70,7 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { // TODO(ajm): concatenate data in frame buffer here. int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)), - apm_->split_sample_rate_hz(), + apm_->proc_split_sample_rate_hz(), mixed_data, frame_size_samples_); if (vad_ret == 0) { @@ -87,7 +87,7 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { } int VoiceDetectionImpl::Enable(bool enable) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); return EnableComponent(enable); } @@ -108,7 +108,7 @@ bool VoiceDetectionImpl::stream_has_voice() const { } int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); if (MapSetting(likelihood) == -1) { return apm_->kBadParameterError; } @@ -122,7 +122,7 @@ VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const { } int VoiceDetectionImpl::set_frame_size_ms(int size) { - CriticalSectionScoped crit_scoped(apm_->crit()); + CriticalSectionScoped crit_scoped(crit_); assert(size == 10); // TODO(ajm): remove when supported. if (size != 10 && size != 20 && @@ -146,7 +146,8 @@ int VoiceDetectionImpl::Initialize() { } using_external_vad_ = false; - frame_size_samples_ = frame_size_ms_ * (apm_->split_sample_rate_hz() / 1000); + frame_size_samples_ = frame_size_ms_ * + apm_->proc_split_sample_rate_hz() / 1000; // TODO(ajm): intialize frame buffer here. return apm_->kNoError; @@ -163,8 +164,8 @@ void* VoiceDetectionImpl::CreateHandle() const { return handle; } -int VoiceDetectionImpl::DestroyHandle(void* handle) const { - return WebRtcVad_Free(static_cast<Handle*>(handle)); +void VoiceDetectionImpl::DestroyHandle(void* handle) const { + WebRtcVad_Free(static_cast<Handle*>(handle)); } int VoiceDetectionImpl::InitializeHandle(void* handle) const { diff --git a/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.h b/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.h index f8f50e8493c..1dfdf20ae92 100644 --- a/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.h +++ b/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.h @@ -15,13 +15,14 @@ #include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { -class AudioProcessingImpl; + class AudioBuffer; +class CriticalSectionWrapper; class VoiceDetectionImpl : public VoiceDetection, public ProcessingComponent { public: - explicit VoiceDetectionImpl(const AudioProcessingImpl* apm); + VoiceDetectionImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit); virtual ~VoiceDetectionImpl(); int ProcessCaptureAudio(AudioBuffer* audio); @@ -46,11 +47,12 @@ class VoiceDetectionImpl : public VoiceDetection, virtual void* CreateHandle() const OVERRIDE; virtual int InitializeHandle(void* handle) const OVERRIDE; virtual int ConfigureHandle(void* handle) const OVERRIDE; - virtual int DestroyHandle(void* handle) const OVERRIDE; + virtual void DestroyHandle(void* handle) const OVERRIDE; virtual int num_handles_required() const OVERRIDE; virtual int GetHandleError(void* handle) const OVERRIDE; - const AudioProcessingImpl* apm_; + const AudioProcessing* apm_; + CriticalSectionWrapper* crit_; bool stream_has_voice_; bool using_external_vad_; Likelihood likelihood_; diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/OWNERS b/chromium/third_party/webrtc/modules/bitrate_controller/OWNERS index 6c7028550b6..cce3a26bc5e 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/OWNERS +++ b/chromium/third_party/webrtc/modules/bitrate_controller/OWNERS @@ -3,3 +3,8 @@ stefan@webrtc.org henrik.lundin@webrtc.org
mflodman@webrtc.org
asapersson@webrtc.org
+ +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc index c2c8616559b..cff5dd18545 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc +++ b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc @@ -18,7 +18,8 @@ namespace webrtc { -class RtcpBandwidthObserverImpl : public RtcpBandwidthObserver { +class BitrateControllerImpl::RtcpBandwidthObserverImpl + : public RtcpBandwidthObserver { public: explicit RtcpBandwidthObserverImpl(BitrateControllerImpl* owner) : owner_(owner) { @@ -76,94 +77,29 @@ class RtcpBandwidthObserverImpl : public RtcpBandwidthObserver { BitrateControllerImpl* owner_; }; -class LowRateStrategy { - public: - LowRateStrategy( - SendSideBandwidthEstimation* bandwidth_estimation, - BitrateControllerImpl::BitrateObserverConfList* bitrate_observers) - : bandwidth_estimation_(bandwidth_estimation), - bitrate_observers_(bitrate_observers) {} - - virtual ~LowRateStrategy() {} - - virtual void LowRateAllocation(uint32_t bitrate, - uint8_t fraction_loss, - uint32_t rtt, - uint32_t sum_min_bitrates) = 0; - - protected: - SendSideBandwidthEstimation* bandwidth_estimation_; - BitrateControllerImpl::BitrateObserverConfList* bitrate_observers_; -}; - -class EnforceMinRateStrategy : public LowRateStrategy { - public: - EnforceMinRateStrategy( - SendSideBandwidthEstimation* bandwidth_estimation, - BitrateControllerImpl::BitrateObserverConfList* bitrate_observers) - : LowRateStrategy(bandwidth_estimation, bitrate_observers) {} - - void LowRateAllocation(uint32_t bitrate, - uint8_t fraction_loss, - uint32_t rtt, - uint32_t sum_min_bitrates) { - // Min bitrate to all observers. - BitrateControllerImpl::BitrateObserverConfList::iterator it; - for (it = bitrate_observers_->begin(); it != bitrate_observers_->end(); - ++it) { - it->first->OnNetworkChanged(it->second->min_bitrate_, fraction_loss, - rtt); - } - // Set sum of min to current send bitrate. - bandwidth_estimation_->SetSendBitrate(sum_min_bitrates); - } -}; - -class NoEnforceMinRateStrategy : public LowRateStrategy { - public: - NoEnforceMinRateStrategy( - SendSideBandwidthEstimation* bandwidth_estimation, - BitrateControllerImpl::BitrateObserverConfList* bitrate_observers) - : LowRateStrategy(bandwidth_estimation, bitrate_observers) {} - - void LowRateAllocation(uint32_t bitrate, - uint8_t fraction_loss, - uint32_t rtt, - uint32_t sum_min_bitrates) { - // Allocate up to |min_bitrate_| to one observer at a time, until - // |bitrate| is depleted. - uint32_t remainder = bitrate; - BitrateControllerImpl::BitrateObserverConfList::iterator it; - for (it = bitrate_observers_->begin(); it != bitrate_observers_->end(); - ++it) { - uint32_t allocation = std::min(remainder, it->second->min_bitrate_); - it->first->OnNetworkChanged(allocation, fraction_loss, rtt); - remainder -= allocation; - } - // Set |bitrate| to current send bitrate. - bandwidth_estimation_->SetSendBitrate(bitrate); - } -}; - BitrateController* BitrateController::CreateBitrateController( + Clock* clock, bool enforce_min_bitrate) { - return new BitrateControllerImpl(enforce_min_bitrate); + return new BitrateControllerImpl(clock, enforce_min_bitrate); } -BitrateControllerImpl::BitrateControllerImpl(bool enforce_min_bitrate) - : critsect_(CriticalSectionWrapper::CreateCriticalSection()) { - if (enforce_min_bitrate) { - low_rate_strategy_.reset(new EnforceMinRateStrategy( - &bandwidth_estimation_, &bitrate_observers_)); - } else { - low_rate_strategy_.reset(new NoEnforceMinRateStrategy( - &bandwidth_estimation_, &bitrate_observers_)); - } -} +BitrateControllerImpl::BitrateControllerImpl(Clock* clock, bool enforce_min_bitrate) + : clock_(clock), + last_bitrate_update_ms_(clock_->TimeInMilliseconds()), + critsect_(CriticalSectionWrapper::CreateCriticalSection()), + bandwidth_estimation_(), + bitrate_observers_(), + enforce_min_bitrate_(enforce_min_bitrate), + reserved_bitrate_bps_(0), + last_bitrate_bps_(0), + last_fraction_loss_(0), + last_rtt_ms_(0), + last_enforce_min_bitrate_(!enforce_min_bitrate_), + bitrate_observers_modified_(false), + last_reserved_bitrate_bps_(0) {} BitrateControllerImpl::~BitrateControllerImpl() { - BitrateObserverConfList::iterator it = - bitrate_observers_.begin(); + BitrateObserverConfList::iterator it = bitrate_observers_.begin(); while (it != bitrate_observers_.end()) { delete it->second; bitrate_observers_.erase(it); @@ -203,26 +139,56 @@ void BitrateControllerImpl::SetBitrateObserver( it->second->start_bitrate_ = start_bitrate; it->second->min_bitrate_ = min_bitrate; it->second->max_bitrate_ = max_bitrate; + // Set the send-side bandwidth to the max of the sum of start bitrates and + // the current estimate, so that if the user wants to immediately use more + // bandwidth, that can be enforced. + uint32_t sum_start_bitrate = 0; + BitrateObserverConfList::iterator it; + for (it = bitrate_observers_.begin(); it != bitrate_observers_.end(); + ++it) { + sum_start_bitrate += it->second->start_bitrate_; + } + uint32_t current_estimate; + uint8_t loss; + uint32_t rtt; + bandwidth_estimation_.CurrentEstimate(¤t_estimate, &loss, &rtt); + bandwidth_estimation_.SetSendBitrate(std::max(sum_start_bitrate, + current_estimate)); } else { // Add new settings. bitrate_observers_.push_back(BitrateObserverConfiguration(observer, new BitrateConfiguration(start_bitrate, min_bitrate, max_bitrate))); + bitrate_observers_modified_ = true; + + // TODO(andresp): This is a ugly way to set start bitrate. + // + // Only change start bitrate if we have exactly one observer. By definition + // you can only have one start bitrate, once we have our first estimate we + // will adapt from there. + if (bitrate_observers_.size() == 1) { + bandwidth_estimation_.SetSendBitrate(start_bitrate); + } } - uint32_t sum_start_bitrate = 0; + + UpdateMinMaxBitrate(); +} + +void BitrateControllerImpl::UpdateMinMaxBitrate() { uint32_t sum_min_bitrate = 0; uint32_t sum_max_bitrate = 0; - - // Summarize all configurations. + BitrateObserverConfList::iterator it; for (it = bitrate_observers_.begin(); it != bitrate_observers_.end(); ++it) { - sum_start_bitrate += it->second->start_bitrate_; sum_min_bitrate += it->second->min_bitrate_; sum_max_bitrate += it->second->max_bitrate_; } - // Only change start bitrate if we have exactly one observer. By definition - // you can only have one start bitrate, once we have our first estimate we - // will adapt from there. - if (bitrate_observers_.size() == 1) { - bandwidth_estimation_.SetSendBitrate(sum_start_bitrate); + if (sum_max_bitrate == 0) { + // No max configured use 1Gbit/s. + sum_max_bitrate = 1000000000; + } + if (enforce_min_bitrate_ == false) { + // If not enforcing min bitrate, allow the bandwidth estimation to + // go as low as 10 kbps. + sum_min_bitrate = std::min(sum_min_bitrate, 10000u); } bandwidth_estimation_.SetMinMaxBitrate(sum_min_bitrate, sum_max_bitrate); @@ -235,31 +201,46 @@ void BitrateControllerImpl::RemoveBitrateObserver(BitrateObserver* observer) { if (it != bitrate_observers_.end()) { delete it->second; bitrate_observers_.erase(it); + bitrate_observers_modified_ = true; } } void BitrateControllerImpl::EnforceMinBitrate(bool enforce_min_bitrate) { CriticalSectionScoped cs(critsect_); - if (enforce_min_bitrate) { - low_rate_strategy_.reset(new EnforceMinRateStrategy( - &bandwidth_estimation_, &bitrate_observers_)); - } else { - low_rate_strategy_.reset(new NoEnforceMinRateStrategy( - &bandwidth_estimation_, &bitrate_observers_)); - } + enforce_min_bitrate_ = enforce_min_bitrate; + UpdateMinMaxBitrate(); +} + +void BitrateControllerImpl::SetReservedBitrate(uint32_t reserved_bitrate_bps) { + CriticalSectionScoped cs(critsect_); + reserved_bitrate_bps_ = reserved_bitrate_bps; + MaybeTriggerOnNetworkChanged(); } void BitrateControllerImpl::OnReceivedEstimatedBitrate(const uint32_t bitrate) { - uint32_t new_bitrate = 0; - uint8_t fraction_lost = 0; - uint16_t rtt = 0; CriticalSectionScoped cs(critsect_); - if (bandwidth_estimation_.UpdateBandwidthEstimate(bitrate, - &new_bitrate, - &fraction_lost, - &rtt)) { - OnNetworkChanged(new_bitrate, fraction_lost, rtt); + bandwidth_estimation_.UpdateReceiverEstimate(bitrate); + MaybeTriggerOnNetworkChanged(); +} + +int32_t BitrateControllerImpl::TimeUntilNextProcess() { + enum { kBitrateControllerUpdateIntervalMs = 25 }; + CriticalSectionScoped cs(critsect_); + int time_since_update_ms = + clock_->TimeInMilliseconds() - last_bitrate_update_ms_; + return std::max(0, kBitrateControllerUpdateIntervalMs - time_since_update_ms); +} + +int32_t BitrateControllerImpl::Process() { + if (TimeUntilNextProcess() > 0) + return 0; + { + CriticalSectionScoped cs(critsect_); + bandwidth_estimation_.UpdateEstimate(clock_->TimeInMilliseconds()); + MaybeTriggerOnNetworkChanged(); } + last_bitrate_update_ms_ = clock_->TimeInMilliseconds(); + return 0; } void BitrateControllerImpl::OnReceivedRtcpReceiverReport( @@ -267,37 +248,63 @@ void BitrateControllerImpl::OnReceivedRtcpReceiverReport( const uint32_t rtt, const int number_of_packets, const uint32_t now_ms) { - uint32_t new_bitrate = 0; - uint8_t loss = fraction_loss; CriticalSectionScoped cs(critsect_); - if (bandwidth_estimation_.UpdatePacketLoss(number_of_packets, rtt, now_ms, - &loss, &new_bitrate)) { - OnNetworkChanged(new_bitrate, loss, rtt); + bandwidth_estimation_.UpdateReceiverBlock( + fraction_loss, rtt, number_of_packets, now_ms); + MaybeTriggerOnNetworkChanged(); +} + +void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() { + uint32_t bitrate; + uint8_t fraction_loss; + uint32_t rtt; + bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt); + bitrate -= std::min(bitrate, reserved_bitrate_bps_); + + if (bitrate_observers_modified_ || + bitrate != last_bitrate_bps_ || + fraction_loss != last_fraction_loss_ || + rtt != last_rtt_ms_ || + last_enforce_min_bitrate_ != enforce_min_bitrate_ || + last_reserved_bitrate_bps_ != reserved_bitrate_bps_) { + last_bitrate_bps_ = bitrate; + last_fraction_loss_ = fraction_loss; + last_rtt_ms_ = rtt; + last_enforce_min_bitrate_ = enforce_min_bitrate_; + last_reserved_bitrate_bps_ = reserved_bitrate_bps_; + bitrate_observers_modified_ = false; + OnNetworkChanged(bitrate, fraction_loss, rtt); } } -// We have the lock here. void BitrateControllerImpl::OnNetworkChanged(const uint32_t bitrate, const uint8_t fraction_loss, const uint32_t rtt) { // Sanity check. - uint32_t number_of_observers = bitrate_observers_.size(); - if (number_of_observers == 0) { + if (bitrate_observers_.empty()) return; - } + uint32_t sum_min_bitrates = 0; BitrateObserverConfList::iterator it; for (it = bitrate_observers_.begin(); it != bitrate_observers_.end(); ++it) { sum_min_bitrates += it->second->min_bitrate_; } - if (bitrate <= sum_min_bitrates) { - return low_rate_strategy_->LowRateAllocation(bitrate, fraction_loss, rtt, - sum_min_bitrates); - } + if (bitrate <= sum_min_bitrates) + return LowRateAllocation(bitrate, fraction_loss, rtt, sum_min_bitrates); + else + return NormalRateAllocation(bitrate, fraction_loss, rtt, sum_min_bitrates); +} + +void BitrateControllerImpl::NormalRateAllocation(uint32_t bitrate, + uint8_t fraction_loss, + uint32_t rtt, + uint32_t sum_min_bitrates) { + uint32_t number_of_observers = bitrate_observers_.size(); uint32_t bitrate_per_observer = (bitrate - sum_min_bitrates) / number_of_observers; // Use map to sort list based on max bitrate. ObserverSortingMap list_max_bitrates; + BitrateObserverConfList::iterator it; for (it = bitrate_observers_.begin(); it != bitrate_observers_.end(); ++it) { list_max_bitrates.insert(std::pair<uint32_t, ObserverConfiguration*>( it->second->max_bitrate_, @@ -328,8 +335,46 @@ void BitrateControllerImpl::OnNetworkChanged(const uint32_t bitrate, } } +void BitrateControllerImpl::LowRateAllocation(uint32_t bitrate, + uint8_t fraction_loss, + uint32_t rtt, + uint32_t sum_min_bitrates) { + if (enforce_min_bitrate_) { + // Min bitrate to all observers. + BitrateControllerImpl::BitrateObserverConfList::iterator it; + for (it = bitrate_observers_.begin(); it != bitrate_observers_.end(); + ++it) { + it->first->OnNetworkChanged(it->second->min_bitrate_, fraction_loss, rtt); + } + // Set sum of min to current send bitrate. + bandwidth_estimation_.SetSendBitrate(sum_min_bitrates); + } else { + // Allocate up to |min_bitrate_| to one observer at a time, until + // |bitrate| is depleted. + uint32_t remainder = bitrate; + BitrateControllerImpl::BitrateObserverConfList::iterator it; + for (it = bitrate_observers_.begin(); it != bitrate_observers_.end(); + ++it) { + uint32_t allocation = std::min(remainder, it->second->min_bitrate_); + it->first->OnNetworkChanged(allocation, fraction_loss, rtt); + remainder -= allocation; + } + // Set |bitrate| to current send bitrate. + bandwidth_estimation_.SetSendBitrate(bitrate); + } +} + bool BitrateControllerImpl::AvailableBandwidth(uint32_t* bandwidth) const { - return bandwidth_estimation_.AvailableBandwidth(bandwidth); + CriticalSectionScoped cs(critsect_); + uint32_t bitrate; + uint8_t fraction_loss; + uint32_t rtt; + bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt); + if (bitrate) { + *bandwidth = bitrate - std::min(bitrate, reserved_bitrate_bps_); + return true; + } + return false; } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h index 62ed6fda5ab..aff127bd142 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h +++ b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h @@ -27,12 +27,30 @@ namespace webrtc { -class RtcpBandwidthObserverImpl; -class LowRateStrategy; - class BitrateControllerImpl : public BitrateController { public: - friend class RtcpBandwidthObserverImpl; + BitrateControllerImpl(Clock* clock, bool enforce_min_bitrate); + virtual ~BitrateControllerImpl(); + + virtual bool AvailableBandwidth(uint32_t* bandwidth) const OVERRIDE; + + virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() OVERRIDE; + + virtual void SetBitrateObserver(BitrateObserver* observer, + const uint32_t start_bitrate, + const uint32_t min_bitrate, + const uint32_t max_bitrate) OVERRIDE; + + virtual void RemoveBitrateObserver(BitrateObserver* observer) OVERRIDE; + + virtual void EnforceMinBitrate(bool enforce_min_bitrate) OVERRIDE; + virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) OVERRIDE; + + virtual int32_t TimeUntilNextProcess() OVERRIDE; + virtual int32_t Process() OVERRIDE; + + private: + class RtcpBandwidthObserverImpl; struct BitrateConfiguration { BitrateConfiguration(uint32_t start_bitrate, @@ -59,23 +77,8 @@ class BitrateControllerImpl : public BitrateController { BitrateObserverConfiguration; typedef std::list<BitrateObserverConfiguration> BitrateObserverConfList; - explicit BitrateControllerImpl(bool enforce_min_bitrate); - virtual ~BitrateControllerImpl(); - - virtual bool AvailableBandwidth(uint32_t* bandwidth) const OVERRIDE; - - virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() OVERRIDE; - - virtual void SetBitrateObserver(BitrateObserver* observer, - const uint32_t start_bitrate, - const uint32_t min_bitrate, - const uint32_t max_bitrate) OVERRIDE; - - virtual void RemoveBitrateObserver(BitrateObserver* observer) OVERRIDE; + void UpdateMinMaxBitrate() EXCLUSIVE_LOCKS_REQUIRED(*critsect_); - virtual void EnforceMinBitrate(bool enforce_min_bitrate) OVERRIDE; - - private: // Called by BitrateObserver's direct from the RTCP module. void OnReceivedEstimatedBitrate(const uint32_t bitrate); @@ -84,18 +87,48 @@ class BitrateControllerImpl : public BitrateController { const int number_of_packets, const uint32_t now_ms); - typedef std::multimap<uint32_t, ObserverConfiguration*> ObserverSortingMap; + void MaybeTriggerOnNetworkChanged() EXCLUSIVE_LOCKS_REQUIRED(*critsect_); - BitrateObserverConfList::iterator - FindObserverConfigurationPair(const BitrateObserver* observer); void OnNetworkChanged(const uint32_t bitrate, const uint8_t fraction_loss, // 0 - 255. - const uint32_t rtt); + const uint32_t rtt) + EXCLUSIVE_LOCKS_REQUIRED(*critsect_); + + void NormalRateAllocation(uint32_t bitrate, + uint8_t fraction_loss, + uint32_t rtt, + uint32_t sum_min_bitrates) + EXCLUSIVE_LOCKS_REQUIRED(*critsect_); + + void LowRateAllocation(uint32_t bitrate, + uint8_t fraction_loss, + uint32_t rtt, + uint32_t sum_min_bitrates) + EXCLUSIVE_LOCKS_REQUIRED(*critsect_); + + typedef std::multimap<uint32_t, ObserverConfiguration*> ObserverSortingMap; + + BitrateObserverConfList::iterator FindObserverConfigurationPair( + const BitrateObserver* observer) EXCLUSIVE_LOCKS_REQUIRED(*critsect_); + + // Used by process thread. + Clock* clock_; + uint32_t last_bitrate_update_ms_; CriticalSectionWrapper* critsect_; - SendSideBandwidthEstimation bandwidth_estimation_; - BitrateObserverConfList bitrate_observers_; - scoped_ptr<LowRateStrategy> low_rate_strategy_; + SendSideBandwidthEstimation bandwidth_estimation_ GUARDED_BY(*critsect_); + BitrateObserverConfList bitrate_observers_ GUARDED_BY(*critsect_); + bool enforce_min_bitrate_ GUARDED_BY(*critsect_); + uint32_t reserved_bitrate_bps_ GUARDED_BY(*critsect_); + + uint32_t last_bitrate_bps_ GUARDED_BY(*critsect_); + uint8_t last_fraction_loss_ GUARDED_BY(*critsect_); + uint32_t last_rtt_ms_ GUARDED_BY(*critsect_); + bool last_enforce_min_bitrate_ GUARDED_BY(*critsect_); + bool bitrate_observers_modified_ GUARDED_BY(*critsect_); + uint32_t last_reserved_bitrate_bps_ GUARDED_BY(*critsect_); + + DISALLOW_IMPLICIT_CONSTRUCTORS(BitrateControllerImpl); }; } // namespace webrtc #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc index 30f85a81cb9..8523d505b24 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc +++ b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc @@ -57,12 +57,12 @@ class TestBitrateObserver: public BitrateObserver { class BitrateControllerTest : public ::testing::Test { protected: - BitrateControllerTest() : enforce_min_bitrate_(true) {} + BitrateControllerTest() : clock_(0), enforce_min_bitrate_(true) {} ~BitrateControllerTest() {} virtual void SetUp() { - controller_ = - BitrateController::CreateBitrateController(enforce_min_bitrate_); + controller_ = BitrateController::CreateBitrateController( + &clock_, enforce_min_bitrate_); bandwidth_observer_ = controller_->CreateRtcpBandwidthObserver(); } @@ -70,6 +70,8 @@ class BitrateControllerTest : public ::testing::Test { delete bandwidth_observer_; delete controller_; } + + webrtc::SimulatedClock clock_; bool enforce_min_bitrate_; BitrateController* controller_; RtcpBandwidthObserver* bandwidth_observer_; @@ -81,58 +83,74 @@ TEST_F(BitrateControllerTest, Basic) { controller_->RemoveBitrateObserver(&bitrate_observer); } +TEST_F(BitrateControllerTest, UpdatingBitrateObserver) { + TestBitrateObserver bitrate_observer; + controller_->SetBitrateObserver(&bitrate_observer, 200000, 100000, 1500000); + clock_.AdvanceTimeMilliseconds(25); + controller_->Process(); + EXPECT_EQ(200000u, bitrate_observer.last_bitrate_); + + controller_->SetBitrateObserver(&bitrate_observer, 1500000, 100000, 1500000); + clock_.AdvanceTimeMilliseconds(25); + controller_->Process(); + EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_); + + controller_->SetBitrateObserver(&bitrate_observer, 500000, 100000, 1500000); + clock_.AdvanceTimeMilliseconds(25); + controller_->Process(); + EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_); +} + TEST_F(BitrateControllerTest, OneBitrateObserverOneRtcpObserver) { TestBitrateObserver bitrate_observer; controller_->SetBitrateObserver(&bitrate_observer, 200000, 100000, 300000); // Receive a high remb, test bitrate inc. bandwidth_observer_->OnReceivedEstimatedBitrate(400000); + EXPECT_EQ(200000u, bitrate_observer.last_bitrate_); + EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); + EXPECT_EQ(0u, bitrate_observer.last_rtt_); - // Test start bitrate. + // Test bitrate increase 8% per second. webrtc::ReportBlockList report_blocks; report_blocks.push_back(CreateReportBlock(1, 2, 0, 1)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 1); - EXPECT_EQ(0u, bitrate_observer.last_bitrate_); + EXPECT_EQ(217000u, bitrate_observer.last_bitrate_); EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); - EXPECT_EQ(0u, bitrate_observer.last_rtt_); + EXPECT_EQ(50u, bitrate_observer.last_rtt_); - // Test bitrate increase 8% per second. report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 21)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 1001); - EXPECT_EQ(217000u, bitrate_observer.last_bitrate_); + EXPECT_EQ(235360u, bitrate_observer.last_bitrate_); EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); EXPECT_EQ(50u, bitrate_observer.last_rtt_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 41)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 2001); - EXPECT_EQ(235360u, bitrate_observer.last_bitrate_); + EXPECT_EQ(255189u, bitrate_observer.last_bitrate_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 61)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 3001); - EXPECT_EQ(255189u, bitrate_observer.last_bitrate_); + EXPECT_EQ(276604u, bitrate_observer.last_bitrate_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 801)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 4001); - EXPECT_EQ(276604u, bitrate_observer.last_bitrate_); + EXPECT_EQ(299732u, bitrate_observer.last_bitrate_); + // Reach max cap. report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 101)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 5001); - EXPECT_EQ(299732u, bitrate_observer.last_bitrate_); - - report_blocks.clear(); - report_blocks.push_back(CreateReportBlock(1, 2, 0, 121)); - bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 6001); - EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); // Max cap. + EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 141)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 7001); - EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); // Max cap. + EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); // Test that a low REMB trigger immediately. bandwidth_observer_->OnReceivedEstimatedBitrate(250000); @@ -154,6 +172,9 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) { // Receive a high remb, test bitrate inc. bandwidth_observer_->OnReceivedEstimatedBitrate(400000); + EXPECT_EQ(200000u, bitrate_observer.last_bitrate_); + EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); + EXPECT_EQ(0u, bitrate_observer.last_rtt_); // Test start bitrate. webrtc::ReportBlockList report_blocks; @@ -161,9 +182,9 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) { bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 1); second_bandwidth_observer->OnReceivedRtcpReceiverReport( report_blocks, 100, 1); - EXPECT_EQ(0u, bitrate_observer.last_bitrate_); + EXPECT_EQ(217000u, bitrate_observer.last_bitrate_); EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); - EXPECT_EQ(0u, bitrate_observer.last_rtt_); + EXPECT_EQ(100u, bitrate_observer.last_rtt_); // Test bitrate increase 8% per second. report_blocks.clear(); @@ -171,7 +192,7 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) { bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 501); second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100, 1001); - EXPECT_EQ(217000u, bitrate_observer.last_bitrate_); + EXPECT_EQ(235360u, bitrate_observer.last_bitrate_); EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); EXPECT_EQ(100u, bitrate_observer.last_rtt_); @@ -180,50 +201,45 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) { report_blocks.push_back(CreateReportBlock(1, 2, 0, 31)); second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100, 1501); - EXPECT_EQ(217000u, bitrate_observer.last_bitrate_); + EXPECT_EQ(235360u, bitrate_observer.last_bitrate_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 41)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 2001); - EXPECT_EQ(235360u, bitrate_observer.last_bitrate_); + EXPECT_EQ(255189u, bitrate_observer.last_bitrate_); // Second report should not change estimate. report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 41)); second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100, 2001); - EXPECT_EQ(235360u, bitrate_observer.last_bitrate_); + EXPECT_EQ(255189u, bitrate_observer.last_bitrate_); // Reports from only one bandwidth observer is ok. report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 61)); second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50, 3001); - EXPECT_EQ(255189u, bitrate_observer.last_bitrate_); + EXPECT_EQ(276604u, bitrate_observer.last_bitrate_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 81)); second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50, 4001); - EXPECT_EQ(276604u, bitrate_observer.last_bitrate_); - - report_blocks.clear(); - report_blocks.push_back(CreateReportBlock(1, 2, 0, 101)); - second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50, - 5001); EXPECT_EQ(299732u, bitrate_observer.last_bitrate_); + // Reach max cap. report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 121)); - second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50, - 6001); - EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); // Max cap. + second_bandwidth_observer->OnReceivedRtcpReceiverReport( + report_blocks, 50, 5001); + EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 141)); - second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50, - 7001); - EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); // Max cap. + second_bandwidth_observer->OnReceivedRtcpReceiverReport( + report_blocks, 50, 6001); + EXPECT_EQ(300000u, bitrate_observer.last_bitrate_); // Test that a low REMB trigger immediately. // We don't care which bandwidth observer that delivers the REMB. @@ -232,8 +248,9 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) { EXPECT_EQ(0, bitrate_observer.last_fraction_loss_); EXPECT_EQ(50u, bitrate_observer.last_rtt_); + // Min cap. bandwidth_observer_->OnReceivedEstimatedBitrate(1000); - EXPECT_EQ(100000u, bitrate_observer.last_bitrate_); // Min cap. + EXPECT_EQ(100000u, bitrate_observer.last_bitrate_); controller_->RemoveBitrateObserver(&bitrate_observer); delete second_bandwidth_observer; } @@ -317,40 +334,33 @@ TEST_F(BitrateControllerTest, TwoBitrateObserversOneRtcpObserver) { controller_->SetBitrateObserver(&bitrate_observer_1, 200000, 100000, 300000); // Receive a high remb, test bitrate inc. + // Test too low start bitrate, hence lower than sum of min. bandwidth_observer_->OnReceivedEstimatedBitrate(400000); + EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_); + EXPECT_EQ(0, bitrate_observer_1.last_fraction_loss_); + EXPECT_EQ(0u, bitrate_observer_1.last_rtt_); - // Test too low start bitrate, hence lower than sum of min. + // Test bitrate increase 8% per second, distributed equally. webrtc::ReportBlockList report_blocks; report_blocks.push_back(CreateReportBlock(1, 2, 0, 1)); - bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 1); - - // Test bitrate increase 8% per second, distributed equally. - report_blocks.clear(); - report_blocks.push_back(CreateReportBlock(1, 2, 0, 21)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 1001); - EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_); + EXPECT_EQ(112500u, bitrate_observer_1.last_bitrate_); EXPECT_EQ(0, bitrate_observer_1.last_fraction_loss_); EXPECT_EQ(50u, bitrate_observer_1.last_rtt_); - EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_); + EXPECT_EQ(212500u, bitrate_observer_2.last_bitrate_); EXPECT_EQ(0, bitrate_observer_2.last_fraction_loss_); EXPECT_EQ(50u, bitrate_observer_2.last_rtt_); report_blocks.clear(); report_blocks.push_back(CreateReportBlock(1, 2, 0, 41)); bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 2001); - EXPECT_EQ(112500u, bitrate_observer_1.last_bitrate_); - EXPECT_EQ(212500u, bitrate_observer_2.last_bitrate_); - - report_blocks.clear(); - report_blocks.push_back(CreateReportBlock(1, 2, 0, 61)); - bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 3001); EXPECT_EQ(126000u, bitrate_observer_1.last_bitrate_); EXPECT_EQ(226000u, bitrate_observer_2.last_bitrate_); report_blocks.clear(); - report_blocks.push_back(CreateReportBlock(1, 2, 0, 81)); - bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 4001); + report_blocks.push_back(CreateReportBlock(1, 2, 0, 61)); + bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, 3001); EXPECT_EQ(140580u, bitrate_observer_1.last_bitrate_); EXPECT_EQ(240580u, bitrate_observer_2.last_bitrate_); @@ -416,6 +426,61 @@ TEST_F(BitrateControllerTest, TwoBitrateObserversOneRtcpObserver) { controller_->RemoveBitrateObserver(&bitrate_observer_2); } +TEST_F(BitrateControllerTest, SetReservedBitrate) { + TestBitrateObserver bitrate_observer; + controller_->SetBitrateObserver(&bitrate_observer, 200000, 100000, 300000); + + // Receive successively lower REMBs, verify the reserved bitrate is deducted. + + controller_->SetReservedBitrate(0); + bandwidth_observer_->OnReceivedEstimatedBitrate(400000); + EXPECT_EQ(200000u, bitrate_observer.last_bitrate_); + controller_->SetReservedBitrate(50000); + bandwidth_observer_->OnReceivedEstimatedBitrate(400000); + EXPECT_EQ(150000u, bitrate_observer.last_bitrate_); + + controller_->SetReservedBitrate(0); + bandwidth_observer_->OnReceivedEstimatedBitrate(250000); + EXPECT_EQ(200000u, bitrate_observer.last_bitrate_); + controller_->SetReservedBitrate(50000); + bandwidth_observer_->OnReceivedEstimatedBitrate(250000); + EXPECT_EQ(150000u, bitrate_observer.last_bitrate_); + + controller_->SetReservedBitrate(0); + bandwidth_observer_->OnReceivedEstimatedBitrate(200000); + EXPECT_EQ(200000u, bitrate_observer.last_bitrate_); + controller_->SetReservedBitrate(30000); + bandwidth_observer_->OnReceivedEstimatedBitrate(200000); + EXPECT_EQ(170000u, bitrate_observer.last_bitrate_); + + controller_->SetReservedBitrate(0); + bandwidth_observer_->OnReceivedEstimatedBitrate(160000); + EXPECT_EQ(160000u, bitrate_observer.last_bitrate_); + controller_->SetReservedBitrate(30000); + bandwidth_observer_->OnReceivedEstimatedBitrate(160000); + EXPECT_EQ(130000u, bitrate_observer.last_bitrate_); + + controller_->SetReservedBitrate(0); + bandwidth_observer_->OnReceivedEstimatedBitrate(120000); + EXPECT_EQ(120000u, bitrate_observer.last_bitrate_); + controller_->SetReservedBitrate(10000); + bandwidth_observer_->OnReceivedEstimatedBitrate(120000); + EXPECT_EQ(110000u, bitrate_observer.last_bitrate_); + + controller_->SetReservedBitrate(0); + bandwidth_observer_->OnReceivedEstimatedBitrate(120000); + EXPECT_EQ(120000u, bitrate_observer.last_bitrate_); + controller_->SetReservedBitrate(50000); + bandwidth_observer_->OnReceivedEstimatedBitrate(120000); + EXPECT_EQ(100000u, bitrate_observer.last_bitrate_); + + controller_->SetReservedBitrate(10000); + bandwidth_observer_->OnReceivedEstimatedBitrate(0); + EXPECT_EQ(100000u, bitrate_observer.last_bitrate_); + + controller_->RemoveBitrateObserver(&bitrate_observer); +} + class BitrateControllerTestNoEnforceMin : public BitrateControllerTest { protected: BitrateControllerTestNoEnforceMin() : BitrateControllerTest() { @@ -434,8 +499,32 @@ TEST_F(BitrateControllerTestNoEnforceMin, OneBitrateObserver) { EXPECT_EQ(150000u, bitrate_observer_1.last_bitrate_); // Low REMB. - bandwidth_observer_->OnReceivedEstimatedBitrate(1000); - EXPECT_EQ(1000u, bitrate_observer_1.last_bitrate_); + bandwidth_observer_->OnReceivedEstimatedBitrate(10000); + EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_); + + // Keeps at least 10 kbps. + bandwidth_observer_->OnReceivedEstimatedBitrate(9000); + EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_); + + controller_->RemoveBitrateObserver(&bitrate_observer_1); +} + +TEST_F(BitrateControllerTestNoEnforceMin, SetReservedBitrate) { + TestBitrateObserver bitrate_observer_1; + controller_->SetBitrateObserver(&bitrate_observer_1, 200000, 100000, 400000); + controller_->SetReservedBitrate(10000); + + // High REMB. + bandwidth_observer_->OnReceivedEstimatedBitrate(150000); + EXPECT_EQ(140000u, bitrate_observer_1.last_bitrate_); + + // Low REMB. + bandwidth_observer_->OnReceivedEstimatedBitrate(15000); + EXPECT_EQ(5000u, bitrate_observer_1.last_bitrate_); + + // Keeps at least 10 kbps. + bandwidth_observer_->OnReceivedEstimatedBitrate(9000); + EXPECT_EQ(0u, bitrate_observer_1.last_bitrate_); controller_->RemoveBitrateObserver(&bitrate_observer_1); } @@ -469,9 +558,15 @@ TEST_F(BitrateControllerTestNoEnforceMin, ThreeBitrateObservers) { EXPECT_EQ(200000u, bitrate_observer_3.last_bitrate_); // Remainder. // Low REMB. - bandwidth_observer_->OnReceivedEstimatedBitrate(1000); + bandwidth_observer_->OnReceivedEstimatedBitrate(10000); // Verify that the first observer gets all the rate, and the rest get zero. - EXPECT_EQ(1000u, bitrate_observer_1.last_bitrate_); + EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_); + EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_); + EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_); + + // Verify it keeps an estimate of at least 10kbps. + bandwidth_observer_->OnReceivedEstimatedBitrate(9000); + EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_); EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_); EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_); diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h index 0f743676581..46d7830300b 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h +++ b/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h @@ -15,6 +15,7 @@ #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ #define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ +#include "webrtc/modules/interface/module.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" namespace webrtc { @@ -35,7 +36,7 @@ class BitrateObserver { virtual ~BitrateObserver() {} }; -class BitrateController { +class BitrateController : public Module { /* * This class collects feedback from all streams sent to a peer (via * RTCPBandwidthObservers). It does one aggregated send side bandwidth @@ -48,7 +49,8 @@ class BitrateController { // When true, the bitrate will never be set lower than the minimum bitrate(s). // When false, the bitrate observers will be allocated rates up to their // respective minimum bitrate, satisfying one observer after the other. - static BitrateController* CreateBitrateController(bool enforce_min_bitrate); + static BitrateController* CreateBitrateController(Clock* clock, + bool enforce_min_bitrate); virtual ~BitrateController() {} virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0; @@ -73,6 +75,8 @@ class BitrateController { // Changes the mode that was set in the constructor. virtual void EnforceMinBitrate(bool enforce_min_bitrate) = 0; + + virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0; }; } // namespace webrtc #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc index ce385db54c1..5da23f06624 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc +++ b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc @@ -10,216 +10,204 @@ #include "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h" -#include <math.h> // sqrt() +#include <cmath> -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { +namespace { +enum { kBweIncreaseIntervalMs = 1000 }; +enum { kBweDecreaseIntervalMs = 300 }; +enum { kLimitNumPackets = 20 }; +enum { kAvgPacketSizeBytes = 1000 }; + +// Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply. +// The formula in RFC 3448, Section 3.1, is used. +uint32_t CalcTfrcBps(uint16_t rtt, uint8_t loss) { + if (rtt == 0 || loss == 0) { + // Input variables out of range. + return 0; + } + double R = static_cast<double>(rtt) / 1000; // RTT in seconds. + int b = 1; // Number of packets acknowledged by a single TCP acknowledgement: + // recommended = 1. + double t_RTO = 4.0 * R; // TCP retransmission timeout value in seconds + // recommended = 4*R. + double p = static_cast<double>(loss) / 255; // Packet loss rate in [0, 1). + double s = static_cast<double>(kAvgPacketSizeBytes); + + // Calculate send rate in bytes/second. + double X = + s / (R * std::sqrt(2 * b * p / 3) + + (t_RTO * (3 * std::sqrt(3 * b * p / 8) * p * (1 + 32 * p * p)))); + + // Convert to bits/second. + return (static_cast<uint32_t>(X * 8)); +} +} SendSideBandwidthEstimation::SendSideBandwidthEstimation() - : critsect_(CriticalSectionWrapper::CreateCriticalSection()), - accumulate_lost_packets_Q8_(0), + : accumulate_lost_packets_Q8_(0), accumulate_expected_packets_(0), bitrate_(0), min_bitrate_configured_(0), max_bitrate_configured_(0), + time_last_receiver_block_ms_(0), last_fraction_loss_(0), - last_round_trip_time_(0), + last_round_trip_time_ms_(0), bwe_incoming_(0), - time_last_increase_(0), - time_last_decrease_(0) { -} + time_last_decrease_ms_(0) {} -SendSideBandwidthEstimation::~SendSideBandwidthEstimation() { - delete critsect_; -} +SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {} -void SendSideBandwidthEstimation::SetSendBitrate(const uint32_t bitrate) { - CriticalSectionScoped cs(critsect_); +void SendSideBandwidthEstimation::SetSendBitrate(uint32_t bitrate) { bitrate_ = bitrate; + + // Clear last sent bitrate history so the new value can be used directly + // and not capped. + min_bitrate_history_.clear(); } -void SendSideBandwidthEstimation::SetMinMaxBitrate(const uint32_t min_bitrate, - const uint32_t max_bitrate) { - CriticalSectionScoped cs(critsect_); +void SendSideBandwidthEstimation::SetMinMaxBitrate(uint32_t min_bitrate, + uint32_t max_bitrate) { min_bitrate_configured_ = min_bitrate; - if (max_bitrate == 0) { - // no max configured use 1Gbit/s - max_bitrate_configured_ = 1000000000; - } else { - max_bitrate_configured_ = max_bitrate; - } + max_bitrate_configured_ = max_bitrate; } -bool SendSideBandwidthEstimation::UpdateBandwidthEstimate( - const uint32_t bandwidth, - uint32_t* new_bitrate, - uint8_t* fraction_lost, - uint16_t* rtt) { - *new_bitrate = 0; - CriticalSectionScoped cs(critsect_); +void SendSideBandwidthEstimation::SetMinBitrate(uint32_t min_bitrate) { + min_bitrate_configured_ = min_bitrate; +} - bwe_incoming_ = bandwidth; +void SendSideBandwidthEstimation::CurrentEstimate(uint32_t* bitrate, + uint8_t* loss, + uint32_t* rtt) const { + *bitrate = bitrate_; + *loss = last_fraction_loss_; + *rtt = last_round_trip_time_ms_; +} - if (bitrate_ == 0) { - // SendSideBandwidthEstimation off - return false; - } - if (bwe_incoming_ > 0 && bitrate_ > bwe_incoming_) { - bitrate_ = bwe_incoming_; - *new_bitrate = bitrate_; - *fraction_lost = last_fraction_loss_; - *rtt = last_round_trip_time_; - return true; - } - return false; +void SendSideBandwidthEstimation::UpdateReceiverEstimate(uint32_t bandwidth) { + bwe_incoming_ = bandwidth; + CapBitrateToThresholds(); } -bool SendSideBandwidthEstimation::UpdatePacketLoss( - const int number_of_packets, - const uint32_t rtt, - const uint32_t now_ms, - uint8_t* loss, - uint32_t* new_bitrate) { - CriticalSectionScoped cs(critsect_); - - if (bitrate_ == 0) { - // SendSideBandwidthEstimation off - return false; - } +void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss, + uint32_t rtt, + int number_of_packets, + uint32_t now_ms) { // Update RTT. - last_round_trip_time_ = rtt; + last_round_trip_time_ms_ = rtt; // Check sequence number diff and weight loss report if (number_of_packets > 0) { // Calculate number of lost packets. - const int num_lost_packets_Q8 = *loss * number_of_packets; + const int num_lost_packets_Q8 = fraction_loss * number_of_packets; // Accumulate reports. accumulate_lost_packets_Q8_ += num_lost_packets_Q8; accumulate_expected_packets_ += number_of_packets; // Report loss if the total report is based on sufficiently many packets. if (accumulate_expected_packets_ >= kLimitNumPackets) { - *loss = accumulate_lost_packets_Q8_ / accumulate_expected_packets_; + last_fraction_loss_ = + accumulate_lost_packets_Q8_ / accumulate_expected_packets_; - // Reset accumulators + // Reset accumulators. accumulate_lost_packets_Q8_ = 0; accumulate_expected_packets_ = 0; } else { - // Report zero loss until we have enough data to estimate - // the loss rate. - return false; + // Early return without updating estimate. + return; } } - // Keep for next time. - last_fraction_loss_ = *loss; - uint32_t bitrate = 0; - if (!ShapeSimple(*loss, rtt, now_ms, &bitrate)) { - // No change. - return false; - } - bitrate_ = bitrate; - *new_bitrate = bitrate; - return true; + time_last_receiver_block_ms_ = now_ms; + UpdateEstimate(now_ms); } -bool SendSideBandwidthEstimation::AvailableBandwidth( - uint32_t* bandwidth) const { - CriticalSectionScoped cs(critsect_); - if (bitrate_ == 0) { - return false; +void SendSideBandwidthEstimation::UpdateEstimate(uint32_t now_ms) { + UpdateMinHistory(now_ms); + + // Only start updating bitrate when receiving receiver blocks. + if (time_last_receiver_block_ms_ != 0) { + if (last_fraction_loss_ <= 5) { + // Loss < 2%: Increase rate by 8% of the min bitrate in the last + // kBweIncreaseIntervalMs. + // Note that by remembering the bitrate over the last second one can + // rampup up one second faster than if only allowed to start ramping + // at 8% per second rate now. E.g.: + // If sending a constant 100kbps it can rampup immediatly to 108kbps + // whenever a receiver report is received with lower packet loss. + // If instead one would do: bitrate_ *= 1.08^(delta time), it would + // take over one second since the lower packet loss to achieve 108kbps. + bitrate_ = static_cast<uint32_t>( + min_bitrate_history_.front().second * 1.08 + 0.5); + + // Add 1 kbps extra, just to make sure that we do not get stuck + // (gives a little extra increase at low rates, negligible at higher + // rates). + bitrate_ += 1000; + + } else if (last_fraction_loss_ <= 26) { + // Loss between 2% - 10%: Do nothing. + + } else { + // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs + + // rtt. + if ((now_ms - time_last_decrease_ms_) >= + static_cast<uint32_t>(kBweDecreaseIntervalMs + + last_round_trip_time_ms_)) { + time_last_decrease_ms_ = now_ms; + + // Reduce rate: + // newRate = rate * (1 - 0.5*lossRate); + // where packetLoss = 256*lossRate; + bitrate_ = static_cast<uint32_t>( + (bitrate_ * static_cast<double>(512 - last_fraction_loss_)) / + 512.0); + + // Calculate what rate TFRC would apply in this situation and to not + // reduce further than it. + bitrate_ = std::max( + bitrate_, + CalcTfrcBps(last_round_trip_time_ms_, last_fraction_loss_)); + } + } } - *bandwidth = bitrate_; - return true; + CapBitrateToThresholds(); } -/* - * Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply. - * The formula in RFC 3448, Section 3.1, is used. - */ -uint32_t SendSideBandwidthEstimation::CalcTFRCbps(uint16_t rtt, uint8_t loss) { - if (rtt == 0 || loss == 0) { - // input variables out of range - return 0; +void SendSideBandwidthEstimation::UpdateMinHistory(uint32_t now_ms) { + // Remove old data points from history. + // Since history precision is in ms, add one so it is able to increase + // bitrate if it is off by as little as 0.5ms. + while (!min_bitrate_history_.empty() && + now_ms - min_bitrate_history_.front().first + 1 > + kBweIncreaseIntervalMs) { + min_bitrate_history_.pop_front(); } - double R = static_cast<double>(rtt) / 1000; // RTT in seconds - int b = 1; // number of packets acknowledged by a single TCP acknowledgement; - // recommended = 1 - double t_RTO = 4.0 * R; // TCP retransmission timeout value in seconds - // recommended = 4*R - double p = static_cast<double>(loss) / 255; // packet loss rate in [0, 1) - double s = static_cast<double>(kAvgPacketSizeBytes); - // calculate send rate in bytes/second - double X = s / (R * sqrt(2 * b * p / 3) + - (t_RTO * (3 * sqrt(3 * b * p / 8) * p * (1 + 32 * p * p)))); + // Typical minimum sliding-window algorithm: Pop values higher than current + // bitrate before pushing it. + while (!min_bitrate_history_.empty() && + bitrate_ <= min_bitrate_history_.back().second) { + min_bitrate_history_.pop_back(); + } - return (static_cast<uint32_t>(X * 8)); // bits/second + min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_)); } -bool SendSideBandwidthEstimation::ShapeSimple(const uint8_t loss, - const uint32_t rtt, - const uint32_t now_ms, - uint32_t* bitrate) { - uint32_t new_bitrate = 0; - bool reducing = false; - - // Limit the rate increases to once a kBWEIncreaseIntervalMs. - if (loss <= 5) { - if ((now_ms - time_last_increase_) < kBWEIncreaseIntervalMs) { - return false; - } - time_last_increase_ = now_ms; - } - // Limit the rate decreases to once a kBWEDecreaseIntervalMs + rtt. - if (loss > 26) { - if ((now_ms - time_last_decrease_) < kBWEDecreaseIntervalMs + rtt) { - return false; - } - time_last_decrease_ = now_ms; - } - - if (loss > 5 && loss <= 26) { - // 2% - 10% - new_bitrate = bitrate_; - } else if (loss > 26) { - // 26/256 ~= 10% - // reduce rate: newRate = rate * (1 - 0.5*lossRate) - // packetLoss = 256*lossRate - new_bitrate = static_cast<uint32_t>((bitrate_ * - static_cast<double>(512 - loss)) / 512.0); - reducing = true; - } else { - // increase rate by 8% - new_bitrate = static_cast<uint32_t>(bitrate_ * 1.08 + 0.5); - - // add 1 kbps extra, just to make sure that we do not get stuck - // (gives a little extra increase at low rates, negligible at higher rates) - new_bitrate += 1000; - } - if (reducing) { - // Calculate what rate TFRC would apply in this situation - // scale loss to Q0 (back to [0, 255]) - uint32_t tfrc_bitrate = CalcTFRCbps(rtt, loss); - if (tfrc_bitrate > new_bitrate) { - // do not reduce further if rate is below TFRC rate - new_bitrate = tfrc_bitrate; - } - } - if (bwe_incoming_ > 0 && new_bitrate > bwe_incoming_) { - new_bitrate = bwe_incoming_; +void SendSideBandwidthEstimation::CapBitrateToThresholds() { + if (bwe_incoming_ > 0 && bitrate_ > bwe_incoming_) { + bitrate_ = bwe_incoming_; } - if (new_bitrate > max_bitrate_configured_) { - new_bitrate = max_bitrate_configured_; + if (bitrate_ > max_bitrate_configured_) { + bitrate_ = max_bitrate_configured_; } - if (new_bitrate < min_bitrate_configured_) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, - "The configured min bitrate (%u kbps) is greater than the " - "estimated available bandwidth (%u kbps).\n", - min_bitrate_configured_ / 1000, new_bitrate / 1000); - new_bitrate = min_bitrate_configured_; + if (bitrate_ < min_bitrate_configured_) { + LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate_ / 1000 + << " kbps is below configured min bitrate " + << min_bitrate_configured_ / 1000 << " kbps."; + bitrate_ = min_bitrate_configured_; } - *bitrate = new_bitrate; - return true; } + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h index 0c1fa94517d..eb675d1ca68 100644 --- a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h +++ b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h @@ -13,6 +13,8 @@ #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ +#include <deque> + #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" @@ -22,38 +24,33 @@ class SendSideBandwidthEstimation { SendSideBandwidthEstimation(); virtual ~SendSideBandwidthEstimation(); - // Call when we receive a RTCP message with TMMBR or REMB - // Return true if new_bitrate is valid. - bool UpdateBandwidthEstimate(const uint32_t bandwidth, - uint32_t* new_bitrate, - uint8_t* fraction_lost, - uint16_t* rtt); - - // Call when we receive a RTCP message with a ReceiveBlock - // Return true if new_bitrate is valid. - bool UpdatePacketLoss(const int number_of_packets, - const uint32_t rtt, - const uint32_t now_ms, - uint8_t* loss, - uint32_t* new_bitrate); - - // Return false if no bandwidth estimate is available - bool AvailableBandwidth(uint32_t* bandwidth) const; - void SetSendBitrate(const uint32_t bitrate); - void SetMinMaxBitrate(const uint32_t min_bitrate, const uint32_t max_bitrate); + void CurrentEstimate(uint32_t* bitrate, uint8_t* loss, uint32_t* rtt) const; - private: - bool ShapeSimple(const uint8_t loss, const uint32_t rtt, - const uint32_t now_ms, uint32_t* bitrate); + // Call periodically to update estimate. + void UpdateEstimate(uint32_t now_ms); + + // Call when we receive a RTCP message with TMMBR or REMB. + void UpdateReceiverEstimate(uint32_t bandwidth); - uint32_t CalcTFRCbps(uint16_t rtt, uint8_t loss); + // Call when we receive a RTCP message with a ReceiveBlock. + void UpdateReceiverBlock(uint8_t fraction_loss, + uint32_t rtt, + int number_of_packets, + uint32_t now_ms); + + void SetSendBitrate(uint32_t bitrate); + void SetMinMaxBitrate(uint32_t min_bitrate, uint32_t max_bitrate); + void SetMinBitrate(uint32_t min_bitrate); + + private: + void CapBitrateToThresholds(); - enum { kBWEIncreaseIntervalMs = 1000 }; - enum { kBWEDecreaseIntervalMs = 300 }; - enum { kLimitNumPackets = 20 }; - enum { kAvgPacketSizeBytes = 1000 }; + // Updates history of min bitrates. + // After this method returns min_bitrate_history_.front().second contains the + // min bitrate used during last kBweIncreaseIntervalMs. + void UpdateMinHistory(uint32_t now_ms); - CriticalSectionWrapper* critsect_; + std::deque<std::pair<uint32_t, uint32_t> > min_bitrate_history_; // incoming filters int accumulate_lost_packets_Q8_; @@ -63,12 +60,12 @@ class SendSideBandwidthEstimation { uint32_t min_bitrate_configured_; uint32_t max_bitrate_configured_; + uint32_t time_last_receiver_block_ms_; uint8_t last_fraction_loss_; - uint16_t last_round_trip_time_; + uint16_t last_round_trip_time_ms_; uint32_t bwe_incoming_; - uint32_t time_last_increase_; - uint32_t time_last_decrease_; + uint32_t time_last_decrease_ms_; }; } // namespace webrtc #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ diff --git a/chromium/third_party/webrtc/modules/desktop_capture/OWNERS b/chromium/third_party/webrtc/modules/desktop_capture/OWNERS index 3276530e93d..e85861b8d3e 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/OWNERS +++ b/chromium/third_party/webrtc/modules/desktop_capture/OWNERS @@ -1,3 +1,8 @@ alexeypa@chromium.org sergeyu@chromium.org wez@chromium.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc index 6ed3ae881f2..2547ba37aa8 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc @@ -81,7 +81,7 @@ DesktopFrameWithCursor::DesktopFrameWithCursor(DesktopFrame* frame, mutable_updated_region()->Swap(frame->mutable_updated_region()); DesktopVector image_pos = position.subtract(cursor.hotspot()); - DesktopRect target_rect = DesktopRect::MakeSize(cursor.image().size()); + DesktopRect target_rect = DesktopRect::MakeSize(cursor.image()->size()); target_rect.Translate(image_pos); DesktopVector target_origin = target_rect.top_left(); target_rect.IntersectWith(DesktopRect::MakeSize(size())); @@ -101,10 +101,10 @@ DesktopFrameWithCursor::DesktopFrameWithCursor(DesktopFrame* frame, target_rect.left() * DesktopFrame::kBytesPerPixel; DesktopVector origin_shift = target_rect.top_left().subtract(target_origin); AlphaBlend(target_rect_data, stride(), - cursor.image().data() + - origin_shift.y() * cursor.image().stride() + + cursor.image()->data() + + origin_shift.y() * cursor.image()->stride() + origin_shift.x() * DesktopFrame::kBytesPerPixel, - cursor.image().stride(), + cursor.image()->stride(), target_rect.size()); } @@ -142,12 +142,16 @@ void DesktopAndCursorComposer::Capture(const DesktopRegion& region) { desktop_capturer_->Capture(region); } +void DesktopAndCursorComposer::SetExcludedWindow(WindowId window) { + desktop_capturer_->SetExcludedWindow(window); +} + SharedMemory* DesktopAndCursorComposer::CreateSharedMemory(size_t size) { return callback_->CreateSharedMemory(size); } void DesktopAndCursorComposer::OnCaptureCompleted(DesktopFrame* frame) { - if (cursor_.get() && cursor_state_ == MouseCursorMonitor::INSIDE) { + if (frame && cursor_.get() && cursor_state_ == MouseCursorMonitor::INSIDE) { DesktopFrameWithCursor* frame_with_cursor = new DesktopFrameWithCursor(frame, *cursor_, cursor_position_); frame = frame_with_cursor; diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h index 4f7c85bde3e..3fac0212d49 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h @@ -34,6 +34,7 @@ class DesktopAndCursorComposer : public DesktopCapturer, // DesktopCapturer interface. virtual void Start(DesktopCapturer::Callback* callback) OVERRIDE; virtual void Capture(const DesktopRegion& region) OVERRIDE; + virtual void SetExcludedWindow(WindowId window) OVERRIDE; private: // DesktopCapturer::Callback interface. diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc index 15d6f546118..b482a29605e 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc @@ -58,6 +58,18 @@ uint32_t BlendPixels(uint32_t dest, uint32_t src) { return b + (g << 8) + (r << 16) + 0xff000000; } +DesktopFrame* CreateTestFrame() { + DesktopFrame* frame = + new BasicDesktopFrame(DesktopSize(kScreenWidth, kScreenHeight)); + uint32_t* data = reinterpret_cast<uint32_t*>(frame->data()); + for (int y = 0; y < kScreenHeight; ++y) { + for (int x = 0; x < kScreenWidth; ++x) { + *(data++) = GetFakeFramePixelValue(DesktopVector(x, y)); + } + } + return frame; +} + class FakeScreenCapturer : public DesktopCapturer { public: FakeScreenCapturer() {} @@ -67,27 +79,17 @@ class FakeScreenCapturer : public DesktopCapturer { } virtual void Capture(const DesktopRegion& region) OVERRIDE { - DesktopFrame* frame = - new BasicDesktopFrame(DesktopSize(kScreenWidth, kScreenHeight)); - uint32_t* data = reinterpret_cast<uint32_t*>(frame->data()); - for (int y = 0; y < kScreenHeight; ++y) { - for (int x = 0; x < kScreenWidth; ++x) { - *(data++) = GetFakeFramePixelValue(DesktopVector(x, y)); - } - } - - last_frame_.reset(SharedDesktopFrame::Wrap(frame)); - - callback_->OnCaptureCompleted(last_frame_->Share()); + callback_->OnCaptureCompleted(next_frame_.release()); } - // Returns last fake captured frame. - SharedDesktopFrame* last_frame() { return last_frame_.get(); } + void SetNextFrame(DesktopFrame* next_frame) { + next_frame_.reset(next_frame); + } private: Callback* callback_; - scoped_ptr<SharedDesktopFrame> last_frame_; + scoped_ptr<DesktopFrame> next_frame_; }; class FakeMouseMonitor : public MouseCursorMonitor { @@ -187,6 +189,20 @@ class DesktopAndCursorComposerTest : public testing::Test, scoped_ptr<DesktopFrame> frame_; }; +// Verify DesktopAndCursorComposer can handle the case when the screen capturer +// fails. +TEST_F(DesktopAndCursorComposerTest, Error) { + blender_.Start(this); + + fake_cursor_->SetHotspot(DesktopVector()); + fake_cursor_->SetState(MouseCursorMonitor::INSIDE, DesktopVector()); + fake_screen_->SetNextFrame(NULL); + + blender_.Capture(DesktopRegion()); + + EXPECT_EQ(frame_, static_cast<DesktopFrame*>(NULL)); +} + TEST_F(DesktopAndCursorComposerTest, Blend) { struct { int x, y; @@ -222,6 +238,10 @@ TEST_F(DesktopAndCursorComposerTest, Blend) { DesktopVector pos(tests[i].x, tests[i].y); fake_cursor_->SetState(state, pos); + scoped_ptr<SharedDesktopFrame> frame( + SharedDesktopFrame::Wrap(CreateTestFrame())); + fake_screen_->SetNextFrame(frame->Share()); + blender_.Capture(DesktopRegion()); VerifyFrame(*frame_, state, pos); @@ -229,9 +249,7 @@ TEST_F(DesktopAndCursorComposerTest, Blend) { // Verify that the cursor is erased before the frame buffer is returned to // the screen capturer. frame_.reset(); - VerifyFrame(*fake_screen_->last_frame(), - MouseCursorMonitor::OUTSIDE, - DesktopVector()); + VerifyFrame(*frame, MouseCursorMonitor::OUTSIDE, DesktopVector()); } } diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi index eb3bc9a29d1..6f4a083015a 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi @@ -36,6 +36,10 @@ "differ_block.h", "mac/desktop_configuration.h", "mac/desktop_configuration.mm", + "mac/desktop_configuration_monitor.h", + "mac/desktop_configuration_monitor.cc", + "mac/osx_version.h", + "mac/osx_version.cc", "mac/scoped_pixel_buffer_object.cc", "mac/scoped_pixel_buffer_object.h", "mouse_cursor.cc", @@ -65,9 +69,17 @@ "win/scoped_gdi_object.h", "win/scoped_thread_desktop.cc", "win/scoped_thread_desktop.h", + "win/screen_capturer_win_gdi.cc", + "win/screen_capturer_win_gdi.h", + "win/screen_capturer_win_magnifier.cc", + "win/screen_capturer_win_magnifier.h", + "win/screen_capture_utils.cc", + "win/screen_capture_utils.h", + "win/window_capture_utils.cc", + "win/window_capture_utils.h", "window_capturer.cc", "window_capturer.h", - "window_capturer_mac.cc", + "window_capturer_mac.mm", "window_capturer_win.cc", "window_capturer_x11.cc", "x11/shared_x_display.h", diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.cc index a4fa02547bd..105853bf94b 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.cc @@ -19,6 +19,10 @@ DesktopCaptureOptions::DesktopCaptureOptions() // XDamage is often broken, so don't use it by default. use_update_notifications_ = false; #endif + +#if defined(WEBRTC_WIN) + allow_use_magnification_api_ = false; +#endif } DesktopCaptureOptions::~DesktopCaptureOptions() {} @@ -29,6 +33,9 @@ DesktopCaptureOptions DesktopCaptureOptions::CreateDefault() { #if defined(USE_X11) result.set_x_display(SharedXDisplay::CreateDefault()); #endif +#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) + result.set_configuration_monitor(new DesktopConfigurationMonitor()); +#endif return result; } diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.h index f0c76b1733f..c6aabd4529d 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_options.h @@ -10,12 +10,17 @@ #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_OPTIONS_H_ #define WEBRTC_MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_OPTIONS_H_ -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/system_wrappers/interface/scoped_refptr.h" #if defined(USE_X11) #include "webrtc/modules/desktop_capture/x11/shared_x_display.h" #endif +#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) +#include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" +#endif + namespace webrtc { // An object that stores initialization parameters for screen and window @@ -38,6 +43,15 @@ class DesktopCaptureOptions { } #endif +#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) + DesktopConfigurationMonitor* configuration_monitor() const { + return configuration_monitor_; + } + void set_configuration_monitor(scoped_refptr<DesktopConfigurationMonitor> m) { + configuration_monitor_ = m; + } +#endif + // Flag indicating that the capturer should use screen change notifications. // Enables/disables use of XDAMAGE in the X11 capturer. bool use_update_notifications() const { return use_update_notifications_; } @@ -52,10 +66,27 @@ class DesktopCaptureOptions { disable_effects_ = disable_effects; } +#if defined(WEBRTC_WIN) + bool allow_use_magnification_api() const { + return allow_use_magnification_api_; + } + void set_allow_use_magnification_api(bool allow) { + allow_use_magnification_api_ = allow; + } +#endif + private: #if defined(USE_X11) scoped_refptr<SharedXDisplay> x_display_; #endif + +#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) + scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_; +#endif + +#if defined(WEBRTC_WIN) + bool allow_use_magnification_api_; +#endif bool use_update_notifications_; bool disable_effects_; }; diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_types.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_types.h index d43ec499191..3e417965531 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_types.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture_types.h @@ -26,6 +26,17 @@ typedef intptr_t WindowId; const WindowId kNullWindowId = 0; +// Type used to identify screens on the desktop. Values are platform-specific: +// - On Windows: integer display device index. +// - On OSX: CGDirectDisplayID cast to intptr_t. +// - On Linux (with X11): TBD. +typedef intptr_t ScreenId; + +// The screen id corresponds to all screen combined together. +const ScreenId kFullDesktopScreenId = -1; + +const ScreenId kInvalidScreenId = -2; + } // namespace webrtc #endif // WEBRTC_MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_TYPES_H_ diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h index bcb664ef859..7ad16364977 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h @@ -13,6 +13,8 @@ #include <stddef.h> +#include "webrtc/modules/desktop_capture/desktop_capture_types.h" + namespace webrtc { class DesktopFrame; @@ -52,6 +54,11 @@ class DesktopCapturer { // the top left corner of the capture target. Pending capture operations are // canceled when DesktopCapturer is deleted. virtual void Capture(const DesktopRegion& region) = 0; + + // Sets the window to be excluded from the captured image in the future + // Capture calls. Used to exclude the screenshare notification window for + // screen capturing. + virtual void SetExcludedWindow(WindowId window) {} }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_geometry.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_geometry.h index e51273d8d21..047eeec3d9c 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_geometry.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_geometry.h @@ -11,8 +11,8 @@ #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_ #define WEBRTC_MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_ +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" namespace webrtc { @@ -58,7 +58,7 @@ class DesktopSize { int32_t width() const { return width_; } int32_t height() const { return height_; } - bool is_empty() const { return width_ <= 0 && height_ <= 0; } + bool is_empty() const { return width_ <= 0 || height_ <= 0; } bool equals(const DesktopSize& other) const { return width_ == other.width_ && height_ == other.height_; diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h index fc7c6ed9e2d..c4528ae3496 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h @@ -14,8 +14,8 @@ #include <map> #include <vector> +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/desktop_capture/desktop_geometry.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/differ.h b/chromium/third_party/webrtc/modules/desktop_capture/differ.h index 8edce80b4ef..0b419d2dded 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/differ.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/differ.h @@ -76,7 +76,7 @@ class Differ { int bytes_per_row_; // Diff information for each block in the image. - scoped_array<DiffInfo> diff_info_; + scoped_ptr<DiffInfo[]> diff_info_; // Dimensions and total size of diff info array. int diff_info_width_; diff --git a/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc index 40fde4dbc4e..da1a21461dc 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc @@ -200,8 +200,8 @@ class DifferTest : public testing::Test { int buffer_size_; // Previous and current screen buffers. - scoped_array<uint8_t> prev_; - scoped_array<uint8_t> curr_; + scoped_ptr<uint8_t[]> prev_; + scoped_ptr<uint8_t[]> curr_; private: DISALLOW_COPY_AND_ASSIGN(DifferTest); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.h b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.h index 433040a04ee..bb2339bb0f1 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.h @@ -52,10 +52,18 @@ struct MacDesktopConfiguration { // increase as you move up the screen) or Carbon-style "top-down" coordinates. static MacDesktopConfiguration GetCurrent(Origin origin); - // Bounds of the desktop in Density-Independent Pixels (DIPs). + // Returns true if the given desktop configuration equals this one. + bool Equals(const MacDesktopConfiguration& other); + + // Returns the pointer to the display configuration with the specified id. + const MacDisplayConfiguration* FindDisplayConfigurationById( + CGDirectDisplayID id); + + // Bounds of the desktop excluding monitors with DPI settings different from + // the main monitor. In Density-Independent Pixels (DIPs). DesktopRect bounds; - // Bounds of the desktop in physical pixels. + // Same as bounds, but expressed in physical pixels. DesktopRect pixel_bounds; // Scale factor from DIPs to physical pixels. diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.mm b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.mm index a917b5dc052..35fa65be2d2 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.mm +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration.mm @@ -110,15 +110,8 @@ MacDesktopConfiguration MacDesktopConfiguration::GetCurrent(Origin origin) { MacDisplayConfiguration display_config = GetConfigurationForScreen([screens objectAtIndex: i]); - // Handling mixed-DPI is hard, so we only return displays that match the - // "primary" display's DPI. The primary display is always the first in the - // list returned by [NSScreen screens]. - if (i == 0) { + if (i == 0) desktop_config.dip_to_pixel_scale = display_config.dip_to_pixel_scale; - } else if (desktop_config.dip_to_pixel_scale != - display_config.dip_to_pixel_scale) { - continue; - } // Cocoa uses bottom-up coordinates, so if the caller wants top-down then // we need to invert the positions of secondary monitors relative to the @@ -126,21 +119,62 @@ MacDesktopConfiguration MacDesktopConfiguration::GetCurrent(Origin origin) { if (i > 0 && origin == TopLeftOrigin) { InvertRectYOrigin(desktop_config.displays[0].bounds, &display_config.bounds); - InvertRectYOrigin(desktop_config.displays[0].pixel_bounds, - &display_config.pixel_bounds); + // |display_bounds| is density dependent, so we need to convert the + // primay monitor's position into the secondary monitor's density context. + float scaling_factor = display_config.dip_to_pixel_scale / + desktop_config.displays[0].dip_to_pixel_scale; + DesktopRect primary_bounds = DesktopRect::MakeLTRB( + desktop_config.displays[0].pixel_bounds.left() * scaling_factor, + desktop_config.displays[0].pixel_bounds.top() * scaling_factor, + desktop_config.displays[0].pixel_bounds.right() * scaling_factor, + desktop_config.displays[0].pixel_bounds.bottom() * scaling_factor); + InvertRectYOrigin(primary_bounds, &display_config.pixel_bounds); } // Add the display to the configuration. desktop_config.displays.push_back(display_config); - // Update the desktop bounds to account for this display. - desktop_config.bounds = - JoinRects(desktop_config.bounds, display_config.bounds); - desktop_config.pixel_bounds = - JoinRects(desktop_config.pixel_bounds, display_config.pixel_bounds); + // Update the desktop bounds to account for this display, unless the current + // display uses different DPI settings. + if (display_config.dip_to_pixel_scale == + desktop_config.dip_to_pixel_scale) { + desktop_config.bounds = + JoinRects(desktop_config.bounds, display_config.bounds); + desktop_config.pixel_bounds = + JoinRects(desktop_config.pixel_bounds, display_config.pixel_bounds); + } } return desktop_config; } +// For convenience of comparing MacDisplayConfigurations in +// MacDesktopConfiguration::Equals. +bool operator==(const MacDisplayConfiguration& left, + const MacDisplayConfiguration& right) { + return left.id == right.id && + left.bounds.equals(right.bounds) && + left.pixel_bounds.equals(right.pixel_bounds) && + left.dip_to_pixel_scale == right.dip_to_pixel_scale; +} + +bool MacDesktopConfiguration::Equals(const MacDesktopConfiguration& other) { + return bounds.equals(other.bounds) && + pixel_bounds.equals(other.pixel_bounds) && + dip_to_pixel_scale == other.dip_to_pixel_scale && + displays == other.displays; +} + +// Finds the display configuration with the specified id. +const MacDisplayConfiguration* +MacDesktopConfiguration::FindDisplayConfigurationById( + CGDirectDisplayID id) { + for (MacDisplayConfigurations::const_iterator it = displays.begin(); + it != displays.end(); ++it) { + if (it->id == id) + return &(*it); + } + return NULL; +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc new file mode 100644 index 00000000000..f0d5c34be65 --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc @@ -0,0 +1,91 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" + +#include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" +#include "webrtc/system_wrappers/interface/event_wrapper.h" +#include "webrtc/system_wrappers/interface/logging.h" + +namespace webrtc { + +// The amount of time allowed for displays to reconfigure. +static const int64_t kDisplayConfigurationEventTimeoutMs = 10 * 1000; + +DesktopConfigurationMonitor::DesktopConfigurationMonitor() + : ref_count_(0), + display_configuration_capture_event_(EventWrapper::Create()) { + CGError err = CGDisplayRegisterReconfigurationCallback( + DesktopConfigurationMonitor::DisplaysReconfiguredCallback, this); + if (err != kCGErrorSuccess) { + LOG(LS_ERROR) << "CGDisplayRegisterReconfigurationCallback " << err; + abort(); + } + display_configuration_capture_event_->Set(); + + desktop_configuration_ = MacDesktopConfiguration::GetCurrent( + MacDesktopConfiguration::TopLeftOrigin); +} + +DesktopConfigurationMonitor::~DesktopConfigurationMonitor() { + CGError err = CGDisplayRemoveReconfigurationCallback( + DesktopConfigurationMonitor::DisplaysReconfiguredCallback, this); + if (err != kCGErrorSuccess) + LOG(LS_ERROR) << "CGDisplayRemoveReconfigurationCallback " << err; +} + +void DesktopConfigurationMonitor::Lock() { + if (!display_configuration_capture_event_->Wait( + kDisplayConfigurationEventTimeoutMs)) { + LOG_F(LS_ERROR) << "Event wait timed out."; + abort(); + } +} + +void DesktopConfigurationMonitor::Unlock() { + display_configuration_capture_event_->Set(); +} + +// static +void DesktopConfigurationMonitor::DisplaysReconfiguredCallback( + CGDirectDisplayID display, + CGDisplayChangeSummaryFlags flags, + void *user_parameter) { + DesktopConfigurationMonitor* monitor = + reinterpret_cast<DesktopConfigurationMonitor*>(user_parameter); + monitor->DisplaysReconfigured(display, flags); +} + +void DesktopConfigurationMonitor::DisplaysReconfigured( + CGDirectDisplayID display, + CGDisplayChangeSummaryFlags flags) { + if (flags & kCGDisplayBeginConfigurationFlag) { + if (reconfiguring_displays_.empty()) { + // If this is the first display to start reconfiguring then wait on + // |display_configuration_capture_event_| to block the capture thread + // from accessing display memory until the reconfiguration completes. + if (!display_configuration_capture_event_->Wait( + kDisplayConfigurationEventTimeoutMs)) { + LOG_F(LS_ERROR) << "Event wait timed out."; + abort(); + } + } + reconfiguring_displays_.insert(display); + } else { + reconfiguring_displays_.erase(display); + if (reconfiguring_displays_.empty()) { + desktop_configuration_ = MacDesktopConfiguration::GetCurrent( + MacDesktopConfiguration::TopLeftOrigin); + display_configuration_capture_event_->Set(); + } + } +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h new file mode 100644 index 00000000000..27143a84e1e --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_MONITOR_H_ +#define WEBRTC_MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_MONITOR_H_ + +#include <ApplicationServices/ApplicationServices.h> + +#include <set> + +#include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" +#include "webrtc/system_wrappers/interface/atomic32.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +class EventWrapper; + +// The class provides functions to synchronize capturing and display +// reconfiguring across threads, and the up-to-date MacDesktopConfiguration. +class DesktopConfigurationMonitor { + public: + DesktopConfigurationMonitor(); + // Acquires a lock on the current configuration. + void Lock(); + // Releases the lock previously acquired. + void Unlock(); + // Returns the current desktop configuration. Should only be called when the + // lock has been acquired. + const MacDesktopConfiguration& desktop_configuration() { + return desktop_configuration_; + } + + void AddRef() { ++ref_count_; } + void Release() { + if (--ref_count_ == 0) + delete this; + } + + private: + static void DisplaysReconfiguredCallback(CGDirectDisplayID display, + CGDisplayChangeSummaryFlags flags, + void *user_parameter); + ~DesktopConfigurationMonitor(); + + void DisplaysReconfigured(CGDirectDisplayID display, + CGDisplayChangeSummaryFlags flags); + + Atomic32 ref_count_; + std::set<CGDirectDisplayID> reconfiguring_displays_; + MacDesktopConfiguration desktop_configuration_; + scoped_ptr<EventWrapper> display_configuration_capture_event_; + + DISALLOW_COPY_AND_ASSIGN(DesktopConfigurationMonitor); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_MONITOR_H_ diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/osx_version.cc b/chromium/third_party/webrtc/modules/desktop_capture/mac/osx_version.cc new file mode 100644 index 00000000000..7466f20342b --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/osx_version.cc @@ -0,0 +1,54 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <sys/utsname.h> + +#include "webrtc/system_wrappers/interface/logging.h" + +namespace webrtc { + +namespace { + +int GetDarwinVersion() { + struct utsname uname_info; + if (uname(&uname_info) != 0) { + LOG(LS_ERROR) << "uname failed"; + return 0; + } + + if (strcmp(uname_info.sysname, "Darwin") != 0) + return 0; + + char* dot; + int result = strtol(uname_info.release, &dot, 10); + if (*dot != '.') { + LOG(LS_ERROR) << "Failed to parse version"; + return 0; + } + + return result; +} + +} // namespace + +bool IsOSLionOrLater() { + static int darwin_version = GetDarwinVersion(); + + // Verify that the version has been parsed correctly. + if (darwin_version < 6) { + LOG_F(LS_ERROR) << "Invalid Darwin version: " << darwin_version; + abort(); + } + + // Darwin major version 11 corresponds to OSX 10.7. + return darwin_version >= 11; +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/osx_version.h b/chromium/third_party/webrtc/modules/desktop_capture/mac/osx_version.h new file mode 100644 index 00000000000..0ba49a4e69e --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/osx_version.h @@ -0,0 +1,16 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +namespace webrtc { + +// Returns true if the OS version >= OSX 10.7. +bool IsOSLionOrLater(); + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h b/chromium/third_party/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h index 73d425aea23..4d1dd1ffd6a 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h @@ -14,7 +14,7 @@ #include <OpenGL/CGLMacro.h> #include <OpenGL/OpenGL.h> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.cc b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.cc index 3f1ab3ddf50..22a9c0ee8c8 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.cc @@ -10,10 +10,14 @@ #include "webrtc/modules/desktop_capture/mouse_cursor.h" +#include <assert.h> + #include "webrtc/modules/desktop_capture/desktop_frame.h" namespace webrtc { +MouseCursor::MouseCursor() {} + MouseCursor::MouseCursor(DesktopFrame* image, const DesktopVector& hotspot) : image_(image), hotspot_(hotspot) { @@ -25,8 +29,10 @@ MouseCursor::~MouseCursor() {} // static MouseCursor* MouseCursor::CopyOf(const MouseCursor& cursor) { - return new MouseCursor(BasicDesktopFrame::CopyOf(cursor.image()), - cursor.hotspot()); + return cursor.image() + ? new MouseCursor(BasicDesktopFrame::CopyOf(*cursor.image()), + cursor.hotspot()) + : new MouseCursor(); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.h b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.h index 4cf770830ce..22887f9ae47 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor.h @@ -11,8 +11,8 @@ #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_H_ #define WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_H_ +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/desktop_capture/desktop_geometry.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { @@ -21,13 +21,19 @@ class DesktopFrame; class MouseCursor { public: + MouseCursor(); + // Takes ownership of |image|. |hotspot| must be within |image| boundaries. MouseCursor(DesktopFrame* image, const DesktopVector& hotspot); + ~MouseCursor(); static MouseCursor* CopyOf(const MouseCursor& cursor); - const DesktopFrame& image() const { return *image_; } + void set_image(DesktopFrame* image) { image_.reset(image); } + const DesktopFrame* image() const { return image_.get(); } + + void set_hotspot(const DesktopVector& hotspot ) { hotspot_ = hotspot; } const DesktopVector& hotspot() const { return hotspot_; } private: diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor.h b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor.h index 9785b736b2a..24dfe72dfa7 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor.h @@ -69,7 +69,8 @@ class MouseCursorMonitor { // // TODO(sergeyu): Provide a way to select a specific screen. static MouseCursorMonitor* CreateForScreen( - const DesktopCaptureOptions& options); + const DesktopCaptureOptions& options, + ScreenId screen); // Initializes the monitor with the |callback|, which must remain valid until // capturer is destroyed. diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm index d3c02896f9e..e8806338197 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm @@ -15,35 +15,59 @@ #include <Cocoa/Cocoa.h> #include <CoreFoundation/CoreFoundation.h> +#include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" +#include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" +#include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" +#include "webrtc/modules/desktop_capture/mac/osx_version.h" #include "webrtc/modules/desktop_capture/mouse_cursor.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/scoped_refptr.h" namespace webrtc { class MouseCursorMonitorMac : public MouseCursorMonitor { public: - MouseCursorMonitorMac(CGWindowID window_id); + MouseCursorMonitorMac(const DesktopCaptureOptions& options, + CGWindowID window_id, + ScreenId screen_id); virtual ~MouseCursorMonitorMac(); virtual void Init(Callback* callback, Mode mode) OVERRIDE; virtual void Capture() OVERRIDE; private: + static void DisplaysReconfiguredCallback(CGDirectDisplayID display, + CGDisplayChangeSummaryFlags flags, + void *user_parameter); + void DisplaysReconfigured(CGDirectDisplayID display, + CGDisplayChangeSummaryFlags flags); + void CaptureImage(); + scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_; CGWindowID window_id_; - + ScreenId screen_id_; Callback* callback_; Mode mode_; - scoped_ptr<MouseCursor> last_cursor_; }; -MouseCursorMonitorMac::MouseCursorMonitorMac(CGWindowID window_id) - : window_id_(window_id), +MouseCursorMonitorMac::MouseCursorMonitorMac( + const DesktopCaptureOptions& options, + CGWindowID window_id, + ScreenId screen_id) + : configuration_monitor_(options.configuration_monitor()), + window_id_(window_id), + screen_id_(screen_id), callback_(NULL), mode_(SHAPE_AND_POSITION) { + assert(window_id == kCGNullWindowID || screen_id == kInvalidScreenId); + if (screen_id != kInvalidScreenId && !IsOSLionOrLater()) { + // Single screen capture is not supported on pre OS X 10.7. + screen_id_ = kFullDesktopScreenId; + } } MouseCursorMonitorMac::~MouseCursorMonitorMac() {} @@ -72,6 +96,21 @@ void MouseCursorMonitorMac::Capture() { DesktopVector position(gc_position.x, gc_position.y); + configuration_monitor_->Lock(); + MacDesktopConfiguration configuration = + configuration_monitor_->desktop_configuration(); + configuration_monitor_->Unlock(); + float scale = 1.0f; + + // Find the dpi to physical pixel scale for the screen where the mouse cursor + // is. + for (MacDisplayConfigurations::iterator it = configuration.displays.begin(); + it != configuration.displays.end(); ++it) { + if (it->bounds.Contains(position)) { + scale = it->dip_to_pixel_scale; + break; + } + } // If we are capturing cursor for a specific window then we need to figure out // if the current mouse position is covered by another window and also adjust // |position| to make it relative to the window origin. @@ -134,10 +173,8 @@ void MouseCursorMonitorMac::Capture() { } } } - CFRelease(window_array); } - if (!found_window) { // If we failed to get list of windows or the window wasn't in the list // pretend that the cursor is outside the window. This can happen, e.g. if @@ -145,8 +182,32 @@ void MouseCursorMonitorMac::Capture() { state = OUTSIDE; position.set(-1, -1); } + } else { + assert(screen_id_ >= kFullDesktopScreenId); + if (screen_id_ != kFullDesktopScreenId) { + // For single screen capturing, convert the position to relative to the + // target screen. + const MacDisplayConfiguration* config = + configuration.FindDisplayConfigurationById( + static_cast<CGDirectDisplayID>(screen_id_)); + if (config) { + if (!config->pixel_bounds.Contains(position)) + state = OUTSIDE; + position = position.subtract(config->bounds.top_left()); + } else { + // The target screen is no longer valid. + state = OUTSIDE; + position.set(-1, -1); + } + } else { + position.subtract(configuration.bounds.top_left()); + } + } + if (state == INSIDE) { + // Convert Density Independent Pixel to physical pixel. + position = DesktopVector(round(position.x() * scale), + round(position.y() * scale)); } - callback_->OnMouseCursorPosition(state, position); } @@ -182,10 +243,10 @@ void MouseCursorMonitorMac::CaptureImage() { // Compare the cursor with the previous one. if (last_cursor_.get() && - last_cursor_->image().size().equals(size) && + last_cursor_->image()->size().equals(size) && last_cursor_->hotspot().equals(hotspot) && - memcmp(last_cursor_->image().data(), src_data, - last_cursor_->image().stride() * size.height()) == 0) { + memcmp(last_cursor_->image()->data(), src_data, + last_cursor_->image()->stride() * size.height()) == 0) { return; } @@ -204,15 +265,15 @@ void MouseCursorMonitorMac::CaptureImage() { callback_->OnMouseCursor(cursor.release()); } - MouseCursorMonitor* MouseCursorMonitor::CreateForWindow( const DesktopCaptureOptions& options, WindowId window) { - return new MouseCursorMonitorMac(window); + return new MouseCursorMonitorMac(options, window, kInvalidScreenId); } MouseCursorMonitor* MouseCursorMonitor::CreateForScreen( - const DesktopCaptureOptions& options) { - return new MouseCursorMonitorMac(kCGNullWindowID); + const DesktopCaptureOptions& options, + ScreenId screen) { + return new MouseCursorMonitorMac(options, kCGNullWindowID, screen); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc index 7aa1b7141d5..3a632cc0d9c 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc @@ -10,7 +10,7 @@ #include "webrtc/modules/desktop_capture/mouse_cursor_monitor.h" -#include <cstddef> +#include <stddef.h> namespace webrtc { @@ -21,7 +21,8 @@ MouseCursorMonitor* MouseCursorMonitor::CreateForWindow( } MouseCursorMonitor* MouseCursorMonitor::CreateForScreen( - const DesktopCaptureOptions& options) { + const DesktopCaptureOptions& options, + ScreenId screen) { return NULL; } diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc index 18bf1ca40e0..c6af2b700e4 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc @@ -50,7 +50,12 @@ class MouseCursorMonitorTest : public testing::Test, // tests. Figure out how to do that without breaking other tests in // modules_unittests and enable these tests on Mac. // https://code.google.com/p/webrtc/issues/detail?id=2532 -#if !defined(WEBRTC_MAC) +// +// Disabled on Windows due to flake, see: +// https://code.google.com/p/webrtc/issues/detail?id=3408 +// Disabled on Linux due to flake, see: +// https://code.google.com/p/webrtc/issues/detail?id=3245 +#if !defined(WEBRTC_MAC) && !defined(WEBRTC_WIN) && !defined(WEBRTC_LINUX) #define MAYBE(x) x #else #define MAYBE(x) DISABLED_##x @@ -58,7 +63,7 @@ class MouseCursorMonitorTest : public testing::Test, TEST_F(MouseCursorMonitorTest, MAYBE(FromScreen)) { scoped_ptr<MouseCursorMonitor> capturer(MouseCursorMonitor::CreateForScreen( - DesktopCaptureOptions::CreateDefault())); + DesktopCaptureOptions::CreateDefault(), webrtc::kFullDesktopScreenId)); assert(capturer.get()); capturer->Init(this, MouseCursorMonitor::SHAPE_AND_POSITION); capturer->Capture(); @@ -66,10 +71,10 @@ TEST_F(MouseCursorMonitorTest, MAYBE(FromScreen)) { EXPECT_TRUE(cursor_image_.get()); EXPECT_GE(cursor_image_->hotspot().x(), 0); EXPECT_LE(cursor_image_->hotspot().x(), - cursor_image_->image().size().width()); + cursor_image_->image()->size().width()); EXPECT_GE(cursor_image_->hotspot().y(), 0); EXPECT_LE(cursor_image_->hotspot().y(), - cursor_image_->image().size().height()); + cursor_image_->image()->size().height()); EXPECT_TRUE(position_received_); EXPECT_EQ(MouseCursorMonitor::INSIDE, state_); @@ -109,7 +114,7 @@ TEST_F(MouseCursorMonitorTest, MAYBE(FromWindow)) { // Make sure that OnMouseCursorPosition() is not called in the SHAPE_ONLY mode. TEST_F(MouseCursorMonitorTest, MAYBE(ShapeOnly)) { scoped_ptr<MouseCursorMonitor> capturer(MouseCursorMonitor::CreateForScreen( - DesktopCaptureOptions::CreateDefault())); + DesktopCaptureOptions::CreateDefault(), webrtc::kFullDesktopScreenId)); assert(capturer.get()); capturer->Init(this, MouseCursorMonitor::SHAPE_ONLY); capturer->Capture(); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc index 82f7d2447fb..fd0b222a3a6 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc @@ -10,9 +10,12 @@ #include "webrtc/modules/desktop_capture/mouse_cursor_monitor.h" +#include <assert.h> + #include "webrtc/modules/desktop_capture/desktop_frame.h" #include "webrtc/modules/desktop_capture/mouse_cursor.h" #include "webrtc/modules/desktop_capture/win/cursor.h" +#include "webrtc/modules/desktop_capture/win/window_capture_utils.h" #include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -20,13 +23,20 @@ namespace webrtc { class MouseCursorMonitorWin : public MouseCursorMonitor { public: explicit MouseCursorMonitorWin(HWND window); + explicit MouseCursorMonitorWin(ScreenId screen); virtual ~MouseCursorMonitorWin(); virtual void Init(Callback* callback, Mode mode) OVERRIDE; virtual void Capture() OVERRIDE; private: + // Get the rect of the currently selected screen, relative to the primary + // display's top-left. If the screen is disabled or disconnected, or any error + // happens, an empty rect is returned. + DesktopRect GetScreenRect(); + HWND window_; + ScreenId screen_; Callback* callback_; Mode mode_; @@ -38,12 +48,23 @@ class MouseCursorMonitorWin : public MouseCursorMonitor { MouseCursorMonitorWin::MouseCursorMonitorWin(HWND window) : window_(window), + screen_(kInvalidScreenId), callback_(NULL), mode_(SHAPE_AND_POSITION), desktop_dc_(NULL), last_cursor_(NULL) { } +MouseCursorMonitorWin::MouseCursorMonitorWin(ScreenId screen) + : window_(NULL), + screen_(screen), + callback_(NULL), + mode_(SHAPE_AND_POSITION), + desktop_dc_(NULL), + last_cursor_(NULL) { + assert(screen >= kFullDesktopScreenId); +} + MouseCursorMonitorWin::~MouseCursorMonitorWin() { if (desktop_dc_) ReleaseDC(NULL, desktop_dc_); @@ -85,28 +106,68 @@ void MouseCursorMonitorWin::Capture() { bool inside = cursor_info.flags == CURSOR_SHOWING; if (window_) { - RECT rect; - if (!GetWindowRect(window_, &rect)) { + DesktopRect original_rect; + DesktopRect cropped_rect; + if (!GetCroppedWindowRect(window_, &cropped_rect, &original_rect)) { position.set(0, 0); inside = false; } else { - position = position.subtract(DesktopVector(rect.left, rect.top)); - if (inside) - inside = (window_ == WindowFromPoint(cursor_info.ptScreenPos)); + if (inside) { + HWND windowUnderCursor = WindowFromPoint(cursor_info.ptScreenPos); + inside = windowUnderCursor ? + (window_ == GetAncestor(windowUnderCursor, GA_ROOT)) : false; + } + position = position.subtract(cropped_rect.top_left()); } + } else { + assert(screen_ != kInvalidScreenId); + DesktopRect rect = GetScreenRect(); + if (inside) + inside = rect.Contains(position); + position = position.subtract(rect.top_left()); } callback_->OnMouseCursorPosition(inside ? INSIDE : OUTSIDE, position); } +DesktopRect MouseCursorMonitorWin::GetScreenRect() { + assert(screen_ != kInvalidScreenId); + if (screen_ == kFullDesktopScreenId) { + return DesktopRect::MakeXYWH( + GetSystemMetrics(SM_XVIRTUALSCREEN), + GetSystemMetrics(SM_YVIRTUALSCREEN), + GetSystemMetrics(SM_CXVIRTUALSCREEN), + GetSystemMetrics(SM_CYVIRTUALSCREEN)); + } + DISPLAY_DEVICE device; + device.cb = sizeof(device); + BOOL result = EnumDisplayDevices(NULL, screen_, &device, 0); + if (!result) + return DesktopRect(); + + DEVMODE device_mode; + device_mode.dmSize = sizeof(device_mode); + device_mode.dmDriverExtra = 0; + result = EnumDisplaySettingsEx( + device.DeviceName, ENUM_CURRENT_SETTINGS, &device_mode, 0); + if (!result) + return DesktopRect(); + + return DesktopRect::MakeXYWH(device_mode.dmPosition.x, + device_mode.dmPosition.y, + device_mode.dmPelsWidth, + device_mode.dmPelsHeight); +} + MouseCursorMonitor* MouseCursorMonitor::CreateForWindow( const DesktopCaptureOptions& options, WindowId window) { return new MouseCursorMonitorWin(reinterpret_cast<HWND>(window)); } MouseCursorMonitor* MouseCursorMonitor::CreateForScreen( - const DesktopCaptureOptions& options) { - return new MouseCursorMonitorWin(NULL); + const DesktopCaptureOptions& options, + ScreenId screen) { + return new MouseCursorMonitorWin(screen); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc index 9114b95f3b4..f09593db9d6 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc @@ -17,6 +17,7 @@ #include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" #include "webrtc/modules/desktop_capture/mouse_cursor.h" +#include "webrtc/modules/desktop_capture/x11/x_error_trap.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" @@ -145,10 +146,12 @@ void MouseCursorMonitorX11::Capture() { Window root_window; Window child_window; unsigned int mask; + + XErrorTrap error_trap(display()); Bool result = XQueryPointer(display(), window_, &root_window, &child_window, &root_x, &root_y, &win_x, &win_y, &mask); CursorState state; - if (!result) { + if (!result || error_trap.GetLastErrorAndDisable() != 0) { state = OUTSIDE; } else { // In screen mode (window_ == root_window) the mouse is always inside. @@ -214,7 +217,8 @@ MouseCursorMonitor* MouseCursorMonitor::CreateForWindow( } MouseCursorMonitor* MouseCursorMonitor::CreateForScreen( - const DesktopCaptureOptions& options) { + const DesktopCaptureOptions& options, + ScreenId screen) { if (!options.x_display()) return NULL; return new MouseCursorMonitorX11( diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc index b045f05267c..45a3507b923 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc @@ -10,6 +10,7 @@ #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h" +#include <assert.h> #include <algorithm> #include "webrtc/modules/desktop_capture/desktop_frame.h" diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h index 9dd3a1050a4..a8d40a72ef8 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h @@ -11,6 +11,9 @@ #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_H_ #define WEBRTC_MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_H_ +#include <vector> + +#include "webrtc/modules/desktop_capture/desktop_capture_types.h" #include "webrtc/modules/desktop_capture/desktop_capturer.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" @@ -40,6 +43,13 @@ struct MouseCursorShape; // Since data can be read while another capture action is happening. class ScreenCapturer : public DesktopCapturer { public: + // Use a struct to represent a screen although it has only an id for now, + // because we may want to add more fields (e.g. description) in the future. + struct Screen { + ScreenId id; + }; + typedef std::vector<Screen> ScreenList; + // Provides callbacks used by the capturer to pass captured video frames and // mouse cursor shapes to the processing pipeline. // @@ -78,6 +88,15 @@ class ScreenCapturer : public DesktopCapturer { // remain valid until the capturer is destroyed. virtual void SetMouseShapeObserver( MouseShapeObserver* mouse_shape_observer) = 0; + + // Get the list of screens (not containing kFullDesktopScreenId). Returns + // false in case of a failure. + virtual bool GetScreenList(ScreenList* screens) = 0; + + // Select the screen to be captured. Returns false in case of a failure (e.g. + // if there is no screen with the specified id). If this is never called, the + // full desktop is captured. + virtual bool SelectScreen(ScreenId id) = 0; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.cc index 75af043c84d..86761c170f0 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.cc @@ -10,6 +10,7 @@ #include "webrtc/modules/desktop_capture/screen_capturer_helper.h" +#include <assert.h> #include <algorithm> #include "webrtc/system_wrappers/interface/logging.h" diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm index 00639c73918..2d5733906f0 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm @@ -19,18 +19,18 @@ #include <IOKit/pwr_mgt/IOPMLib.h> #include <OpenGL/CGLMacro.h> #include <OpenGL/OpenGL.h> -#include <sys/utsname.h> #include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" #include "webrtc/modules/desktop_capture/desktop_geometry.h" #include "webrtc/modules/desktop_capture/desktop_region.h" #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" +#include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" +#include "webrtc/modules/desktop_capture/mac/osx_version.h" #include "webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h" #include "webrtc/modules/desktop_capture/mouse_cursor_shape.h" #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h" #include "webrtc/modules/desktop_capture/screen_capturer_helper.h" -#include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/tick_util.h" @@ -63,7 +63,8 @@ DesktopRect ScaleAndRoundCGRect(const CGRect& rect, float scale) { static_cast<int>(ceil((rect.origin.y + rect.size.height) * scale))); } -// Copy pixels in the |rect| from |src_place| to |dest_plane|. +// Copy pixels in the |rect| from |src_place| to |dest_plane|. |rect| should be +// relative to the origin of |src_plane| and |dest_plane|. void CopyRect(const uint8_t* src_plane, int src_plane_stride, uint8_t* dest_plane, @@ -87,46 +88,110 @@ void CopyRect(const uint8_t* src_plane, } } -int GetDarwinVersion() { - struct utsname uname_info; - if (uname(&uname_info) != 0) { - LOG(LS_ERROR) << "uname failed"; - return 0; +// Returns an array of CGWindowID for all the on-screen windows except +// |window_to_exclude|, or NULL if the window is not found or it fails. The +// caller should release the returned CFArrayRef. +CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) { + if (!window_to_exclude) + return NULL; + + CFArrayRef all_windows = CGWindowListCopyWindowInfo( + kCGWindowListOptionOnScreenOnly, kCGNullWindowID); + if (!all_windows) + return NULL; + + CFMutableArrayRef returned_array = CFArrayCreateMutable( + NULL, CFArrayGetCount(all_windows), NULL); + + bool found = false; + for (CFIndex i = 0; i < CFArrayGetCount(all_windows); ++i) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(all_windows, i)); + + CFNumberRef id_ref = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowNumber)); + + CGWindowID id; + CFNumberGetValue(id_ref, kCFNumberIntType, &id); + if (id == window_to_exclude) { + found = true; + continue; + } + CFArrayAppendValue(returned_array, reinterpret_cast<void *>(id)); } + CFRelease(all_windows); - if (strcmp(uname_info.sysname, "Darwin") != 0) - return 0; - - char* dot; - int result = strtol(uname_info.release, &dot, 10); - if (*dot != '.') { - LOG(LS_ERROR) << "Failed to parse version"; - return 0; + if (!found) { + CFRelease(returned_array); + returned_array = NULL; } - - return result; + return returned_array; } -bool IsOSLionOrLater() { - static int darwin_version = GetDarwinVersion(); - - // Verify that the version has been parsed correctly. - if (darwin_version < 6) { - LOG_F(LS_ERROR) << "Invalid Darwin version: " << darwin_version; - abort(); +// Returns the bounds of |window| in physical pixels, enlarged by a small amount +// on four edges to take account of the border/shadow effects. +DesktopRect GetExcludedWindowPixelBounds(CGWindowID window, + float dip_to_pixel_scale) { + // The amount of pixels to add to the actual window bounds to take into + // account of the border/shadow effects. + static const int kBorderEffectSize = 20; + CGRect rect; + CGWindowID ids[1]; + ids[0] = window; + + CFArrayRef window_id_array = + CFArrayCreate(NULL, reinterpret_cast<const void **>(&ids), 1, NULL); + CFArrayRef window_array = + CGWindowListCreateDescriptionFromArray(window_id_array); + + if (CFArrayGetCount(window_array) > 0) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, 0)); + CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>( + CFDictionaryGetValue(window, kCGWindowBounds)); + CGRectMakeWithDictionaryRepresentation(bounds_ref, &rect); } - // Darwin major version 11 corresponds to OSX 10.7. - return darwin_version >= 11; + CFRelease(window_id_array); + CFRelease(window_array); + + rect.origin.x -= kBorderEffectSize; + rect.origin.y -= kBorderEffectSize; + rect.size.width += kBorderEffectSize * 2; + rect.size.height += kBorderEffectSize * 2; + // |rect| is in DIP, so convert to physical pixels. + return ScaleAndRoundCGRect(rect, dip_to_pixel_scale); } -// The amount of time allowed for displays to reconfigure. -const int64_t kDisplayConfigurationEventTimeoutMs = 10 * 1000; +// Create an image of the given region using the given |window_list|. +// |pixel_bounds| should be in the primary display's coordinate in physical +// pixels. The caller should release the returned CGImageRef and CFDataRef. +CGImageRef CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds, + float dip_to_pixel_scale, + CFArrayRef window_list, + CFDataRef* data_ref) { + CGRect window_bounds; + // The origin is in DIP while the size is in physical pixels. That's what + // CGWindowListCreateImageFromArray expects. + window_bounds.origin.x = pixel_bounds.left() / dip_to_pixel_scale; + window_bounds.origin.y = pixel_bounds.top() / dip_to_pixel_scale; + window_bounds.size.width = pixel_bounds.width(); + window_bounds.size.height = pixel_bounds.height(); + + CGImageRef excluded_image = CGWindowListCreateImageFromArray( + window_bounds, window_list, kCGWindowImageDefault); + + CGDataProviderRef provider = CGImageGetDataProvider(excluded_image); + *data_ref = CGDataProviderCopyData(provider); + assert(*data_ref); + return excluded_image; +} // A class to perform video frame capturing for mac. class ScreenCapturerMac : public ScreenCapturer { public: - ScreenCapturerMac(); + explicit ScreenCapturerMac( + scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor); virtual ~ScreenCapturerMac(); bool Init(); @@ -134,8 +199,11 @@ class ScreenCapturerMac : public ScreenCapturer { // Overridden from ScreenCapturer: virtual void Start(Callback* callback) OVERRIDE; virtual void Capture(const DesktopRegion& region) OVERRIDE; + virtual void SetExcludedWindow(WindowId window) OVERRIDE; virtual void SetMouseShapeObserver( MouseShapeObserver* mouse_shape_observer) OVERRIDE; + virtual bool GetScreenList(ScreenList* screens) OVERRIDE; + virtual bool SelectScreen(ScreenId id) OVERRIDE; private: void CaptureCursor(); @@ -145,7 +213,8 @@ class ScreenCapturerMac : public ScreenCapturer { void GlBlitSlow(const DesktopFrame& frame); void CgBlitPreLion(const DesktopFrame& frame, const DesktopRegion& region); - void CgBlitPostLion(const DesktopFrame& frame, + // Returns false if the selected screen is no longer valid. + bool CgBlitPostLion(const DesktopFrame& frame, const DesktopRegion& region); // Called when the screen configuration is changed. @@ -158,8 +227,6 @@ class ScreenCapturerMac : public ScreenCapturer { void ScreenUpdateMove(CGScreenUpdateMoveDelta delta, size_t count, const CGRect *rect_array); - void DisplaysReconfigured(CGDirectDisplayID display, - CGDisplayChangeSummaryFlags flags); static void ScreenRefreshCallback(CGRectCount count, const CGRect *rect_array, void *user_parameter); @@ -167,12 +234,10 @@ class ScreenCapturerMac : public ScreenCapturer { size_t count, const CGRect *rect_array, void *user_parameter); - static void DisplaysReconfiguredCallback(CGDirectDisplayID display, - CGDisplayChangeSummaryFlags flags, - void *user_parameter); - void ReleaseBuffers(); + DesktopFrame* CreateFrame(); + Callback* callback_; MouseShapeObserver* mouse_shape_observer_; @@ -185,6 +250,16 @@ class ScreenCapturerMac : public ScreenCapturer { // Current display configuration. MacDesktopConfiguration desktop_config_; + // Currently selected display, or 0 if the full desktop is selected. On OS X + // 10.6 and before, this is always 0. + CGDirectDisplayID current_display_; + + // The physical pixel bounds of the current screen. + DesktopRect screen_pixel_bounds_; + + // The dip to physical pixel scale of the current screen. + float dip_to_pixel_scale_; + // A thread-safe list of invalid rectangles, and the size of the most // recently captured screen. ScreenCapturerHelper helper_; @@ -195,13 +270,8 @@ class ScreenCapturerMac : public ScreenCapturer { // Contains an invalid region from the previous capture. DesktopRegion last_invalid_region_; - // Used to ensure that frame captures do not take place while displays - // are being reconfigured. - scoped_ptr<EventWrapper> display_configuration_capture_event_; - - // Records the Ids of attached displays which are being reconfigured. - // Accessed on the thread on which we are notified of display events. - std::set<CGDirectDisplayID> reconfiguring_displays_; + // Monitoring display reconfiguration. + scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_; // Power management assertion to prevent the screen from sleeping. IOPMAssertionID power_assertion_id_display_; @@ -217,6 +287,8 @@ class ScreenCapturerMac : public ScreenCapturer { void* opengl_library_; CGLSetFullScreenFunc cgl_set_full_screen_; + CGWindowID excluded_window_; + DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac); }; @@ -243,24 +315,14 @@ class InvertedDesktopFrame : public DesktopFrame { DISALLOW_COPY_AND_ASSIGN(InvertedDesktopFrame); }; -DesktopFrame* CreateFrame( - const MacDesktopConfiguration& desktop_config) { - - DesktopSize size(desktop_config.pixel_bounds.width(), - desktop_config.pixel_bounds.height()); - scoped_ptr<DesktopFrame> frame(new BasicDesktopFrame(size)); - - frame->set_dpi(DesktopVector( - kStandardDPI * desktop_config.dip_to_pixel_scale, - kStandardDPI * desktop_config.dip_to_pixel_scale)); - return frame.release(); -} - -ScreenCapturerMac::ScreenCapturerMac() +ScreenCapturerMac::ScreenCapturerMac( + scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor) : callback_(NULL), mouse_shape_observer_(NULL), cgl_context_(NULL), - display_configuration_capture_event_(EventWrapper::Create()), + current_display_(0), + dip_to_pixel_scale_(1.0f), + desktop_config_monitor_(desktop_config_monitor), power_assertion_id_display_(kIOPMNullAssertionID), power_assertion_id_user_(kIOPMNullAssertionID), app_services_library_(NULL), @@ -268,8 +330,8 @@ ScreenCapturerMac::ScreenCapturerMac() cg_display_bytes_per_row_(NULL), cg_display_bits_per_pixel_(NULL), opengl_library_(NULL), - cgl_set_full_screen_(NULL) { - display_configuration_capture_event_->Set(); + cgl_set_full_screen_(NULL), + excluded_window_(0) { } ScreenCapturerMac::~ScreenCapturerMac() { @@ -284,11 +346,6 @@ ScreenCapturerMac::~ScreenCapturerMac() { ReleaseBuffers(); UnregisterRefreshAndMoveHandlers(); - CGError err = CGDisplayRemoveReconfigurationCallback( - ScreenCapturerMac::DisplaysReconfiguredCallback, this); - if (err != kCGErrorSuccess) - LOG(LS_ERROR) << "CGDisplayRemoveReconfigurationCallback " << err; - dlclose(app_services_library_); dlclose(opengl_library_); } @@ -297,14 +354,9 @@ bool ScreenCapturerMac::Init() { if (!RegisterRefreshAndMoveHandlers()) { return false; } - - CGError err = CGDisplayRegisterReconfigurationCallback( - ScreenCapturerMac::DisplaysReconfiguredCallback, this); - if (err != kCGErrorSuccess) { - LOG(LS_ERROR) << "CGDisplayRegisterReconfigurationCallback " << err; - return false; - } - + desktop_config_monitor_->Lock(); + desktop_config_ = desktop_config_monitor_->desktop_configuration(); + desktop_config_monitor_->Unlock(); ScreenConfigurationChanged(); return true; } @@ -343,20 +395,22 @@ void ScreenCapturerMac::Start(Callback* callback) { &power_assertion_id_user_); } -void ScreenCapturerMac::Capture( - const DesktopRegion& region_to_capture) { +void ScreenCapturerMac::Capture(const DesktopRegion& region_to_capture) { TickTime capture_start_time = TickTime::Now(); queue_.MoveToNextFrame(); - // Wait until the display configuration is stable. If one or more displays - // are reconfiguring then |display_configuration_capture_event_| will not be - // set until the reconfiguration completes. - // TODO(wez): Replace this with an early-exit (See crbug.com/104542). - if (!display_configuration_capture_event_->Wait( - kDisplayConfigurationEventTimeoutMs)) { - LOG_F(LS_ERROR) << "Event wait timed out."; - abort(); + desktop_config_monitor_->Lock(); + MacDesktopConfiguration new_config = + desktop_config_monitor_->desktop_configuration(); + if (!desktop_config_.Equals(new_config)) { + desktop_config_ = new_config; + // If the display configuraiton has changed then refresh capturer data + // structures. Occasionally, the refresh and move handlers are lost when + // the screen mode changes, so re-register them here. + UnregisterRefreshAndMoveHandlers(); + RegisterRefreshAndMoveHandlers(); + ScreenConfigurationChanged(); } DesktopRegion region; @@ -366,7 +420,7 @@ void ScreenCapturerMac::Capture( // Note that we can't reallocate other buffers at this point, since the caller // may still be reading from them. if (!queue_.current_frame()) - queue_.ReplaceCurrentFrame(CreateFrame(desktop_config_)); + queue_.ReplaceCurrentFrame(CreateFrame()); DesktopFrame* current_frame = queue_.current_frame(); @@ -374,7 +428,10 @@ void ScreenCapturerMac::Capture( if (IsOSLionOrLater()) { // Lion requires us to use their new APIs for doing screen capture. These // APIS currently crash on 10.6.8 if there is no monitor attached. - CgBlitPostLion(*current_frame, region); + if (!CgBlitPostLion(*current_frame, region)) { + callback_->OnCaptureCompleted(NULL); + return; + } } else if (cgl_context_) { flip = true; if (pixel_buffer_object_.get() != 0) { @@ -398,7 +455,7 @@ void ScreenCapturerMac::Capture( // Signal that we are done capturing data from the display framebuffer, // and accessing display structures. - display_configuration_capture_event_->Set(); + desktop_config_monitor_->Unlock(); // Capture the current cursor shape and notify |callback_| if it has changed. CaptureCursor(); @@ -408,6 +465,10 @@ void ScreenCapturerMac::Capture( callback_->OnCaptureCompleted(new_frame); } +void ScreenCapturerMac::SetExcludedWindow(WindowId window) { + excluded_window_ = window; +} + void ScreenCapturerMac::SetMouseShapeObserver( MouseShapeObserver* mouse_shape_observer) { assert(!mouse_shape_observer_); @@ -415,6 +476,47 @@ void ScreenCapturerMac::SetMouseShapeObserver( mouse_shape_observer_ = mouse_shape_observer; } +bool ScreenCapturerMac::GetScreenList(ScreenList* screens) { + assert(screens->size() == 0); + if (!IsOSLionOrLater()) { + // Single monitor cast is not supported on pre OS X 10.7. + Screen screen; + screen.id = kFullDesktopScreenId; + screens->push_back(screen); + return true; + } + + for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin(); + it != desktop_config_.displays.end(); ++it) { + Screen screen; + screen.id = static_cast<ScreenId>(it->id); + screens->push_back(screen); + } + return true; +} + +bool ScreenCapturerMac::SelectScreen(ScreenId id) { + if (!IsOSLionOrLater()) { + // Ignore the screen selection on unsupported OS. + assert(!current_display_); + return id == kFullDesktopScreenId; + } + + if (id == kFullDesktopScreenId) { + current_display_ = 0; + } else { + const MacDisplayConfiguration* config = + desktop_config_.FindDisplayConfigurationById( + static_cast<CGDirectDisplayID>(id)); + if (!config) + return false; + current_display_ = config->id; + } + + ScreenConfigurationChanged(); + return true; +} + void ScreenCapturerMac::CaptureCursor() { if (!mouse_shape_observer_) return; @@ -608,7 +710,7 @@ void ScreenCapturerMac::CgBlitPreLion(const DesktopFrame& frame, } } -void ScreenCapturerMac::CgBlitPostLion(const DesktopFrame& frame, +bool ScreenCapturerMac::CgBlitPostLion(const DesktopFrame& frame, const DesktopRegion& region) { // Copy the entire contents of the previous capture buffer, to capture over. // TODO(wez): Get rid of this as per crbug.com/145064, or implement @@ -619,13 +721,40 @@ void ScreenCapturerMac::CgBlitPostLion(const DesktopFrame& frame, frame.stride() * frame.size().height()); } - for (size_t i = 0; i < desktop_config_.displays.size(); ++i) { - const MacDisplayConfiguration& display_config = desktop_config_.displays[i]; + MacDisplayConfigurations displays_to_capture; + if (current_display_) { + // Capturing a single screen. Note that the screen id may change when + // screens are added or removed. + const MacDisplayConfiguration* config = + desktop_config_.FindDisplayConfigurationById(current_display_); + if (config) { + displays_to_capture.push_back(*config); + } else { + LOG(LS_ERROR) << "The selected screen cannot be found for capturing."; + return false; + } + } else { + // Capturing the whole desktop. + displays_to_capture = desktop_config_.displays; + } + // Create the window list once for all displays. + CFArrayRef window_list = CreateWindowListWithExclusion(excluded_window_); + + for (size_t i = 0; i < displays_to_capture.size(); ++i) { + const MacDisplayConfiguration& display_config = displays_to_capture[i]; + + // Capturing mixed-DPI on one surface is hard, so we only return displays + // that match the "primary" display's DPI. The primary display is always + // the first in the list. + if (i > 0 && display_config.dip_to_pixel_scale != + displays_to_capture[0].dip_to_pixel_scale) { + continue; + } // Determine the display's position relative to the desktop, in pixels. DesktopRect display_bounds = display_config.pixel_bounds; - display_bounds.Translate(-desktop_config_.pixel_bounds.left(), - -desktop_config_.pixel_bounds.top()); + display_bounds.Translate(-screen_pixel_bounds_.left(), + -screen_pixel_bounds_.top()); // Determine which parts of the blit region, if any, lay within the monitor. DesktopRegion copy_region = region; @@ -636,6 +765,26 @@ void ScreenCapturerMac::CgBlitPostLion(const DesktopFrame& frame, // Translate the region to be copied into display-relative coordinates. copy_region.Translate(-display_bounds.left(), -display_bounds.top()); + DesktopRect excluded_window_bounds; + CGImageRef excluded_image = NULL; + CFDataRef excluded_window_region_data = NULL; + if (excluded_window_ && window_list) { + // Get the region of the excluded window relative the primary display. + excluded_window_bounds = GetExcludedWindowPixelBounds( + excluded_window_, display_config.dip_to_pixel_scale); + excluded_window_bounds.IntersectWith(display_config.pixel_bounds); + + // Create the image under the excluded window first, because it's faster + // than captuing the whole display. + if (!excluded_window_bounds.is_empty()) { + excluded_image = CreateExcludedWindowRegionImage( + excluded_window_bounds, + display_config.dip_to_pixel_scale, + window_list, + &excluded_window_region_data); + } + } + // Create an image containing a snapshot of the display. CGImageRef image = CGDisplayCreateImage(display_config.id); if (image == NULL) @@ -665,26 +814,58 @@ void ScreenCapturerMac::CgBlitPostLion(const DesktopFrame& frame, i.rect()); } + // Copy the region of the excluded window to the frame. + if (excluded_image) { + assert(excluded_window_region_data); + display_base_address = CFDataGetBytePtr(excluded_window_region_data); + src_bytes_per_row = CGImageGetBytesPerRow(excluded_image); + + // Translate the bounds relative to the desktop, because |frame| data + // starts from the desktop top-left corner. + DesktopRect window_bounds_relative_to_desktop(excluded_window_bounds); + window_bounds_relative_to_desktop.Translate( + -screen_pixel_bounds_.left(), -screen_pixel_bounds_.top()); + out_ptr = frame.data() + + (window_bounds_relative_to_desktop.left() * src_bytes_per_pixel) + + (window_bounds_relative_to_desktop.top() * frame.stride()); + + CopyRect(display_base_address, + src_bytes_per_row, + out_ptr, + frame.stride(), + src_bytes_per_pixel, + DesktopRect::MakeSize(excluded_window_bounds.size())); + CFRelease(excluded_window_region_data); + CFRelease(excluded_image); + } + CFRelease(data); CFRelease(image); } + if (window_list) + CFRelease(window_list); + return true; } void ScreenCapturerMac::ScreenConfigurationChanged() { + if (current_display_) { + const MacDisplayConfiguration* config = + desktop_config_.FindDisplayConfigurationById(current_display_); + screen_pixel_bounds_ = config ? config->pixel_bounds : DesktopRect(); + dip_to_pixel_scale_ = config ? config->dip_to_pixel_scale : 1.0f; + } else { + screen_pixel_bounds_ = desktop_config_.pixel_bounds; + dip_to_pixel_scale_ = desktop_config_.dip_to_pixel_scale; + } + // Release existing buffers, which will be of the wrong size. ReleaseBuffers(); // Clear the dirty region, in case the display is down-sizing. helper_.ClearInvalidRegion(); - // Refresh the cached desktop configuration. - desktop_config_ = MacDesktopConfiguration::GetCurrent( - MacDesktopConfiguration::TopLeftOrigin); - // Re-mark the entire desktop as dirty. - helper_.InvalidateScreen( - DesktopSize(desktop_config_.pixel_bounds.width(), - desktop_config_.pixel_bounds.height())); + helper_.InvalidateScreen(screen_pixel_bounds_.size()); // Make sure the frame buffers will be reallocated. queue_.Reset(); @@ -765,8 +946,8 @@ void ScreenCapturerMac::ScreenConfigurationChanged() { (*cgl_set_full_screen_)(cgl_context_); CGLSetCurrentContext(cgl_context_); - size_t buffer_size = desktop_config_.pixel_bounds.width() * - desktop_config_.pixel_bounds.height() * + size_t buffer_size = screen_pixel_bounds_.width() * + screen_pixel_bounds_.height() * sizeof(uint32_t); pixel_buffer_object_.Init(cgl_context_, buffer_size); } @@ -798,20 +979,17 @@ void ScreenCapturerMac::UnregisterRefreshAndMoveHandlers() { void ScreenCapturerMac::ScreenRefresh(CGRectCount count, const CGRect* rect_array) { - if (desktop_config_.pixel_bounds.is_empty()) + if (screen_pixel_bounds_.is_empty()) return; DesktopRegion region; - + DesktopVector translate_vector = + DesktopVector().subtract(screen_pixel_bounds_.top_left()); for (CGRectCount i = 0; i < count; ++i) { // Convert from Density-Independent Pixel to physical pixel coordinates. - DesktopRect rect = - ScaleAndRoundCGRect(rect_array[i], desktop_config_.dip_to_pixel_scale); - + DesktopRect rect = ScaleAndRoundCGRect(rect_array[i], dip_to_pixel_scale_); // Translate from local desktop to capturer framebuffer coordinates. - rect.Translate(-desktop_config_.pixel_bounds.left(), - -desktop_config_.pixel_bounds.top()); - + rect.Translate(translate_vector); region.AddRect(rect); } @@ -831,45 +1009,12 @@ void ScreenCapturerMac::ScreenUpdateMove(CGScreenUpdateMoveDelta delta, ScreenRefresh(count, refresh_rects); } -void ScreenCapturerMac::DisplaysReconfigured( - CGDirectDisplayID display, - CGDisplayChangeSummaryFlags flags) { - if (flags & kCGDisplayBeginConfigurationFlag) { - if (reconfiguring_displays_.empty()) { - // If this is the first display to start reconfiguring then wait on - // |display_configuration_capture_event_| to block the capture thread - // from accessing display memory until the reconfiguration completes. - if (!display_configuration_capture_event_->Wait( - kDisplayConfigurationEventTimeoutMs)) { - LOG_F(LS_ERROR) << "Event wait timed out."; - abort(); - } - } - - reconfiguring_displays_.insert(display); - } else { - reconfiguring_displays_.erase(display); - - if (reconfiguring_displays_.empty()) { - // If no other displays are reconfiguring then refresh capturer data - // structures and un-block the capturer thread. Occasionally, the - // refresh and move handlers are lost when the screen mode changes, - // so re-register them here (the same does not appear to be true for - // the reconfiguration handler itself). - UnregisterRefreshAndMoveHandlers(); - RegisterRefreshAndMoveHandlers(); - ScreenConfigurationChanged(); - display_configuration_capture_event_->Set(); - } - } -} - void ScreenCapturerMac::ScreenRefreshCallback(CGRectCount count, const CGRect* rect_array, void* user_parameter) { ScreenCapturerMac* capturer = reinterpret_cast<ScreenCapturerMac*>(user_parameter); - if (capturer->desktop_config_.pixel_bounds.is_empty()) + if (capturer->screen_pixel_bounds_.is_empty()) capturer->ScreenConfigurationChanged(); capturer->ScreenRefresh(count, rect_array); } @@ -884,20 +1029,24 @@ void ScreenCapturerMac::ScreenUpdateMoveCallback( capturer->ScreenUpdateMove(delta, count, rect_array); } -void ScreenCapturerMac::DisplaysReconfiguredCallback( - CGDirectDisplayID display, - CGDisplayChangeSummaryFlags flags, - void* user_parameter) { - ScreenCapturerMac* capturer = - reinterpret_cast<ScreenCapturerMac*>(user_parameter); - capturer->DisplaysReconfigured(display, flags); +DesktopFrame* ScreenCapturerMac::CreateFrame() { + scoped_ptr<DesktopFrame> frame( + new BasicDesktopFrame(screen_pixel_bounds_.size())); + + frame->set_dpi(DesktopVector(kStandardDPI * dip_to_pixel_scale_, + kStandardDPI * dip_to_pixel_scale_)); + return frame.release(); } } // namespace // static ScreenCapturer* ScreenCapturer::Create(const DesktopCaptureOptions& options) { - scoped_ptr<ScreenCapturerMac> capturer(new ScreenCapturerMac()); + if (!options.configuration_monitor()) + return NULL; + + scoped_ptr<ScreenCapturerMac> capturer( + new ScreenCapturerMac(options.configuration_monitor())); if (!capturer->Init()) capturer.reset(); return capturer.release(); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h index 17673b5cc0e..aa0e808eb69 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h @@ -26,6 +26,8 @@ class MockScreenCapturer : public ScreenCapturer { MOCK_METHOD1(Capture, void(const DesktopRegion& region)); MOCK_METHOD1(SetMouseShapeObserver, void( MouseShapeObserver* mouse_shape_observer)); + MOCK_METHOD1(GetScreenList, bool(ScreenList* screens)); + MOCK_METHOD1(SelectScreen, bool(ScreenId id)); private: DISALLOW_COPY_AND_ASSIGN(MockScreenCapturer); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc index b4ae128085a..50ff7a2853a 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc @@ -59,6 +59,15 @@ SharedMemory* ScreenCapturerTest::CreateSharedMemory(size_t size) { return new FakeSharedMemory(new char[size], size); } +TEST_F(ScreenCapturerTest, GetScreenListAndSelectScreen) { + webrtc::ScreenCapturer::ScreenList screens; + EXPECT_TRUE(capturer_->GetScreenList(&screens)); + for(webrtc::ScreenCapturer::ScreenList::iterator it = screens.begin(); + it != screens.end(); ++it) { + EXPECT_TRUE(capturer_->SelectScreen(it->id)); + } +} + TEST_F(ScreenCapturerTest, StartCapturer) { capturer_->SetMouseShapeObserver(&mouse_observer_); capturer_->Start(&callback_); @@ -97,7 +106,7 @@ TEST_F(ScreenCapturerTest, Capture) { delete frame; } -#if defined(OS_WIN) +#if defined(WEBRTC_WIN) TEST_F(ScreenCapturerTest, UseSharedBuffers) { DesktopFrame* frame = NULL; @@ -120,6 +129,20 @@ TEST_F(ScreenCapturerTest, UseSharedBuffers) { delete frame; } -#endif // defined(OS_WIN) +TEST_F(ScreenCapturerTest, UseMagnifier) { + DesktopCaptureOptions options(DesktopCaptureOptions::CreateDefault()); + options.set_allow_use_magnification_api(true); + capturer_.reset(ScreenCapturer::Create(options)); + + DesktopFrame* frame = NULL; + EXPECT_CALL(callback_, OnCaptureCompleted(_)).WillOnce(SaveArg<0>(&frame)); + + capturer_->Start(&callback_); + capturer_->Capture(DesktopRegion()); + ASSERT_TRUE(frame); + delete frame; +} + +#endif // defined(WEBRTC_WIN) } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_win.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_win.cc index a9bcd48f9c1..5950795d470 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_win.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_win.cc @@ -10,364 +10,20 @@ #include "webrtc/modules/desktop_capture/screen_capturer.h" -#include <windows.h> - #include "webrtc/modules/desktop_capture/desktop_capture_options.h" -#include "webrtc/modules/desktop_capture/desktop_frame.h" -#include "webrtc/modules/desktop_capture/desktop_frame_win.h" -#include "webrtc/modules/desktop_capture/desktop_region.h" -#include "webrtc/modules/desktop_capture/differ.h" -#include "webrtc/modules/desktop_capture/mouse_cursor.h" -#include "webrtc/modules/desktop_capture/mouse_cursor_shape.h" -#include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h" -#include "webrtc/modules/desktop_capture/screen_capturer_helper.h" -#include "webrtc/modules/desktop_capture/win/cursor.h" -#include "webrtc/modules/desktop_capture/win/desktop.h" -#include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h" -#include "webrtc/system_wrappers/interface/logging.h" -#include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/tick_util.h" +#include "webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h" +#include "webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h" namespace webrtc { -namespace { - -// Constants from dwmapi.h. -const UINT DWM_EC_DISABLECOMPOSITION = 0; -const UINT DWM_EC_ENABLECOMPOSITION = 1; - -typedef HRESULT (WINAPI * DwmEnableCompositionFunc)(UINT); - -const wchar_t kDwmapiLibraryName[] = L"dwmapi.dll"; - -// ScreenCapturerWin captures 32bit RGB using GDI. -// -// ScreenCapturerWin is double-buffered as required by ScreenCapturer. -class ScreenCapturerWin : public ScreenCapturer { - public: - ScreenCapturerWin(const DesktopCaptureOptions& options); - virtual ~ScreenCapturerWin(); - - // Overridden from ScreenCapturer: - virtual void Start(Callback* callback) OVERRIDE; - virtual void Capture(const DesktopRegion& region) OVERRIDE; - virtual void SetMouseShapeObserver( - MouseShapeObserver* mouse_shape_observer) OVERRIDE; - - private: - // Make sure that the device contexts match the screen configuration. - void PrepareCaptureResources(); - - // Captures the current screen contents into the current buffer. - void CaptureImage(); - - // Capture the current cursor shape. - void CaptureCursor(); - - Callback* callback_; - MouseShapeObserver* mouse_shape_observer_; - - // A thread-safe list of invalid rectangles, and the size of the most - // recently captured screen. - ScreenCapturerHelper helper_; - - // Snapshot of the last cursor bitmap we sent to the client. This is used - // to diff against the current cursor so we only send a cursor-change - // message when the shape has changed. - MouseCursorShape last_cursor_; - - ScopedThreadDesktop desktop_; - - // GDI resources used for screen capture. - HDC desktop_dc_; - HDC memory_dc_; - - // Queue of the frames buffers. - ScreenCaptureFrameQueue queue_; - - // Rectangle describing the bounds of the desktop device context. - DesktopRect desktop_dc_rect_; - - // Class to calculate the difference between two screen bitmaps. - scoped_ptr<Differ> differ_; - - HMODULE dwmapi_library_; - DwmEnableCompositionFunc composition_func_; - - // Used to suppress duplicate logging of SetThreadExecutionState errors. - bool set_thread_execution_state_failed_; - - DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWin); -}; - -ScreenCapturerWin::ScreenCapturerWin(const DesktopCaptureOptions& options) - : callback_(NULL), - mouse_shape_observer_(NULL), - desktop_dc_(NULL), - memory_dc_(NULL), - dwmapi_library_(NULL), - composition_func_(NULL), - set_thread_execution_state_failed_(false) { - if (options.disable_effects()) { - // Load dwmapi.dll dynamically since it is not available on XP. - if (!dwmapi_library_) - dwmapi_library_ = LoadLibrary(kDwmapiLibraryName); - - if (dwmapi_library_) { - composition_func_ = reinterpret_cast<DwmEnableCompositionFunc>( - GetProcAddress(dwmapi_library_, "DwmEnableComposition")); - } - } -} - -ScreenCapturerWin::~ScreenCapturerWin() { - if (desktop_dc_) - ReleaseDC(NULL, desktop_dc_); - if (memory_dc_) - DeleteDC(memory_dc_); - - // Restore Aero. - if (composition_func_) - (*composition_func_)(DWM_EC_ENABLECOMPOSITION); - - if (dwmapi_library_) - FreeLibrary(dwmapi_library_); -} - -void ScreenCapturerWin::Capture(const DesktopRegion& region) { - TickTime capture_start_time = TickTime::Now(); - - queue_.MoveToNextFrame(); - - // Request that the system not power-down the system, or the display hardware. - if (!SetThreadExecutionState(ES_DISPLAY_REQUIRED | ES_SYSTEM_REQUIRED)) { - if (!set_thread_execution_state_failed_) { - set_thread_execution_state_failed_ = true; - LOG_F(LS_WARNING) << "Failed to make system & display power assertion: " - << GetLastError(); - } - } - - // Make sure the GDI capture resources are up-to-date. - PrepareCaptureResources(); - - // Copy screen bits to the current buffer. - CaptureImage(); - - const DesktopFrame* current_frame = queue_.current_frame(); - const DesktopFrame* last_frame = queue_.previous_frame(); - if (last_frame) { - // Make sure the differencer is set up correctly for these previous and - // current screens. - if (!differ_.get() || - (differ_->width() != current_frame->size().width()) || - (differ_->height() != current_frame->size().height()) || - (differ_->bytes_per_row() != current_frame->stride())) { - differ_.reset(new Differ(current_frame->size().width(), - current_frame->size().height(), - DesktopFrame::kBytesPerPixel, - current_frame->stride())); - } - - // Calculate difference between the two last captured frames. - DesktopRegion region; - differ_->CalcDirtyRegion(last_frame->data(), current_frame->data(), - ®ion); - helper_.InvalidateRegion(region); - } else { - // No previous frame is available. Invalidate the whole screen. - helper_.InvalidateScreen(current_frame->size()); - } - - helper_.set_size_most_recent(current_frame->size()); - - // Emit the current frame. - DesktopFrame* frame = queue_.current_frame()->Share(); - frame->set_dpi(DesktopVector( - GetDeviceCaps(desktop_dc_, LOGPIXELSX), - GetDeviceCaps(desktop_dc_, LOGPIXELSY))); - frame->mutable_updated_region()->Clear(); - helper_.TakeInvalidRegion(frame->mutable_updated_region()); - frame->set_capture_time_ms( - (TickTime::Now() - capture_start_time).Milliseconds()); - callback_->OnCaptureCompleted(frame); - - // Check for cursor shape update. - CaptureCursor(); -} - -void ScreenCapturerWin::SetMouseShapeObserver( - MouseShapeObserver* mouse_shape_observer) { - assert(!mouse_shape_observer_); - assert(mouse_shape_observer); - - mouse_shape_observer_ = mouse_shape_observer; -} - -void ScreenCapturerWin::Start(Callback* callback) { - assert(!callback_); - assert(callback); - - callback_ = callback; - - // Vote to disable Aero composited desktop effects while capturing. Windows - // will restore Aero automatically if the process exits. This has no effect - // under Windows 8 or higher. See crbug.com/124018. - if (composition_func_) - (*composition_func_)(DWM_EC_DISABLECOMPOSITION); -} - -void ScreenCapturerWin::PrepareCaptureResources() { - // Switch to the desktop receiving user input if different from the current - // one. - scoped_ptr<Desktop> input_desktop(Desktop::GetInputDesktop()); - if (input_desktop.get() != NULL && !desktop_.IsSame(*input_desktop)) { - // Release GDI resources otherwise SetThreadDesktop will fail. - if (desktop_dc_) { - ReleaseDC(NULL, desktop_dc_); - desktop_dc_ = NULL; - } - - if (memory_dc_) { - DeleteDC(memory_dc_); - memory_dc_ = NULL; - } - - // If SetThreadDesktop() fails, the thread is still assigned a desktop. - // So we can continue capture screen bits, just from the wrong desktop. - desktop_.SetThreadDesktop(input_desktop.release()); - - // Re-assert our vote to disable Aero. - // See crbug.com/124018 and crbug.com/129906. - if (composition_func_ != NULL) { - (*composition_func_)(DWM_EC_DISABLECOMPOSITION); - } - } - - // If the display bounds have changed then recreate GDI resources. - // TODO(wez): Also check for pixel format changes. - DesktopRect screen_rect(DesktopRect::MakeXYWH( - GetSystemMetrics(SM_XVIRTUALSCREEN), - GetSystemMetrics(SM_YVIRTUALSCREEN), - GetSystemMetrics(SM_CXVIRTUALSCREEN), - GetSystemMetrics(SM_CYVIRTUALSCREEN))); - if (!screen_rect.equals(desktop_dc_rect_)) { - if (desktop_dc_) { - ReleaseDC(NULL, desktop_dc_); - desktop_dc_ = NULL; - } - if (memory_dc_) { - DeleteDC(memory_dc_); - memory_dc_ = NULL; - } - desktop_dc_rect_ = DesktopRect(); - } - - if (desktop_dc_ == NULL) { - assert(memory_dc_ == NULL); - - // Create GDI device contexts to capture from the desktop into memory. - desktop_dc_ = GetDC(NULL); - if (!desktop_dc_) - abort(); - memory_dc_ = CreateCompatibleDC(desktop_dc_); - if (!memory_dc_) - abort(); - desktop_dc_rect_ = screen_rect; - - // Make sure the frame buffers will be reallocated. - queue_.Reset(); - - helper_.ClearInvalidRegion(); - } -} - -void ScreenCapturerWin::CaptureImage() { - // If the current buffer is from an older generation then allocate a new one. - // Note that we can't reallocate other buffers at this point, since the caller - // may still be reading from them. - if (!queue_.current_frame()) { - assert(desktop_dc_ != NULL); - assert(memory_dc_ != NULL); - - DesktopSize size = DesktopSize( - desktop_dc_rect_.width(), desktop_dc_rect_.height()); - - size_t buffer_size = size.width() * size.height() * - DesktopFrame::kBytesPerPixel; - SharedMemory* shared_memory = - callback_->CreateSharedMemory(buffer_size); - scoped_ptr<DesktopFrameWin> buffer( - DesktopFrameWin::Create(size, shared_memory, desktop_dc_)); - queue_.ReplaceCurrentFrame(buffer.release()); - } - - // Select the target bitmap into the memory dc and copy the rect from desktop - // to memory. - DesktopFrameWin* current = static_cast<DesktopFrameWin*>( - queue_.current_frame()->GetUnderlyingFrame()); - HGDIOBJ previous_object = SelectObject(memory_dc_, current->bitmap()); - if (previous_object != NULL) { - BitBlt(memory_dc_, - 0, 0, desktop_dc_rect_.width(), desktop_dc_rect_.height(), - desktop_dc_, - desktop_dc_rect_.left(), desktop_dc_rect_.top(), - SRCCOPY | CAPTUREBLT); - - // Select back the previously selected object to that the device contect - // could be destroyed independently of the bitmap if needed. - SelectObject(memory_dc_, previous_object); - } -} - -void ScreenCapturerWin::CaptureCursor() { - CURSORINFO cursor_info; - cursor_info.cbSize = sizeof(CURSORINFO); - if (!GetCursorInfo(&cursor_info)) { - LOG_F(LS_ERROR) << "Unable to get cursor info. Error = " << GetLastError(); - return; - } - - // Note that |cursor_info.hCursor| does not need to be freed. - scoped_ptr<MouseCursor> cursor_image( - CreateMouseCursorFromHCursor(desktop_dc_, cursor_info.hCursor)); - if (!cursor_image.get()) - return; - - scoped_ptr<MouseCursorShape> cursor(new MouseCursorShape); - cursor->hotspot = cursor_image->hotspot(); - cursor->size = cursor_image->image().size(); - uint8_t* current_row = cursor_image->image().data(); - for (int y = 0; y < cursor_image->image().size().height(); ++y) { - cursor->data.append(current_row, - current_row + cursor_image->image().size().width() * - DesktopFrame::kBytesPerPixel); - current_row += cursor_image->image().stride(); - } - - // Compare the current cursor with the last one we sent to the client. If - // they're the same, then don't bother sending the cursor again. - if (last_cursor_.size.equals(cursor->size) && - last_cursor_.hotspot.equals(cursor->hotspot) && - last_cursor_.data == cursor->data) { - return; - } - - LOG(LS_VERBOSE) << "Sending updated cursor: " << cursor->size.width() << "x" - << cursor->size.height(); - - // Record the last cursor image that we sent to the client. - last_cursor_ = *cursor; - - if (mouse_shape_observer_) - mouse_shape_observer_->OnCursorShapeChanged(cursor.release()); -} - -} // namespace - // static ScreenCapturer* ScreenCapturer::Create(const DesktopCaptureOptions& options) { - return new ScreenCapturerWin(options); + scoped_ptr<ScreenCapturer> gdi_capturer(new ScreenCapturerWinGdi(options)); + + if (options.allow_use_magnification_api()) + return new ScreenCapturerWinMagnifier(gdi_capturer.Pass()); + + return gdi_capturer.release(); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc index c5a4c8cb17a..4d07d98c628 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc @@ -57,6 +57,8 @@ class ScreenCapturerLinux : public ScreenCapturer, // ScreenCapturer interface. virtual void SetMouseShapeObserver( MouseShapeObserver* mouse_shape_observer) OVERRIDE; + virtual bool GetScreenList(ScreenList* screens) OVERRIDE; + virtual bool SelectScreen(ScreenId id) OVERRIDE; private: Display* display() { return options_.x_display()->display(); } @@ -310,6 +312,20 @@ void ScreenCapturerLinux::SetMouseShapeObserver( mouse_shape_observer_ = mouse_shape_observer; } +bool ScreenCapturerLinux::GetScreenList(ScreenList* screens) { + DCHECK(screens->size() == 0); + // TODO(jiayl): implement screen enumeration. + Screen default_screen; + default_screen.id = 0; + screens->push_back(default_screen); + return true; +} + +bool ScreenCapturerLinux::SelectScreen(ScreenId id) { + // TODO(jiayl): implement screen selection. + return true; +} + bool ScreenCapturerLinux::HandleXEvent(const XEvent& event) { if (use_damage_ && (event.type == damage_event_base_ + XDamageNotify)) { const XDamageNotifyEvent* damage_event = diff --git a/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h b/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h index bb43b28b1c7..7870d833f15 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h @@ -17,8 +17,8 @@ #include <windows.h> #endif +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/cursor.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/cursor.cc index 11bb2dbb6d0..00055c44ad8 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/win/cursor.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/cursor.cc @@ -137,7 +137,7 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) { int width = bitmap_info.bmWidth; int height = bitmap_info.bmHeight; - scoped_array<uint32_t> mask_data(new uint32_t[width * height]); + scoped_ptr<uint32_t[]> mask_data(new uint32_t[width * height]); // Get pixel data from |scoped_mask| converting it to 32bpp along the way. // GetDIBits() sets the alpha component of every pixel to 0. diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/cursor_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/cursor_unittest.cc index f590bd255ce..b046ace315a 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/win/cursor_unittest.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/cursor_unittest.cc @@ -58,15 +58,15 @@ bool ConvertToMouseShapeAndCompare(unsigned left, unsigned right) { int width = bitmap_info.bmWidth; int height = bitmap_info.bmHeight; - EXPECT_TRUE(DesktopSize(width, height).equals(mouse_shape->image().size())); + EXPECT_TRUE(DesktopSize(width, height).equals(mouse_shape->image()->size())); // Get the pixels from |scoped_color|. int size = width * height; - scoped_array<uint32_t> data(new uint32_t[size]); + scoped_ptr<uint32_t[]> data(new uint32_t[size]); EXPECT_TRUE(GetBitmapBits(scoped_color, size * sizeof(uint32_t), data.get())); // Compare the 32bpp image in |mouse_shape| with the one loaded from |right|. - return memcmp(data.get(), mouse_shape->image().data(), + return memcmp(data.get(), mouse_shape->image()->data(), size * sizeof(uint32_t)) == 0; } diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/desktop.h b/chromium/third_party/webrtc/modules/desktop_capture/win/desktop.h index bdc490c7286..fda56ca8d81 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/win/desktop.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/desktop.h @@ -11,10 +11,10 @@ #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_H_ #define WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_H_ -#include <string> #include <windows.h> +#include <string> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_gdi_object.h b/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_gdi_object.h index 0ca35c526ab..366df6d4ff0 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_gdi_object.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_gdi_object.h @@ -13,7 +13,7 @@ #include <windows.h> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h b/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h index 39514237ea7..f12731d975e 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h @@ -13,7 +13,7 @@ #include <windows.h> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capture_utils.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capture_utils.cc new file mode 100644 index 00000000000..1b335452779 --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capture_utils.cc @@ -0,0 +1,92 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/desktop_capture/win/screen_capture_utils.h" + +#include <assert.h> +#include <windows.h> + +namespace webrtc { + +bool GetScreenList(ScreenCapturer::ScreenList* screens) { + assert(screens->size() == 0); + + BOOL enum_result = TRUE; + for (int device_index = 0;; ++device_index) { + DISPLAY_DEVICE device; + device.cb = sizeof(device); + enum_result = EnumDisplayDevices(NULL, device_index, &device, 0); + + // |enum_result| is 0 if we have enumerated all devices. + if (!enum_result) + break; + + // We only care about active displays. + if (!(device.StateFlags & DISPLAY_DEVICE_ACTIVE)) + continue; + + ScreenCapturer::Screen screen; + screen.id = device_index; + screens->push_back(screen); + } + return true; +} + +bool IsScreenValid(ScreenId screen, std::wstring* device_key) { + if (screen == kFullDesktopScreenId) { + *device_key = L""; + return true; + } + + DISPLAY_DEVICE device; + device.cb = sizeof(device); + BOOL enum_result = EnumDisplayDevices(NULL, screen, &device, 0); + if (enum_result) + *device_key = device.DeviceKey; + + return !!enum_result; +} + +DesktopRect GetScreenRect(ScreenId screen, const std::wstring& device_key) { + if (screen == kFullDesktopScreenId) { + return DesktopRect::MakeXYWH(GetSystemMetrics(SM_XVIRTUALSCREEN), + GetSystemMetrics(SM_YVIRTUALSCREEN), + GetSystemMetrics(SM_CXVIRTUALSCREEN), + GetSystemMetrics(SM_CYVIRTUALSCREEN)); + } + + DISPLAY_DEVICE device; + device.cb = sizeof(device); + BOOL result = EnumDisplayDevices(NULL, screen, &device, 0); + if (!result) + return DesktopRect(); + + // Verifies the device index still maps to the same display device, to make + // sure we are capturing the same device when devices are added or removed. + // DeviceKey is documented as reserved, but it actually contains the registry + // key for the device and is unique for each monitor, while DeviceID is not. + if (device_key != device.DeviceKey) + return DesktopRect(); + + DEVMODE device_mode; + device_mode.dmSize = sizeof(device_mode); + device_mode.dmDriverExtra = 0; + result = EnumDisplaySettingsEx( + device.DeviceName, ENUM_CURRENT_SETTINGS, &device_mode, 0); + if (!result) + return DesktopRect(); + + return DesktopRect::MakeXYWH(device_mode.dmPosition.x, + device_mode.dmPosition.y, + device_mode.dmPelsWidth, + device_mode.dmPelsHeight); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capture_utils.h b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capture_utils.h new file mode 100644 index 00000000000..42473e047b3 --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capture_utils.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_ +#define WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_ + +#include "webrtc/modules/desktop_capture/screen_capturer.h" + +namespace webrtc { + +// Output the list of active screens into |screens|. Returns true if succeeded, +// or false if it fails to enumerate the display devices. +bool GetScreenList(ScreenCapturer::ScreenList* screens); + +// Returns true if |screen| is a valid screen. The screen device key is +// returned through |device_key| if the screen is valid. The device key can be +// used in GetScreenRect to verify the screen matches the previously obtained +// id. +bool IsScreenValid(ScreenId screen, std::wstring* device_key); + +// Get the rect of the screen identified by |screen|, relative to the primary +// display's top-left. If the screen device key does not match |device_key|, or +// the screen does not exist, or any error happens, an empty rect is returned. +DesktopRect GetScreenRect(ScreenId screen, const std::wstring& device_key); + +} // namespace webrtc + +#endif // WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_ diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc new file mode 100644 index 00000000000..9cb3681fd2c --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc @@ -0,0 +1,324 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h" + +#include <assert.h> + +#include "webrtc/modules/desktop_capture/desktop_capture_options.h" +#include "webrtc/modules/desktop_capture/desktop_frame.h" +#include "webrtc/modules/desktop_capture/desktop_frame_win.h" +#include "webrtc/modules/desktop_capture/desktop_region.h" +#include "webrtc/modules/desktop_capture/differ.h" +#include "webrtc/modules/desktop_capture/mouse_cursor.h" +#include "webrtc/modules/desktop_capture/win/cursor.h" +#include "webrtc/modules/desktop_capture/win/desktop.h" +#include "webrtc/modules/desktop_capture/win/screen_capture_utils.h" +#include "webrtc/system_wrappers/interface/logging.h" +#include "webrtc/system_wrappers/interface/tick_util.h" + +namespace webrtc { + +namespace { + +// Constants from dwmapi.h. +const UINT DWM_EC_DISABLECOMPOSITION = 0; +const UINT DWM_EC_ENABLECOMPOSITION = 1; + +const wchar_t kDwmapiLibraryName[] = L"dwmapi.dll"; + +} // namespace + +ScreenCapturerWinGdi::ScreenCapturerWinGdi(const DesktopCaptureOptions& options) + : callback_(NULL), + mouse_shape_observer_(NULL), + current_screen_id_(kFullDesktopScreenId), + desktop_dc_(NULL), + memory_dc_(NULL), + dwmapi_library_(NULL), + composition_func_(NULL), + set_thread_execution_state_failed_(false) { + if (options.disable_effects()) { + // Load dwmapi.dll dynamically since it is not available on XP. + if (!dwmapi_library_) + dwmapi_library_ = LoadLibrary(kDwmapiLibraryName); + + if (dwmapi_library_) { + composition_func_ = reinterpret_cast<DwmEnableCompositionFunc>( + GetProcAddress(dwmapi_library_, "DwmEnableComposition")); + } + } +} + +ScreenCapturerWinGdi::~ScreenCapturerWinGdi() { + if (desktop_dc_) + ReleaseDC(NULL, desktop_dc_); + if (memory_dc_) + DeleteDC(memory_dc_); + + // Restore Aero. + if (composition_func_) + (*composition_func_)(DWM_EC_ENABLECOMPOSITION); + + if (dwmapi_library_) + FreeLibrary(dwmapi_library_); +} + +void ScreenCapturerWinGdi::Capture(const DesktopRegion& region) { + TickTime capture_start_time = TickTime::Now(); + + queue_.MoveToNextFrame(); + + // Request that the system not power-down the system, or the display hardware. + if (!SetThreadExecutionState(ES_DISPLAY_REQUIRED | ES_SYSTEM_REQUIRED)) { + if (!set_thread_execution_state_failed_) { + set_thread_execution_state_failed_ = true; + LOG_F(LS_WARNING) << "Failed to make system & display power assertion: " + << GetLastError(); + } + } + + // Make sure the GDI capture resources are up-to-date. + PrepareCaptureResources(); + + if (!CaptureImage()) { + callback_->OnCaptureCompleted(NULL); + return; + } + + const DesktopFrame* current_frame = queue_.current_frame(); + const DesktopFrame* last_frame = queue_.previous_frame(); + if (last_frame && last_frame->size().equals(current_frame->size())) { + // Make sure the differencer is set up correctly for these previous and + // current screens. + if (!differ_.get() || + (differ_->width() != current_frame->size().width()) || + (differ_->height() != current_frame->size().height()) || + (differ_->bytes_per_row() != current_frame->stride())) { + differ_.reset(new Differ(current_frame->size().width(), + current_frame->size().height(), + DesktopFrame::kBytesPerPixel, + current_frame->stride())); + } + + // Calculate difference between the two last captured frames. + DesktopRegion region; + differ_->CalcDirtyRegion(last_frame->data(), current_frame->data(), + ®ion); + helper_.InvalidateRegion(region); + } else { + // No previous frame is available, or the screen is resized. Invalidate the + // whole screen. + helper_.InvalidateScreen(current_frame->size()); + } + + helper_.set_size_most_recent(current_frame->size()); + + // Emit the current frame. + DesktopFrame* frame = queue_.current_frame()->Share(); + frame->set_dpi(DesktopVector( + GetDeviceCaps(desktop_dc_, LOGPIXELSX), + GetDeviceCaps(desktop_dc_, LOGPIXELSY))); + frame->mutable_updated_region()->Clear(); + helper_.TakeInvalidRegion(frame->mutable_updated_region()); + frame->set_capture_time_ms( + (TickTime::Now() - capture_start_time).Milliseconds()); + callback_->OnCaptureCompleted(frame); + + // Check for cursor shape update. + CaptureCursor(); +} + +void ScreenCapturerWinGdi::SetMouseShapeObserver( + MouseShapeObserver* mouse_shape_observer) { + assert(!mouse_shape_observer_); + assert(mouse_shape_observer); + + mouse_shape_observer_ = mouse_shape_observer; +} + +bool ScreenCapturerWinGdi::GetScreenList(ScreenList* screens) { + return webrtc::GetScreenList(screens); +} + +bool ScreenCapturerWinGdi::SelectScreen(ScreenId id) { + bool valid = IsScreenValid(id, ¤t_device_key_); + if (valid) + current_screen_id_ = id; + return valid; +} + +void ScreenCapturerWinGdi::Start(Callback* callback) { + assert(!callback_); + assert(callback); + + callback_ = callback; + + // Vote to disable Aero composited desktop effects while capturing. Windows + // will restore Aero automatically if the process exits. This has no effect + // under Windows 8 or higher. See crbug.com/124018. + if (composition_func_) + (*composition_func_)(DWM_EC_DISABLECOMPOSITION); +} + +void ScreenCapturerWinGdi::PrepareCaptureResources() { + // Switch to the desktop receiving user input if different from the current + // one. + scoped_ptr<Desktop> input_desktop(Desktop::GetInputDesktop()); + if (input_desktop.get() != NULL && !desktop_.IsSame(*input_desktop)) { + // Release GDI resources otherwise SetThreadDesktop will fail. + if (desktop_dc_) { + ReleaseDC(NULL, desktop_dc_); + desktop_dc_ = NULL; + } + + if (memory_dc_) { + DeleteDC(memory_dc_); + memory_dc_ = NULL; + } + + // If SetThreadDesktop() fails, the thread is still assigned a desktop. + // So we can continue capture screen bits, just from the wrong desktop. + desktop_.SetThreadDesktop(input_desktop.release()); + + // Re-assert our vote to disable Aero. + // See crbug.com/124018 and crbug.com/129906. + if (composition_func_ != NULL) { + (*composition_func_)(DWM_EC_DISABLECOMPOSITION); + } + } + + // If the display bounds have changed then recreate GDI resources. + // TODO(wez): Also check for pixel format changes. + DesktopRect screen_rect(DesktopRect::MakeXYWH( + GetSystemMetrics(SM_XVIRTUALSCREEN), + GetSystemMetrics(SM_YVIRTUALSCREEN), + GetSystemMetrics(SM_CXVIRTUALSCREEN), + GetSystemMetrics(SM_CYVIRTUALSCREEN))); + if (!screen_rect.equals(desktop_dc_rect_)) { + if (desktop_dc_) { + ReleaseDC(NULL, desktop_dc_); + desktop_dc_ = NULL; + } + if (memory_dc_) { + DeleteDC(memory_dc_); + memory_dc_ = NULL; + } + desktop_dc_rect_ = DesktopRect(); + } + + if (desktop_dc_ == NULL) { + assert(memory_dc_ == NULL); + + // Create GDI device contexts to capture from the desktop into memory. + desktop_dc_ = GetDC(NULL); + if (!desktop_dc_) + abort(); + memory_dc_ = CreateCompatibleDC(desktop_dc_); + if (!memory_dc_) + abort(); + + desktop_dc_rect_ = screen_rect; + + // Make sure the frame buffers will be reallocated. + queue_.Reset(); + + helper_.ClearInvalidRegion(); + } +} + +bool ScreenCapturerWinGdi::CaptureImage() { + DesktopRect screen_rect = + GetScreenRect(current_screen_id_, current_device_key_); + if (screen_rect.is_empty()) + return false; + + DesktopSize size = screen_rect.size(); + // If the current buffer is from an older generation then allocate a new one. + // Note that we can't reallocate other buffers at this point, since the caller + // may still be reading from them. + if (!queue_.current_frame() || + !queue_.current_frame()->size().equals(screen_rect.size())) { + assert(desktop_dc_ != NULL); + assert(memory_dc_ != NULL); + + size_t buffer_size = size.width() * size.height() * + DesktopFrame::kBytesPerPixel; + SharedMemory* shared_memory = callback_->CreateSharedMemory(buffer_size); + + scoped_ptr<DesktopFrame> buffer; + buffer.reset( + DesktopFrameWin::Create(size, shared_memory, desktop_dc_)); + queue_.ReplaceCurrentFrame(buffer.release()); + } + + // Select the target bitmap into the memory dc and copy the rect from desktop + // to memory. + DesktopFrameWin* current = static_cast<DesktopFrameWin*>( + queue_.current_frame()->GetUnderlyingFrame()); + HGDIOBJ previous_object = SelectObject(memory_dc_, current->bitmap()); + if (previous_object != NULL) { + BitBlt(memory_dc_, + 0, 0, screen_rect.width(), screen_rect.height(), + desktop_dc_, + screen_rect.left(), screen_rect.top(), + SRCCOPY | CAPTUREBLT); + + // Select back the previously selected object to that the device contect + // could be destroyed independently of the bitmap if needed. + SelectObject(memory_dc_, previous_object); + } + return true; +} + +void ScreenCapturerWinGdi::CaptureCursor() { + CURSORINFO cursor_info; + cursor_info.cbSize = sizeof(CURSORINFO); + if (!GetCursorInfo(&cursor_info)) { + LOG_F(LS_ERROR) << "Unable to get cursor info. Error = " << GetLastError(); + return; + } + + // Note that |cursor_info.hCursor| does not need to be freed. + scoped_ptr<MouseCursor> cursor_image( + CreateMouseCursorFromHCursor(desktop_dc_, cursor_info.hCursor)); + if (!cursor_image.get()) + return; + + scoped_ptr<MouseCursorShape> cursor(new MouseCursorShape); + cursor->hotspot = cursor_image->hotspot(); + cursor->size = cursor_image->image()->size(); + uint8_t* current_row = cursor_image->image()->data(); + for (int y = 0; y < cursor_image->image()->size().height(); ++y) { + cursor->data.append(current_row, + current_row + cursor_image->image()->size().width() * + DesktopFrame::kBytesPerPixel); + current_row += cursor_image->image()->stride(); + } + + // Compare the current cursor with the last one we sent to the client. If + // they're the same, then don't bother sending the cursor again. + if (last_cursor_.size.equals(cursor->size) && + last_cursor_.hotspot.equals(cursor->hotspot) && + last_cursor_.data == cursor->data) { + return; + } + + LOG(LS_VERBOSE) << "Sending updated cursor: " << cursor->size.width() << "x" + << cursor->size.height(); + + // Record the last cursor image that we sent to the client. + last_cursor_ = *cursor; + + if (mouse_shape_observer_) + mouse_shape_observer_->OnCursorShapeChanged(cursor.release()); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h new file mode 100644 index 00000000000..2db87d097ca --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h @@ -0,0 +1,99 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_GDI_H_ +#define WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_GDI_H_ + +#include "webrtc/modules/desktop_capture/screen_capturer.h" + +#include <windows.h> + +#include "webrtc/modules/desktop_capture/mouse_cursor_shape.h" +#include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h" +#include "webrtc/modules/desktop_capture/screen_capturer_helper.h" +#include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +class Differ; +class MouseShapeObserver; + +// ScreenCapturerWinGdi captures 32bit RGB using GDI. +// +// ScreenCapturerWinGdi is double-buffered as required by ScreenCapturer. +class ScreenCapturerWinGdi : public ScreenCapturer { + public: + explicit ScreenCapturerWinGdi(const DesktopCaptureOptions& options); + virtual ~ScreenCapturerWinGdi(); + + // Overridden from ScreenCapturer: + virtual void Start(Callback* callback) OVERRIDE; + virtual void Capture(const DesktopRegion& region) OVERRIDE; + virtual void SetMouseShapeObserver( + MouseShapeObserver* mouse_shape_observer) OVERRIDE; + virtual bool GetScreenList(ScreenList* screens) OVERRIDE; + virtual bool SelectScreen(ScreenId id) OVERRIDE; + + private: + typedef HRESULT (WINAPI * DwmEnableCompositionFunc)(UINT); + + // Make sure that the device contexts match the screen configuration. + void PrepareCaptureResources(); + + // Captures the current screen contents into the current buffer. Returns true + // if succeeded. + bool CaptureImage(); + + // Capture the current cursor shape. + void CaptureCursor(); + + Callback* callback_; + MouseShapeObserver* mouse_shape_observer_; + ScreenId current_screen_id_; + std::wstring current_device_key_; + + // A thread-safe list of invalid rectangles, and the size of the most + // recently captured screen. + ScreenCapturerHelper helper_; + + // Snapshot of the last cursor bitmap we sent to the client. This is used + // to diff against the current cursor so we only send a cursor-change + // message when the shape has changed. + MouseCursorShape last_cursor_; + + ScopedThreadDesktop desktop_; + + // GDI resources used for screen capture. + HDC desktop_dc_; + HDC memory_dc_; + + // Queue of the frames buffers. + ScreenCaptureFrameQueue queue_; + + // Rectangle describing the bounds of the desktop device context, relative to + // the primary display's top-left. + DesktopRect desktop_dc_rect_; + + // Class to calculate the difference between two screen bitmaps. + scoped_ptr<Differ> differ_; + + HMODULE dwmapi_library_; + DwmEnableCompositionFunc composition_func_; + + // Used to suppress duplicate logging of SetThreadExecutionState errors. + bool set_thread_execution_state_failed_; + + DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWinGdi); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_GDI_H_ diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc new file mode 100644 index 00000000000..042cb937acc --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc @@ -0,0 +1,461 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h" + +#include <assert.h> + +#include "webrtc/modules/desktop_capture/desktop_capture_options.h" +#include "webrtc/modules/desktop_capture/desktop_frame.h" +#include "webrtc/modules/desktop_capture/desktop_frame_win.h" +#include "webrtc/modules/desktop_capture/desktop_region.h" +#include "webrtc/modules/desktop_capture/differ.h" +#include "webrtc/modules/desktop_capture/mouse_cursor.h" +#include "webrtc/modules/desktop_capture/win/cursor.h" +#include "webrtc/modules/desktop_capture/win/desktop.h" +#include "webrtc/modules/desktop_capture/win/screen_capture_utils.h" +#include "webrtc/system_wrappers/interface/logging.h" +#include "webrtc/system_wrappers/interface/tick_util.h" + +namespace webrtc { + +// kMagnifierWindowClass has to be "Magnifier" according to the Magnification +// API. The other strings can be anything. +static LPCTSTR kMagnifierHostClass = L"ScreenCapturerWinMagnifierHost"; +static LPCTSTR kHostWindowName = L"MagnifierHost"; +static LPCTSTR kMagnifierWindowClass = L"Magnifier"; +static LPCTSTR kMagnifierWindowName = L"MagnifierWindow"; + +Atomic32 ScreenCapturerWinMagnifier::tls_index_(TLS_OUT_OF_INDEXES); + +ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier( + scoped_ptr<ScreenCapturer> fallback_capturer) + : fallback_capturer_(fallback_capturer.Pass()), + fallback_capturer_started_(false), + callback_(NULL), + current_screen_id_(kFullDesktopScreenId), + excluded_window_(NULL), + set_thread_execution_state_failed_(false), + desktop_dc_(NULL), + mag_lib_handle_(NULL), + mag_initialize_func_(NULL), + mag_uninitialize_func_(NULL), + set_window_source_func_(NULL), + set_window_filter_list_func_(NULL), + set_image_scaling_callback_func_(NULL), + host_window_(NULL), + magnifier_window_(NULL), + magnifier_initialized_(false), + magnifier_capture_succeeded_(true) { +} + +ScreenCapturerWinMagnifier::~ScreenCapturerWinMagnifier() { + // DestroyWindow must be called before MagUninitialize. magnifier_window_ is + // destroyed automatically when host_window_ is destroyed. + if (host_window_) + DestroyWindow(host_window_); + + if (magnifier_initialized_) + mag_uninitialize_func_(); + + if (mag_lib_handle_) + FreeLibrary(mag_lib_handle_); + + if (desktop_dc_) + ReleaseDC(NULL, desktop_dc_); +} + +void ScreenCapturerWinMagnifier::Start(Callback* callback) { + assert(!callback_); + assert(callback); + callback_ = callback; + + InitializeMagnifier(); +} + +void ScreenCapturerWinMagnifier::Capture(const DesktopRegion& region) { + TickTime capture_start_time = TickTime::Now(); + + queue_.MoveToNextFrame(); + + // Request that the system not power-down the system, or the display hardware. + if (!SetThreadExecutionState(ES_DISPLAY_REQUIRED | ES_SYSTEM_REQUIRED)) { + if (!set_thread_execution_state_failed_) { + set_thread_execution_state_failed_ = true; + LOG_F(LS_WARNING) << "Failed to make system & display power assertion: " + << GetLastError(); + } + } + // Switch to the desktop receiving user input if different from the current + // one. + scoped_ptr<Desktop> input_desktop(Desktop::GetInputDesktop()); + if (input_desktop.get() != NULL && !desktop_.IsSame(*input_desktop)) { + // Release GDI resources otherwise SetThreadDesktop will fail. + if (desktop_dc_) { + ReleaseDC(NULL, desktop_dc_); + desktop_dc_ = NULL; + } + // If SetThreadDesktop() fails, the thread is still assigned a desktop. + // So we can continue capture screen bits, just from the wrong desktop. + desktop_.SetThreadDesktop(input_desktop.release()); + } + + bool succeeded = false; + + // Do not try to use the magnfiier if it's capturing non-primary screen, or it + // failed before. + if (magnifier_initialized_ && IsCapturingPrimaryScreenOnly() && + magnifier_capture_succeeded_) { + DesktopRect rect = GetScreenRect(current_screen_id_, current_device_key_); + CreateCurrentFrameIfNecessary(rect.size()); + + // CaptureImage may fail in some situations, e.g. windows8 metro mode. + succeeded = CaptureImage(rect); + } + + // Defer to the fallback capturer if magnifier capturer did not work. + if (!succeeded) { + LOG_F(LS_WARNING) << "Switching to the fallback screen capturer."; + StartFallbackCapturer(); + fallback_capturer_->Capture(region); + return; + } + + const DesktopFrame* current_frame = queue_.current_frame(); + const DesktopFrame* last_frame = queue_.previous_frame(); + if (last_frame && last_frame->size().equals(current_frame->size())) { + // Make sure the differencer is set up correctly for these previous and + // current screens. + if (!differ_.get() || (differ_->width() != current_frame->size().width()) || + (differ_->height() != current_frame->size().height()) || + (differ_->bytes_per_row() != current_frame->stride())) { + differ_.reset(new Differ(current_frame->size().width(), + current_frame->size().height(), + DesktopFrame::kBytesPerPixel, + current_frame->stride())); + } + + // Calculate difference between the two last captured frames. + DesktopRegion region; + differ_->CalcDirtyRegion( + last_frame->data(), current_frame->data(), ®ion); + helper_.InvalidateRegion(region); + } else { + // No previous frame is available, or the screen is resized. Invalidate the + // whole screen. + helper_.InvalidateScreen(current_frame->size()); + } + + helper_.set_size_most_recent(current_frame->size()); + + // Emit the current frame. + DesktopFrame* frame = queue_.current_frame()->Share(); + frame->set_dpi(DesktopVector(GetDeviceCaps(desktop_dc_, LOGPIXELSX), + GetDeviceCaps(desktop_dc_, LOGPIXELSY))); + frame->mutable_updated_region()->Clear(); + helper_.TakeInvalidRegion(frame->mutable_updated_region()); + frame->set_capture_time_ms( + (TickTime::Now() - capture_start_time).Milliseconds()); + callback_->OnCaptureCompleted(frame); +} + +void ScreenCapturerWinMagnifier::SetMouseShapeObserver( + MouseShapeObserver* mouse_shape_observer) { + assert(false); // NOTREACHED(); +} + +bool ScreenCapturerWinMagnifier::GetScreenList(ScreenList* screens) { + return webrtc::GetScreenList(screens); +} + +bool ScreenCapturerWinMagnifier::SelectScreen(ScreenId id) { + bool valid = IsScreenValid(id, ¤t_device_key_); + + // Set current_screen_id_ even if the fallback capturer is being used, so we + // can switch back to the magnifier when possible. + if (valid) + current_screen_id_ = id; + + if (fallback_capturer_started_) + fallback_capturer_->SelectScreen(id); + + return valid; +} + +void ScreenCapturerWinMagnifier::SetExcludedWindow(WindowId excluded_window) { + excluded_window_ = (HWND)excluded_window; + if (excluded_window_ && magnifier_initialized_) { + set_window_filter_list_func_( + magnifier_window_, MW_FILTERMODE_EXCLUDE, 1, &excluded_window_); + } +} + +bool ScreenCapturerWinMagnifier::CaptureImage(const DesktopRect& rect) { + assert(magnifier_initialized_); + + // Set the magnifier control to cover the captured rect. The content of the + // magnifier control will be the captured image. + BOOL result = SetWindowPos(magnifier_window_, + NULL, + rect.left(), rect.top(), + rect.width(), rect.height(), + 0); + if (!result) { + LOG_F(LS_WARNING) << "Failed to call SetWindowPos: " << GetLastError() + << ". Rect = {" << rect.left() << ", " << rect.top() + << ", " << rect.right() << ", " << rect.bottom() << "}"; + return false; + } + + magnifier_capture_succeeded_ = false; + + RECT native_rect = {rect.left(), rect.top(), rect.right(), rect.bottom()}; + + // OnCaptured will be called via OnMagImageScalingCallback and fill in the + // frame before set_window_source_func_ returns. + result = set_window_source_func_(magnifier_window_, native_rect); + + if (!result) { + LOG_F(LS_WARNING) << "Failed to call MagSetWindowSource: " << GetLastError() + << ". Rect = {" << rect.left() << ", " << rect.top() + << ", " << rect.right() << ", " << rect.bottom() << "}"; + return false; + } + + return magnifier_capture_succeeded_; +} + +BOOL ScreenCapturerWinMagnifier::OnMagImageScalingCallback( + HWND hwnd, + void* srcdata, + MAGIMAGEHEADER srcheader, + void* destdata, + MAGIMAGEHEADER destheader, + RECT unclipped, + RECT clipped, + HRGN dirty) { + assert(tls_index_.Value() != TLS_OUT_OF_INDEXES); + + ScreenCapturerWinMagnifier* owner = + reinterpret_cast<ScreenCapturerWinMagnifier*>( + TlsGetValue(tls_index_.Value())); + + owner->OnCaptured(srcdata, srcheader); + + return TRUE; +} + +bool ScreenCapturerWinMagnifier::InitializeMagnifier() { + assert(!magnifier_initialized_); + + desktop_dc_ = GetDC(NULL); + + mag_lib_handle_ = LoadLibrary(L"Magnification.dll"); + if (!mag_lib_handle_) + return false; + + // Initialize Magnification API function pointers. + mag_initialize_func_ = reinterpret_cast<MagInitializeFunc>( + GetProcAddress(mag_lib_handle_, "MagInitialize")); + mag_uninitialize_func_ = reinterpret_cast<MagUninitializeFunc>( + GetProcAddress(mag_lib_handle_, "MagUninitialize")); + set_window_source_func_ = reinterpret_cast<MagSetWindowSourceFunc>( + GetProcAddress(mag_lib_handle_, "MagSetWindowSource")); + set_window_filter_list_func_ = reinterpret_cast<MagSetWindowFilterListFunc>( + GetProcAddress(mag_lib_handle_, "MagSetWindowFilterList")); + set_image_scaling_callback_func_ = + reinterpret_cast<MagSetImageScalingCallbackFunc>( + GetProcAddress(mag_lib_handle_, "MagSetImageScalingCallback")); + + if (!mag_initialize_func_ || !mag_uninitialize_func_ || + !set_window_source_func_ || !set_window_filter_list_func_ || + !set_image_scaling_callback_func_) { + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "library functions missing."; + return false; + } + + BOOL result = mag_initialize_func_(); + if (!result) { + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "error from MagInitialize " << GetLastError(); + return false; + } + + HMODULE hInstance = NULL; + result = GetModuleHandleExA(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS | + GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT, + reinterpret_cast<char*>(&DefWindowProc), + &hInstance); + if (!result) { + mag_uninitialize_func_(); + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "error from GetModulehandleExA " << GetLastError(); + return false; + } + + // Register the host window class. See the MSDN documentation of the + // Magnification API for more infomation. + WNDCLASSEX wcex = {}; + wcex.cbSize = sizeof(WNDCLASSEX); + wcex.lpfnWndProc = &DefWindowProc; + wcex.hInstance = hInstance; + wcex.hCursor = LoadCursor(NULL, IDC_ARROW); + wcex.lpszClassName = kMagnifierHostClass; + + // Ignore the error which may happen when the class is already registered. + RegisterClassEx(&wcex); + + // Create the host window. + host_window_ = CreateWindowEx(WS_EX_LAYERED, + kMagnifierHostClass, + kHostWindowName, + 0, + 0, 0, 0, 0, + NULL, + NULL, + hInstance, + NULL); + if (!host_window_) { + mag_uninitialize_func_(); + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "error from creating host window " << GetLastError(); + return false; + } + + // Create the magnifier control. + magnifier_window_ = CreateWindow(kMagnifierWindowClass, + kMagnifierWindowName, + WS_CHILD | WS_VISIBLE, + 0, 0, 0, 0, + host_window_, + NULL, + hInstance, + NULL); + if (!magnifier_window_) { + mag_uninitialize_func_(); + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "error from creating magnifier window " + << GetLastError(); + return false; + } + + // Hide the host window. + ShowWindow(host_window_, SW_HIDE); + + // Set the scaling callback to receive captured image. + result = set_image_scaling_callback_func_( + magnifier_window_, + &ScreenCapturerWinMagnifier::OnMagImageScalingCallback); + if (!result) { + mag_uninitialize_func_(); + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "error from MagSetImageScalingCallback " + << GetLastError(); + return false; + } + + if (excluded_window_) { + result = set_window_filter_list_func_( + magnifier_window_, MW_FILTERMODE_EXCLUDE, 1, &excluded_window_); + if (!result) { + mag_uninitialize_func_(); + LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " + << "error from MagSetWindowFilterList " + << GetLastError(); + return false; + } + } + + if (tls_index_.Value() == TLS_OUT_OF_INDEXES) { + // More than one threads may get here at the same time, but only one will + // write to tls_index_ using CompareExchange. + DWORD new_tls_index = TlsAlloc(); + if (!tls_index_.CompareExchange(new_tls_index, TLS_OUT_OF_INDEXES)) + TlsFree(new_tls_index); + } + + assert(tls_index_.Value() != TLS_OUT_OF_INDEXES); + TlsSetValue(tls_index_.Value(), this); + + magnifier_initialized_ = true; + return true; +} + +void ScreenCapturerWinMagnifier::OnCaptured(void* data, + const MAGIMAGEHEADER& header) { + DesktopFrame* current_frame = queue_.current_frame(); + + // Verify the format. + // TODO(jiayl): support capturing sources with pixel formats other than RGBA. + int captured_bytes_per_pixel = header.cbSize / header.width / header.height; + if (header.format != GUID_WICPixelFormat32bppRGBA || + header.width != static_cast<UINT>(current_frame->size().width()) || + header.height != static_cast<UINT>(current_frame->size().height()) || + header.stride != static_cast<UINT>(current_frame->stride()) || + captured_bytes_per_pixel != DesktopFrame::kBytesPerPixel) { + LOG_F(LS_WARNING) << "Output format does not match the captured format: " + << "width = " << header.width << ", " + << "height = " << header.height << ", " + << "stride = " << header.stride << ", " + << "bpp = " << captured_bytes_per_pixel << ", " + << "pixel format RGBA ? " + << (header.format == GUID_WICPixelFormat32bppRGBA) << "."; + return; + } + + // Copy the data into the frame. + current_frame->CopyPixelsFrom( + reinterpret_cast<uint8_t*>(data), + header.stride, + DesktopRect::MakeXYWH(0, 0, header.width, header.height)); + + magnifier_capture_succeeded_ = true; +} + +void ScreenCapturerWinMagnifier::CreateCurrentFrameIfNecessary( + const DesktopSize& size) { + // If the current buffer is from an older generation then allocate a new one. + // Note that we can't reallocate other buffers at this point, since the caller + // may still be reading from them. + if (!queue_.current_frame() || !queue_.current_frame()->size().equals(size)) { + size_t buffer_size = + size.width() * size.height() * DesktopFrame::kBytesPerPixel; + SharedMemory* shared_memory = callback_->CreateSharedMemory(buffer_size); + + scoped_ptr<DesktopFrame> buffer; + if (shared_memory) { + buffer.reset(new SharedMemoryDesktopFrame( + size, size.width() * DesktopFrame::kBytesPerPixel, shared_memory)); + } else { + buffer.reset(new BasicDesktopFrame(size)); + } + queue_.ReplaceCurrentFrame(buffer.release()); + } +} + +bool ScreenCapturerWinMagnifier::IsCapturingPrimaryScreenOnly() const { + if (current_screen_id_ != kFullDesktopScreenId) + return current_screen_id_ == 0; // the primary screen is always '0'. + + return GetSystemMetrics(SM_CMONITORS) == 1; +} + +void ScreenCapturerWinMagnifier::StartFallbackCapturer() { + assert(fallback_capturer_); + if (!fallback_capturer_started_) { + fallback_capturer_started_ = true; + + fallback_capturer_->Start(callback_); + fallback_capturer_->SelectScreen(current_screen_id_); + } +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h new file mode 100644 index 00000000000..b6d559083eb --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h @@ -0,0 +1,159 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_MAGNIFIER_H_ +#define WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_MAGNIFIER_H_ + +#include <windows.h> +#include <magnification.h> +#include <wincodec.h> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h" +#include "webrtc/modules/desktop_capture/screen_capturer.h" +#include "webrtc/modules/desktop_capture/screen_capturer_helper.h" +#include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h" +#include "webrtc/system_wrappers/interface/atomic32.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +class DesktopFrame; +class DesktopRect; +class Differ; +class MouseShapeObserver; + +// Captures the screen using the Magnification API to support window exclusion. +// Each capturer must run on a dedicated thread because it uses thread local +// storage for redirecting the library callback. Also the thread must have a UI +// message loop to handle the window messages for the magnifier window. +class ScreenCapturerWinMagnifier : public ScreenCapturer { + public: + // |fallback_capturer| will be used to capture the screen if a non-primary + // screen is being captured, or the OS does not support Magnification API, or + // the magnifier capturer fails (e.g. in Windows8 Metro mode). + explicit ScreenCapturerWinMagnifier( + scoped_ptr<ScreenCapturer> fallback_capturer); + virtual ~ScreenCapturerWinMagnifier(); + + // Overridden from ScreenCapturer: + virtual void Start(Callback* callback) OVERRIDE; + virtual void Capture(const DesktopRegion& region) OVERRIDE; + virtual void SetMouseShapeObserver( + MouseShapeObserver* mouse_shape_observer) OVERRIDE; + virtual bool GetScreenList(ScreenList* screens) OVERRIDE; + virtual bool SelectScreen(ScreenId id) OVERRIDE; + virtual void SetExcludedWindow(WindowId window) OVERRIDE; + + private: + typedef BOOL(WINAPI* MagImageScalingCallback)(HWND hwnd, + void* srcdata, + MAGIMAGEHEADER srcheader, + void* destdata, + MAGIMAGEHEADER destheader, + RECT unclipped, + RECT clipped, + HRGN dirty); + typedef BOOL(WINAPI* MagInitializeFunc)(void); + typedef BOOL(WINAPI* MagUninitializeFunc)(void); + typedef BOOL(WINAPI* MagSetWindowSourceFunc)(HWND hwnd, RECT rect); + typedef BOOL(WINAPI* MagSetWindowFilterListFunc)(HWND hwnd, + DWORD dwFilterMode, + int count, + HWND* pHWND); + typedef BOOL(WINAPI* MagSetImageScalingCallbackFunc)( + HWND hwnd, + MagImageScalingCallback callback); + + static BOOL WINAPI OnMagImageScalingCallback(HWND hwnd, + void* srcdata, + MAGIMAGEHEADER srcheader, + void* destdata, + MAGIMAGEHEADER destheader, + RECT unclipped, + RECT clipped, + HRGN dirty); + + // Captures the screen within |rect| in the desktop coordinates. Returns true + // if succeeded. + // It can only capture the primary screen for now. The magnification library + // crashes under some screen configurations (e.g. secondary screen on top of + // primary screen) if it tries to capture a non-primary screen. The caller + // must make sure not calling it on non-primary screens. + bool CaptureImage(const DesktopRect& rect); + + // Helper method for setting up the magnifier control. Returns true if + // succeeded. + bool InitializeMagnifier(); + + // Called by OnMagImageScalingCallback to output captured data. + void OnCaptured(void* data, const MAGIMAGEHEADER& header); + + // Makes sure the current frame exists and matches |size|. + void CreateCurrentFrameIfNecessary(const DesktopSize& size); + + // Returns true if we are capturing the primary screen only. + bool IsCapturingPrimaryScreenOnly() const; + + // Start the fallback capturer and select the screen. + void StartFallbackCapturer(); + + static Atomic32 tls_index_; + + scoped_ptr<ScreenCapturer> fallback_capturer_; + bool fallback_capturer_started_; + Callback* callback_; + ScreenId current_screen_id_; + std::wstring current_device_key_; + HWND excluded_window_; + + // A thread-safe list of invalid rectangles, and the size of the most + // recently captured screen. + ScreenCapturerHelper helper_; + + // Queue of the frames buffers. + ScreenCaptureFrameQueue queue_; + + // Class to calculate the difference between two screen bitmaps. + scoped_ptr<Differ> differ_; + + // Used to suppress duplicate logging of SetThreadExecutionState errors. + bool set_thread_execution_state_failed_; + + ScopedThreadDesktop desktop_; + + // Used for getting the screen dpi. + HDC desktop_dc_; + + HMODULE mag_lib_handle_; + MagInitializeFunc mag_initialize_func_; + MagUninitializeFunc mag_uninitialize_func_; + MagSetWindowSourceFunc set_window_source_func_; + MagSetWindowFilterListFunc set_window_filter_list_func_; + MagSetImageScalingCallbackFunc set_image_scaling_callback_func_; + + // The hidden window hosting the magnifier control. + HWND host_window_; + // The magnifier control that captures the screen. + HWND magnifier_window_; + + // True if the magnifier control has been successfully initialized. + bool magnifier_initialized_; + + // True if the last OnMagImageScalingCallback was called and handled + // successfully. Reset at the beginning of each CaptureImage call. + bool magnifier_capture_succeeded_; + + DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWinMagnifier); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_MAGNIFIER_H_ diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.cc new file mode 100644 index 00000000000..03e021954b1 --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.cc @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/desktop_capture/win/window_capture_utils.h" + +namespace webrtc { + +bool +GetCroppedWindowRect(HWND window, + DesktopRect* cropped_rect, + DesktopRect* original_rect) { + RECT rect; + if (!GetWindowRect(window, &rect)) { + return false; + } + WINDOWPLACEMENT window_placement; + window_placement.length = sizeof(window_placement); + if (!GetWindowPlacement(window, &window_placement)) { + return false; + } + + *original_rect = DesktopRect::MakeLTRB( + rect.left, rect.top, rect.right, rect.bottom); + + if (window_placement.showCmd & SW_SHOWMAXIMIZED) { + DesktopSize border = DesktopSize(GetSystemMetrics(SM_CXSIZEFRAME), + GetSystemMetrics(SM_CYSIZEFRAME)); + *cropped_rect = DesktopRect::MakeLTRB( + rect.left + border.width(), + rect.top, + rect.right - border.width(), + rect.bottom - border.height()); + } else { + *cropped_rect = *original_rect; + } + return true; +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h b/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h new file mode 100644 index 00000000000..2a3a470c59e --- /dev/null +++ b/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h @@ -0,0 +1,25 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <windows.h> + +#include "webrtc/modules/desktop_capture/desktop_geometry.h" + +namespace webrtc { + +// Output the window rect, with the left/right/bottom frame border cropped if +// the window is maximized. |cropped_rect| is the cropped rect relative to the +// desktop. |original_rect| is the original rect returned from GetWindowRect. +// Returns true if all API calls succeeded. +bool GetCroppedWindowRect(HWND window, + DesktopRect* cropped_rect, + DesktopRect* original_rect); + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h index 478c8ee99c3..ad75c88d5d8 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h @@ -11,12 +11,12 @@ #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WINDOW_CAPTURER_H_ #define WEBRTC_MODULES_DESKTOP_CAPTURE_WINDOW_CAPTURER_H_ -#include <vector> #include <string> +#include <vector> +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/desktop_capture/desktop_capture_types.h" #include "webrtc/modules/desktop_capture/desktop_capturer.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -49,6 +49,14 @@ class WindowCapturer : public DesktopCapturer { // Select window to be captured. Returns false in case of a failure (e.g. if // there is no window with the specified id). virtual bool SelectWindow(WindowId id) = 0; + + // Bring the selected window to the front. Returns false in case of a + // failure or no window selected. + // TODO(jiayl): remove the default impl when FakeWindowCapturer is updated in + // Chromium. + virtual bool BringSelectedWindowToFront() { + return true; + } }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.mm index 6268fc01156..d177fc40c7d 100755..100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.mm @@ -12,6 +12,7 @@ #include <assert.h> #include <ApplicationServices/ApplicationServices.h> +#include <Cocoa/Cocoa.h> #include <CoreFoundation/CoreFoundation.h> #include "webrtc/modules/desktop_capture/desktop_frame.h" @@ -41,6 +42,18 @@ bool CFStringRefToUtf8(const CFStringRef string, std::string* str_utf8) { return true; } +bool IsWindowValid(CGWindowID id) { + CFArrayRef window_id_array = + CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL); + CFArrayRef window_array = + CGWindowListCreateDescriptionFromArray(window_id_array); + bool valid = window_array && CFArrayGetCount(window_array); + CFRelease(window_id_array); + CFRelease(window_array); + + return valid; +} + class WindowCapturerMac : public WindowCapturer { public: WindowCapturerMac(); @@ -49,6 +62,7 @@ class WindowCapturerMac : public WindowCapturer { // WindowCapturer interface. virtual bool GetWindowList(WindowList* windows) OVERRIDE; virtual bool SelectWindow(WindowId id) OVERRIDE; + virtual bool BringSelectedWindowToFront() OVERRIDE; // DesktopCapturer interface. virtual void Start(Callback* callback) OVERRIDE; @@ -113,24 +127,47 @@ bool WindowCapturerMac::GetWindowList(WindowList* windows) { } bool WindowCapturerMac::SelectWindow(WindowId id) { - // Request description for the specified window to make sure |id| is valid. + if (!IsWindowValid(id)) + return false; + window_id_ = id; + return true; +} + +bool WindowCapturerMac::BringSelectedWindowToFront() { + if (!window_id_) + return false; + CGWindowID ids[1]; - ids[0] = id; + ids[0] = window_id_; CFArrayRef window_id_array = CFArrayCreate(NULL, reinterpret_cast<const void **>(&ids), 1, NULL); + CFArrayRef window_array = CGWindowListCreateDescriptionFromArray(window_id_array); - int results_count = window_array ? CFArrayGetCount(window_array) : 0; - CFRelease(window_id_array); - CFRelease(window_array); - - if (results_count == 0) { + if (window_array == NULL || 0 == CFArrayGetCount(window_array)) { // Could not find the window. It might have been closed. + LOG(LS_INFO) << "Window not found"; + CFRelease(window_id_array); return false; } - window_id_ = id; - return true; + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, 0)); + CFNumberRef pid_ref = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowOwnerPID)); + + int pid; + CFNumberGetValue(pid_ref, kCFNumberIntType, &pid); + + // TODO(jiayl): this will bring the process main window to the front. We + // should find a way to bring only the window to the front. + bool result = + [[NSRunningApplication runningApplicationWithProcessIdentifier: pid] + activateWithOptions: NSApplicationActivateIgnoringOtherApps]; + + CFRelease(window_id_array); + CFRelease(window_array); + return result; } void WindowCapturerMac::Start(Callback* callback) { @@ -141,12 +178,16 @@ void WindowCapturerMac::Start(Callback* callback) { } void WindowCapturerMac::Capture(const DesktopRegion& region) { + if (!IsWindowValid(window_id_)) { + callback_->OnCaptureCompleted(NULL); + return; + } + CGImageRef window_image = CGWindowListCreateImage( CGRectNull, kCGWindowListOptionIncludingWindow, window_id_, kCGWindowImageBoundsIgnoreFraming); if (!window_image) { - CFRelease(window_image); callback_->OnCaptureCompleted(NULL); return; } diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc index 7bb1247ea9d..5f9010d2fd3 100755 --- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc @@ -26,6 +26,7 @@ class WindowCapturerNull : public WindowCapturer { // WindowCapturer interface. virtual bool GetWindowList(WindowList* windows) OVERRIDE; virtual bool SelectWindow(WindowId id) OVERRIDE; + virtual bool BringSelectedWindowToFront() OVERRIDE; // DesktopCapturer interface. virtual void Start(Callback* callback) OVERRIDE; @@ -54,6 +55,11 @@ bool WindowCapturerNull::SelectWindow(WindowId id) { return false; } +bool WindowCapturerNull::BringSelectedWindowToFront() { + // Not implemented yet. + return false; +} + void WindowCapturerNull::Start(Callback* callback) { assert(!callback_); assert(callback); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc index 95f41db73b2..a0021856203 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc @@ -11,9 +11,9 @@ #include "webrtc/modules/desktop_capture/window_capturer.h" #include <assert.h> -#include <windows.h> #include "webrtc/modules/desktop_capture/desktop_frame_win.h" +#include "webrtc/modules/desktop_capture/win/window_capture_utils.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" @@ -89,6 +89,7 @@ class WindowCapturerWin : public WindowCapturer { // WindowCapturer interface. virtual bool GetWindowList(WindowList* windows) OVERRIDE; virtual bool SelectWindow(WindowId id) OVERRIDE; + virtual bool BringSelectedWindowToFront() OVERRIDE; // DesktopCapturer interface. virtual void Start(Callback* callback) OVERRIDE; @@ -157,6 +158,16 @@ bool WindowCapturerWin::SelectWindow(WindowId id) { return true; } +bool WindowCapturerWin::BringSelectedWindowToFront() { + if (!window_) + return false; + + if (!IsWindow(window_) || !IsWindowVisible(window_) || IsIconic(window_)) + return false; + + return SetForegroundWindow(window_) != 0; +} + void WindowCapturerWin::Start(Callback* callback) { assert(!callback_); assert(callback); @@ -171,15 +182,16 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) { return; } - // Stop capturing if the window has been minimized or hidden. - if (IsIconic(window_) || !IsWindowVisible(window_)) { + // Stop capturing if the window has been closed or hidden. + if (!IsWindow(window_) || !IsWindowVisible(window_)) { callback_->OnCaptureCompleted(NULL); return; } - RECT rect; - if (!GetWindowRect(window_, &rect)) { - LOG(LS_WARNING) << "Failed to get window size: " << GetLastError(); + DesktopRect original_rect; + DesktopRect cropped_rect; + if (!GetCroppedWindowRect(window_, &cropped_rect, &original_rect)) { + LOG(LS_WARNING) << "Failed to get window info: " << GetLastError(); callback_->OnCaptureCompleted(NULL); return; } @@ -192,8 +204,7 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) { } scoped_ptr<DesktopFrameWin> frame(DesktopFrameWin::Create( - DesktopSize(rect.right - rect.left, rect.bottom - rect.top), - NULL, window_dc)); + cropped_rect.size(), NULL, window_dc)); if (!frame.get()) { ReleaseDC(window_, window_dc); callback_->OnCaptureCompleted(NULL); @@ -201,7 +212,7 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) { } HDC mem_dc = CreateCompatibleDC(window_dc); - SelectObject(mem_dc, frame->bitmap()); + HGDIOBJ previous_object = SelectObject(mem_dc, frame->bitmap()); BOOL result = FALSE; // When desktop composition (Aero) is enabled each window is rendered to a @@ -217,21 +228,24 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) { // When composition is enabled the DC returned by GetWindowDC() doesn't always // have window frame rendered correctly. Windows renders it only once and then // caches the result between captures. We hack it around by calling - // PrintWindow() whenever window size changes - it somehow affects what we - // get from BitBlt() on the subsequent captures. + // PrintWindow() whenever window size changes, including the first time of + // capturing - it somehow affects what we get from BitBlt() on the subsequent + // captures. - if (!IsAeroEnabled() || - (!previous_size_.is_empty() && !previous_size_.equals(frame->size()))) { + if (!IsAeroEnabled() || !previous_size_.equals(frame->size())) { result = PrintWindow(window_, mem_dc, 0); } // Aero is enabled or PrintWindow() failed, use BitBlt. if (!result) { result = BitBlt(mem_dc, 0, 0, frame->size().width(), frame->size().height(), - window_dc, 0, 0, SRCCOPY); + window_dc, + cropped_rect.left() - original_rect.left(), + cropped_rect.top() - original_rect.top(), + SRCCOPY); } - SelectObject(mem_dc, NULL); + SelectObject(mem_dc, previous_object); DeleteDC(mem_dc); ReleaseDC(window_, window_dc); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc index 5a14356fd16..b641c932180 100755 --- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc @@ -10,13 +10,14 @@ #include "webrtc/modules/desktop_capture/window_capturer.h" +#include <assert.h> #include <string.h> #include <X11/Xatom.h> #include <X11/extensions/Xcomposite.h> #include <X11/extensions/Xrender.h> #include <X11/Xutil.h> + #include <algorithm> -#include <cassert> #include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" @@ -84,7 +85,8 @@ class XWindowProperty { DISALLOW_COPY_AND_ASSIGN(XWindowProperty); }; -class WindowCapturerLinux : public WindowCapturer { +class WindowCapturerLinux : public WindowCapturer, + public SharedXDisplay::XEventHandler { public: WindowCapturerLinux(const DesktopCaptureOptions& options); virtual ~WindowCapturerLinux(); @@ -92,11 +94,15 @@ class WindowCapturerLinux : public WindowCapturer { // WindowCapturer interface. virtual bool GetWindowList(WindowList* windows) OVERRIDE; virtual bool SelectWindow(WindowId id) OVERRIDE; + virtual bool BringSelectedWindowToFront() OVERRIDE; // DesktopCapturer interface. virtual void Start(Callback* callback) OVERRIDE; virtual void Capture(const DesktopRegion& region) OVERRIDE; + // SharedXDisplay::XEventHandler interface. + virtual bool HandleXEvent(const XEvent& event) OVERRIDE; + private: Display* display() { return x_display_->display(); } @@ -146,9 +152,13 @@ WindowCapturerLinux::WindowCapturerLinux(const DesktopCaptureOptions& options) } else { LOG(LS_INFO) << "Xcomposite extension not available or too old."; } + + x_display_->AddEventHandler(ConfigureNotify, this); } -WindowCapturerLinux::~WindowCapturerLinux() {} +WindowCapturerLinux::~WindowCapturerLinux() { + x_display_->RemoveEventHandler(ConfigureNotify, this); +} bool WindowCapturerLinux::GetWindowList(WindowList* windows) { WindowList result; @@ -194,6 +204,9 @@ bool WindowCapturerLinux::SelectWindow(WindowId id) { if (!x_server_pixel_buffer_.Init(display(), id)) return false; + // Tell the X server to send us window resizing events. + XSelectInput(display(), id, StructureNotifyMask); + selected_window_ = id; // In addition to needing X11 server-side support for Xcomposite, it actually @@ -208,6 +221,55 @@ bool WindowCapturerLinux::SelectWindow(WindowId id) { return true; } +bool WindowCapturerLinux::BringSelectedWindowToFront() { + if (!selected_window_) + return false; + + unsigned int num_children; + ::Window* children; + ::Window parent; + ::Window root; + // Find the root window to pass event to. + int status = XQueryTree( + display(), selected_window_, &root, &parent, &children, &num_children); + if (status == 0) { + LOG(LS_ERROR) << "Failed to query for the root window."; + return false; + } + + if (children) + XFree(children); + + XRaiseWindow(display(), selected_window_); + + // Some window managers (e.g., metacity in GNOME) consider it illegal to + // raise a window without also giving it input focus with + // _NET_ACTIVE_WINDOW, so XRaiseWindow() on its own isn't enough. + Atom atom = XInternAtom(display(), "_NET_ACTIVE_WINDOW", True); + if (atom != None) { + XEvent xev; + xev.xclient.type = ClientMessage; + xev.xclient.serial = 0; + xev.xclient.send_event = True; + xev.xclient.window = selected_window_; + xev.xclient.message_type = atom; + + // The format member is set to 8, 16, or 32 and specifies whether the + // data should be viewed as a list of bytes, shorts, or longs. + xev.xclient.format = 32; + + memset(xev.xclient.data.l, 0, sizeof(xev.xclient.data.l)); + + XSendEvent(display(), + root, + False, + SubstructureRedirectMask | SubstructureNotifyMask, + &xev); + } + XFlush(display()); + return true; +} + void WindowCapturerLinux::Start(Callback* callback) { assert(!callback_); assert(callback); @@ -216,6 +278,14 @@ void WindowCapturerLinux::Start(Callback* callback) { } void WindowCapturerLinux::Capture(const DesktopRegion& region) { + if (!x_server_pixel_buffer_.IsWindowValid()) { + LOG(LS_INFO) << "The window is no longer valid."; + callback_->OnCaptureCompleted(NULL); + return; + } + + x_display_->ProcessPendingXEvents(); + if (!has_composite_extension_) { // Without the Xcomposite extension we capture when the whole window is // visible on screen and not covered by any other window. This is not @@ -235,6 +305,20 @@ void WindowCapturerLinux::Capture(const DesktopRegion& region) { callback_->OnCaptureCompleted(frame); } +bool WindowCapturerLinux::HandleXEvent(const XEvent& event) { + if (event.type == ConfigureNotify) { + XConfigureEvent xce = event.xconfigure; + if (!DesktopSize(xce.width, xce.height).equals( + x_server_pixel_buffer_.window_size())) { + if (!x_server_pixel_buffer_.Init(display(), selected_window_)) { + LOG(LS_ERROR) << "Failed to initialize pixel buffer after resizing."; + } + return true; + } + } + return false; +} + ::Window WindowCapturerLinux::GetApplicationWindow(::Window window) { // Get WM_STATE property of the window. XWindowProperty<uint32_t> window_state(display(), window, wm_state_atom_); diff --git a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_error_trap.h b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_error_trap.h index fd8346928c9..aa771145d59 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_error_trap.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_error_trap.h @@ -13,7 +13,7 @@ #include <X11/Xlib.h> -#include "webrtc/system_wrappers/interface/constructor_magic.h" +#include "webrtc/base/constructormagic.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc index 6983a6dcced..be00fa7697e 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc +++ b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc @@ -213,6 +213,18 @@ bool XServerPixelBuffer::InitPixmaps(int depth) { return true; } +bool XServerPixelBuffer::IsWindowValid() const { + XWindowAttributes attributes; + { + XErrorTrap error_trap(display_); + if (!XGetWindowAttributes(display_, window_, &attributes) || + error_trap.GetLastErrorAndDisable() != 0) { + return false; + } + } + return true; +} + void XServerPixelBuffer::Synchronize() { if (shm_segment_info_ && !shm_pixmap_) { // XShmGetImage can fail if the display is being reconfigured. diff --git a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h index b81096c8110..98f263f3a88 100644 --- a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h +++ b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h @@ -40,6 +40,9 @@ class XServerPixelBuffer { // Returns the size of the window the buffer was initialized for. const DesktopSize& window_size() { return window_size_; } + // Returns true if the window can be found. + bool IsWindowValid() const; + // If shared memory is being used without pixmaps, synchronize this pixel // buffer with the root window contents (otherwise, this is a no-op). // This is to avoid doing a full-screen capture for each individual diff --git a/chromium/third_party/webrtc/modules/interface/module_common_types.h b/chromium/third_party/webrtc/modules/interface/module_common_types.h index 2494d68b9ff..2c947071045 100644 --- a/chromium/third_party/webrtc/modules/interface/module_common_types.h +++ b/chromium/third_party/webrtc/modules/interface/module_common_types.h @@ -16,8 +16,8 @@ #include <algorithm> +#include "webrtc/base/constructormagic.h" #include "webrtc/common_types.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" #ifdef _WIN32 @@ -27,27 +27,6 @@ namespace webrtc { -struct RTPHeaderExtension { - bool hasTransmissionTimeOffset; - int32_t transmissionTimeOffset; - bool hasAbsoluteSendTime; - uint32_t absoluteSendTime; -}; - -struct RTPHeader { - bool markerBit; - uint8_t payloadType; - uint16_t sequenceNumber; - uint32_t timestamp; - uint32_t ssrc; - uint8_t numCSRCs; - uint32_t arrOfCSRCs[kRtpCsrcSize]; - uint8_t paddingLength; - uint16_t headerLength; - int payload_type_frequency; - RTPHeaderExtension extension; -}; - struct RTPAudioHeader { uint8_t numEnergy; // number of valid entries in arrOfEnergy uint8_t arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel @@ -124,6 +103,8 @@ struct WebRtcRTPHeader { RTPHeader header; FrameType frameType; RTPTypeHeader type; + // NTP time of the capture time in local timebase in milliseconds. + int64_t ntp_time_ms; }; class RTPFragmentationHeader { @@ -686,6 +667,10 @@ class AudioFrame { AudioFrame(); virtual ~AudioFrame() {} + // Resets all members to their default state (except does not modify the + // contents of |data_|). + void Reset(); + // |interleaved_| is not changed by this method. void UpdateFrame(int id, uint32_t timestamp, const int16_t* data, int samples_per_channel, int sample_rate_hz, @@ -703,13 +688,24 @@ class AudioFrame { AudioFrame& operator-=(const AudioFrame& rhs); int id_; + // RTP timestamp of the first sample in the AudioFrame. uint32_t timestamp_; + // Time since the first frame in milliseconds. + // -1 represents an uninitialized value. + int64_t elapsed_time_ms_; + // NTP time of the estimated capture time in local timebase in milliseconds. + // -1 represents an uninitialized value. + int64_t ntp_time_ms_; int16_t data_[kMaxDataSizeSamples]; int samples_per_channel_; int sample_rate_hz_; int num_channels_; SpeechType speech_type_; VADActivity vad_activity_; + // Note that there is no guarantee that |energy_| is correct. Any user of this + // member must verify that the value is correct. + // TODO(henrike) Remove |energy_|. + // See https://code.google.com/p/webrtc/issues/detail?id=3315. uint32_t energy_; bool interleaved_; @@ -718,16 +714,25 @@ class AudioFrame { }; inline AudioFrame::AudioFrame() - : id_(-1), - timestamp_(0), - data_(), - samples_per_channel_(0), - sample_rate_hz_(0), - num_channels_(1), - speech_type_(kUndefined), - vad_activity_(kVadUnknown), - energy_(0xffffffff), - interleaved_(true) {} + : data_() { + Reset(); +} + +inline void AudioFrame::Reset() { + id_ = -1; + // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize + // to an invalid value, or add a new member to indicate invalidity. + timestamp_ = 0; + elapsed_time_ms_ = -1; + ntp_time_ms_ = -1; + samples_per_channel_ = 0; + sample_rate_hz_ = 0; + num_channels_ = 0; + speech_type_ = kUndefined; + vad_activity_ = kVadUnknown; + energy_ = 0xffffffff; + interleaved_ = true; +} inline void AudioFrame::UpdateFrame(int id, uint32_t timestamp, const int16_t* data, @@ -758,6 +763,8 @@ inline void AudioFrame::CopyFrom(const AudioFrame& src) { id_ = src.id_; timestamp_ = src.timestamp_; + elapsed_time_ms_ = src.elapsed_time_ms_; + ntp_time_ms_ = src.ntp_time_ms_; samples_per_channel_ = src.samples_per_channel_; sample_rate_hz_ = src.sample_rate_hz_; speech_type_ = src.speech_type_; diff --git a/chromium/third_party/webrtc/modules/media_file/source/OWNERS b/chromium/third_party/webrtc/modules/media_file/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/media_file/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/media_file/source/avi_file.cc b/chromium/third_party/webrtc/modules/media_file/source/avi_file.cc index 92c51acce60..19baaa3b218 100644 --- a/chromium/third_party/webrtc/modules/media_file/source/avi_file.cc +++ b/chromium/third_party/webrtc/modules/media_file/source/avi_file.cc @@ -23,7 +23,6 @@ #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/file_wrapper.h" -#include "webrtc/system_wrappers/interface/list_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" // http://msdn2.microsoft.com/en-us/library/ms779636.aspx @@ -178,8 +177,7 @@ AviFile::AviFile() _videoCodecConfigParamsLength(0), _videoStreamDataChunkPrefix(0), _audioStreamDataChunkPrefix(0), - _created(false), - _indexList(new ListWrapper()) + _created(false) { ResetComplexMembers(); } @@ -188,7 +186,6 @@ AviFile::~AviFile() { Close(); - delete _indexList; delete[] _videoCodecConfigParams; delete _crit; } @@ -1712,21 +1709,11 @@ uint32_t AviFile::StreamAndTwoCharCodeToTag(int32_t streamNum, void AviFile::ClearIndexList() { - while (!_indexList->Empty()) - { - ListItem* listItem = _indexList->First(); - if (listItem == 0) - { - break; - } - - AVIINDEXENTRY* item = static_cast<AVIINDEXENTRY*>(listItem->GetItem()); - if (item != NULL) - { - delete item; - } - _indexList->PopFront(); - } + for (IndexList::iterator iter = _indexList.begin(); + iter != _indexList.end(); ++iter) { + delete *iter; + } + _indexList.clear(); } void AviFile::AddChunkToIndexList(uint32_t inChunkId, @@ -1734,7 +1721,7 @@ void AviFile::AddChunkToIndexList(uint32_t inChunkId, uint32_t inOffset, uint32_t inSize) { - _indexList->PushBack(new AVIINDEXENTRY(inChunkId, inFlags, inOffset, + _indexList.push_back(new AVIINDEXENTRY(inChunkId, inFlags, inOffset, inSize)); } @@ -1747,19 +1734,13 @@ void AviFile::WriteIndex() _bytesWritten += PutLE32(0); const size_t idxChunkSize = _bytesWritten; - for (ListItem* listItem = _indexList->First(); - listItem != NULL; - listItem = _indexList->Next(listItem)) - { - const AVIINDEXENTRY* item = - static_cast<AVIINDEXENTRY*>(listItem->GetItem()); - if (item != NULL) - { - _bytesWritten += PutLE32(item->ckid); - _bytesWritten += PutLE32(item->dwFlags); - _bytesWritten += PutLE32(item->dwChunkOffset); - _bytesWritten += PutLE32(item->dwChunkLength); - } + for (IndexList::iterator iter = _indexList.begin(); + iter != _indexList.end(); ++iter) { + const AVIINDEXENTRY* item = *iter; + _bytesWritten += PutLE32(item->ckid); + _bytesWritten += PutLE32(item->dwFlags); + _bytesWritten += PutLE32(item->dwChunkOffset); + _bytesWritten += PutLE32(item->dwChunkLength); } PutLE32LengthFromCurrent(static_cast<long>(idxChunkSize)); } diff --git a/chromium/third_party/webrtc/modules/media_file/source/avi_file.h b/chromium/third_party/webrtc/modules/media_file/source/avi_file.h index a55fc187481..d8b10626dff 100644 --- a/chromium/third_party/webrtc/modules/media_file/source/avi_file.h +++ b/chromium/third_party/webrtc/modules/media_file/source/avi_file.h @@ -14,12 +14,12 @@ #define WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_ #include <stdio.h> +#include <list> #include "webrtc/typedefs.h" namespace webrtc { class CriticalSectionWrapper; -class ListWrapper; struct AVISTREAMHEADER { @@ -194,6 +194,7 @@ private: void WriteIndex(); private: + typedef std::list<AVIINDEXENTRY*> IndexList; struct AVIMAINHEADER { AVIMAINHEADER(); @@ -269,7 +270,7 @@ private: uint32_t _audioStreamDataChunkPrefix; bool _created; - ListWrapper* _indexList; // Elements are of type AVIINDEXENTRY. + IndexList _indexList; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/media_file/source/media_file_utility.cc b/chromium/third_party/webrtc/modules/media_file/source/media_file_utility.cc index 85df0b3a40a..e8883c9a4ba 100644 --- a/chromium/third_party/webrtc/modules/media_file/source/media_file_utility.cc +++ b/chromium/third_party/webrtc/modules/media_file/source/media_file_utility.cc @@ -2521,6 +2521,7 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName, break; } #endif + break; } case kFileFormatPreencodedFile: { diff --git a/chromium/third_party/webrtc/modules/modules.gyp b/chromium/third_party/webrtc/modules/modules.gyp index ef354abc0fe..8dec125b093 100644 --- a/chromium/third_party/webrtc/modules/modules.gyp +++ b/chromium/third_party/webrtc/modules/modules.gyp @@ -16,9 +16,8 @@ 'audio_coding/codecs/isac/main/source/isac.gypi', 'audio_coding/codecs/isac/fix/source/isacfix.gypi', 'audio_coding/codecs/pcm16b/pcm16b.gypi', - 'audio_coding/main/source/audio_coding_module.gypi', + 'audio_coding/main/acm2/audio_coding_module.gypi', 'audio_coding/neteq/neteq.gypi', - 'audio_coding/neteq4/neteq.gypi', 'audio_conference_mixer/source/audio_conference_mixer.gypi', 'audio_device/audio_device.gypi', 'audio_processing/audio_processing.gypi', @@ -43,6 +42,7 @@ 'includes': [ 'audio_coding/codecs/isac/isac_test.gypi', 'audio_coding/codecs/isac/isacfix_test.gypi', + 'audio_coding/codecs/tools/audio_codec_speed_tests.gypi', 'audio_processing/audio_processing_tests.gypi', 'rtp_rtcp/test/testFec/test_fec.gypi', 'video_coding/main/source/video_coding_test.gypi', @@ -70,15 +70,13 @@ 'dependencies': [ 'audio_coding_module', 'audio_processing', - 'audioproc_unittest_proto', 'bitrate_controller', 'CNG', 'desktop_capture', 'iSACFix', 'media_file', - 'NetEq', - 'NetEq4', - 'NetEq4TestTools', + 'neteq', + 'neteq_test_tools', 'neteq_unittest_tools', 'paced_sender', 'PCM16B', # Needed by NetEq tests. @@ -99,13 +97,14 @@ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', '<(webrtc_root)/test/test.gyp:test_support_main', '<(webrtc_root)/test/test.gyp:frame_generator', + '<(webrtc_root)/test/test.gyp:rtcp_packet_parser', ], 'sources': [ 'audio_coding/main/acm2/acm_receiver_unittest.cc', + 'audio_coding/main/acm2/audio_coding_module_unittest.cc', 'audio_coding/main/acm2/call_statistics_unittest.cc', 'audio_coding/main/acm2/initial_delay_manager_unittest.cc', 'audio_coding/main/acm2/nack_unittest.cc', - 'audio_coding/main/source/acm_neteq_unittest.cc', 'audio_coding/codecs/cng/cng_unittest.cc', 'audio_coding/codecs/isac/fix/source/filters_unittest.cc', 'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc', @@ -113,47 +112,48 @@ 'audio_coding/codecs/isac/fix/source/transform_unittest.cc', 'audio_coding/codecs/isac/main/source/isac_unittest.cc', 'audio_coding/codecs/opus/opus_unittest.cc', - 'audio_coding/neteq4/audio_multi_vector_unittest.cc', - 'audio_coding/neteq4/audio_vector_unittest.cc', - 'audio_coding/neteq4/background_noise_unittest.cc', - 'audio_coding/neteq4/buffer_level_filter_unittest.cc', - 'audio_coding/neteq4/comfort_noise_unittest.cc', - 'audio_coding/neteq4/decision_logic_unittest.cc', - 'audio_coding/neteq4/decoder_database_unittest.cc', - 'audio_coding/neteq4/delay_manager_unittest.cc', - 'audio_coding/neteq4/delay_peak_detector_unittest.cc', - 'audio_coding/neteq4/dsp_helper_unittest.cc', - 'audio_coding/neteq4/dtmf_buffer_unittest.cc', - 'audio_coding/neteq4/dtmf_tone_generator_unittest.cc', - 'audio_coding/neteq4/expand_unittest.cc', - 'audio_coding/neteq4/merge_unittest.cc', - 'audio_coding/neteq4/neteq_external_decoder_unittest.cc', - 'audio_coding/neteq4/neteq_impl_unittest.cc', - 'audio_coding/neteq4/neteq_stereo_unittest.cc', - 'audio_coding/neteq4/neteq_unittest.cc', - 'audio_coding/neteq4/normal_unittest.cc', - 'audio_coding/neteq4/packet_buffer_unittest.cc', - 'audio_coding/neteq4/payload_splitter_unittest.cc', - 'audio_coding/neteq4/post_decode_vad_unittest.cc', - 'audio_coding/neteq4/random_vector_unittest.cc', - 'audio_coding/neteq4/sync_buffer_unittest.cc', - 'audio_coding/neteq4/timestamp_scaler_unittest.cc', - 'audio_coding/neteq4/time_stretch_unittest.cc', - 'audio_coding/neteq4/mock/mock_audio_decoder.h', - 'audio_coding/neteq4/mock/mock_audio_vector.h', - 'audio_coding/neteq4/mock/mock_buffer_level_filter.h', - 'audio_coding/neteq4/mock/mock_decoder_database.h', - 'audio_coding/neteq4/mock/mock_delay_manager.h', - 'audio_coding/neteq4/mock/mock_delay_peak_detector.h', - 'audio_coding/neteq4/mock/mock_dtmf_buffer.h', - 'audio_coding/neteq4/mock/mock_dtmf_tone_generator.h', - 'audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h', - 'audio_coding/neteq4/mock/mock_packet_buffer.h', - 'audio_coding/neteq4/mock/mock_payload_splitter.h', + 'audio_coding/neteq/audio_classifier_unittest.cc', + 'audio_coding/neteq/audio_multi_vector_unittest.cc', + 'audio_coding/neteq/audio_vector_unittest.cc', + 'audio_coding/neteq/background_noise_unittest.cc', + 'audio_coding/neteq/buffer_level_filter_unittest.cc', + 'audio_coding/neteq/comfort_noise_unittest.cc', + 'audio_coding/neteq/decision_logic_unittest.cc', + 'audio_coding/neteq/decoder_database_unittest.cc', + 'audio_coding/neteq/delay_manager_unittest.cc', + 'audio_coding/neteq/delay_peak_detector_unittest.cc', + 'audio_coding/neteq/dsp_helper_unittest.cc', + 'audio_coding/neteq/dtmf_buffer_unittest.cc', + 'audio_coding/neteq/dtmf_tone_generator_unittest.cc', + 'audio_coding/neteq/expand_unittest.cc', + 'audio_coding/neteq/merge_unittest.cc', + 'audio_coding/neteq/neteq_external_decoder_unittest.cc', + 'audio_coding/neteq/neteq_impl_unittest.cc', + 'audio_coding/neteq/neteq_stereo_unittest.cc', + 'audio_coding/neteq/neteq_unittest.cc', + 'audio_coding/neteq/normal_unittest.cc', + 'audio_coding/neteq/packet_buffer_unittest.cc', + 'audio_coding/neteq/payload_splitter_unittest.cc', + 'audio_coding/neteq/post_decode_vad_unittest.cc', + 'audio_coding/neteq/random_vector_unittest.cc', + 'audio_coding/neteq/sync_buffer_unittest.cc', + 'audio_coding/neteq/timestamp_scaler_unittest.cc', + 'audio_coding/neteq/time_stretch_unittest.cc', + 'audio_coding/neteq/mock/mock_audio_decoder.h', + 'audio_coding/neteq/mock/mock_audio_vector.h', + 'audio_coding/neteq/mock/mock_buffer_level_filter.h', + 'audio_coding/neteq/mock/mock_decoder_database.h', + 'audio_coding/neteq/mock/mock_delay_manager.h', + 'audio_coding/neteq/mock/mock_delay_peak_detector.h', + 'audio_coding/neteq/mock/mock_dtmf_buffer.h', + 'audio_coding/neteq/mock/mock_dtmf_tone_generator.h', + 'audio_coding/neteq/mock/mock_external_decoder_pcm16b.h', + 'audio_coding/neteq/mock/mock_packet_buffer.h', + 'audio_coding/neteq/mock/mock_payload_splitter.h', + 'audio_coding/neteq/tools/packet_unittest.cc', 'audio_processing/aec/system_delay_unittest.cc', 'audio_processing/aec/echo_cancellation_unittest.cc', 'audio_processing/echo_cancellation_impl_unittest.cc', - 'audio_processing/test/audio_processing_unittest.cc', 'audio_processing/utility/delay_estimator_unittest.cc', 'audio_processing/utility/ring_buffer_unittest.cc', 'bitrate_controller/bitrate_controller_unittest.cc', @@ -167,19 +167,19 @@ 'desktop_capture/screen_capturer_mock_objects.h', 'desktop_capture/screen_capturer_unittest.cc', 'desktop_capture/window_capturer_unittest.cc', - "desktop_capture/win/cursor_unittest.cc", - "desktop_capture/win/cursor_unittest_resources.h", - "desktop_capture/win/cursor_unittest_resources.rc", + 'desktop_capture/win/cursor_unittest.cc', + 'desktop_capture/win/cursor_unittest_resources.h', + 'desktop_capture/win/cursor_unittest_resources.rc', 'media_file/source/media_file_unittest.cc', 'module_common_types_unittest.cc', 'pacing/paced_sender_unittest.cc', + 'remote_bitrate_estimator/bwe_simulations.cc', 'remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h', 'remote_bitrate_estimator/rate_statistics_unittest.cc', 'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc', 'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc', 'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h', 'remote_bitrate_estimator/remote_bitrate_estimators_test.cc', - 'remote_bitrate_estimator/rtp_to_ntp_unittest.cc', 'remote_bitrate_estimator/test/bwe_test_baselinefile.cc', 'remote_bitrate_estimator/test/bwe_test_baselinefile.h', 'remote_bitrate_estimator/test/bwe_test_fileutils.cc', @@ -199,9 +199,11 @@ 'rtp_rtcp/source/nack_rtx_unittest.cc', 'rtp_rtcp/source/producer_fec_unittest.cc', 'rtp_rtcp/source/receive_statistics_unittest.cc', + 'rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc', 'rtp_rtcp/source/rtcp_format_remb_unittest.cc', - 'rtp_rtcp/source/rtcp_sender_unittest.cc', + 'rtp_rtcp/source/rtcp_packet_unittest.cc', 'rtp_rtcp/source/rtcp_receiver_unittest.cc', + 'rtp_rtcp/source/rtcp_sender_unittest.cc', 'rtp_rtcp/source/rtp_fec_unittest.cc', 'rtp_rtcp/source/rtp_format_vp8_unittest.cc', 'rtp_rtcp/source/rtp_format_vp8_test_helper.cc', @@ -219,6 +221,7 @@ 'rtp_rtcp/test/testAPI/test_api_rtcp.cc', 'rtp_rtcp/test/testAPI/test_api_video.cc', 'utility/source/audio_frame_operations_unittest.cc', + 'utility/source/file_player_unittests.cc', 'video_coding/codecs/test/packet_manipulator_unittest.cc', 'video_coding/codecs/test/stats_unittest.cc', 'video_coding/codecs/test/videoprocessor_unittest.cc', @@ -278,6 +281,14 @@ }], ['enable_protobuf==1', { 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ], + 'dependencies': [ + 'audioproc_unittest_proto', + ], + 'sources': [ + 'audio_processing/audio_processing_impl_unittest.cc', + 'audio_processing/test/audio_processing_unittest.cc', + 'audio_processing/test/test_utils.h', + ], }], ['build_libvpx==1', { 'dependencies': [ @@ -286,7 +297,7 @@ }], # TODO(henrike): remove build_with_chromium==1 when the bots are # using Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { + ['build_with_chromium==1 and OS=="android"', { 'dependencies': [ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', ], @@ -326,13 +337,14 @@ 'audio_coding/main/test/EncodeDecodeTest.cc', 'audio_coding/main/test/iSACTest.cc', 'audio_coding/main/test/opus_test.cc', + 'audio_coding/main/test/PacketLossTest.cc', 'audio_coding/main/test/PCMFile.cc', 'audio_coding/main/test/RTPFile.cc', 'audio_coding/main/test/SpatialAudio.cc', 'audio_coding/main/test/TestAllCodecs.cc', 'audio_coding/main/test/target_delay_unittest.cc', 'audio_coding/main/test/Tester.cc', - 'audio_coding/main/test/TestFEC.cc', + 'audio_coding/main/test/TestRedFec.cc', 'audio_coding/main/test/TestStereo.cc', 'audio_coding/main/test/TestVADDTX.cc', 'audio_coding/main/test/TimedTrace.cc', @@ -346,7 +358,7 @@ 'conditions': [ # TODO(henrike): remove build_with_chromium==1 when the bots are # using Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { + ['build_with_chromium==1 and OS=="android"', { 'dependencies': [ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', ], @@ -357,7 +369,7 @@ 'conditions': [ # TODO(henrike): remove build_with_chromium==1 when the bots are using # Chromium's buildbots. - ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', { + ['build_with_chromium==1 and OS=="android"', { 'targets': [ { 'target_name': 'modules_unittests_apk_target', diff --git a/chromium/third_party/webrtc/modules/modules_tests.isolate b/chromium/third_party/webrtc/modules/modules_tests.isolate index 7a051f66eb6..e5055f0d91e 100644 --- a/chromium/third_party/webrtc/modules/modules_tests.isolate +++ b/chromium/third_party/webrtc/modules/modules_tests.isolate @@ -8,32 +8,30 @@ { 'conditions': [ ['OS=="android"', { - # When doing Android builds, the WebRTC code is put in third_party/webrtc - # of a Chromium checkout, this is one level above the standalone build. 'variables': { 'isolate_dependency_untracked': [ - '../../../data/', - '../../../resources/', + '<(DEPTH)/data/', + '<(DEPTH)/resources/', ], }, }], ['OS=="linux" or OS=="mac" or OS=="win"', { 'variables': { 'command': [ - '../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/modules_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_tracked': [ - '../../DEPS', - '../../resources/audio_coding/testfile32kHz.pcm', - '../../resources/audio_coding/teststereo32kHz.pcm', - '../../resources/foreman_cif.yuv', - '../../resources/paris_qcif.yuv', - '../../testing/test_env.py', + '<(DEPTH)/DEPS', + '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm', + '<(DEPTH)/resources/audio_coding/teststereo32kHz.pcm', + '<(DEPTH)/resources/foreman_cif.yuv', + '<(DEPTH)/resources/paris_qcif.yuv', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/modules_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_untracked': [ - '../../tools/swarming_client/', + '<(DEPTH)/tools/swarming_client/', ], }, }], diff --git a/chromium/third_party/webrtc/modules/modules_unittests.isolate b/chromium/third_party/webrtc/modules/modules_unittests.isolate index aa91238caaf..09ace1c1816 100644 --- a/chromium/third_party/webrtc/modules/modules_unittests.isolate +++ b/chromium/third_party/webrtc/modules/modules_unittests.isolate @@ -8,77 +8,105 @@ { 'conditions': [ ['OS=="android"', { - # When doing Android builds, the WebRTC code is put in third_party/webrtc - # of a Chromium checkout, this is one level above the standalone build. 'variables': { 'isolate_dependency_untracked': [ - '../../../data/', - '../../../resources/', + '<(DEPTH)/data/', + '<(DEPTH)/resources/', + ], + 'isolate_dependency_tracked': [ + '<(DEPTH)/resources/short_mixed_mono_48.dat', + '<(DEPTH)/resources/short_mixed_mono_48.pcm', + '<(DEPTH)/resources/short_mixed_stereo_48.dat', + '<(DEPTH)/resources/short_mixed_stereo_48.pcm', ], }, }], ['OS=="linux" or OS=="mac" or OS=="win"', { 'variables': { 'command': [ - '../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/modules_unittests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_tracked': [ - '../../DEPS', - '../../data/audio_processing/output_data_float.pb', - '../../data/voice_engine/audio_tiny48.wav', - '../../resources/audio_coding/neteq4_network_stats.dat', - '../../resources/audio_coding/neteq4_rtcp_stats.dat', - '../../resources/audio_coding/neteq4_universal_ref.pcm', - '../../resources/audio_coding/neteq_network_stats.dat', - '../../resources/audio_coding/neteq_rtcp_stats.dat', - '../../resources/audio_coding/neteq_universal_new.rtp', - '../../resources/audio_coding/neteq_universal_ref.pcm', - '../../resources/audio_coding/testfile32kHz.pcm', - '../../resources/deflicker_before_cif_short.yuv', - '../../resources/far16_stereo.pcm', - '../../resources/far32_stereo.pcm', - '../../resources/far8_stereo.pcm', - '../../resources/foremanColorEnhanced_cif_short.yuv', - '../../resources/foreman_cif.yuv', - '../../resources/foreman_cif_short.yuv', - '../../resources/near16_stereo.pcm', - '../../resources/near32_stereo.pcm', - '../../resources/near8_stereo.pcm', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_TOF.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin', - '../../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin', - '../../resources/video_coding/frame-ethernet-ii.pcap', - '../../resources/video_coding/frame-loopback.pcap', - '../../resources/video_coding/pltype103.rtp', - '../../resources/video_coding/ssrcs-2.pcap', - '../../resources/video_coding/ssrcs-3.pcap', - '../../testing/test_env.py', + '<(DEPTH)/DEPS', + '<(DEPTH)/data/audio_processing/output_data_float.pb', + '<(DEPTH)/data/voice_engine/audio_tiny48.wav', + '<(DEPTH)/resources/att-downlink.rx', + '<(DEPTH)/resources/att-uplink.rx', + '<(DEPTH)/resources/audio_coding/neteq4_network_stats.dat', + '<(DEPTH)/resources/audio_coding/neteq4_rtcp_stats.dat', + '<(DEPTH)/resources/audio_coding/neteq4_universal_ref.pcm', + '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_32.pcm', + '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_64.pcm', + '<(DEPTH)/resources/audio_coding/neteq_network_stats.dat', + '<(DEPTH)/resources/audio_coding/neteq_rtcp_stats.dat', + '<(DEPTH)/resources/audio_coding/neteq_universal_new.rtp', + '<(DEPTH)/resources/audio_coding/neteq_universal_ref.pcm', + '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm', + '<(DEPTH)/resources/deflicker_before_cif_short.yuv', + '<(DEPTH)/resources/far16_stereo.pcm', + '<(DEPTH)/resources/far32_stereo.pcm', + '<(DEPTH)/resources/far44_stereo.pcm', + '<(DEPTH)/resources/far48_stereo.pcm', + '<(DEPTH)/resources/far8_stereo.pcm', + '<(DEPTH)/resources/foremanColorEnhanced_cif_short.yuv', + '<(DEPTH)/resources/foreman_cif.yuv', + '<(DEPTH)/resources/foreman_cif_short.yuv', + '<(DEPTH)/resources/near16_stereo.pcm', + '<(DEPTH)/resources/near32_stereo.pcm', + '<(DEPTH)/resources/near44_stereo.pcm', + '<(DEPTH)/resources/near48_stereo.pcm', + '<(DEPTH)/resources/near8_stereo.pcm', + '<(DEPTH)/resources/ref03.aecdump', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_TOF.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin', + '<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin', + '<(DEPTH)/resources/short_mixed_mono_48.dat', + '<(DEPTH)/resources/short_mixed_mono_48.pcm', + '<(DEPTH)/resources/short_mixed_stereo_48.dat', + '<(DEPTH)/resources/short_mixed_stereo_48.pcm', + '<(DEPTH)/resources/sprint-downlink.rx', + '<(DEPTH)/resources/sprint-uplink.rx', + '<(DEPTH)/resources/synthetic-trace.rx', + '<(DEPTH)/resources/tmobile-downlink.rx', + '<(DEPTH)/resources/tmobile-uplink.rx', + '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav', + '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav', + '<(DEPTH)/resources/verizon3g-downlink.rx', + '<(DEPTH)/resources/verizon3g-uplink.rx', + '<(DEPTH)/resources/verizon4g-downlink.rx', + '<(DEPTH)/resources/verizon4g-uplink.rx', + '<(DEPTH)/resources/video_coding/frame-ethernet-ii.pcap', + '<(DEPTH)/resources/video_coding/frame-loopback.pcap', + '<(DEPTH)/resources/video_coding/pltype103.rtp', + '<(DEPTH)/resources/video_coding/ssrcs-2.pcap', + '<(DEPTH)/resources/video_coding/ssrcs-3.pcap', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/modules_unittests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_untracked': [ - '../../tools/swarming_client/', + '<(DEPTH)/tools/swarming_client/', ], }, }], diff --git a/chromium/third_party/webrtc/modules/pacing/OWNERS b/chromium/third_party/webrtc/modules/pacing/OWNERS index 933a045009b..1426abc6215 100644 --- a/chromium/third_party/webrtc/modules/pacing/OWNERS +++ b/chromium/third_party/webrtc/modules/pacing/OWNERS @@ -2,3 +2,8 @@ pwestin@webrtc.org stefan@webrtc.org mflodman@webrtc.org asapersson@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/pacing/include/paced_sender.h b/chromium/third_party/webrtc/modules/pacing/include/paced_sender.h index 045469009f0..95f1a86e8f6 100644 --- a/chromium/third_party/webrtc/modules/pacing/include/paced_sender.h +++ b/chromium/third_party/webrtc/modules/pacing/include/paced_sender.h @@ -49,14 +49,14 @@ class PacedSender : public Module { bool retransmission) = 0; // Called when it's a good time to send a padding data. virtual int TimeToSendPadding(int bytes) = 0; + protected: virtual ~Callback() {} }; static const int kDefaultMaxQueueLengthMs = 2000; - PacedSender(Callback* callback, int target_bitrate_kbps, - float pace_multiplier); + PacedSender(Callback* callback, int max_bitrate_kbps, int min_bitrate_kbps); virtual ~PacedSender(); @@ -71,13 +71,9 @@ class PacedSender : public Module { // Resume sending packets. void Resume(); - // Set the pacing target bitrate and the bitrate up to which we are allowed to - // pad. We will send padding packets to increase the total bitrate until we - // reach |pad_up_to_bitrate_kbps|. If the media bitrate is above - // |pad_up_to_bitrate_kbps| no padding will be sent. - void UpdateBitrate(int target_bitrate_kbps, - int max_padding_bitrate_kbps, - int pad_up_to_bitrate_kbps); + // Set target bitrates for the pacer. Padding packets will be utilized to + // reach |min_bitrate| unless enough media packets are available. + void UpdateBitrate(int max_bitrate_kbps, int min_bitrate_kbps); // Returns true if we send the packet now, else it will add the packet // information to the queue and call TimeToSendPacket when it's time to send. @@ -119,7 +115,6 @@ class PacedSender : public Module { void UpdateMediaBytesSent(int num_bytes); Callback* callback_; - const float pace_multiplier_; bool enabled_; bool paused_; int max_queue_length_ms_; @@ -128,12 +123,9 @@ class PacedSender : public Module { // we can pace out during the current interval. scoped_ptr<paced_sender::IntervalBudget> media_budget_; // This is the padding budget, keeping track of how many bits of padding we're - // allowed to send out during the current interval. + // allowed to send out during the current interval. This budget will be + // utilized when there's no media to send. scoped_ptr<paced_sender::IntervalBudget> padding_budget_; - // Media and padding share this budget, therefore no padding will be sent if - // media uses all of this budget. This is used to avoid padding above a given - // bitrate. - scoped_ptr<paced_sender::IntervalBudget> pad_up_to_bitrate_budget_; TickTime time_last_update_; TickTime time_last_send_; diff --git a/chromium/third_party/webrtc/modules/pacing/paced_sender.cc b/chromium/third_party/webrtc/modules/pacing/paced_sender.cc index c46bd04ed73..e9f9bddced1 100644 --- a/chromium/third_party/webrtc/modules/pacing/paced_sender.cc +++ b/chromium/third_party/webrtc/modules/pacing/paced_sender.cc @@ -120,19 +120,16 @@ class IntervalBudget { }; } // namespace paced_sender -PacedSender::PacedSender(Callback* callback, int target_bitrate_kbps, - float pace_multiplier) +PacedSender::PacedSender(Callback* callback, + int max_bitrate_kbps, + int min_bitrate_kbps) : callback_(callback), - pace_multiplier_(pace_multiplier), - enabled_(false), + enabled_(true), paused_(false), max_queue_length_ms_(kDefaultMaxQueueLengthMs), critsect_(CriticalSectionWrapper::CreateCriticalSection()), - media_budget_(new paced_sender::IntervalBudget( - pace_multiplier_ * target_bitrate_kbps)), - padding_budget_(new paced_sender::IntervalBudget(0)), - // No padding until UpdateBitrate is called. - pad_up_to_bitrate_budget_(new paced_sender::IntervalBudget(0)), + media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)), + padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)), time_last_update_(TickTime::Now()), capture_time_ms_last_queued_(0), capture_time_ms_last_sent_(0), @@ -165,13 +162,11 @@ bool PacedSender::Enabled() const { return enabled_; } -void PacedSender::UpdateBitrate(int target_bitrate_kbps, - int max_padding_bitrate_kbps, - int pad_up_to_bitrate_kbps) { +void PacedSender::UpdateBitrate(int max_bitrate_kbps, + int min_bitrate_kbps) { CriticalSectionScoped cs(critsect_.get()); - media_budget_->set_target_rate_kbps(pace_multiplier_ * target_bitrate_kbps); - padding_budget_->set_target_rate_kbps(max_padding_bitrate_kbps); - pad_up_to_bitrate_budget_->set_target_rate_kbps(pad_up_to_bitrate_kbps); + media_budget_->set_target_rate_kbps(max_bitrate_kbps); + padding_budget_->set_target_rate_kbps(min_bitrate_kbps); } bool PacedSender::SendPacket(Priority priority, uint32_t ssrc, @@ -273,24 +268,21 @@ int32_t PacedSender::Process() { if (high_priority_packets_->empty() && normal_priority_packets_->empty() && low_priority_packets_->empty() && - padding_budget_->bytes_remaining() > 0 && - pad_up_to_bitrate_budget_->bytes_remaining() > 0) { - int padding_needed = std::min( - padding_budget_->bytes_remaining(), - pad_up_to_bitrate_budget_->bytes_remaining()); + padding_budget_->bytes_remaining() > 0) { + int padding_needed = padding_budget_->bytes_remaining(); critsect_->Leave(); int bytes_sent = callback_->TimeToSendPadding(padding_needed); critsect_->Enter(); media_budget_->UseBudget(bytes_sent); padding_budget_->UseBudget(bytes_sent); - pad_up_to_bitrate_budget_->UseBudget(bytes_sent); } } return 0; } // MUST have critsect_ when calling. -bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) { +bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) + EXCLUSIVE_LOCKS_REQUIRED(critsect_.get()) { paced_sender::Packet packet = GetNextPacketFromList(packet_list); critsect_->Leave(); @@ -323,7 +315,6 @@ bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) { void PacedSender::UpdateBytesPerInterval(uint32_t delta_time_ms) { media_budget_->IncreaseBudget(delta_time_ms); padding_budget_->IncreaseBudget(delta_time_ms); - pad_up_to_bitrate_budget_->IncreaseBudget(delta_time_ms); } // MUST have critsect_ when calling. @@ -387,7 +378,7 @@ paced_sender::Packet PacedSender::GetNextPacketFromList( void PacedSender::UpdateMediaBytesSent(int num_bytes) { time_last_send_ = TickTime::Now(); media_budget_->UseBudget(num_bytes); - pad_up_to_bitrate_budget_->UseBudget(num_bytes); + padding_budget_->UseBudget(num_bytes); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc b/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc index f8dcdfc6985..9763c80070d 100644 --- a/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc +++ b/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc @@ -59,9 +59,8 @@ class PacedSenderTest : public ::testing::Test { srand(0); TickTime::UseFakeClock(123456); // Need to initialize PacedSender after we initialize clock. - send_bucket_.reset(new PacedSender(&callback_, kTargetBitrate, - kPaceMultiplier)); - send_bucket_->SetStatus(true); + send_bucket_.reset( + new PacedSender(&callback_, kPaceMultiplier * kTargetBitrate, 0)); } void SendAndExpectPacket(PacedSender::Priority priority, @@ -209,7 +208,7 @@ TEST_F(PacedSenderTest, Padding) { uint32_t ssrc = 12345; uint16_t sequence_number = 1234; - send_bucket_->UpdateBitrate(kTargetBitrate, kTargetBitrate, kTargetBitrate); + send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); // Due to the multiplicative factor we can send 3 packets not 2 packets. SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, TickTime::MillisecondTimestamp(), 250, false); @@ -235,7 +234,7 @@ TEST_F(PacedSenderTest, Padding) { TEST_F(PacedSenderTest, NoPaddingWhenDisabled) { send_bucket_->SetStatus(false); - send_bucket_->UpdateBitrate(kTargetBitrate, kTargetBitrate, kTargetBitrate); + send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); // No padding is expected since the pacer is disabled. EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); @@ -255,7 +254,7 @@ TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) { int64_t capture_time_ms = 56789; const int kTimeStep = 5; const int64_t kBitrateWindow = 100; - send_bucket_->UpdateBitrate(kTargetBitrate, kTargetBitrate, kTargetBitrate); + send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); int64_t start_time = TickTime::MillisecondTimestamp(); while (TickTime::MillisecondTimestamp() - start_time < kBitrateWindow) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, @@ -267,27 +266,6 @@ TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) { } } -TEST_F(PacedSenderTest, VerifyMaxPaddingBitrate) { - uint32_t ssrc = 12345; - uint16_t sequence_number = 1234; - int64_t capture_time_ms = 56789; - const int kTimeStep = 5; - const int64_t kBitrateWindow = 100; - const int kTargetBitrate = 1500; - const int kMaxPaddingBitrate = 800; - send_bucket_->UpdateBitrate(kTargetBitrate, kMaxPaddingBitrate, - kTargetBitrate); - int64_t start_time = TickTime::MillisecondTimestamp(); - while (TickTime::MillisecondTimestamp() - start_time < kBitrateWindow) { - SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - capture_time_ms, 250, false); - TickTime::AdvanceFakeClock(kTimeStep); - EXPECT_CALL(callback_, TimeToSendPadding(500)).Times(1). - WillOnce(Return(250)); - send_bucket_->Process(); - } -} - TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) { uint32_t ssrc = 12345; uint16_t sequence_number = 1234; @@ -295,10 +273,9 @@ TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) { const int kTimeStep = 5; const int64_t kBitrateWindow = 10000; PacedSenderPadding callback; - send_bucket_.reset(new PacedSender(&callback, kTargetBitrate, - kPaceMultiplier)); - send_bucket_->SetStatus(true); - send_bucket_->UpdateBitrate(kTargetBitrate, kTargetBitrate, kTargetBitrate); + send_bucket_.reset( + new PacedSender(&callback, kPaceMultiplier * kTargetBitrate, 0)); + send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); int64_t start_time = TickTime::MillisecondTimestamp(); int media_bytes = 0; while (TickTime::MillisecondTimestamp() - start_time < kBitrateWindow) { @@ -421,7 +398,8 @@ TEST_F(PacedSenderTest, Pause) { EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); - EXPECT_CALL(callback_, TimeToSendPacket(_, _, second_capture_time_ms, false)) + EXPECT_CALL( + callback_, TimeToSendPacket(_, _, second_capture_time_ms, false)) .Times(1) .WillRepeatedly(Return(true)); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); @@ -496,7 +474,7 @@ TEST_F(PacedSenderTest, MaxQueueLength) { uint16_t sequence_number = 1234; EXPECT_EQ(0, send_bucket_->QueueInMs()); - send_bucket_->UpdateBitrate(30, 0, 0); + send_bucket_->UpdateBitrate(kPaceMultiplier * 30, 0); for (int i = 0; i < 30; ++i) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, @@ -525,7 +503,7 @@ TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) { uint16_t sequence_number = 1234; EXPECT_EQ(0, send_bucket_->QueueInMs()); - send_bucket_->UpdateBitrate(30, 0, 0); + send_bucket_->UpdateBitrate(kPaceMultiplier * 30, 0); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number, diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/OWNERS b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/OWNERS index b705ede2a00..eab2b8e0041 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/OWNERS +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/OWNERS @@ -2,4 +2,8 @@ pwestin@webrtc.org stefan@webrtc.org henrik.lundin@webrtc.org mflodman@webrtc.org -asapersson@webrtc.org
\ No newline at end of file +asapersson@webrtc.org +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc new file mode 100644 index 00000000000..6b208e4999d --- /dev/null +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc @@ -0,0 +1,162 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "gtest/gtest.h" +#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" +#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h" +#include "webrtc/test/testsupport/fileutils.h" + +using std::string; + +namespace webrtc { +namespace testing { +namespace bwe { +#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE +BweTestConfig::EstimatorConfig CreateEstimatorConfig( + int flow_id, bool plot_delay, bool plot_estimate) { + static const AbsoluteSendTimeRemoteBitrateEstimatorFactory factory = + AbsoluteSendTimeRemoteBitrateEstimatorFactory(); + + return BweTestConfig::EstimatorConfig("AST", flow_id, &factory, kAimdControl, + plot_delay, plot_estimate); +} + +BweTestConfig MakeAdaptiveBweTestConfig() { + BweTestConfig result; + result.estimator_configs.push_back(CreateEstimatorConfig(0, true, true)); + return result; +} + +BweTestConfig MakeMultiFlowBweTestConfig(int flow_count) { + BweTestConfig result; + for (int i = 0; i < flow_count; ++i) { + result.estimator_configs.push_back(CreateEstimatorConfig(i, false, true)); + } + return result; +} + +// This test fixture is used to instantiate tests running with adaptive video +// senders. +class BweSimulation : public BweTest, + public ::testing::TestWithParam<BweTestConfig> { + public: + BweSimulation() : BweTest() {} + virtual ~BweSimulation() {} + + virtual void SetUp() { + const BweTestConfig& config = GetParam(); + SetupTestFromConfig(config); + } + + private: + DISALLOW_COPY_AND_ASSIGN(BweSimulation); +}; + +INSTANTIATE_TEST_CASE_P(VideoSendersTest, BweSimulation, + ::testing::Values(MakeAdaptiveBweTestConfig())); + +TEST_P(BweSimulation, SprintUplinkTest) { + VerboseLogging(true); + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + RateCounterFilter counter1(this, "sender_output"); + TraceBasedDeliveryFilter filter(this, "link_capacity"); + RateCounterFilter counter2(this, "receiver_input"); + ASSERT_TRUE(filter.Init(test::ResourcePath("sprint-uplink", "rx"))); + RunFor(60 * 1000); +} + +TEST_P(BweSimulation, Verizon4gDownlinkTest) { + VerboseLogging(true); + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + RateCounterFilter counter1(this, "sender_output"); + TraceBasedDeliveryFilter filter(this, "link_capacity"); + RateCounterFilter counter2(this, "receiver_input"); + ASSERT_TRUE(filter.Init(test::ResourcePath("verizon4g-downlink", "rx"))); + RunFor(22 * 60 * 1000); +} + +TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) { + VerboseLogging(true); + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + ChokeFilter filter(this); + RateCounterFilter counter(this, "receiver_input"); + filter.SetCapacity(1000); + filter.SetMaxDelay(500); + RunFor(60 * 1000); + filter.SetCapacity(500); + RunFor(60 * 1000); + filter.SetCapacity(1000); + RunFor(60 * 1000); +} + +TEST_P(BweSimulation, Choke200kbps30kbps200kbps) { + VerboseLogging(true); + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + ChokeFilter filter(this); + RateCounterFilter counter(this, "receiver_input"); + filter.SetCapacity(200); + filter.SetMaxDelay(500); + RunFor(60 * 1000); + filter.SetCapacity(30); + RunFor(60 * 1000); + filter.SetCapacity(200); + RunFor(60 * 1000); +} + +TEST_P(BweSimulation, GoogleWifiTrace3Mbps) { + VerboseLogging(true); + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + RateCounterFilter counter1(this, "sender_output"); + TraceBasedDeliveryFilter filter(this, "link_capacity"); + filter.SetMaxDelay(500); + RateCounterFilter counter2(this, "receiver_input"); + ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx"))); + RunFor(300 * 1000); +} + +class MultiFlowBweSimulation : public BweSimulation { + public: + MultiFlowBweSimulation() : BweSimulation() {} + virtual ~MultiFlowBweSimulation() {} + + private: + DISALLOW_COPY_AND_ASSIGN(MultiFlowBweSimulation); +}; + +INSTANTIATE_TEST_CASE_P(VideoSendersTest, MultiFlowBweSimulation, + ::testing::Values(MakeMultiFlowBweTestConfig(3))); + +TEST_P(MultiFlowBweSimulation, SelfFairnessTest) { + VerboseLogging(true); + const int kAllFlowIds[] = {0, 1, 2}; + const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]); + scoped_ptr<AdaptiveVideoSender> senders[kNumFlows]; + for (size_t i = 0; i < kNumFlows; ++i) { + senders[i].reset(new AdaptiveVideoSender(kAllFlowIds[i], this, 30, 300, 0, + 0)); + } + // Second and third flow. + ChokeFilter choke(this, CreateFlowIds(&kAllFlowIds[1], 2)); + choke.SetCapacity(1500); + // First flow. + ChokeFilter choke2(this, CreateFlowIds(&kAllFlowIds[0], 1)); + choke2.SetCapacity(1000); + + scoped_ptr<RateCounterFilter> rate_counters[kNumFlows]; + for (size_t i = 0; i < kNumFlows; ++i) { + rate_counters[i].reset(new RateCounterFilter( + this, CreateFlowIds(&kAllFlowIds[i], 1), "receiver_input")); + } + RunFor(30 * 60 * 1000); +} +#endif // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE +} // namespace bwe +} // namespace testing +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h index 170ec765c1e..e61e903558f 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h @@ -25,6 +25,11 @@ namespace webrtc { class Clock; +enum RateControlType { + kMimdControl, + kAimdControl +}; + // RemoteBitrateObserver is used to signal changes in bitrate estimates for // the incoming streams. class RemoteBitrateObserver { @@ -37,6 +42,27 @@ class RemoteBitrateObserver { virtual ~RemoteBitrateObserver() {} }; +struct ReceiveBandwidthEstimatorStats { + ReceiveBandwidthEstimatorStats() : total_propagation_time_delta_ms(0) {} + + // The "propagation_time_delta" of a frame is defined as (d_arrival - d_sent), + // where d_arrival is the delta of the arrival times of the frame and the + // previous frame, d_sent is the delta of the sent times of the frame and + // the previous frame. The sent time is calculated from the RTP timestamp. + + // |total_propagation_time_delta_ms| is the sum of the propagation_time_deltas + // of all received frames, except that it's is adjusted to 0 when it becomes + // negative. + int total_propagation_time_delta_ms; + // The propagation_time_deltas for the frames arrived in the last + // kProcessIntervalMs using the clock passed to + // RemoteBitrateEstimatorFactory::Create. + std::vector<int> recent_propagation_time_delta_ms; + // The arrival times for the frames arrived in the last kProcessIntervalMs + // using the clock passed to RemoteBitrateEstimatorFactory::Create. + std::vector<int64_t> recent_arrival_time_ms; +}; + class RemoteBitrateEstimator : public CallStatsObserver, public Module { public: virtual ~RemoteBitrateEstimator() {} @@ -45,6 +71,7 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module { // estimate and the over-use detector. If an over-use is detected the // remote bitrate estimate will be updated. Note that |payload_size| is the // packet size excluding headers. + // Note that |arrival_time_ms| can be of an arbitrary time base. virtual void IncomingPacket(int64_t arrival_time_ms, int payload_size, const RTPHeader& header) = 0; @@ -58,6 +85,9 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module { virtual bool LatestEstimate(std::vector<unsigned int>* ssrcs, unsigned int* bitrate_bps) const = 0; + // Returns true if the statistics are available. + virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const = 0; + protected: static const int kProcessIntervalMs = 1000; static const int kStreamTimeOutMs = 2000; @@ -70,6 +100,7 @@ struct RemoteBitrateEstimatorFactory { virtual RemoteBitrateEstimator* Create( RemoteBitrateObserver* observer, Clock* clock, + RateControlType control_type, uint32_t min_bitrate_bps) const; }; @@ -81,6 +112,7 @@ struct AbsoluteSendTimeRemoteBitrateEstimatorFactory virtual RemoteBitrateEstimator* Create( RemoteBitrateObserver* observer, Clock* clock, + RateControlType control_type, uint32_t min_bitrate_bps) const; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h deleted file mode 100644 index 7928abfacbc..00000000000 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h +++ /dev/null @@ -1,43 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_ -#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_ - -#include <list> - -#include "webrtc/typedefs.h" - -namespace webrtc { - -namespace synchronization { - -struct RtcpMeasurement { - RtcpMeasurement(); - RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac, uint32_t timestamp); - uint32_t ntp_secs; - uint32_t ntp_frac; - uint32_t rtp_timestamp; -}; - -typedef std::list<RtcpMeasurement> RtcpList; - -// Converts an RTP timestamp to the NTP domain in milliseconds using two -// (RTP timestamp, NTP timestamp) pairs. -bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp, - int64_t* timestamp_in_ms); - -// Returns 1 there has been a forward wrap around, 0 if there has been no wrap -// around and -1 if there has been a backwards wrap around (i.e. reordering). -int CheckForWrapArounds(uint32_t rtp_timestamp, uint32_t rtcp_rtp_timestamp); -} // namespace synchronization -} // namespace webrtc - -#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_ diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc index 86f6cb8ee3c..9baaa9c9134 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc @@ -10,19 +10,12 @@ #include <math.h> #include <stdlib.h> // fabsf -#if _WIN32 -#include <windows.h> -#endif #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h" #include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/interface/trace.h" -#ifdef WEBRTC_BWE_MATLAB -extern MatlabEngine eng; // global variable defined elsewhere -#endif - enum { kOverUsingTimeThreshold = 100 }; enum { kMinFramePeriodHistoryLength = 60 }; @@ -43,74 +36,20 @@ OveruseDetector::OveruseDetector(const OverUseDetectorOptions& options) prev_offset_(0.0), time_over_using_(-1), over_use_counter_(0), - hypothesis_(kBwNormal), - time_of_last_received_packet_(-1) -#ifdef WEBRTC_BWE_MATLAB - , plots_() -#endif - { + hypothesis_(kBwNormal) { memcpy(E_, options_.initial_e, sizeof(E_)); memcpy(process_noise_, options_.initial_process_noise, sizeof(process_noise_)); } OveruseDetector::~OveruseDetector() { -#ifdef WEBRTC_BWE_MATLAB - if (plots_.plot1_) { - eng.DeletePlot(plots_.plot1_); - plots_.plot1_ = NULL; - } - if (plots_.plot2_) { - eng.DeletePlot(plots_.plot2_); - plots_.plot2_ = NULL; - } - if (plots_.plot3_) { - eng.DeletePlot(plots_.plot3_); - plots_.plot3_ = NULL; - } - if (plots_.plot4_) { - eng.DeletePlot(plots_.plot4_); - plots_.plot4_ = NULL; - } -#endif - ts_delta_hist_.clear(); } void OveruseDetector::Update(uint16_t packet_size, int64_t timestamp_ms, uint32_t timestamp, - const int64_t now_ms) { - time_of_last_received_packet_ = now_ms; -#ifdef WEBRTC_BWE_MATLAB - // Create plots - const int64_t startTimeMs = nowMS; - if (plots_.plot1_ == NULL) { - plots_.plot1_ = eng.NewPlot(new MatlabPlot()); - plots_.plot1_->AddLine(1000, "b.", "scatter"); - } - if (plots_.plot2_ == NULL) { - plots_.plot2_ = eng.NewPlot(new MatlabPlot()); - plots_.plot2_->AddTimeLine(30, "b", "offset", startTimeMs); - plots_.plot2_->AddTimeLine(30, "r--", "limitPos", startTimeMs); - plots_.plot2_->AddTimeLine(30, "k.", "trigger", startTimeMs); - plots_.plot2_->AddTimeLine(30, "ko", "detection", startTimeMs); - // plots_.plot2_->AddTimeLine(30, "g", "slowMean", startTimeMs); - } - if (plots_.plot3_ == NULL) { - plots_.plot3_ = eng.NewPlot(new MatlabPlot()); - plots_.plot3_->AddTimeLine(30, "b", "noiseVar", startTimeMs); - } - if (plots_.plot4_ == NULL) { - plots_.plot4_ = eng.NewPlot(new MatlabPlot()); - // plots_.plot4_->AddTimeLine(60, "b", "p11", startTimeMs); - // plots_.plot4_->AddTimeLine(60, "r", "p12", startTimeMs); - plots_.plot4_->AddTimeLine(60, "g", "p22", startTimeMs); - // plots_.plot4_->AddTimeLine(60, "g--", "p22_hat", startTimeMs); - // plots_.plot4_->AddTimeLine(30, "b.-", "deltaFs", startTimeMs); - } - -#endif + const int64_t arrival_time_ms) { bool new_timestamp = (timestamp != current_frame_.timestamp); if (timestamp_ms >= 0) { if (prev_frame_.timestamp_ms == -1 && current_frame_.timestamp_ms == -1) { @@ -127,8 +66,6 @@ void OveruseDetector::Update(uint16_t packet_size, return; } else if (new_timestamp) { // First packet of a later frame, the previous frame sample is ready. - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "Frame complete at %I64i", - current_frame_.complete_time_ms); if (prev_frame_.complete_time_ms >= 0) { // This is our second frame. int64_t t_delta = 0; double ts_delta = 0; @@ -143,7 +80,7 @@ void OveruseDetector::Update(uint16_t packet_size, } // Accumulate the frame size current_frame_.size += packet_size; - current_frame_.complete_time_ms = now_ms; + current_frame_.complete_time_ms = arrival_time_ms; } BandwidthUsage OveruseDetector::State() const { @@ -168,10 +105,6 @@ void OveruseDetector::SetRateControlRegion(RateControlRegion region) { } } -int64_t OveruseDetector::time_of_last_received_packet() const { - return time_of_last_received_packet_; -} - void OveruseDetector::SwitchTimeBase() { current_frame_.size = 0; current_frame_.complete_time_ms = -1; @@ -249,10 +182,10 @@ void OveruseDetector::UpdateKalman(int64_t t_delta, const double residual = t_ts_delta - slope_*h[0] - offset_; const bool stable_state = - (BWE_MIN(num_of_deltas_, 60) * fabsf(offset_) < threshold_); + (BWE_MIN(num_of_deltas_, 60) * fabs(offset_) < threshold_); // We try to filter out very late frames. For instance periodic key // frames doesn't fit the Gaussian model well. - if (fabsf(residual) < 3 * sqrt(var_noise_)) { + if (fabs(residual) < 3 * sqrt(var_noise_)) { UpdateNoiseEstimate(residual, min_frame_period, stable_state); } else { UpdateNoiseEstimate(3 * sqrt(var_noise_), min_frame_period, stable_state); @@ -279,39 +212,11 @@ void OveruseDetector::UpdateKalman(int64_t t_delta, E_[0][0] * E_[1][1] - E_[0][1] * E_[1][0] >= 0 && E_[0][0] >= 0); -#ifdef WEBRTC_BWE_MATLAB - // plots_.plot4_->Append("p11",E_[0][0]); - // plots_.plot4_->Append("p12",E_[0][1]); - plots_.plot4_->Append("p22", E_[1][1]); - // plots_.plot4_->Append("p22_hat", 0.5*(process_noise_[1] + - // sqrt(process_noise_[1]*(process_noise_[1] + 4*var_noise_)))); - // plots_.plot4_->Append("deltaFs", fsDelta); - plots_.plot4_->Plot(); -#endif slope_ = slope_ + K[0] * residual; prev_offset_ = offset_; offset_ = offset_ + K[1] * residual; Detect(ts_delta); - -#ifdef WEBRTC_BWE_MATLAB - plots_.plot1_->Append("scatter", - static_cast<double>(current_frame_.size) - prev_frame_.size, - static_cast<double>(t_delta - ts_delta)); - plots_.plot1_->MakeTrend("scatter", "slope", slope_, offset_, "k-"); - plots_.plot1_->MakeTrend("scatter", "thresholdPos", - slope_, offset_ + 2 * sqrt(var_noise_), "r-"); - plots_.plot1_->MakeTrend("scatter", "thresholdNeg", - slope_, offset_ - 2 * sqrt(var_noise_), "r-"); - plots_.plot1_->Plot(); - - plots_.plot2_->Append("offset", offset_); - plots_.plot2_->Append("limitPos", threshold_/BWE_MIN(num_of_deltas_, 60)); - plots_.plot2_->Plot(); - - plots_.plot3_->Append("noiseVar", var_noise_); - plots_.plot3_->Plot(); -#endif } double OveruseDetector::UpdateMinFramePeriod(double ts_delta) { @@ -358,7 +263,7 @@ BandwidthUsage OveruseDetector::Detect(double ts_delta) { return kBwNormal; } const double T = BWE_MIN(num_of_deltas_, 60) * offset_; - if (fabsf(T) > threshold_) { + if (fabs(T) > threshold_) { if (offset_ > 0) { if (time_over_using_ == -1) { // Initialize the timer. Assume that we've been @@ -373,38 +278,17 @@ BandwidthUsage OveruseDetector::Detect(double ts_delta) { if (time_over_using_ > kOverUsingTimeThreshold && over_use_counter_ > 1) { if (offset_ >= prev_offset_) { -#ifdef _DEBUG - if (hypothesis_ != kBwOverusing) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwOverusing"); - } -#endif time_over_using_ = 0; over_use_counter_ = 0; hypothesis_ = kBwOverusing; -#ifdef WEBRTC_BWE_MATLAB - plots_.plot2_->Append("detection", offset_); // plot it later -#endif } } -#ifdef WEBRTC_BWE_MATLAB - plots_.plot2_->Append("trigger", offset_); // plot it later -#endif } else { -#ifdef _DEBUG - if (hypothesis_ != kBwUnderusing) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwUnderUsing"); - } -#endif time_over_using_ = -1; over_use_counter_ = 0; hypothesis_ = kBwUnderusing; } } else { -#ifdef _DEBUG - if (hypothesis_ != kBwNormal) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwNormal"); - } -#endif time_over_using_ = -1; over_use_counter_ = 0; hypothesis_ = kBwNormal; diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.h index 57f4ddf04f7..9c565e45f18 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.h +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.h @@ -16,10 +16,6 @@ #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/typedefs.h" -#ifdef WEBRTC_BWE_MATLAB -#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h" -#endif - namespace webrtc { enum RateControlRegion; @@ -32,11 +28,10 @@ class OveruseDetector { void Update(uint16_t packet_size, int64_t timestamp_ms, uint32_t rtp_timestamp, - int64_t now_ms); + int64_t arrival_time_ms); BandwidthUsage State() const; double NoiseVar() const; void SetRateControlRegion(RateControlRegion region); - int64_t time_of_last_received_packet() const; private: struct FrameSample { @@ -52,16 +47,6 @@ class OveruseDetector { int64_t timestamp_ms; }; - struct DebugPlots { -#ifdef WEBRTC_BWE_MATLAB - DebugPlots() : plot1(NULL), plot2(NULL), plot3(NULL), plot4(NULL) {} - MatlabPlot* plot1; - MatlabPlot* plot2; - MatlabPlot* plot3; - MatlabPlot* plot4; -#endif - }; - // Returns true if |timestamp| represent a time which is later than // |prev_timestamp|. static bool InOrderTimestamp(uint32_t timestamp, uint32_t prev_timestamp); @@ -103,10 +88,6 @@ class OveruseDetector { double time_over_using_; uint16_t over_use_counter_; BandwidthUsage hypothesis_; - int64_t time_of_last_received_packet_; -#ifdef WEBRTC_BWE_MATLAB - DebugPlots plots_; -#endif }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.cc index 4a9b4488108..48485ffb551 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.cc @@ -10,6 +10,8 @@ #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h" +#include <assert.h> + namespace webrtc { RateStatistics::RateStatistics(uint32_t window_size_ms, float scale) diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.h index 429669059a2..f97371bd621 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.h +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.h @@ -34,7 +34,7 @@ class RateStatistics { // Counters are kept in buckets (circular buffer), with one bucket // per millisecond. const int num_buckets_; - scoped_array<uint32_t> buckets_; + scoped_ptr<uint32_t[]> buckets_; // Total count recorded in buckets. uint32_t accumulated_count_; diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi index bbd353fcdd4..c2f1b3da475 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi @@ -21,10 +21,66 @@ 'sources': [ 'include/bwe_defines.h', 'include/remote_bitrate_estimator.h', - 'include/rtp_to_ntp.h', 'rate_statistics.cc', 'rate_statistics.h', - 'rtp_to_ntp.cc', + ], # source + }, + { + 'target_name': 'bwe_tools_util', + 'type': 'static_library', + 'dependencies': [ + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + 'rtp_rtcp', + ], + 'sources': [ + 'tools/bwe_rtp.cc', + 'tools/bwe_rtp.h', + ], + }, + { + 'target_name': 'bwe_rtp_to_text', + 'type': 'executable', + 'includes': [ + '../rtp_rtcp/source/rtp_rtcp.gypi', + ], + 'dependencies': [ + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', + 'bwe_tools_util', + 'rtp_rtcp', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + 'include', + ], + }, + 'sources': [ + 'tools/rtp_to_text.cc', + '<(webrtc_root)/modules/video_coding/main/test/rtp_file_reader.cc', + '<(webrtc_root)/modules/video_coding/main/test/rtp_file_reader.h', + ], # source + }, + { + 'target_name': 'bwe_rtp_play', + 'type': 'executable', + 'includes': [ + '../rtp_rtcp/source/rtp_rtcp.gypi', + ], + 'dependencies': [ + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', + 'bwe_tools_util', + 'rtp_rtcp', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + 'include', + ], + }, + 'sources': [ + 'tools/bwe_rtp_play.cc', + '<(webrtc_root)/modules/video_coding/main/test/rtp_file_reader.cc', + '<(webrtc_root)/modules/video_coding/main/test/rtp_file_reader.h', ], # source }, ], # targets diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc index a544ee5d034..08422d28b23 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc @@ -16,8 +16,8 @@ #include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -55,11 +55,31 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator { virtual bool LatestEstimate(std::vector<unsigned int>* ssrcs, unsigned int* bitrate_bps) const OVERRIDE; + virtual bool GetStats( + ReceiveBandwidthEstimatorStats* output) const OVERRIDE; + private: - typedef std::map<unsigned int, OveruseDetector> SsrcOveruseDetectorMap; + // Map from SSRC to over-use detector and last incoming packet time in + // milliseconds, taken from clock_. + typedef std::map<unsigned int, std::pair<OveruseDetector, int64_t> > + SsrcOveruseDetectorMap; + + static OveruseDetector* GetDetector( + const SsrcOveruseDetectorMap::iterator it) { + return &it->second.first; + } + + static int64_t GetPacketTimeMs(const SsrcOveruseDetectorMap::iterator it) { + return it->second.second; + } + + static void SetPacketTimeMs(SsrcOveruseDetectorMap::iterator it, + int64_t time_ms) { + it->second.second = time_ms; + } // Triggers a new estimate calculation. - void UpdateEstimate(int64_t time_now); + void UpdateEstimate(int64_t now_ms); void GetSsrcs(std::vector<unsigned int>* ssrcs) const; @@ -92,6 +112,7 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket( uint32_t ssrc = header.ssrc; uint32_t rtp_timestamp = header.timestamp + header.extension.transmissionTimeOffset; + int64_t now_ms = clock_->TimeInMilliseconds(); CriticalSectionScoped cs(crit_sect_.get()); SsrcOveruseDetectorMap::iterator it = overuse_detectors_.find(ssrc); if (it == overuse_detectors_.end()) { @@ -102,22 +123,23 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket( // automatically cleaned up when we have one RemoteBitrateEstimator per REMB // group. std::pair<SsrcOveruseDetectorMap::iterator, bool> insert_result = - overuse_detectors_.insert(std::make_pair(ssrc, OveruseDetector( - OverUseDetectorOptions()))); + overuse_detectors_.insert(std::make_pair(ssrc, + std::make_pair(OveruseDetector(OverUseDetectorOptions()), now_ms))); it = insert_result.first; } - OveruseDetector* overuse_detector = &it->second; - incoming_bitrate_.Update(payload_size, arrival_time_ms); + SetPacketTimeMs(it, now_ms); + OveruseDetector* overuse_detector = GetDetector(it); + incoming_bitrate_.Update(payload_size, now_ms); const BandwidthUsage prior_state = overuse_detector->State(); overuse_detector->Update(payload_size, -1, rtp_timestamp, arrival_time_ms); if (overuse_detector->State() == kBwOverusing) { - unsigned int incoming_bitrate = incoming_bitrate_.Rate(arrival_time_ms); + unsigned int incoming_bitrate = incoming_bitrate_.Rate(now_ms); if (prior_state != kBwOverusing || - remote_rate_.TimeToReduceFurther(arrival_time_ms, incoming_bitrate)) { + remote_rate_.TimeToReduceFurther(now_ms, incoming_bitrate)) { // The first overuse should immediately trigger a new estimate. // We also have to update the estimate immediately if we are overusing // and the target bitrate is too high compared to what we are receiving. - UpdateEstimate(arrival_time_ms); + UpdateEstimate(now_ms); } } } @@ -126,8 +148,9 @@ int32_t RemoteBitrateEstimatorSingleStream::Process() { if (TimeUntilNextProcess() > 0) { return 0; } - UpdateEstimate(clock_->TimeInMilliseconds()); - last_process_time_ = clock_->TimeInMilliseconds(); + int64_t now_ms = clock_->TimeInMilliseconds(); + UpdateEstimate(now_ms); + last_process_time_ = now_ms; return 0; } @@ -138,25 +161,24 @@ int32_t RemoteBitrateEstimatorSingleStream::TimeUntilNextProcess() { return last_process_time_ + kProcessIntervalMs - clock_->TimeInMilliseconds(); } -void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t time_now) { +void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) { CriticalSectionScoped cs(crit_sect_.get()); BandwidthUsage bw_state = kBwNormal; double sum_noise_var = 0.0; SsrcOveruseDetectorMap::iterator it = overuse_detectors_.begin(); while (it != overuse_detectors_.end()) { - const int64_t time_of_last_received_packet = - it->second.time_of_last_received_packet(); - if (time_of_last_received_packet >= 0 && - time_now - time_of_last_received_packet > kStreamTimeOutMs) { + if (GetPacketTimeMs(it) >= 0 && + now_ms - GetPacketTimeMs(it) > kStreamTimeOutMs) { // This over-use detector hasn't received packets for |kStreamTimeOutMs| // milliseconds and is considered stale. overuse_detectors_.erase(it++); } else { - sum_noise_var += it->second.NoiseVar(); + OveruseDetector* overuse_detector = GetDetector(it); + sum_noise_var += overuse_detector->NoiseVar(); // Make sure that we trigger an over-use if any of the over-use detectors // is detecting over-use. - if (it->second.State() > bw_state) { - bw_state = it->second.State(); + if (overuse_detector->State() > bw_state) { + bw_state = overuse_detector->State(); } ++it; } @@ -169,17 +191,17 @@ void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t time_now) { double mean_noise_var = sum_noise_var / static_cast<double>(overuse_detectors_.size()); const RateControlInput input(bw_state, - incoming_bitrate_.Rate(time_now), + incoming_bitrate_.Rate(now_ms), mean_noise_var); - const RateControlRegion region = remote_rate_.Update(&input, time_now); - unsigned int target_bitrate = remote_rate_.UpdateBandwidthEstimate(time_now); + const RateControlRegion region = remote_rate_.Update(&input, now_ms); + unsigned int target_bitrate = remote_rate_.UpdateBandwidthEstimate(now_ms); if (remote_rate_.ValidEstimate()) { std::vector<unsigned int> ssrcs; GetSsrcs(&ssrcs); observer_->OnReceiveBitrateChanged(ssrcs, target_bitrate); } for (it = overuse_detectors_.begin(); it != overuse_detectors_.end(); ++it) { - it->second.SetRateControlRegion(region); + GetDetector(it)->SetRateControlRegion(region); } } @@ -210,6 +232,12 @@ bool RemoteBitrateEstimatorSingleStream::LatestEstimate( return true; } +bool RemoteBitrateEstimatorSingleStream::GetStats( + ReceiveBandwidthEstimatorStats* output) const { + // Not implemented. + return false; +} + void RemoteBitrateEstimatorSingleStream::GetSsrcs( std::vector<unsigned int>* ssrcs) const { assert(ssrcs); @@ -225,9 +253,9 @@ void RemoteBitrateEstimatorSingleStream::GetSsrcs( RemoteBitrateEstimator* RemoteBitrateEstimatorFactory::Create( RemoteBitrateObserver* observer, Clock* clock, + RateControlType control_type, uint32_t min_bitrate_bps) const { - WEBRTC_TRACE(kTraceStateInfo, kTraceRemoteBitrateEstimator, -1, - "RemoteBitrateEstimatorFactory: Instantiating."); + LOG(LS_INFO) << "RemoteBitrateEstimatorFactory: Instantiating."; return new RemoteBitrateEstimatorSingleStream(observer, clock, min_bitrate_bps); } @@ -235,9 +263,10 @@ RemoteBitrateEstimator* RemoteBitrateEstimatorFactory::Create( RemoteBitrateEstimator* AbsoluteSendTimeRemoteBitrateEstimatorFactory::Create( RemoteBitrateObserver* observer, Clock* clock, + RateControlType control_type, uint32_t min_bitrate_bps) const { - WEBRTC_TRACE(kTraceStateInfo, kTraceRemoteBitrateEstimator, -1, - "AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating."); + LOG(LS_INFO) << "AbsoluteSendTimeRemoteBitrateEstimatorFactory: " + "Instantiating."; return new RemoteBitrateEstimatorSingleStream(observer, clock, min_bitrate_bps); } diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc index a3e44d87ef5..f67c7f34fc0 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc @@ -10,8 +10,8 @@ #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" namespace webrtc { @@ -24,6 +24,7 @@ class RemoteBitrateEstimatorSingleTest : public RemoteBitrateEstimatorTest { bitrate_estimator_.reset(RemoteBitrateEstimatorFactory().Create( bitrate_observer_.get(), &clock_, + kMimdControl, kRemoteBitrateEstimatorMinBitrateBps)); } protected: @@ -35,7 +36,7 @@ TEST_F(RemoteBitrateEstimatorSingleTest, InitialBehavior) { } TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseReordering) { - RateIncreaseReorderingTestHelper(); + RateIncreaseReorderingTestHelper(498136); } TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseRtpTimestamps) { diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc index 88ffe061b99..1b38a1ea306 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc @@ -226,7 +226,8 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc, header.ssrc = ssrc; header.timestamp = rtp_timestamp; header.extension.absoluteSendTime = absolute_send_time; - bitrate_estimator_->IncomingPacket(arrival_time, payload_size, header); + bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs, + payload_size, header); } // Generates a frame of packets belonging to a stream at a given bitrate and @@ -245,6 +246,10 @@ bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(unsigned int ssrc, while (!packets.empty()) { testing::RtpStream::RtpPacket* packet = packets.front(); bitrate_observer_->Reset(); + // The simulated clock should match the time of packet->arrival_time + // since both are used in IncomingPacket(). + clock_.AdvanceTimeMicroseconds(packet->arrival_time - + clock_.TimeInMicroseconds()); IncomingPacket(packet->ssrc, packet->size, (packet->arrival_time + 500) / 1000, @@ -256,8 +261,6 @@ bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(unsigned int ssrc, overuse = true; EXPECT_LE(bitrate_observer_->latest_bitrate(), bitrate_bps); } - clock_.AdvanceTimeMicroseconds(packet->arrival_time - - clock_.TimeInMicroseconds()); delete packet; packets.pop_front(); } @@ -341,9 +344,14 @@ void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper( EXPECT_TRUE(bitrate_observer_->updated()); bitrate_observer_->Reset(); EXPECT_EQ(bitrate_observer_->latest_bitrate(), bitrate_bps); + bitrate_estimator_->RemoveStream(kDefaultSsrc); + EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); + ASSERT_EQ(0u, ssrcs.size()); + EXPECT_EQ(0u, bitrate_bps); } -void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper() { +void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper( + uint32_t expected_bitrate_bps) { const int kFramerate = 50; // 50 fps to avoid rounding errors. const int kFrameIntervalMs = 1000 / kFramerate; const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate); @@ -364,7 +372,7 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper() { } bitrate_estimator_->Process(); EXPECT_TRUE(bitrate_observer_->updated()); - EXPECT_EQ(498136u, bitrate_observer_->latest_bitrate()); + EXPECT_EQ(expected_bitrate_bps, bitrate_observer_->latest_bitrate()); for (int i = 0; i < 10; ++i) { clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs); timestamp += 2 * 90 * kFrameIntervalMs; @@ -379,7 +387,7 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper() { } bitrate_estimator_->Process(); EXPECT_TRUE(bitrate_observer_->updated()); - EXPECT_EQ(498136u, bitrate_observer_->latest_bitrate()); + EXPECT_EQ(expected_bitrate_bps, bitrate_observer_->latest_bitrate()); } // Make sure we initially increase the bitrate as expected. @@ -486,5 +494,21 @@ void RemoteBitrateEstimatorTest::CapacityDropTestHelper( EXPECT_EQ(expected_bitrate_drop_delta, bitrate_drop_time - overuse_start_time); + + // Remove stream one by one. + unsigned int latest_bps = 0; + std::vector<unsigned int> ssrcs; + for (int i = 0; i < number_of_streams; i++) { + EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &latest_bps)); + EXPECT_EQ(number_of_streams - i, static_cast<int>(ssrcs.size())); + EXPECT_EQ(bitrate_bps, latest_bps); + for (int j = i; j < number_of_streams; j++) { + EXPECT_EQ(kDefaultSsrc + j, ssrcs[j - i]); + } + bitrate_estimator_->RemoveStream(kDefaultSsrc + i); + } + EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &latest_bps)); + EXPECT_EQ(0u, ssrcs.size()); + EXPECT_EQ(0u, latest_bps); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h index 64830dab70f..1d748c57b9f 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h @@ -16,9 +16,9 @@ #include <utility> #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { @@ -190,7 +190,7 @@ class RemoteBitrateEstimatorTest : public ::testing::Test { unsigned int target_bitrate); void InitialBehaviorTestHelper(unsigned int expected_converge_bitrate); - void RateIncreaseReorderingTestHelper(); + void RateIncreaseReorderingTestHelper(unsigned int expected_bitrate); void RateIncreaseRtpTimestampsTestHelper(); void CapacityDropTestHelper(int number_of_streams, bool wrap_time_stamp, @@ -198,6 +198,7 @@ class RemoteBitrateEstimatorTest : public ::testing::Test { unsigned int expected_bitrate_drop_delta); static const unsigned int kDefaultSsrc; + static const int kArrivalTimeClockOffsetMs = 60000; SimulatedClock clock_; // Time at the receiver. scoped_ptr<testing::TestBitrateObserver> bitrate_observer_; diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc index ed8e5c555ee..67b60848148 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc @@ -8,90 +8,95 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <sstream> + #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/test/testsupport/perf_test.h" + +using std::string; namespace webrtc { namespace testing { namespace bwe { +enum Estimator { kAbsSendTime, kTransmissionOffset }; -std::vector<const PacketSenderFactory*> VideoSenderFactories(uint32_t count) { - class VideoPacketSenderFactory : public PacketSenderFactory { - public: - VideoPacketSenderFactory(float fps, uint32_t kbps, uint32_t ssrc, - float frame_offset) - : fps_(fps), - kbps_(kbps), - ssrc_(ssrc), - frame_offset_(frame_offset) { - } - virtual ~VideoPacketSenderFactory() {} - virtual PacketSender* Create() const { - return new VideoSender(NULL, fps_, kbps_, ssrc_, frame_offset_); - } - private: - float fps_; - uint32_t kbps_; - uint32_t ssrc_; - float frame_offset_; - }; - - static const VideoPacketSenderFactory factories[] = { - VideoPacketSenderFactory(30.00f, 150, 0x1234, 0.13f), - VideoPacketSenderFactory(15.00f, 500, 0x2345, 0.16f), - VideoPacketSenderFactory(30.00f, 1200, 0x3456, 0.26f), - VideoPacketSenderFactory(7.49f, 150, 0x4567, 0.05f), - VideoPacketSenderFactory(7.50f, 150, 0x5678, 0.15f), - VideoPacketSenderFactory(7.51f, 150, 0x6789, 0.25f), - VideoPacketSenderFactory(15.02f, 150, 0x7890, 0.27f), - VideoPacketSenderFactory(15.03f, 150, 0x8901, 0.38f), - VideoPacketSenderFactory(30.02f, 150, 0x9012, 0.39f), - VideoPacketSenderFactory(30.03f, 150, 0x0123, 0.52f) - }; - assert(count <= sizeof(factories) / sizeof(factories[0])); - - std::vector<const PacketSenderFactory*> result; - for (uint32_t i = 0; i < count; ++i) { - result.push_back(&factories[i]); - } - return result; -} - -std::vector<BweTestConfig::EstimatorConfig> EstimatorConfigs() { +BweTestConfig::EstimatorConfig EstimatorConfigs(Estimator estimator, + int flow_id) { static const RemoteBitrateEstimatorFactory factories[] = { RemoteBitrateEstimatorFactory(), AbsoluteSendTimeRemoteBitrateEstimatorFactory() }; - - std::vector<BweTestConfig::EstimatorConfig> result; - result.push_back(BweTestConfig::EstimatorConfig("TOF", &factories[0])); - result.push_back(BweTestConfig::EstimatorConfig("AST", &factories[1])); - return result; + switch (estimator) { + case kTransmissionOffset: + return BweTestConfig::EstimatorConfig("TOF", flow_id, &factories[0], + kMimdControl, false, false); + case kAbsSendTime: + return BweTestConfig::EstimatorConfig("AST", flow_id, &factories[1], + kMimdControl, false, false); + } + assert(false); + return BweTestConfig::EstimatorConfig(); } -BweTestConfig MakeBweTestConfig(uint32_t sender_count) { - BweTestConfig result = { - VideoSenderFactories(sender_count), EstimatorConfigs() - }; +struct DefaultBweTestConfig { + BweTestConfig bwe_test_config; + size_t number_of_senders; +}; + +DefaultBweTestConfig MakeBweTestConfig(uint32_t sender_count, + Estimator estimator) { + DefaultBweTestConfig result; + result.bwe_test_config.estimator_configs.push_back( + EstimatorConfigs(estimator, 0)); + result.number_of_senders = sender_count; return result; } -INSTANTIATE_TEST_CASE_P(VideoSendersTest, BweTest, - ::testing::Values(MakeBweTestConfig(1), - MakeBweTestConfig(3))); +class DefaultBweTest : public BweTest, + public ::testing::TestWithParam<DefaultBweTestConfig> { + public: + DefaultBweTest() : packet_senders_() {} + virtual ~DefaultBweTest() {} + + virtual void SetUp() { + const DefaultBweTestConfig& config = GetParam(); + SetupTestFromConfig(config.bwe_test_config); + for (size_t i = 0; i < config.number_of_senders; ++i) { + packet_senders_.push_back(new VideoSender(0, this, 30, 300, 0, 0)); + } + } + + virtual void TearDown() { + while (!packet_senders_.empty()) { + delete packet_senders_.front(); + packet_senders_.pop_front(); + } + } + + protected: + std::list<PacketSender*> packet_senders_; +}; + +INSTANTIATE_TEST_CASE_P(VideoSendersTest, DefaultBweTest, + ::testing::Values(MakeBweTestConfig(1, kAbsSendTime), + MakeBweTestConfig(3, kAbsSendTime), + MakeBweTestConfig(1, kTransmissionOffset), + MakeBweTestConfig(3, kTransmissionOffset))); -TEST_P(BweTest, UnlimitedSpeed) { +TEST_P(DefaultBweTest, UnlimitedSpeed) { VerboseLogging(false); RunFor(10 * 60 * 1000); } -TEST_P(BweTest, SteadyLoss) { +TEST_P(DefaultBweTest, DISABLED_SteadyLoss) { LossFilter loss(this); loss.SetLoss(20.0); RunFor(10 * 60 * 1000); } -TEST_P(BweTest, IncreasingLoss1) { +TEST_P(DefaultBweTest, IncreasingLoss1) { LossFilter loss(this); for (int i = 0; i < 76; ++i) { loss.SetLoss(i); @@ -99,13 +104,13 @@ TEST_P(BweTest, IncreasingLoss1) { } } -TEST_P(BweTest, SteadyDelay) { +TEST_P(DefaultBweTest, SteadyDelay) { DelayFilter delay(this); delay.SetDelay(1000); RunFor(10 * 60 * 1000); } -TEST_P(BweTest, IncreasingDelay1) { +TEST_P(DefaultBweTest, DISABLED_IncreasingDelay1) { DelayFilter delay(this); RunFor(10 * 60 * 1000); for (int i = 0; i < 30 * 2; ++i) { @@ -115,7 +120,7 @@ TEST_P(BweTest, IncreasingDelay1) { RunFor(10 * 60 * 1000); } -TEST_P(BweTest, IncreasingDelay2) { +TEST_P(DefaultBweTest, IncreasingDelay2) { DelayFilter delay(this); RateCounterFilter counter(this); RunFor(1 * 60 * 1000); @@ -127,7 +132,7 @@ TEST_P(BweTest, IncreasingDelay2) { RunFor(10 * 60 * 1000); } -TEST_P(BweTest, JumpyDelay1) { +TEST_P(DefaultBweTest, JumpyDelay1) { DelayFilter delay(this); RunFor(10 * 60 * 1000); for (int i = 1; i < 200; ++i) { @@ -140,14 +145,14 @@ TEST_P(BweTest, JumpyDelay1) { RunFor(10 * 60 * 1000); } -TEST_P(BweTest, SteadyJitter) { +TEST_P(DefaultBweTest, SteadyJitter) { JitterFilter jitter(this); RateCounterFilter counter(this); jitter.SetJitter(20); RunFor(2 * 60 * 1000); } -TEST_P(BweTest, IncreasingJitter1) { +TEST_P(DefaultBweTest, IncreasingJitter1) { JitterFilter jitter(this); for (int i = 0; i < 2 * 60 * 2; ++i) { jitter.SetJitter(i); @@ -156,7 +161,7 @@ TEST_P(BweTest, IncreasingJitter1) { RunFor(10 * 60 * 1000); } -TEST_P(BweTest, IncreasingJitter2) { +TEST_P(DefaultBweTest, IncreasingJitter2) { JitterFilter jitter(this); RunFor(30 * 1000); for (int i = 1; i < 51; ++i) { @@ -167,13 +172,13 @@ TEST_P(BweTest, IncreasingJitter2) { RunFor(10 * 60 * 1000); } -TEST_P(BweTest, SteadyReorder) { +TEST_P(DefaultBweTest, SteadyReorder) { ReorderFilter reorder(this); reorder.SetReorder(20.0); RunFor(10 * 60 * 1000); } -TEST_P(BweTest, IncreasingReorder1) { +TEST_P(DefaultBweTest, IncreasingReorder1) { ReorderFilter reorder(this); for (int i = 0; i < 76; ++i) { reorder.SetReorder(i); @@ -181,13 +186,13 @@ TEST_P(BweTest, IncreasingReorder1) { } } -TEST_P(BweTest, SteadyChoke) { +TEST_P(DefaultBweTest, DISABLED_SteadyChoke) { ChokeFilter choke(this); choke.SetCapacity(140); RunFor(10 * 60 * 1000); } -TEST_P(BweTest, IncreasingChoke1) { +TEST_P(DefaultBweTest, DISABLED_IncreasingChoke1) { ChokeFilter choke(this); for (int i = 1200; i >= 100; i -= 100) { choke.SetCapacity(i); @@ -195,7 +200,7 @@ TEST_P(BweTest, IncreasingChoke1) { } } -TEST_P(BweTest, IncreasingChoke2) { +TEST_P(DefaultBweTest, DISABLED_IncreasingChoke2) { ChokeFilter choke(this); RunFor(60 * 1000); for (int i = 1200; i >= 100; i -= 20) { @@ -204,7 +209,7 @@ TEST_P(BweTest, IncreasingChoke2) { } } -TEST_P(BweTest, Multi1) { +TEST_P(DefaultBweTest, DISABLED_Multi1) { DelayFilter delay(this); ChokeFilter choke(this); RateCounterFilter counter(this); @@ -219,7 +224,7 @@ TEST_P(BweTest, Multi1) { RunFor(5 * 60 * 1000); } -TEST_P(BweTest, Multi2) { +TEST_P(DefaultBweTest, Multi2) { ChokeFilter choke(this); JitterFilter jitter(this); RateCounterFilter counter(this); @@ -227,6 +232,108 @@ TEST_P(BweTest, Multi2) { jitter.SetJitter(120); RunFor(5 * 60 * 1000); } + +// This test fixture is used to instantiate tests running with adaptive video +// senders. +class BweFeedbackTest : public BweTest, + public ::testing::TestWithParam<BweTestConfig> { + public: + BweFeedbackTest() : BweTest() {} + virtual ~BweFeedbackTest() {} + + virtual void SetUp() { + BweTestConfig config; + config.estimator_configs.push_back(EstimatorConfigs(kAbsSendTime, 0)); + SetupTestFromConfig(config); + } + + void PrintResults(double max_throughput_kbps, Stats<double> throughput_kbps, + Stats<double> delay_ms) { + double utilization = throughput_kbps.GetMean() / max_throughput_kbps; + webrtc::test::PrintResult("BwePerformance", + GetTestName(), + "Utilization", + utilization * 100.0, + "%", + false); + std::stringstream ss; + ss << throughput_kbps.GetStdDev() / throughput_kbps.GetMean(); + webrtc::test::PrintResult("BwePerformance", + GetTestName(), + "Utilization var coeff", + ss.str(), + "", + false); + webrtc::test::PrintResult("BwePerformance", + GetTestName(), + "Average delay", + delay_ms.AsString(), + "ms", + false); + } + + private: + DISALLOW_COPY_AND_ASSIGN(BweFeedbackTest); +}; + +TEST_F(BweFeedbackTest, Choke1000kbps500kbps1000kbps) { + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + ChokeFilter filter(this); + RateCounterFilter counter(this, "receiver_input"); + const int kHighCapacityKbps = 1000; + const int kLowCapacityKbps = 500; + filter.SetCapacity(kHighCapacityKbps); + filter.SetMaxDelay(500); + RunFor(60 * 1000); + filter.SetCapacity(kLowCapacityKbps); + RunFor(60 * 1000); + filter.SetCapacity(kHighCapacityKbps); + RunFor(60 * 1000); + PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0, + counter.GetBitrateStats(), filter.GetDelayStats()); +} + +TEST_F(BweFeedbackTest, Choke200kbps30kbps200kbps) { + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + ChokeFilter filter(this); + RateCounterFilter counter(this, "receiver_input"); + const int kHighCapacityKbps = 200; + const int kLowCapacityKbps = 30; + filter.SetCapacity(kHighCapacityKbps); + filter.SetMaxDelay(500); + RunFor(60 * 1000); + filter.SetCapacity(kLowCapacityKbps); + RunFor(60 * 1000); + filter.SetCapacity(kHighCapacityKbps); + RunFor(60 * 1000); + + PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0, + counter.GetBitrateStats(), filter.GetDelayStats()); +} + +TEST_F(BweFeedbackTest, Verizon4gDownlinkTest) { + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + RateCounterFilter counter1(this, "sender_output"); + TraceBasedDeliveryFilter filter(this, "link_capacity"); + RateCounterFilter counter2(this, "receiver_input"); + ASSERT_TRUE(filter.Init(test::ResourcePath("verizon4g-downlink", "rx"))); + RunFor(22 * 60 * 1000); + PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(), + filter.GetDelayStats()); +} + +// webrtc:3277 +TEST_F(BweFeedbackTest, DISABLED_GoogleWifiTrace3Mbps) { + AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); + RateCounterFilter counter1(this, "sender_output"); + TraceBasedDeliveryFilter filter(this, "link_capacity"); + filter.SetMaxDelay(500); + RateCounterFilter counter2(this, "receiver_input"); + ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx"))); + RunFor(300 * 1000); + PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(), + filter.GetDelayStats()); +} } // namespace bwe } // namespace testing } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc index 994abdbee89..dda36a765e9 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc @@ -122,8 +122,6 @@ RateControlRegion RemoteRateControl::Update(const RateControlInput* input, } updated_ = true; current_input_ = *input; - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Incoming rate = %u kbps", - input->_incomingBitRate/1000); return rate_control_region_; } @@ -158,18 +156,11 @@ uint32_t RemoteRateControl::ChangeBitRate(uint32_t current_bit_rate, ChangeRegion(kRcAboveMax); } } - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "BWE: Response time: %f + %i + 10*33\n", - avg_change_period_, rtt_); const uint32_t response_time = static_cast<uint32_t>(avg_change_period_ + 0.5f) + rtt_ + 300; double alpha = RateIncreaseFactor(now_ms, last_bit_rate_change_, response_time, noise_var); - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "BWE: avg_change_period_ = %f ms; RTT = %u ms", avg_change_period_, - rtt_); - current_bit_rate = static_cast<uint32_t>(current_bit_rate * alpha) + 1000; if (max_hold_rate_ > 0 && beta_ * max_hold_rate_ > current_bit_rate) { current_bit_rate = static_cast<uint32_t>(beta_ * max_hold_rate_); @@ -178,9 +169,6 @@ uint32_t RemoteRateControl::ChangeBitRate(uint32_t current_bit_rate, recovery = true; } max_hold_rate_ = 0; - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "BWE: Increase rate to current_bit_rate = %u kbps", - current_bit_rate / 1000); last_bit_rate_change_ = now_ms; break; } @@ -207,10 +195,6 @@ uint32_t RemoteRateControl::ChangeBitRate(uint32_t current_bit_rate, } UpdateMaxBitRateEstimate(incoming_bit_rate_kbps); - - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "BWE: Decrease rate to current_bit_rate = %u kbps", - current_bit_rate / 1000); } // Stay on hold until the pipes are cleared. ChangeState(kRcHold); @@ -251,8 +235,6 @@ double RemoteRateControl::RateIncreaseFactor(int64_t now_ms, alpha = 1.3; } - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: alpha = %f", alpha); - if (last_ms > -1) { alpha = pow(alpha, (now_ms - last_ms) / 1000.0); } @@ -341,45 +323,5 @@ void RemoteRateControl::ChangeRegion(RateControlRegion region) { void RemoteRateControl::ChangeState(RateControlState new_state) { came_from_state_ = rate_control_state_; rate_control_state_ = new_state; - char state1[15]; - char state2[15]; - char state3[15]; - StateStr(came_from_state_, state1); - StateStr(rate_control_state_, state2); - StateStr(current_input_._bwState, state3); - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "\t%s => %s due to %s\n", state1, state2, state3); -} - -void RemoteRateControl::StateStr(RateControlState state, char* str) { - switch (state) { - case kRcDecrease: - strncpy(str, "DECREASE", 9); - break; - case kRcHold: - strncpy(str, "HOLD", 5); - break; - case kRcIncrease: - strncpy(str, "INCREASE", 9); - break; - default: - assert(false); - } -} - -void RemoteRateControl::StateStr(BandwidthUsage state, char* str) { - switch (state) { - case kBwNormal: - strncpy(str, "NORMAL", 7); - break; - case kBwOverusing: - strncpy(str, "OVER USING", 11); - break; - case kBwUnderusing: - strncpy(str, "UNDER USING", 12); - break; - default: - assert(false); - } } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h index b525834eca1..d02c6d56b50 100644 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h @@ -53,8 +53,6 @@ class RemoteRateControl { void ChangeState(const RateControlInput& input, int64_t now_ms); void ChangeState(RateControlState new_state); void ChangeRegion(RateControlRegion region); - static void StateStr(RateControlState state, char* str); - static void StateStr(BandwidthUsage state, char* str); uint32_t min_configured_bit_rate_; uint32_t max_configured_bit_rate_; diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc deleted file mode 100644 index 109edae7cc5..00000000000 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc +++ /dev/null @@ -1,119 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h" - -#include "webrtc/system_wrappers/interface/clock.h" - -#include <assert.h> - -namespace webrtc { - -namespace synchronization { - -RtcpMeasurement::RtcpMeasurement() - : ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {} - -RtcpMeasurement::RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac, - uint32_t timestamp) - : ntp_secs(ntp_secs), ntp_frac(ntp_frac), rtp_timestamp(timestamp) {} - -// Calculates the RTP timestamp frequency from two pairs of NTP and RTP -// timestamps. -bool CalculateFrequency( - int64_t rtcp_ntp_ms1, - uint32_t rtp_timestamp1, - int64_t rtcp_ntp_ms2, - uint32_t rtp_timestamp2, - double* frequency_khz) { - if (rtcp_ntp_ms1 <= rtcp_ntp_ms2) { - return false; - } - *frequency_khz = static_cast<double>(rtp_timestamp1 - rtp_timestamp2) / - static_cast<double>(rtcp_ntp_ms1 - rtcp_ntp_ms2); - return true; -} - -// Detects if there has been a wraparound between |old_timestamp| and -// |new_timestamp|, and compensates by adding 2^32 if that is the case. -bool CompensateForWrapAround(uint32_t new_timestamp, - uint32_t old_timestamp, - int64_t* compensated_timestamp) { - assert(compensated_timestamp); - int64_t wraps = synchronization::CheckForWrapArounds(new_timestamp, - old_timestamp); - if (wraps < 0) { - // Reordering, don't use this packet. - return false; - } - *compensated_timestamp = new_timestamp + (wraps << 32); - return true; -} - -// Converts |rtp_timestamp| to the NTP time base using the NTP and RTP timestamp -// pairs in |rtcp|. The converted timestamp is returned in -// |rtp_timestamp_in_ms|. This function compensates for wrap arounds in RTP -// timestamps and returns false if it can't do the conversion due to reordering. -bool RtpToNtpMs(int64_t rtp_timestamp, - const synchronization::RtcpList& rtcp, - int64_t* rtp_timestamp_in_ms) { - assert(rtcp.size() == 2); - int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs, - rtcp.front().ntp_frac); - int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs, - rtcp.back().ntp_frac); - int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp; - int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp; - if (!CompensateForWrapAround(rtcp_timestamp_new, - rtcp_timestamp_old, - &rtcp_timestamp_new)) { - return false; - } - double freq_khz; - if (!CalculateFrequency(rtcp_ntp_ms_new, - rtcp_timestamp_new, - rtcp_ntp_ms_old, - rtcp_timestamp_old, - &freq_khz)) { - return false; - } - double offset = rtcp_timestamp_new - freq_khz * rtcp_ntp_ms_new; - int64_t rtp_timestamp_unwrapped; - if (!CompensateForWrapAround(rtp_timestamp, rtcp_timestamp_old, - &rtp_timestamp_unwrapped)) { - return false; - } - double rtp_timestamp_ntp_ms = (static_cast<double>(rtp_timestamp_unwrapped) - - offset) / freq_khz + 0.5f; - if (rtp_timestamp_ntp_ms < 0) { - return false; - } - *rtp_timestamp_in_ms = rtp_timestamp_ntp_ms; - return true; -} - -int CheckForWrapArounds(uint32_t new_timestamp, uint32_t old_timestamp) { - if (new_timestamp < old_timestamp) { - // This difference should be less than -2^31 if we have had a wrap around - // (e.g. |new_timestamp| = 1, |rtcp_rtp_timestamp| = 2^32 - 1). Since it is - // cast to a int32_t, it should be positive. - if (static_cast<int32_t>(new_timestamp - old_timestamp) > 0) { - // Forward wrap around. - return 1; - } - } else if (static_cast<int32_t>(old_timestamp - new_timestamp) > 0) { - // This difference should be less than -2^31 if we have had a backward wrap - // around. Since it is cast to a int32_t, it should be positive. - return -1; - } - return 0; -} -} // namespace synchronization -} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc deleted file mode 100644 index aff314aaa53..00000000000 --- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc +++ /dev/null @@ -1,163 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h" - -namespace webrtc { - -TEST(WrapAroundTests, NoWrap) { - EXPECT_EQ(0, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0xFFFFFFFE)); - EXPECT_EQ(0, synchronization::CheckForWrapArounds(1, 0)); - EXPECT_EQ(0, synchronization::CheckForWrapArounds(0x00010000, 0x0000FFFF)); -} - -TEST(WrapAroundTests, ForwardWrap) { - EXPECT_EQ(1, synchronization::CheckForWrapArounds(0, 0xFFFFFFFF)); - EXPECT_EQ(1, synchronization::CheckForWrapArounds(0, 0xFFFF0000)); - EXPECT_EQ(1, synchronization::CheckForWrapArounds(0x0000FFFF, 0xFFFFFFFF)); - EXPECT_EQ(1, synchronization::CheckForWrapArounds(0x0000FFFF, 0xFFFF0000)); -} - -TEST(WrapAroundTests, BackwardWrap) { - EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0)); - EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFF0000, 0)); - EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0x0000FFFF)); - EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFF0000, 0x0000FFFF)); -} - -TEST(WrapAroundTests, OldRtcpWrapped) { - synchronization::RtcpList rtcp; - uint32_t ntp_sec = 0; - uint32_t ntp_frac = 0; - uint32_t timestamp = 0; - const uint32_t kOneMsInNtpFrac = 4294967; - const uint32_t kTimestampTicksPerMs = 90; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp -= kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp -= kTimestampTicksPerMs; - int64_t timestamp_in_ms = -1; - // This expected to fail since it's highly unlikely that the older RTCP - // has a much smaller RTP timestamp than the newer. - EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms)); -} - -TEST(WrapAroundTests, NewRtcpWrapped) { - synchronization::RtcpList rtcp; - uint32_t ntp_sec = 0; - uint32_t ntp_frac = 0; - uint32_t timestamp = 0xFFFFFFFF; - const uint32_t kOneMsInNtpFrac = 4294967; - const uint32_t kTimestampTicksPerMs = 90; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp += kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - int64_t timestamp_in_ms = -1; - EXPECT_TRUE(synchronization::RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp, - ×tamp_in_ms)); - // Since this RTP packet has the same timestamp as the RTCP packet constructed - // at time 0 it should be mapped to 0 as well. - EXPECT_EQ(0, timestamp_in_ms); -} - -TEST(WrapAroundTests, RtpWrapped) { - const uint32_t kOneMsInNtpFrac = 4294967; - const uint32_t kTimestampTicksPerMs = 90; - synchronization::RtcpList rtcp; - uint32_t ntp_sec = 0; - uint32_t ntp_frac = 0; - uint32_t timestamp = 0xFFFFFFFF - 2 * kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp += kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp += kTimestampTicksPerMs; - int64_t timestamp_in_ms = -1; - EXPECT_TRUE(synchronization::RtpToNtpMs(timestamp, rtcp, - ×tamp_in_ms)); - // Since this RTP packet has the same timestamp as the RTCP packet constructed - // at time 0 it should be mapped to 0 as well. - EXPECT_EQ(2, timestamp_in_ms); -} - -TEST(WrapAroundTests, OldRtp_RtcpsWrapped) { - const uint32_t kOneMsInNtpFrac = 4294967; - const uint32_t kTimestampTicksPerMs = 90; - synchronization::RtcpList rtcp; - uint32_t ntp_sec = 0; - uint32_t ntp_frac = 0; - uint32_t timestamp = 0; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp += kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp -= 2*kTimestampTicksPerMs; - int64_t timestamp_in_ms = -1; - EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp, - ×tamp_in_ms)); -} - -TEST(WrapAroundTests, OldRtp_NewRtcpWrapped) { - const uint32_t kOneMsInNtpFrac = 4294967; - const uint32_t kTimestampTicksPerMs = 90; - synchronization::RtcpList rtcp; - uint32_t ntp_sec = 0; - uint32_t ntp_frac = 0; - uint32_t timestamp = 0xFFFFFFFF; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp += kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp -= kTimestampTicksPerMs; - int64_t timestamp_in_ms = -1; - EXPECT_TRUE(synchronization::RtpToNtpMs(timestamp, rtcp, - ×tamp_in_ms)); - // Constructed at the same time as the first RTCP and should therefore be - // mapped to zero. - EXPECT_EQ(0, timestamp_in_ms); -} - -TEST(WrapAroundTests, OldRtp_OldRtcpWrapped) { - const uint32_t kOneMsInNtpFrac = 4294967; - const uint32_t kTimestampTicksPerMs = 90; - synchronization::RtcpList rtcp; - uint32_t ntp_sec = 0; - uint32_t ntp_frac = 0; - uint32_t timestamp = 0; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp -= kTimestampTicksPerMs; - rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac, - timestamp)); - ntp_frac += kOneMsInNtpFrac; - timestamp += 2*kTimestampTicksPerMs; - int64_t timestamp_in_ms = -1; - EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp, - ×tamp_in_ms)); -} -}; // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc new file mode 100644 index 00000000000..40fa6df8ffb --- /dev/null +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc @@ -0,0 +1,74 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" + +#include <stdio.h> +#include <string> + +#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/video_coding/main/test/rtp_file_reader.h" +#include "webrtc/modules/video_coding/main/test/rtp_player.h" + +using webrtc::rtpplayer::RtpPacketSourceInterface; + +const int kMinBitrateBps = 30000; + +bool ParseArgsAndSetupEstimator(int argc, + char** argv, + webrtc::Clock* clock, + webrtc::RemoteBitrateObserver* observer, + RtpPacketSourceInterface** rtp_reader, + webrtc::RtpHeaderParser** parser, + webrtc::RemoteBitrateEstimator** estimator, + std::string* estimator_used) { + *rtp_reader = webrtc::rtpplayer::CreateRtpFileReader(argv[3]); + if (!*rtp_reader) { + fprintf(stderr, "Cannot open input file %s\n", argv[3]); + return false; + } + fprintf(stderr, "Input file: %s\n\n", argv[3]); + webrtc::RTPExtensionType extension = webrtc::kRtpExtensionAbsoluteSendTime; + + if (strncmp("tsoffset", argv[1], 8) == 0) { + extension = webrtc::kRtpExtensionTransmissionTimeOffset; + fprintf(stderr, "Extension: toffset\n"); + } else { + fprintf(stderr, "Extension: abs\n"); + } + int id = atoi(argv[2]); + + // Setup the RTP header parser and the bitrate estimator. + *parser = webrtc::RtpHeaderParser::Create(); + (*parser)->RegisterRtpHeaderExtension(extension, id); + if (estimator) { + switch (extension) { + case webrtc::kRtpExtensionAbsoluteSendTime: { + webrtc::AbsoluteSendTimeRemoteBitrateEstimatorFactory factory; + *estimator = factory.Create(observer, clock, webrtc::kAimdControl, + kMinBitrateBps); + *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator"; + break; + } + case webrtc::kRtpExtensionTransmissionTimeOffset: { + webrtc::RemoteBitrateEstimatorFactory factory; + *estimator = factory.Create(observer, clock, webrtc::kAimdControl, + kMinBitrateBps); + *estimator_used = "RemoteBitrateEstimator"; + break; + } + default: + assert(false); + } + } + return true; +} diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h new file mode 100644 index 00000000000..714457d5668 --- /dev/null +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_ +#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_ + +#include <string> + +namespace webrtc { +class Clock; +class RemoteBitrateEstimator; +class RemoteBitrateObserver; +class RtpHeaderParser; +namespace rtpplayer { +class RtpPacketSourceInterface; +} +} + +bool ParseArgsAndSetupEstimator( + int argc, + char** argv, + webrtc::Clock* clock, + webrtc::RemoteBitrateObserver* observer, + webrtc::rtpplayer::RtpPacketSourceInterface** rtp_reader, + webrtc::RtpHeaderParser** parser, + webrtc::RemoteBitrateEstimator** estimator, + std::string* estimator_used); + +#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_ diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc new file mode 100644 index 00000000000..9ea3f08eab5 --- /dev/null +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc @@ -0,0 +1,116 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <stdio.h> + +#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" +#include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/video_coding/main/test/rtp_file_reader.h" +#include "webrtc/modules/video_coding/main/test/rtp_player.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +using webrtc::rtpplayer::RtpPacketSourceInterface; + +class Observer : public webrtc::RemoteBitrateObserver { + public: + explicit Observer(webrtc::Clock* clock) : clock_(clock) {} + + // Called when a receive channel group has a new bitrate estimate for the + // incoming streams. + virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, + unsigned int bitrate) { + printf("[%u] Num SSRCs: %d, bitrate: %u\n", + static_cast<uint32_t>(clock_->TimeInMilliseconds()), + static_cast<int>(ssrcs.size()), bitrate); + } + + virtual ~Observer() {} + + private: + webrtc::Clock* clock_; +}; + +int main(int argc, char** argv) { + if (argc < 4) { + printf("Usage: bwe_rtp_play <extension type> <extension id> " + "<input_file.rtp>\n"); + printf("<extension type> can either be:\n" + " abs for absolute send time or\n" + " tsoffset for timestamp offset.\n" + "<extension id> is the id associated with the extension.\n"); + return -1; + } + RtpPacketSourceInterface* reader; + webrtc::RemoteBitrateEstimator* estimator; + webrtc::RtpHeaderParser* parser; + std::string estimator_used; + webrtc::SimulatedClock clock(0); + Observer observer(&clock); + if (!ParseArgsAndSetupEstimator(argc, argv, &clock, &observer, &reader, + &parser, &estimator, &estimator_used)) { + return -1; + } + webrtc::scoped_ptr<RtpPacketSourceInterface> rtp_reader(reader); + webrtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser); + webrtc::scoped_ptr<webrtc::RemoteBitrateEstimator> rbe(estimator); + + // Process the file. + int packet_counter = 0; + int64_t next_process_time_ms = 0; + int64_t next_rtp_time_ms = 0; + int64_t first_rtp_time_ms = -1; + const uint32_t kMaxPacketSize = 1500; + uint8_t packet_buffer[kMaxPacketSize]; + uint8_t* packet = packet_buffer; + int non_zero_abs_send_time = 0; + int non_zero_ts_offsets = 0; + while (true) { + uint32_t next_rtp_time; + if (next_rtp_time_ms <= clock.TimeInMilliseconds()) { + uint32_t packet_length = kMaxPacketSize; + if (rtp_reader->NextPacket(packet, &packet_length, + &next_rtp_time) == -1) { + break; + } + if (first_rtp_time_ms == -1) + first_rtp_time_ms = next_rtp_time; + next_rtp_time_ms = next_rtp_time - first_rtp_time_ms; + webrtc::RTPHeader header; + parser->Parse(packet, packet_length, &header); + if (header.extension.absoluteSendTime != 0) + ++non_zero_abs_send_time; + if (header.extension.transmissionTimeOffset != 0) + ++non_zero_ts_offsets; + rbe->IncomingPacket(clock.TimeInMilliseconds(), + packet_length - header.headerLength, + header); + ++packet_counter; + } + next_process_time_ms = rbe->TimeUntilNextProcess() + + clock.TimeInMilliseconds(); + if (next_process_time_ms <= clock.TimeInMilliseconds()) { + rbe->Process(); + } + int time_until_next_event = + std::min(next_process_time_ms, next_rtp_time_ms) - + clock.TimeInMilliseconds(); + clock.AdvanceTimeMilliseconds(std::max(time_until_next_event, 0)); + } + printf("Parsed %d packets\nTime passed: %u ms\n", packet_counter, + static_cast<uint32_t>(clock.TimeInMilliseconds())); + printf("Estimator used: %s\n", estimator_used.c_str()); + printf("Packets with non-zero absolute send time: %d\n", + non_zero_abs_send_time); + printf("Packets with non-zero timestamp offset: %d\n", + non_zero_ts_offsets); + return 0; +} diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc new file mode 100644 index 00000000000..af4a4d4ee7b --- /dev/null +++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc @@ -0,0 +1,80 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <stdio.h> +#include <sstream> + +#include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/video_coding/main/test/rtp_file_reader.h" +#include "webrtc/modules/video_coding/main/test/rtp_player.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +using webrtc::rtpplayer::RtpPacketSourceInterface; + +int main(int argc, char** argv) { + if (argc < 4) { + fprintf(stderr, "Usage: rtp_to_text <extension type> <extension id>" + " <input_file.rtp> [-t]\n"); + fprintf(stderr, "<extension type> can either be:\n" + " abs for absolute send time or\n" + " tsoffset for timestamp offset.\n" + "<extension id> is the id associated with the extension.\n" + " -t is an optional flag, if set only packet arrival time will be" + " output.\n"); + return -1; + } + RtpPacketSourceInterface* reader; + webrtc::RtpHeaderParser* parser; + if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser, + NULL, NULL)) { + return -1; + } + bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0); + webrtc::scoped_ptr<RtpPacketSourceInterface> rtp_reader(reader); + webrtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser); + fprintf(stdout, "seqnum timestamp ts_offset abs_sendtime recvtime " + "markerbit ssrc size\n"); + int packet_counter = 0; + static const uint32_t kMaxPacketSize = 1500; + uint8_t packet_buffer[kMaxPacketSize]; + uint8_t* packet = packet_buffer; + uint32_t packet_length = kMaxPacketSize; + uint32_t time_ms = 0; + int non_zero_abs_send_time = 0; + int non_zero_ts_offsets = 0; + while (rtp_reader->NextPacket(packet, &packet_length, &time_ms) == 0) { + webrtc::RTPHeader header; + parser->Parse(packet, packet_length, &header); + if (header.extension.absoluteSendTime != 0) + ++non_zero_abs_send_time; + if (header.extension.transmissionTimeOffset != 0) + ++non_zero_ts_offsets; + if (arrival_time_only) { + std::stringstream ss; + ss << static_cast<int64_t>(time_ms) * 1000000; + fprintf(stdout, "%s\n", ss.str().c_str()); + } else { + fprintf(stdout, "%u %u %d %u %u %d %u %u\n", header.sequenceNumber, + header.timestamp, header.extension.transmissionTimeOffset, + header.extension.absoluteSendTime, time_ms, header.markerBit, + header.ssrc, packet_length); + } + packet_length = kMaxPacketSize; + ++packet_counter; + } + fprintf(stderr, "Parsed %d packets\n", packet_counter); + fprintf(stderr, "Packets with non-zero absolute send time: %d\n", + non_zero_abs_send_time); + fprintf(stderr, "Packets with non-zero timestamp offset: %d\n", + non_zero_ts_offsets); + return 0; +} diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/fec_receiver.h index 97b200f0777..e2ef4b1e972 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/fec_receiver.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/fec_receiver.h @@ -18,7 +18,7 @@ namespace webrtc { class FecReceiver { public: - static FecReceiver* Create(int32_t id, RtpData* callback); + static FecReceiver* Create(RtpData* callback); virtual ~FecReceiver() {} diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/receive_statistics.h index 707adaa0cd3..6f2ea4fb3e6 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/receive_statistics.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/receive_statistics.h @@ -23,24 +23,9 @@ class Clock; class StreamStatistician { public: - struct Statistics { - Statistics() - : fraction_lost(0), - cumulative_lost(0), - extended_max_sequence_number(0), - jitter(0), - max_jitter(0) {} - - uint8_t fraction_lost; - uint32_t cumulative_lost; - uint32_t extended_max_sequence_number; - uint32_t jitter; - uint32_t max_jitter; - }; - virtual ~StreamStatistician(); - virtual bool GetStatistics(Statistics* statistics, bool reset) = 0; + virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0; virtual void GetDataCounters(uint32_t* bytes_received, uint32_t* packets_received) const = 0; virtual uint32_t BitrateReceived() const = 0; @@ -66,9 +51,13 @@ class ReceiveStatistics : public Module { static ReceiveStatistics* Create(Clock* clock); // Updates the receive statistics with this packet. - virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes, + virtual void IncomingPacket(const RTPHeader& rtp_header, + size_t bytes, bool retransmitted) = 0; + // Increment counter for number of FEC packets received. + virtual void FecPacketReceived(uint32_t ssrc) = 0; + // Returns a map of all statisticians which have seen an incoming packet // during the last two seconds. virtual StatisticianMap GetActiveStatisticians() const = 0; @@ -78,17 +67,31 @@ class ReceiveStatistics : public Module { // Sets the max reordering threshold in number of packets. virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0; + + // Called on new RTCP stats creation. + virtual void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) = 0; + + // Called on new RTP stats creation. + virtual void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) = 0; }; class NullReceiveStatistics : public ReceiveStatistics { public: - virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes, + virtual void IncomingPacket(const RTPHeader& rtp_header, + size_t bytes, bool retransmitted) OVERRIDE; + virtual void FecPacketReceived(uint32_t ssrc) OVERRIDE; virtual StatisticianMap GetActiveStatisticians() const OVERRIDE; virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE; virtual int32_t TimeUntilNextProcess() OVERRIDE; virtual int32_t Process() OVERRIDE; virtual void SetMaxReorderingThreshold(int max_reordering_threshold) OVERRIDE; + virtual void RegisterRtcpStatisticsCallback(RtcpStatisticsCallback* callback) + OVERRIDE; + virtual void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) OVERRIDE; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h new file mode 100644 index 00000000000..25f0f2ecf98 --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h @@ -0,0 +1,50 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_ + +#include "webrtc/system_wrappers/interface/rtp_to_ntp.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +class Clock; +class RtpRtcp; +class TimestampExtrapolator; + +// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP +// time in local timebase. +// Note that it needs to be trained with at least 2 RTCP SR (by calling +// |UpdateRtcpTimestamp|) before it can be used. +class RemoteNtpTimeEstimator { + public: + explicit RemoteNtpTimeEstimator(Clock* clock); + + ~RemoteNtpTimeEstimator(); + + // Updates the estimator with the timestamp from newly received RTCP SR for + // |ssrc|. The RTCP SR is read from |rtp_rtcp|. + bool UpdateRtcpTimestamp(uint32_t ssrc, RtpRtcp* rtp_rtcp); + + // Estimates the NTP timestamp in local timebase from |rtp_timestamp|. + // Returns the NTP timestamp in ms when success. -1 if failed. + int64_t Estimate(uint32_t rtp_timestamp); + + private: + Clock* clock_; + scoped_ptr<TimestampExtrapolator> ts_extrapolator_; + RtcpList rtcp_list_; + DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_ diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h index 3ea4dcd1beb..965f4b02421 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h @@ -54,8 +54,7 @@ class RTPPayloadStrategy { class RTPPayloadRegistry { public: // The registry takes ownership of the strategy. - RTPPayloadRegistry(const int32_t id, - RTPPayloadStrategy* rtp_payload_strategy); + RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy); ~RTPPayloadRegistry(); int32_t RegisterReceivePayload( @@ -76,10 +75,10 @@ class RTPPayloadRegistry { const uint32_t rate, int8_t* payload_type) const; - void SetRtxStatus(bool enable, uint32_t ssrc); - bool RtxEnabled() const; + void SetRtxSsrc(uint32_t ssrc); + void SetRtxPayloadType(int payload_type); bool IsRtx(const RTPHeader& header) const; @@ -153,7 +152,6 @@ class RTPPayloadRegistry { scoped_ptr<CriticalSectionWrapper> crit_sect_; ModuleRTPUtility::PayloadTypeMap payload_type_map_; - int32_t id_; scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_; int8_t red_payload_type_; int8_t ulpfec_payload_type_; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h index 67dad0d977b..95c565f01bf 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h @@ -213,7 +213,7 @@ class RtpRtcp : public Module { * * return -1 on failure else 0 */ - virtual int32_t SetSSRC(const uint32_t ssrc) = 0; + virtual void SetSSRC(const uint32_t ssrc) = 0; /* * Get CSRC @@ -249,10 +249,14 @@ class RtpRtcp : public Module { virtual int32_t SetCSRCStatus(const bool include) = 0; /* - * Turn on/off sending RTX (RFC 4588) on a specific SSRC. + * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination + * of values of the enumerator RtxMode. */ - virtual int32_t SetRTXSendStatus(int modes, bool set_ssrc, - uint32_t ssrc) = 0; + virtual void SetRTXSendStatus(int modes) = 0; + + // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, + // only the SSRC is set. + virtual void SetRtxSsrc(uint32_t ssrc) = 0; // Sets the payload type to use when sending RTX packets. Note that this // doesn't enable RTX, only the payload type is set. @@ -261,8 +265,8 @@ class RtpRtcp : public Module { /* * Get status of sending RTX (RFC 4588) on a specific SSRC. */ - virtual int32_t RTXSendStatus(int* modes, uint32_t* ssrc, - int* payloadType) const = 0; + virtual void RTXSendStatus(int* modes, uint32_t* ssrc, + int* payloadType) const = 0; /* * sends kRtcpByeCode when going from true to false @@ -508,6 +512,13 @@ class RtpRtcp : public Module { virtual int32_t RemoveRTCPReportBlock(const uint32_t SSRC) = 0; /* + * Get number of sent and received RTCP packet types. + */ + virtual void GetRtcpPacketTypeCounters( + RtcpPacketTypeCounter* packets_sent, + RtcpPacketTypeCounter* packets_received) const = 0; + + /* * (APP) Application specific data * * return -1 on failure else 0 @@ -664,25 +675,6 @@ class RtpRtcp : public Module { int8_t& payloadType) const = 0; /* - * Set status and ID for header-extension-for-audio-level-indication. - * See http://tools.ietf.org/html/rfc6464 for more details. - * - * return -1 on failure else 0 - */ - virtual int32_t SetRTPAudioLevelIndicationStatus( - const bool enable, - const uint8_t ID) = 0; - - /* - * Get status and ID for header-extension-for-audio-level-indication. - * - * return -1 on failure else 0 - */ - virtual int32_t GetRTPAudioLevelIndicationStatus( - bool& enable, - uint8_t& ID) const = 0; - - /* * Store the audio level in dBov for header-extension-for-audio-level- * indication. * This API shall be called before transmision of an RTP packet to ensure diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h index b66e927ae59..6f99f938de6 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h @@ -27,6 +27,9 @@ namespace webrtc { const int kVideoPayloadTypeFrequency = 90000; +// Minimum RTP header size in bytes. +const uint8_t kRtpHeaderSize = 12; + struct AudioPayload { uint32_t frequency; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h index 42c7b4eab91..03156c79df4 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h @@ -77,17 +77,19 @@ class MockRtpRtcp : public RtpRtcp { MOCK_CONST_METHOD0(SSRC, uint32_t()); MOCK_METHOD1(SetSSRC, - int32_t(const uint32_t ssrc)); + void(const uint32_t ssrc)); MOCK_CONST_METHOD1(CSRCs, int32_t(uint32_t arrOfCSRC[kRtpCsrcSize])); MOCK_METHOD2(SetCSRCs, int32_t(const uint32_t arrOfCSRC[kRtpCsrcSize], const uint8_t arrLength)); MOCK_METHOD1(SetCSRCStatus, int32_t(const bool include)); - MOCK_METHOD3(SetRTXSendStatus, - int32_t(int modes, bool setSSRC, uint32_t ssrc)); + MOCK_METHOD1(SetRTXSendStatus, + void(int modes)); MOCK_CONST_METHOD3(RTXSendStatus, - int32_t(int* modes, uint32_t* ssrc, int* payload_type)); + void(int* modes, uint32_t* ssrc, int* payload_type)); + MOCK_METHOD1(SetRtxSsrc, + void(uint32_t)); MOCK_METHOD1(SetRtxSendPayloadType, void(int)); MOCK_METHOD1(SetSendingStatus, @@ -168,6 +170,8 @@ class MockRtpRtcp : public RtpRtcp { int32_t(const uint32_t SSRC, const RTCPReportBlock* receiveBlock)); MOCK_METHOD1(RemoveRTCPReportBlock, int32_t(const uint32_t SSRC)); + MOCK_CONST_METHOD2(GetRtcpPacketTypeCounters, + void(RtcpPacketTypeCounter*, RtcpPacketTypeCounter*)); MOCK_METHOD4(SetRTCPApplicationSpecificData, int32_t(const uint8_t subType, const uint32_t name, const uint8_t* data, const uint16_t length)); MOCK_METHOD1(SetRTCPVoIPMetrics, diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/OWNERS b/chromium/third_party/webrtc/modules/rtp_rtcp/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc index 0dc142f867c..0d6c174a3b9 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc @@ -16,20 +16,19 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" // RFC 5109 namespace webrtc { -FecReceiver* FecReceiver::Create(int32_t id, RtpData* callback) { - return new FecReceiverImpl(id, callback); +FecReceiver* FecReceiver::Create(RtpData* callback) { + return new FecReceiverImpl(callback); } -FecReceiverImpl::FecReceiverImpl(const int32_t id, RtpData* callback) - : id_(id), - crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), +FecReceiverImpl::FecReceiverImpl(RtpData* callback) + : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), recovered_packet_callback_(callback), - fec_(new ForwardErrorCorrection(id)) {} + fec_(new ForwardErrorCorrection()) {} FecReceiverImpl::~FecReceiverImpl() { while (!received_packet_list_.empty()) { @@ -103,8 +102,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket( if (timestamp_offset != 0) { // |timestampOffset| should be 0. However, it's possible this is the first // location a corrupt payload can be caught, so don't assert. - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "Corrupt payload found in %s", __FUNCTION__); + LOG(LS_WARNING) << "Corrupt payload found."; delete received_packet; return -1; } @@ -223,6 +221,7 @@ int32_t FecReceiverImpl::ProcessReceivedFec() { crit_sect_->Enter(); } if (fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_) != 0) { + crit_sect_->Leave(); return -1; } assert(received_packet_list_.empty()); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h index 03421235c3d..b876bedc9c3 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h @@ -25,7 +25,7 @@ class CriticalSectionWrapper; class FecReceiverImpl : public FecReceiver { public: - FecReceiverImpl(const int32_t id, RtpData* callback); + FecReceiverImpl(RtpData* callback); virtual ~FecReceiverImpl(); virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header, @@ -36,7 +36,6 @@ class FecReceiverImpl : public FecReceiver { virtual int32_t ProcessReceivedFec() OVERRIDE; private: - int id_; scoped_ptr<CriticalSectionWrapper> crit_sect_; RtpData* recovered_packet_callback_; ForwardErrorCorrection* fec_; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc index 2e8846c3b29..0b1244941c3 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc @@ -39,8 +39,8 @@ class MockRtpData : public RtpData { class ReceiverFecTest : public ::testing::Test { protected: virtual void SetUp() { - fec_ = new ForwardErrorCorrection(0); - receiver_fec_ = FecReceiver::Create(0, &rtp_data_callback_); + fec_ = new ForwardErrorCorrection(); + receiver_fec_ = FecReceiver::Create(&rtp_data_callback_); generator_ = new FrameGenerator(); } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_test_helper.h index e3c3581be73..e6426ea7eea 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_test_helper.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_test_helper.h @@ -17,9 +17,6 @@ namespace webrtc { enum { - kRtpHeaderSize = 12 -}; -enum { kFecPayloadType = 96 }; enum { diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc index 189e1b052f9..31303c8ad19 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc @@ -17,15 +17,13 @@ #include <algorithm> #include <iterator> +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { -// Minimum RTP header size in bytes. -const uint8_t kRtpHeaderSize = 12; - // FEC header size in bytes. const uint8_t kFecHeaderSize = 10; @@ -84,9 +82,8 @@ ForwardErrorCorrection::ReceivedPacket::~ReceivedPacket() {} ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() {} ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() {} -ForwardErrorCorrection::ForwardErrorCorrection(int32_t id) - : id_(id), - generated_fec_packets_(kMaxMediaPackets), +ForwardErrorCorrection::ForwardErrorCorrection() + : generated_fec_packets_(kMaxMediaPackets), fec_packet_received_(false) {} ForwardErrorCorrection::~ForwardErrorCorrection() {} @@ -114,43 +111,23 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list, bool use_unequal_protection, FecMaskType fec_mask_type, PacketList* fec_packet_list) { - if (media_packet_list.empty()) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s media packet list is empty", __FUNCTION__); - return -1; - } - if (!fec_packet_list->empty()) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s FEC packet list is not empty", __FUNCTION__); - return -1; - } const uint16_t num_media_packets = media_packet_list.size(); - bool l_bit = (num_media_packets > 8 * kMaskSizeLBitClear); - int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear; + + // Sanity check arguments. + assert(num_media_packets > 0); + assert(num_important_packets >= 0 && + num_important_packets <= num_media_packets); + assert(fec_packet_list->empty()); if (num_media_packets > kMaxMediaPackets) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s can only protect %d media packets per frame; %d requested", - __FUNCTION__, kMaxMediaPackets, num_media_packets); + LOG(LS_WARNING) << "Can't protect " << num_media_packets + << " media packets per frame. Max is " << kMaxMediaPackets; return -1; } - // Error checking on the number of important packets. - // Can't have more important packets than media packets. - if (num_important_packets > num_media_packets) { - WEBRTC_TRACE( - kTraceError, kTraceRtpRtcp, id_, - "Number of important packets (%d) greater than number of media " - "packets (%d)", - num_important_packets, num_media_packets); - return -1; - } - if (num_important_packets < 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "Number of important packets (%d) less than zero", - num_important_packets); - return -1; - } + bool l_bit = (num_media_packets > 8 * kMaskSizeLBitClear); + int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear; + // Do some error checking on the media packets. PacketList::const_iterator media_list_it = media_packet_list.begin(); while (media_list_it != media_packet_list.end()) { @@ -158,20 +135,16 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list, assert(media_packet); if (media_packet->length < kRtpHeaderSize) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s media packet (%d bytes) is smaller than RTP header", - __FUNCTION__, media_packet->length); + LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes " + << "is smaller than RTP header."; return -1; } // Ensure our FEC packets will fit in a typical MTU. if (media_packet->length + PacketOverhead() + kTransportOverhead > IP_PACKET_SIZE) { - WEBRTC_TRACE( - kTraceError, kTraceRtpRtcp, id_, - "%s media packet (%d bytes) with overhead is larger than MTU(%d)", - __FUNCTION__, media_packet->length, IP_PACKET_SIZE); - return -1; + LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes " + << "with overhead is larger than " << IP_PACKET_SIZE; } media_list_it++; } @@ -584,9 +557,7 @@ void ForwardErrorCorrection::InsertFECPacket( } if (fec_packet->protected_pkt_list.empty()) { // All-zero packet mask; we can discard this FEC packet. - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "FEC packet %u has an all-zero packet mask.", - fec_packet->seq_num, __FUNCTION__); + LOG(LS_WARNING) << "FEC packet has an all-zero packet mask."; delete fec_packet; } else { AssignRecoveredPackets(fec_packet, recovered_packet_list); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.h index 8910fe477d3..bb790f356f2 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.h @@ -117,8 +117,7 @@ class ForwardErrorCorrection { typedef std::list<ReceivedPacket*> ReceivedPacketList; typedef std::list<RecoveredPacket*> RecoveredPacketList; - // \param[in] id Module ID - ForwardErrorCorrection(int32_t id); + ForwardErrorCorrection(); virtual ~ForwardErrorCorrection(); @@ -304,7 +303,6 @@ class ForwardErrorCorrection { static void DiscardOldPackets(RecoveredPacketList* recovered_packet_list); static uint16_t ParseSequenceNumber(uint8_t* packet); - int32_t id_; std::vector<Packet> generated_fec_packets_; FecPacketList fec_packet_list_; bool fec_packet_received_; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h index ddac3cd113a..f82e46d5723 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h @@ -8,6 +8,9 @@ * be found in the AUTHORS file in the root of the source tree. */ +#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_ +#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_ + #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/typedefs.h" @@ -64,3 +67,4 @@ void GeneratePacketMasks(int num_media_packets, int num_fec_packets, } // namespace internal } // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_ diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc index 8c6cc5434cd..209af40bc43 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -164,7 +164,7 @@ class RtxLoopBackTransport : public webrtc::Transport { class RtpRtcpRtxNackTest : public ::testing::Test { protected: RtpRtcpRtxNackTest() - : rtp_payload_registry_(0, RTPPayloadStrategy::CreateStrategy(false)), + : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)), rtp_rtcp_module_(NULL), transport_(kTestSsrc + 1), receiver_(), @@ -188,7 +188,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test { kTestId, &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_)); - EXPECT_EQ(0, rtp_rtcp_module_->SetSSRC(kTestSsrc)); + rtp_rtcp_module_->SetSSRC(kTestSsrc); EXPECT_EQ(0, rtp_rtcp_module_->SetRTCPStatus(kRtcpCompound)); rtp_receiver_->SetNACKStatus(kNackRtcp); EXPECT_EQ(0, rtp_rtcp_module_->SetStorePacketsStatus(true, 600)); @@ -253,9 +253,9 @@ class RtpRtcpRtxNackTest : public ::testing::Test { } void RunRtxTest(RtxMode rtx_method, int loss) { - rtp_payload_registry_.SetRtxStatus(true, kTestSsrc + 1); - EXPECT_EQ(0, rtp_rtcp_module_->SetRTXSendStatus(rtx_method, true, - kTestSsrc + 1)); + rtp_payload_registry_.SetRtxSsrc(kTestSsrc + 1); + rtp_rtcp_module_->SetRTXSendStatus(rtx_method); + rtp_rtcp_module_->SetRtxSsrc(kTestSsrc + 1); transport_.DropEveryNthPacket(loss); uint32_t timestamp = 3000; uint16_t nack_list[kVideoNackListSize]; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc index ada7d70266b..baa3827949c 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc @@ -39,7 +39,7 @@ void VerifyHeader(uint16_t seq_num, class ProducerFecTest : public ::testing::Test { protected: virtual void SetUp() { - fec_ = new ForwardErrorCorrection(0); + fec_ = new ForwardErrorCorrection(); producer_ = new ProducerFec(fec_); generator_ = new FrameGenerator; } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc index 3ed44b83701..aa7c9c57156 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -24,14 +24,16 @@ const int kStatisticsProcessIntervalMs = 1000; StreamStatistician::~StreamStatistician() {} -StreamStatisticianImpl::StreamStatisticianImpl(Clock* clock) +StreamStatisticianImpl::StreamStatisticianImpl( + Clock* clock, + RtcpStatisticsCallback* rtcp_callback, + StreamDataCountersCallback* rtp_callback) : clock_(clock), - crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + stream_lock_(CriticalSectionWrapper::CreateCriticalSection()), incoming_bitrate_(clock, NULL), ssrc_(0), max_reordering_threshold_(kDefaultMaxReorderingThreshold), jitter_q4_(0), - jitter_max_q4_(0), cumulative_loss_(0), jitter_q4_transmission_time_offset_(0), last_receive_time_ms_(0), @@ -42,53 +44,53 @@ StreamStatisticianImpl::StreamStatisticianImpl(Clock* clock) received_seq_first_(0), received_seq_max_(0), received_seq_wraps_(0), - first_packet_(true), received_packet_overhead_(12), - received_byte_count_(0), - received_retransmitted_packets_(0), - received_inorder_packet_count_(0), last_report_inorder_packets_(0), last_report_old_packets_(0), last_report_seq_max_(0), - last_reported_statistics_() {} + rtcp_callback_(rtcp_callback), + rtp_callback_(rtp_callback) {} void StreamStatisticianImpl::ResetStatistics() { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); last_report_inorder_packets_ = 0; last_report_old_packets_ = 0; last_report_seq_max_ = 0; - memset(&last_reported_statistics_, 0, sizeof(last_reported_statistics_)); + last_reported_statistics_ = RtcpStatistics(); jitter_q4_ = 0; - jitter_max_q4_ = 0; cumulative_loss_ = 0; jitter_q4_transmission_time_offset_ = 0; received_seq_wraps_ = 0; received_seq_max_ = 0; received_seq_first_ = 0; - received_byte_count_ = 0; - received_retransmitted_packets_ = 0; - received_inorder_packet_count_ = 0; - first_packet_ = true; + receive_counters_ = StreamDataCounters(); } void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header, size_t bytes, bool retransmitted) { - CriticalSectionScoped cs(crit_sect_.get()); + UpdateCounters(header, bytes, retransmitted); + NotifyRtpCallback(); +} + +void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header, + size_t bytes, + bool retransmitted) { + CriticalSectionScoped cs(stream_lock_.get()); bool in_order = InOrderPacketInternal(header.sequenceNumber); ssrc_ = header.ssrc; incoming_bitrate_.Update(bytes); - received_byte_count_ += bytes; + receive_counters_.bytes += + bytes - (header.paddingLength + header.headerLength); + receive_counters_.header_bytes += header.headerLength; + receive_counters_.padding_bytes += header.paddingLength; + ++receive_counters_.packets; + if (!in_order && retransmitted) { + ++receive_counters_.retransmitted_packets; + } - if (first_packet_) { - first_packet_ = false; - // This is the first received report. + if (receive_counters_.packets == 1) { received_seq_first_ = header.sequenceNumber; - received_seq_max_ = header.sequenceNumber; - received_inorder_packet_count_ = 1; - clock_->CurrentNtp(last_receive_time_secs_, last_receive_time_frac_); - last_receive_time_ms_ = clock_->TimeInMilliseconds(); - return; } // Count only the new packets received. That is, if packets 1, 2, 3, 5, 4, 6 @@ -98,66 +100,27 @@ void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header, uint32_t receive_time_secs; uint32_t receive_time_frac; clock_->CurrentNtp(receive_time_secs, receive_time_frac); - received_inorder_packet_count_++; // Wrong if we use RetransmitOfOldPacket. - int32_t seq_diff = header.sequenceNumber - received_seq_max_; - if (seq_diff < 0) { + if (receive_counters_.packets > 1 && + received_seq_max_ > header.sequenceNumber) { // Wrap around detected. received_seq_wraps_++; } // New max. received_seq_max_ = header.sequenceNumber; + // If new time stamp and more than one in-order packet received, calculate + // new jitter statistics. if (header.timestamp != last_received_timestamp_ && - received_inorder_packet_count_ > 1) { - uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( - receive_time_secs, receive_time_frac, header.payload_type_frequency); - uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( - last_receive_time_secs_, last_receive_time_frac_, - header.payload_type_frequency); - int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) - - (header.timestamp - last_received_timestamp_); - - time_diff_samples = abs(time_diff_samples); - - // lib_jingle sometimes deliver crazy jumps in TS for the same stream. - // If this happens, don't update jitter value. Use 5 secs video frequency - // as the threshold. - if (time_diff_samples < 450000) { - // Note we calculate in Q4 to avoid using float. - int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_; - jitter_q4_ += ((jitter_diff_q4 + 8) >> 4); - } - - // Extended jitter report, RFC 5450. - // Actual network jitter, excluding the source-introduced jitter. - int32_t time_diff_samples_ext = - (receive_time_rtp - last_receive_time_rtp) - - ((header.timestamp + - header.extension.transmissionTimeOffset) - - (last_received_timestamp_ + - last_received_transmission_time_offset_)); - - time_diff_samples_ext = abs(time_diff_samples_ext); - - if (time_diff_samples_ext < 450000) { - int32_t jitter_diffQ4TransmissionTimeOffset = - (time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_; - jitter_q4_transmission_time_offset_ += - ((jitter_diffQ4TransmissionTimeOffset + 8) >> 4); - } + (receive_counters_.packets - receive_counters_.retransmitted_packets) > + 1) { + UpdateJitter(header, receive_time_secs, receive_time_frac); } last_received_timestamp_ = header.timestamp; last_receive_time_secs_ = receive_time_secs; last_receive_time_frac_ = receive_time_frac; last_receive_time_ms_ = clock_->TimeInMilliseconds(); - } else { - if (retransmitted) { - received_retransmitted_packets_++; - } else { - received_inorder_packet_count_++; - } } uint16_t packet_oh = header.headerLength + header.paddingLength; @@ -167,29 +130,113 @@ void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header, received_packet_overhead_ = (15 * received_packet_overhead_ + packet_oh) >> 4; } +void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header, + uint32_t receive_time_secs, + uint32_t receive_time_frac) { + uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( + receive_time_secs, receive_time_frac, header.payload_type_frequency); + uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( + last_receive_time_secs_, last_receive_time_frac_, + header.payload_type_frequency); + int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) - + (header.timestamp - last_received_timestamp_); + + time_diff_samples = abs(time_diff_samples); + + // lib_jingle sometimes deliver crazy jumps in TS for the same stream. + // If this happens, don't update jitter value. Use 5 secs video frequency + // as the threshold. + if (time_diff_samples < 450000) { + // Note we calculate in Q4 to avoid using float. + int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_; + jitter_q4_ += ((jitter_diff_q4 + 8) >> 4); + } + + // Extended jitter report, RFC 5450. + // Actual network jitter, excluding the source-introduced jitter. + int32_t time_diff_samples_ext = + (receive_time_rtp - last_receive_time_rtp) - + ((header.timestamp + + header.extension.transmissionTimeOffset) - + (last_received_timestamp_ + + last_received_transmission_time_offset_)); + + time_diff_samples_ext = abs(time_diff_samples_ext); + + if (time_diff_samples_ext < 450000) { + int32_t jitter_diffQ4TransmissionTimeOffset = + (time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_; + jitter_q4_transmission_time_offset_ += + ((jitter_diffQ4TransmissionTimeOffset + 8) >> 4); + } +} + +void StreamStatisticianImpl::NotifyRtpCallback() { + StreamDataCounters data; + uint32_t ssrc; + { + CriticalSectionScoped cs(stream_lock_.get()); + data = receive_counters_; + ssrc = ssrc_; + } + rtp_callback_->DataCountersUpdated(data, ssrc); +} + +void StreamStatisticianImpl::NotifyRtcpCallback() { + RtcpStatistics data; + uint32_t ssrc; + { + CriticalSectionScoped cs(stream_lock_.get()); + data = last_reported_statistics_; + ssrc = ssrc_; + } + rtcp_callback_->StatisticsUpdated(data, ssrc); +} + +void StreamStatisticianImpl::FecPacketReceived() { + { + CriticalSectionScoped cs(stream_lock_.get()); + ++receive_counters_.fec_packets; + } + NotifyRtpCallback(); +} + void StreamStatisticianImpl::SetMaxReorderingThreshold( int max_reordering_threshold) { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); max_reordering_threshold_ = max_reordering_threshold; } -bool StreamStatisticianImpl::GetStatistics(Statistics* statistics, bool reset) { - CriticalSectionScoped cs(crit_sect_.get()); - if (received_seq_first_ == 0 && received_byte_count_ == 0) { - // We have not received anything. - return false; - } - - if (!reset) { - if (last_report_inorder_packets_ == 0) { - // No report. +bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics, + bool reset) { + { + CriticalSectionScoped cs(stream_lock_.get()); + if (received_seq_first_ == 0 && receive_counters_.bytes == 0) { + // We have not received anything. return false; } - // Just get last report. - *statistics = last_reported_statistics_; - return true; + + if (!reset) { + if (last_report_inorder_packets_ == 0) { + // No report. + return false; + } + // Just get last report. + *statistics = last_reported_statistics_; + return true; + } + + *statistics = CalculateRtcpStatistics(); } + NotifyRtcpCallback(); + + return true; +} + +RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() { + RtcpStatistics stats; + if (last_report_inorder_packets_ == 0) { // First time we send a report. last_report_seq_max_ = received_seq_first_ - 1; @@ -206,7 +253,8 @@ bool StreamStatisticianImpl::GetStatistics(Statistics* statistics, bool reset) { // Number of received RTP packets since last report, counts all packets but // not re-transmissions. uint32_t rec_since_last = - received_inorder_packet_count_ - last_report_inorder_packets_; + (receive_counters_.packets - receive_counters_.retransmitted_packets) - + last_report_inorder_packets_; // With NACK we don't know the expected retransmissions during the last // second. We know how many "old" packets we have received. We just count @@ -218,7 +266,7 @@ bool StreamStatisticianImpl::GetStatistics(Statistics* statistics, bool reset) { // re-transmitted. We use RTT to decide if a packet is re-ordered or // re-transmitted. uint32_t retransmitted_packets = - received_retransmitted_packets_ - last_report_old_packets_; + receive_counters_.retransmitted_packets - last_report_old_packets_; rec_since_last += retransmitted_packets; int32_t missing = 0; @@ -231,64 +279,60 @@ bool StreamStatisticianImpl::GetStatistics(Statistics* statistics, bool reset) { local_fraction_lost = static_cast<uint8_t>(255 * missing / exp_since_last); } - statistics->fraction_lost = local_fraction_lost; + stats.fraction_lost = local_fraction_lost; // We need a counter for cumulative loss too. cumulative_loss_ += missing; - - if (jitter_q4_ > jitter_max_q4_) { - jitter_max_q4_ = jitter_q4_; - } - statistics->cumulative_lost = cumulative_loss_; - statistics->extended_max_sequence_number = (received_seq_wraps_ << 16) + - received_seq_max_; + stats.cumulative_lost = cumulative_loss_; + stats.extended_max_sequence_number = + (received_seq_wraps_ << 16) + received_seq_max_; // Note: internal jitter value is in Q4 and needs to be scaled by 1/16. - statistics->jitter = jitter_q4_ >> 4; - statistics->max_jitter = jitter_max_q4_ >> 4; - if (reset) { - // Store this report. - last_reported_statistics_ = *statistics; - - // Only for report blocks in RTCP SR and RR. - last_report_inorder_packets_ = received_inorder_packet_count_; - last_report_old_packets_ = received_retransmitted_packets_; - last_report_seq_max_ = received_seq_max_; - } - return true; + stats.jitter = jitter_q4_ >> 4; + + // Store this report. + last_reported_statistics_ = stats; + + // Only for report blocks in RTCP SR and RR. + last_report_inorder_packets_ = + receive_counters_.packets - receive_counters_.retransmitted_packets; + last_report_old_packets_ = receive_counters_.retransmitted_packets; + last_report_seq_max_ = received_seq_max_; + + return stats; } void StreamStatisticianImpl::GetDataCounters( uint32_t* bytes_received, uint32_t* packets_received) const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); if (bytes_received) { - *bytes_received = received_byte_count_; + *bytes_received = receive_counters_.bytes + receive_counters_.header_bytes + + receive_counters_.padding_bytes; } if (packets_received) { - *packets_received = - received_retransmitted_packets_ + received_inorder_packet_count_; + *packets_received = receive_counters_.packets; } } uint32_t StreamStatisticianImpl::BitrateReceived() const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); return incoming_bitrate_.BitrateNow(); } void StreamStatisticianImpl::ProcessBitrate() { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); incoming_bitrate_.Process(); } void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs, uint32_t* frac) const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); *secs = last_receive_time_secs_; *frac = last_receive_time_frac_; } bool StreamStatisticianImpl::IsRetransmitOfOldPacket( const RTPHeader& header, int min_rtt) const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); if (InOrderPacketInternal(header.sequenceNumber)) { return false; } @@ -323,7 +367,7 @@ bool StreamStatisticianImpl::IsRetransmitOfOldPacket( } bool StreamStatisticianImpl::IsPacketInOrder(uint16_t sequence_number) const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(stream_lock_.get()); return InOrderPacketInternal(sequence_number); } @@ -348,8 +392,10 @@ ReceiveStatistics* ReceiveStatistics::Create(Clock* clock) { ReceiveStatisticsImpl::ReceiveStatisticsImpl(Clock* clock) : clock_(clock), - crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - last_rate_update_ms_(0) {} + receive_statistics_lock_(CriticalSectionWrapper::CreateCriticalSection()), + last_rate_update_ms_(0), + rtcp_stats_callback_(NULL), + rtp_stats_callback_(NULL) {} ReceiveStatisticsImpl::~ReceiveStatisticsImpl() { while (!statisticians_.empty()) { @@ -359,20 +405,31 @@ ReceiveStatisticsImpl::~ReceiveStatisticsImpl() { } void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header, - size_t bytes, bool old_packet) { - CriticalSectionScoped cs(crit_sect_.get()); - StatisticianImplMap::iterator it = statisticians_.find(header.ssrc); - if (it == statisticians_.end()) { - std::pair<StatisticianImplMap::iterator, uint32_t> insert_result = - statisticians_.insert(std::make_pair( - header.ssrc, new StreamStatisticianImpl(clock_))); - it = insert_result.first; + size_t bytes, + bool retransmitted) { + StatisticianImplMap::iterator it; + { + CriticalSectionScoped cs(receive_statistics_lock_.get()); + it = statisticians_.find(header.ssrc); + if (it == statisticians_.end()) { + std::pair<StatisticianImplMap::iterator, uint32_t> insert_result = + statisticians_.insert(std::make_pair( + header.ssrc, new StreamStatisticianImpl(clock_, this, this))); + it = insert_result.first; + } } - statisticians_[header.ssrc]->IncomingPacket(header, bytes, old_packet); + it->second->IncomingPacket(header, bytes, retransmitted); +} + +void ReceiveStatisticsImpl::FecPacketReceived(uint32_t ssrc) { + CriticalSectionScoped cs(receive_statistics_lock_.get()); + StatisticianImplMap::iterator it = statisticians_.find(ssrc); + assert(it != statisticians_.end()); + it->second->FecPacketReceived(); } void ReceiveStatisticsImpl::ChangeSsrc(uint32_t from_ssrc, uint32_t to_ssrc) { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(receive_statistics_lock_.get()); StatisticianImplMap::iterator from_it = statisticians_.find(from_ssrc); if (from_it == statisticians_.end()) return; @@ -383,7 +440,7 @@ void ReceiveStatisticsImpl::ChangeSsrc(uint32_t from_ssrc, uint32_t to_ssrc) { } StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(receive_statistics_lock_.get()); StatisticianMap active_statisticians; for (StatisticianImplMap::const_iterator it = statisticians_.begin(); it != statisticians_.end(); ++it) { @@ -400,7 +457,7 @@ StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const { StreamStatistician* ReceiveStatisticsImpl::GetStatistician( uint32_t ssrc) const { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(receive_statistics_lock_.get()); StatisticianImplMap::const_iterator it = statisticians_.find(ssrc); if (it == statisticians_.end()) return NULL; @@ -409,7 +466,7 @@ StreamStatistician* ReceiveStatisticsImpl::GetStatistician( void ReceiveStatisticsImpl::SetMaxReorderingThreshold( int max_reordering_threshold) { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(receive_statistics_lock_.get()); for (StatisticianImplMap::iterator it = statisticians_.begin(); it != statisticians_.end(); ++it) { it->second->SetMaxReorderingThreshold(max_reordering_threshold); @@ -417,7 +474,7 @@ void ReceiveStatisticsImpl::SetMaxReorderingThreshold( } int32_t ReceiveStatisticsImpl::Process() { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(receive_statistics_lock_.get()); for (StatisticianImplMap::iterator it = statisticians_.begin(); it != statisticians_.end(); ++it) { it->second->ProcessBitrate(); @@ -427,17 +484,50 @@ int32_t ReceiveStatisticsImpl::Process() { } int32_t ReceiveStatisticsImpl::TimeUntilNextProcess() { - CriticalSectionScoped cs(crit_sect_.get()); + CriticalSectionScoped cs(receive_statistics_lock_.get()); int time_since_last_update = clock_->TimeInMilliseconds() - last_rate_update_ms_; return std::max(kStatisticsProcessIntervalMs - time_since_last_update, 0); } +void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) { + CriticalSectionScoped cs(receive_statistics_lock_.get()); + if (callback != NULL) + assert(rtcp_stats_callback_ == NULL); + rtcp_stats_callback_ = callback; +} + +void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics, + uint32_t ssrc) { + CriticalSectionScoped cs(receive_statistics_lock_.get()); + if (rtcp_stats_callback_) { + rtcp_stats_callback_->StatisticsUpdated(statistics, ssrc); + } +} + +void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) { + CriticalSectionScoped cs(receive_statistics_lock_.get()); + if (callback != NULL) + assert(rtp_stats_callback_ == NULL); + rtp_stats_callback_ = callback; +} + +void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats, + uint32_t ssrc) { + CriticalSectionScoped cs(receive_statistics_lock_.get()); + if (rtp_stats_callback_) { + rtp_stats_callback_->DataCountersUpdated(stats, ssrc); + } +} void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header, size_t bytes, bool retransmitted) {} +void NullReceiveStatistics::FecPacketReceived(uint32_t ssrc) {} + StatisticianMap NullReceiveStatistics::GetActiveStatisticians() const { return StatisticianMap(); } @@ -454,4 +544,10 @@ int32_t NullReceiveStatistics::TimeUntilNextProcess() { return 0; } int32_t NullReceiveStatistics::Process() { return 0; } +void NullReceiveStatistics::RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) {} + +void NullReceiveStatistics::RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) {} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h index 0af074c4a13..4aa41f349e5 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h @@ -25,37 +25,48 @@ class CriticalSectionWrapper; class StreamStatisticianImpl : public StreamStatistician { public: - explicit StreamStatisticianImpl(Clock* clock); - + StreamStatisticianImpl(Clock* clock, + RtcpStatisticsCallback* rtcp_callback, + StreamDataCountersCallback* rtp_callback); virtual ~StreamStatisticianImpl() {} - virtual bool GetStatistics(Statistics* statistics, bool reset) OVERRIDE; + virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) OVERRIDE; virtual void GetDataCounters(uint32_t* bytes_received, uint32_t* packets_received) const OVERRIDE; virtual uint32_t BitrateReceived() const OVERRIDE; virtual void ResetStatistics() OVERRIDE; virtual bool IsRetransmitOfOldPacket(const RTPHeader& header, - int min_rtt) const OVERRIDE; + int min_rtt) const OVERRIDE; virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE; - void IncomingPacket(const RTPHeader& rtp_header, size_t bytes, + void IncomingPacket(const RTPHeader& rtp_header, + size_t bytes, bool retransmitted); + void FecPacketReceived(); void SetMaxReorderingThreshold(int max_reordering_threshold); void ProcessBitrate(); virtual void LastReceiveTimeNtp(uint32_t* secs, uint32_t* frac) const; private: bool InOrderPacketInternal(uint16_t sequence_number) const; + RtcpStatistics CalculateRtcpStatistics(); + void UpdateJitter(const RTPHeader& header, + uint32_t receive_time_secs, + uint32_t receive_time_frac); + void UpdateCounters(const RTPHeader& rtp_header, + size_t bytes, + bool retransmitted); + void NotifyRtpCallback() LOCKS_EXCLUDED(stream_lock_.get()); + void NotifyRtcpCallback() LOCKS_EXCLUDED(stream_lock_.get()); Clock* clock_; - scoped_ptr<CriticalSectionWrapper> crit_sect_; + scoped_ptr<CriticalSectionWrapper> stream_lock_; Bitrate incoming_bitrate_; uint32_t ssrc_; int max_reordering_threshold_; // In number of packets or sequence numbers. // Stats on received RTP packets. uint32_t jitter_q4_; - uint32_t jitter_max_q4_; uint32_t cumulative_loss_; uint32_t jitter_q4_transmission_time_offset_; @@ -67,30 +78,34 @@ class StreamStatisticianImpl : public StreamStatistician { uint16_t received_seq_first_; uint16_t received_seq_max_; uint16_t received_seq_wraps_; - bool first_packet_; // Current counter values. uint16_t received_packet_overhead_; - uint32_t received_byte_count_; - uint32_t received_retransmitted_packets_; - uint32_t received_inorder_packet_count_; + StreamDataCounters receive_counters_; // Counter values when we sent the last report. uint32_t last_report_inorder_packets_; uint32_t last_report_old_packets_; uint16_t last_report_seq_max_; - Statistics last_reported_statistics_; + RtcpStatistics last_reported_statistics_; + + RtcpStatisticsCallback* const rtcp_callback_; + StreamDataCountersCallback* const rtp_callback_; }; -class ReceiveStatisticsImpl : public ReceiveStatistics { +class ReceiveStatisticsImpl : public ReceiveStatistics, + public RtcpStatisticsCallback, + public StreamDataCountersCallback { public: explicit ReceiveStatisticsImpl(Clock* clock); ~ReceiveStatisticsImpl(); // Implement ReceiveStatistics. - virtual void IncomingPacket(const RTPHeader& header, size_t bytes, - bool old_packet) OVERRIDE; + virtual void IncomingPacket(const RTPHeader& header, + size_t bytes, + bool retransmitted) OVERRIDE; + virtual void FecPacketReceived(uint32_t ssrc) OVERRIDE; virtual StatisticianMap GetActiveStatisticians() const OVERRIDE; virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE; virtual void SetMaxReorderingThreshold(int max_reordering_threshold) OVERRIDE; @@ -101,13 +116,27 @@ class ReceiveStatisticsImpl : public ReceiveStatistics { void ChangeSsrc(uint32_t from_ssrc, uint32_t to_ssrc); + virtual void RegisterRtcpStatisticsCallback(RtcpStatisticsCallback* callback) + OVERRIDE; + + virtual void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) OVERRIDE; + private: + virtual void StatisticsUpdated(const RtcpStatistics& statistics, + uint32_t ssrc) OVERRIDE; + virtual void DataCountersUpdated(const StreamDataCounters& counters, + uint32_t ssrc) OVERRIDE; + typedef std::map<uint32_t, StreamStatisticianImpl*> StatisticianImplMap; Clock* clock_; - scoped_ptr<CriticalSectionWrapper> crit_sect_; + scoped_ptr<CriticalSectionWrapper> receive_statistics_lock_; int64_t last_rate_update_ms_; StatisticianImplMap statisticians_; + + RtcpStatisticsCallback* rtcp_stats_callback_; + StreamDataCountersCallback* rtp_stats_callback_; }; } // namespace webrtc #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc index be8f2fcdcc9..f0b9dedde57 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc @@ -28,10 +28,10 @@ class ReceiveStatisticsTest : public ::testing::Test { receive_statistics_(ReceiveStatistics::Create(&clock_)) { memset(&header1_, 0, sizeof(header1_)); header1_.ssrc = kSsrc1; - header1_.sequenceNumber = 0; + header1_.sequenceNumber = 100; memset(&header2_, 0, sizeof(header2_)); header2_.ssrc = kSsrc2; - header2_.sequenceNumber = 0; + header2_.sequenceNumber = 100; } protected: @@ -131,4 +131,173 @@ TEST_F(ReceiveStatisticsTest, ActiveStatisticians) { EXPECT_EQ(200u, bytes_received); EXPECT_EQ(2u, packets_received); } + +TEST_F(ReceiveStatisticsTest, RtcpCallbacks) { + class TestCallback : public RtcpStatisticsCallback { + public: + TestCallback() + : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {} + virtual ~TestCallback() {} + + virtual void StatisticsUpdated(const RtcpStatistics& statistics, + uint32_t ssrc) { + ssrc_ = ssrc; + stats_ = statistics; + ++num_calls_; + } + + uint32_t num_calls_; + uint32_t ssrc_; + RtcpStatistics stats_; + } callback; + + receive_statistics_->RegisterRtcpStatisticsCallback(&callback); + + // Add some arbitrary data, with loss and jitter. + header1_.sequenceNumber = 1; + clock_.AdvanceTimeMilliseconds(7); + header1_.timestamp += 3; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); + header1_.sequenceNumber += 2; + clock_.AdvanceTimeMilliseconds(9); + header1_.timestamp += 9; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); + --header1_.sequenceNumber; + clock_.AdvanceTimeMilliseconds(13); + header1_.timestamp += 47; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, true); + header1_.sequenceNumber += 3; + clock_.AdvanceTimeMilliseconds(11); + header1_.timestamp += 17; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); + ++header1_.sequenceNumber; + + EXPECT_EQ(0u, callback.num_calls_); + + // Call GetStatistics, simulating a timed rtcp sender thread. + RtcpStatistics statistics; + receive_statistics_->GetStatistician(kSsrc1) + ->GetStatistics(&statistics, true); + + EXPECT_EQ(1u, callback.num_calls_); + EXPECT_EQ(callback.ssrc_, kSsrc1); + EXPECT_EQ(statistics.cumulative_lost, callback.stats_.cumulative_lost); + EXPECT_EQ(statistics.extended_max_sequence_number, + callback.stats_.extended_max_sequence_number); + EXPECT_EQ(statistics.fraction_lost, callback.stats_.fraction_lost); + EXPECT_EQ(statistics.jitter, callback.stats_.jitter); + EXPECT_EQ(51, statistics.fraction_lost); + EXPECT_EQ(1u, statistics.cumulative_lost); + EXPECT_EQ(5u, statistics.extended_max_sequence_number); + EXPECT_EQ(4u, statistics.jitter); + + receive_statistics_->RegisterRtcpStatisticsCallback(NULL); + + // Add some more data. + header1_.sequenceNumber = 1; + clock_.AdvanceTimeMilliseconds(7); + header1_.timestamp += 3; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); + header1_.sequenceNumber += 2; + clock_.AdvanceTimeMilliseconds(9); + header1_.timestamp += 9; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); + --header1_.sequenceNumber; + clock_.AdvanceTimeMilliseconds(13); + header1_.timestamp += 47; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, true); + header1_.sequenceNumber += 3; + clock_.AdvanceTimeMilliseconds(11); + header1_.timestamp += 17; + receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); + ++header1_.sequenceNumber; + + receive_statistics_->GetStatistician(kSsrc1) + ->GetStatistics(&statistics, true); + + // Should not have been called after deregister. + EXPECT_EQ(1u, callback.num_calls_); +} + +TEST_F(ReceiveStatisticsTest, RtpCallbacks) { + class TestCallback : public StreamDataCountersCallback { + public: + TestCallback() + : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {} + virtual ~TestCallback() {} + + virtual void DataCountersUpdated(const StreamDataCounters& counters, + uint32_t ssrc) { + ssrc_ = ssrc; + stats_ = counters; + ++num_calls_; + } + + void ExpectMatches(uint32_t num_calls, + uint32_t ssrc, + uint32_t bytes, + uint32_t padding, + uint32_t packets, + uint32_t retransmits, + uint32_t fec) { + EXPECT_EQ(num_calls, num_calls_); + EXPECT_EQ(ssrc, ssrc_); + EXPECT_EQ(bytes, stats_.bytes); + EXPECT_EQ(padding, stats_.padding_bytes); + EXPECT_EQ(packets, stats_.packets); + EXPECT_EQ(retransmits, stats_.retransmitted_packets); + EXPECT_EQ(fec, stats_.fec_packets); + } + + uint32_t num_calls_; + uint32_t ssrc_; + StreamDataCounters stats_; + } callback; + + receive_statistics_->RegisterRtpStatisticsCallback(&callback); + + const uint32_t kHeaderLength = 20; + const uint32_t kPaddingLength = 9; + + // One packet of size kPacketSize1. + header1_.headerLength = kHeaderLength; + receive_statistics_->IncomingPacket( + header1_, kPacketSize1 + kHeaderLength, false); + callback.ExpectMatches(1, kSsrc1, kPacketSize1, 0, 1, 0, 0); + + ++header1_.sequenceNumber; + clock_.AdvanceTimeMilliseconds(5); + header1_.paddingLength = 9; + // Another packet of size kPacketSize1 with 9 bytes padding. + receive_statistics_->IncomingPacket( + header1_, kPacketSize1 + kHeaderLength + kPaddingLength, false); + callback.ExpectMatches(2, kSsrc1, 2 * kPacketSize1, kPaddingLength, 2, 0, 0); + + clock_.AdvanceTimeMilliseconds(5); + // Retransmit last packet. + receive_statistics_->IncomingPacket( + header1_, kPacketSize1 + kHeaderLength + kPaddingLength, true); + callback.ExpectMatches( + 3, kSsrc1, 3 * kPacketSize1, kPaddingLength * 2, 3, 1, 0); + + header1_.paddingLength = 0; + ++header1_.sequenceNumber; + clock_.AdvanceTimeMilliseconds(5); + // One recovered packet. + receive_statistics_->IncomingPacket( + header1_, kPacketSize1 + kHeaderLength, false); + receive_statistics_->FecPacketReceived(kSsrc1); + callback.ExpectMatches( + 5, kSsrc1, 4 * kPacketSize1, kPaddingLength * 2, 4, 1, 1); + + receive_statistics_->RegisterRtpStatisticsCallback(NULL); + + // New stats, but callback should not be called. + ++header1_.sequenceNumber; + clock_.AdvanceTimeMilliseconds(5); + receive_statistics_->IncomingPacket( + header1_, kPacketSize1 + kHeaderLength, true); + callback.ExpectMatches( + 5, kSsrc1, 4 * kPacketSize1, kPaddingLength * 2, 4, 1, 1); +} } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc new file mode 100644 index 00000000000..0d71c26b63c --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc @@ -0,0 +1,85 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" + +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h" + +namespace webrtc { + +// TODO(wu): Refactor this class so that it can be shared with +// vie_sync_module.cc. +RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock) + : clock_(clock), + ts_extrapolator_( + new TimestampExtrapolator(clock_->TimeInMilliseconds())) { +} + +RemoteNtpTimeEstimator::~RemoteNtpTimeEstimator() {} + +bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(uint32_t ssrc, + RtpRtcp* rtp_rtcp) { + assert(rtp_rtcp); + uint16_t rtt = 0; + rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL); + if (rtt == 0) { + // Waiting for valid rtt. + return true; + } + // Update RTCP list + uint32_t ntp_secs = 0; + uint32_t ntp_frac = 0; + uint32_t rtp_timestamp = 0; + if (0 != rtp_rtcp->RemoteNTP(&ntp_secs, + &ntp_frac, + NULL, + NULL, + &rtp_timestamp)) { + // Waiting for RTCP. + return true; + } + bool new_rtcp_sr = false; + if (!UpdateRtcpList( + ntp_secs, ntp_frac, rtp_timestamp, &rtcp_list_, &new_rtcp_sr)) { + return false; + } + if (!new_rtcp_sr) { + // No new RTCP SR since last time this function was called. + return true; + } + // Update extrapolator with the new arrival time. + // The extrapolator assumes the TimeInMilliseconds time. + int64_t receiver_arrival_time_ms = clock_->TimeInMilliseconds(); + int64_t sender_send_time_ms = Clock::NtpToMs(ntp_secs, ntp_frac); + int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90; + ts_extrapolator_->Update(receiver_arrival_time_ms, sender_arrival_time_90k); + return true; +} + +int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) { + if (rtcp_list_.size() < 2) { + // We need two RTCP SR reports to calculate NTP. + return -1; + } + int64_t sender_capture_ntp_ms = 0; + if (!RtpToNtpMs(rtp_timestamp, rtcp_list_, &sender_capture_ntp_ms)) { + return -1; + } + uint32_t timestamp = sender_capture_ntp_ms * 90; + int64_t receiver_capture_ms = + ts_extrapolator_->ExtrapolateLocalTime(timestamp); + int64_t ntp_offset = + clock_->CurrentNtpInMilliseconds() - clock_->TimeInMilliseconds(); + return receiver_capture_ms + ntp_offset; +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc new file mode 100644 index 00000000000..63cedf03afe --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc @@ -0,0 +1,112 @@ +/* +* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +* +* Use of this source code is governed by a BSD-style license +* that can be found in the LICENSE file in the root of the source +* tree. An additional intellectual property rights grant can be found +* in the file PATENTS. All contributing project authors may +* be found in the AUTHORS file in the root of the source tree. +*/ + +#include "testing/gmock/include/gmock/gmock.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/common_types.h" +#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" +#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" + +using ::testing::_; +using ::testing::DoAll; +using ::testing::Return; +using ::testing::SetArgPointee; + +namespace webrtc { + +static const int kTestRtt = 10; +static const int64_t kLocalClockInitialTimeMs = 123; +static const int64_t kRemoteClockInitialTimeMs = 345; +static const uint32_t kTimestampOffset = 567; +static const int kTestSsrc = 789; + +class RemoteNtpTimeEstimatorTest : public ::testing::Test { + protected: + RemoteNtpTimeEstimatorTest() + : local_clock_(kLocalClockInitialTimeMs * 1000), + remote_clock_(kRemoteClockInitialTimeMs * 1000), + estimator_(&local_clock_) {} + ~RemoteNtpTimeEstimatorTest() {} + + void AdvanceTimeMilliseconds(int64_t ms) { + local_clock_.AdvanceTimeMilliseconds(ms); + remote_clock_.AdvanceTimeMilliseconds(ms); + } + + uint32_t GetRemoteTimestamp() { + return static_cast<uint32_t>(remote_clock_.TimeInMilliseconds()) * 90 + + kTimestampOffset; + } + + void SendRtcpSr() { + uint32_t rtcp_timestamp = GetRemoteTimestamp(); + uint32_t ntp_seconds; + uint32_t ntp_fractions; + remote_clock_.CurrentNtp(ntp_seconds, ntp_fractions); + + AdvanceTimeMilliseconds(kTestRtt / 2); + ReceiveRtcpSr(rtcp_timestamp, ntp_seconds, ntp_fractions); + } + + void UpdateRtcpTimestamp(MockRtpRtcp* rtp_rtcp, bool expected_result) { + if (rtp_rtcp) { + EXPECT_CALL(*rtp_rtcp, RTT(_, _, _, _, _)) + .WillOnce(DoAll(SetArgPointee<1>(kTestRtt), + Return(0))); + } + EXPECT_EQ(expected_result, + estimator_.UpdateRtcpTimestamp(kTestSsrc, rtp_rtcp)); + } + + void ReceiveRtcpSr(uint32_t rtcp_timestamp, + uint32_t ntp_seconds, + uint32_t ntp_fractions) { + EXPECT_CALL(rtp_rtcp_, RemoteNTP(_, _, _, _, _)) + .WillOnce(DoAll(SetArgPointee<0>(ntp_seconds), + SetArgPointee<1>(ntp_fractions), + SetArgPointee<4>(rtcp_timestamp), + Return(0))); + + UpdateRtcpTimestamp(&rtp_rtcp_, true); + } + + SimulatedClock local_clock_; + SimulatedClock remote_clock_; + MockRtpRtcp rtp_rtcp_; + RemoteNtpTimeEstimator estimator_; +}; + +TEST_F(RemoteNtpTimeEstimatorTest, Estimate) { + // Failed without any RTCP SR, where RemoteNTP returns without valid NTP. + EXPECT_CALL(rtp_rtcp_, RemoteNTP(_, _, _, _, _)).WillOnce(Return(0)); + UpdateRtcpTimestamp(&rtp_rtcp_, false); + + AdvanceTimeMilliseconds(1000); + // Remote peer sends first RTCP SR. + SendRtcpSr(); + + // Remote sends a RTP packet. + AdvanceTimeMilliseconds(15); + uint32_t rtp_timestamp = GetRemoteTimestamp(); + int64_t capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds(); + + // Local peer needs at least 2 RTCP SR to calculate the capture time. + const int64_t kNotEnoughRtcpSr = -1; + EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp)); + + AdvanceTimeMilliseconds(800); + // Remote sends second RTCP SR. + SendRtcpSr(); + + // Local peer gets enough RTCP SR to calculate the capture time. + EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp)); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc index 68f61371395..88463e4718f 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc @@ -69,6 +69,7 @@ class RtcpFormatRembTest : public ::testing::Test { RemoteBitrateEstimatorFactory().Create( &remote_bitrate_observer_, system_clock_, + kMimdControl, kRemoteBitrateEstimatorMinBitrateBps)) {} virtual void SetUp(); virtual void TearDown(); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc new file mode 100644 index 00000000000..a4cdfd95dee --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc @@ -0,0 +1,695 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" + +#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" +#include "webrtc/system_wrappers/interface/logging.h" + +using webrtc::RTCPUtility::PT_APP; +using webrtc::RTCPUtility::PT_BYE; +using webrtc::RTCPUtility::PT_IJ; +using webrtc::RTCPUtility::PT_PSFB; +using webrtc::RTCPUtility::PT_RR; +using webrtc::RTCPUtility::PT_RTPFB; +using webrtc::RTCPUtility::PT_SDES; +using webrtc::RTCPUtility::PT_SR; +using webrtc::RTCPUtility::PT_XR; + +using webrtc::RTCPUtility::RTCPPacketAPP; +using webrtc::RTCPUtility::RTCPPacketBYE; +using webrtc::RTCPUtility::RTCPPacketPSFBAPP; +using webrtc::RTCPUtility::RTCPPacketPSFBFIR; +using webrtc::RTCPUtility::RTCPPacketPSFBFIRItem; +using webrtc::RTCPUtility::RTCPPacketPSFBPLI; +using webrtc::RTCPUtility::RTCPPacketPSFBREMBItem; +using webrtc::RTCPUtility::RTCPPacketPSFBRPSI; +using webrtc::RTCPUtility::RTCPPacketPSFBSLI; +using webrtc::RTCPUtility::RTCPPacketPSFBSLIItem; +using webrtc::RTCPUtility::RTCPPacketReportBlockItem; +using webrtc::RTCPUtility::RTCPPacketRR; +using webrtc::RTCPUtility::RTCPPacketRTPFBNACK; +using webrtc::RTCPUtility::RTCPPacketRTPFBNACKItem; +using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBN; +using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBNItem; +using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBR; +using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem; +using webrtc::RTCPUtility::RTCPPacketSR; +using webrtc::RTCPUtility::RTCPPacketXRDLRRReportBlockItem; +using webrtc::RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem; +using webrtc::RTCPUtility::RTCPPacketXR; +using webrtc::RTCPUtility::RTCPPacketXRVOIPMetricItem; + +namespace webrtc { +namespace rtcp { +namespace { +// Unused SSRC of media source, set to 0. +const uint32_t kUnusedMediaSourceSsrc0 = 0; + +void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) { + buffer[(*offset)++] = value; +} +void AssignUWord16(uint8_t* buffer, size_t* offset, uint16_t value) { + ModuleRTPUtility::AssignUWord16ToBuffer(buffer + *offset, value); + *offset += 2; +} +void AssignUWord24(uint8_t* buffer, size_t* offset, uint32_t value) { + ModuleRTPUtility::AssignUWord24ToBuffer(buffer + *offset, value); + *offset += 3; +} +void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) { + ModuleRTPUtility::AssignUWord32ToBuffer(buffer + *offset, value); + *offset += 4; +} + +size_t BlockToHeaderLength(size_t length_in_bytes) { + // Length in 32-bit words minus 1. + assert(length_in_bytes > 0); + assert(length_in_bytes % 4 == 0); + return (length_in_bytes / 4) - 1; +} + +// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications. +// +// RTP header format. +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| RC/FMT | PT | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateHeader(uint8_t count_or_format, // Depends on packet type. + uint8_t packet_type, + size_t length, + uint8_t* buffer, + size_t* pos) { + assert(length <= 0xffff); + const uint8_t kVersion = 2; + AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format); + AssignUWord8(buffer, pos, packet_type); + AssignUWord16(buffer, pos, length); +} + +// Sender report (SR) (RFC 3550). +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| RC | PT=SR=200 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of sender | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// | NTP timestamp, most significant word | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | NTP timestamp, least significant word | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | RTP timestamp | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | sender's packet count | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | sender's octet count | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ + +void CreateSenderReport(const RTCPPacketSR& sr, + size_t length, + uint8_t* buffer, + size_t* pos) { + CreateHeader(sr.NumberOfReportBlocks, PT_SR, length, buffer, pos); + AssignUWord32(buffer, pos, sr.SenderSSRC); + AssignUWord32(buffer, pos, sr.NTPMostSignificant); + AssignUWord32(buffer, pos, sr.NTPLeastSignificant); + AssignUWord32(buffer, pos, sr.RTPTimestamp); + AssignUWord32(buffer, pos, sr.SenderPacketCount); + AssignUWord32(buffer, pos, sr.SenderOctetCount); +} + +// Receiver report (RR), header (RFC 3550). +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| RC | PT=RR=201 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of packet sender | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ + +void CreateReceiverReport(const RTCPPacketRR& rr, + size_t length, + uint8_t* buffer, + size_t* pos) { + CreateHeader(rr.NumberOfReportBlocks, PT_RR, length, buffer, pos); + AssignUWord32(buffer, pos, rr.SenderSSRC); +} + +// Report block (RFC 3550). +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// | SSRC_1 (SSRC of first source) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | fraction lost | cumulative number of packets lost | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | extended highest sequence number received | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | interarrival jitter | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | last SR (LSR) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | delay since last SR (DLSR) | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ + +void CreateReportBlocks(const std::vector<RTCPPacketReportBlockItem>& blocks, + uint8_t* buffer, + size_t* pos) { + for (std::vector<RTCPPacketReportBlockItem>::const_iterator + it = blocks.begin(); it != blocks.end(); ++it) { + AssignUWord32(buffer, pos, (*it).SSRC); + AssignUWord8(buffer, pos, (*it).FractionLost); + AssignUWord24(buffer, pos, (*it).CumulativeNumOfPacketsLost); + AssignUWord32(buffer, pos, (*it).ExtendedHighestSequenceNumber); + AssignUWord32(buffer, pos, (*it).Jitter); + AssignUWord32(buffer, pos, (*it).LastSR); + AssignUWord32(buffer, pos, (*it).DelayLastSR); + } +} + +// Transmission Time Offsets in RTP Streams (RFC 5450). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// hdr |V=2|P| RC | PT=IJ=195 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | inter-arrival jitter | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// . . +// . . +// . . +// | inter-arrival jitter | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateIj(const std::vector<uint32_t>& ij_items, + uint8_t* buffer, + size_t* pos) { + size_t length = ij_items.size(); + CreateHeader(length, PT_IJ, length, buffer, pos); + for (std::vector<uint32_t>::const_iterator it = ij_items.begin(); + it != ij_items.end(); ++it) { + AssignUWord32(buffer, pos, *it); + } +} + +// Source Description (SDES) (RFC 3550). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// header |V=2|P| SC | PT=SDES=202 | length | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// chunk | SSRC/CSRC_1 | +// 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SDES items | +// | ... | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// chunk | SSRC/CSRC_2 | +// 2 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SDES items | +// | ... | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// +// Canonical End-Point Identifier SDES Item (CNAME) +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | CNAME=1 | length | user and domain name ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateSdes(const std::vector<Sdes::Chunk>& chunks, + size_t length, + uint8_t* buffer, + size_t* pos) { + CreateHeader(chunks.size(), PT_SDES, length, buffer, pos); + const uint8_t kSdesItemType = 1; + for (std::vector<Sdes::Chunk>::const_iterator it = chunks.begin(); + it != chunks.end(); ++it) { + AssignUWord32(buffer, pos, (*it).ssrc); + AssignUWord8(buffer, pos, kSdesItemType); + AssignUWord8(buffer, pos, (*it).name.length()); + memcpy(buffer + *pos, (*it).name.data(), (*it).name.length()); + *pos += (*it).name.length(); + memset(buffer + *pos, 0, (*it).null_octets); + *pos += (*it).null_octets; + } +} + +// Bye packet (BYE) (RFC 3550). +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| SC | PT=BYE=203 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC/CSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// : ... : +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// (opt) | length | reason for leaving ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateBye(const RTCPPacketBYE& bye, + const std::vector<uint32_t>& csrcs, + size_t length, + uint8_t* buffer, + size_t* pos) { + CreateHeader(length, PT_BYE, length, buffer, pos); + AssignUWord32(buffer, pos, bye.SenderSSRC); + for (std::vector<uint32_t>::const_iterator it = csrcs.begin(); + it != csrcs.end(); ++it) { + AssignUWord32(buffer, pos, *it); + } +} + +// Application-Defined packet (APP) (RFC 3550). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| subtype | PT=APP=204 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC/CSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | name (ASCII) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | application-dependent data ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateApp(const RTCPPacketAPP& app, + uint32_t ssrc, + size_t length, + uint8_t* buffer, + size_t* pos) { + CreateHeader(app.SubType, PT_APP, length, buffer, pos); + AssignUWord32(buffer, pos, ssrc); + AssignUWord32(buffer, pos, app.Name); + memcpy(buffer + *pos, app.Data, app.Size); + *pos += app.Size; +} + +// RFC 4585: Feedback format. +// +// Common packet format: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| FMT | PT | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of packet sender | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of media source | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// : Feedback Control Information (FCI) : +// : +// + +// Picture loss indication (PLI) (RFC 4585). +// +// FCI: no feedback control information. + +void CreatePli(const RTCPPacketPSFBPLI& pli, + size_t length, + uint8_t* buffer, + size_t* pos) { + const uint8_t kFmt = 1; + CreateHeader(kFmt, PT_PSFB, length, buffer, pos); + AssignUWord32(buffer, pos, pli.SenderSSRC); + AssignUWord32(buffer, pos, pli.MediaSSRC); +} + +// Slice loss indication (SLI) (RFC 4585). +// +// FCI: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | First | Number | PictureID | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateSli(const RTCPPacketPSFBSLI& sli, + const RTCPPacketPSFBSLIItem& sli_item, + size_t length, + uint8_t* buffer, + size_t* pos) { + const uint8_t kFmt = 2; + CreateHeader(kFmt, PT_PSFB, length, buffer, pos); + AssignUWord32(buffer, pos, sli.SenderSSRC); + AssignUWord32(buffer, pos, sli.MediaSSRC); + + AssignUWord8(buffer, pos, sli_item.FirstMB >> 5); + AssignUWord8(buffer, pos, (sli_item.FirstMB << 3) + + ((sli_item.NumberOfMB >> 10) & 0x07)); + AssignUWord8(buffer, pos, sli_item.NumberOfMB >> 2); + AssignUWord8(buffer, pos, (sli_item.NumberOfMB << 6) + sli_item.PictureId); +} + +// Generic NACK (RFC 4585). +// +// FCI: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PID | BLP | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateNack(const RTCPPacketRTPFBNACK& nack, + const std::vector<RTCPPacketRTPFBNACKItem>& nack_fields, + size_t length, + uint8_t* buffer, + size_t* pos) { + const uint8_t kFmt = 1; + CreateHeader(kFmt, PT_RTPFB, length, buffer, pos); + AssignUWord32(buffer, pos, nack.SenderSSRC); + AssignUWord32(buffer, pos, nack.MediaSSRC); + for (std::vector<RTCPPacketRTPFBNACKItem>::const_iterator + it = nack_fields.begin(); it != nack_fields.end(); ++it) { + AssignUWord16(buffer, pos, (*it).PacketID); + AssignUWord16(buffer, pos, (*it).BitMask); + } +} + +// Reference picture selection indication (RPSI) (RFC 4585). +// +// FCI: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PB |0| Payload Type| Native RPSI bit string | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | defined per codec ... | Padding (0) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateRpsi(const RTCPPacketPSFBRPSI& rpsi, + uint8_t padding_bytes, + size_t length, + uint8_t* buffer, + size_t* pos) { + // Native bit string should be a multiple of 8 bits. + assert(rpsi.NumberOfValidBits % 8 == 0); + const uint8_t kFmt = 3; + CreateHeader(kFmt, PT_PSFB, length, buffer, pos); + AssignUWord32(buffer, pos, rpsi.SenderSSRC); + AssignUWord32(buffer, pos, rpsi.MediaSSRC); + AssignUWord8(buffer, pos, padding_bytes * 8); + AssignUWord8(buffer, pos, rpsi.PayloadType); + memcpy(buffer + *pos, rpsi.NativeBitString, rpsi.NumberOfValidBits / 8); + *pos += rpsi.NumberOfValidBits / 8; + memset(buffer + *pos, 0, padding_bytes); + *pos += padding_bytes; +} + +// Full intra request (FIR) (RFC 5104). +// +// FCI: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Seq nr. | Reserved | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +void CreateFir(const RTCPPacketPSFBFIR& fir, + const RTCPPacketPSFBFIRItem& fir_item, + size_t length, + uint8_t* buffer, + size_t* pos) { + const uint8_t kFmt = 4; + CreateHeader(kFmt, PT_PSFB, length, buffer, pos); + AssignUWord32(buffer, pos, fir.SenderSSRC); + AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0); + AssignUWord32(buffer, pos, fir_item.SSRC); + AssignUWord8(buffer, pos, fir_item.CommandSequenceNumber); + AssignUWord24(buffer, pos, 0); +} +} // namespace + +void RtcpPacket::Append(RtcpPacket* packet) { + assert(packet); + appended_packets_.push_back(packet); +} + +RawPacket RtcpPacket::Build() const { + size_t length = 0; + uint8_t packet[IP_PACKET_SIZE]; + CreateAndAddAppended(packet, &length, IP_PACKET_SIZE); + return RawPacket(packet, length); +} + +void RtcpPacket::Build(uint8_t* packet, + size_t* length, + size_t max_length) const { + *length = 0; + CreateAndAddAppended(packet, length, max_length); +} + +void RtcpPacket::CreateAndAddAppended(uint8_t* packet, + size_t* length, + size_t max_length) const { + Create(packet, length, max_length); + for (std::vector<RtcpPacket*>::const_iterator it = appended_packets_.begin(); + it != appended_packets_.end(); ++it) { + (*it)->CreateAndAddAppended(packet, length, max_length); + } +} + +void Empty::Create(uint8_t* packet, size_t* length, size_t max_length) const { +} + +void SenderReport::Create(uint8_t* packet, + size_t* length, + size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateSenderReport(sr_, BlockToHeaderLength(BlockLength()), packet, length); + CreateReportBlocks(report_blocks_, packet, length); +} + +void SenderReport::WithReportBlock(ReportBlock* block) { + assert(block); + if (report_blocks_.size() >= kMaxNumberOfReportBlocks) { + LOG(LS_WARNING) << "Max report blocks reached."; + return; + } + report_blocks_.push_back(block->report_block_); + sr_.NumberOfReportBlocks = report_blocks_.size(); +} + +void ReceiverReport::Create(uint8_t* packet, + size_t* length, + size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateReceiverReport(rr_, BlockToHeaderLength(BlockLength()), packet, length); + CreateReportBlocks(report_blocks_, packet, length); +} + +void ReceiverReport::WithReportBlock(ReportBlock* block) { + assert(block); + if (report_blocks_.size() >= kMaxNumberOfReportBlocks) { + LOG(LS_WARNING) << "Max report blocks reached."; + return; + } + report_blocks_.push_back(block->report_block_); + rr_.NumberOfReportBlocks = report_blocks_.size(); +} + +void Ij::Create(uint8_t* packet, size_t* length, size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateIj(ij_items_, packet, length); +} + +void Ij::WithJitterItem(uint32_t jitter) { + if (ij_items_.size() >= kMaxNumberOfIjItems) { + LOG(LS_WARNING) << "Max inter-arrival jitter items reached."; + return; + } + ij_items_.push_back(jitter); +} + +void Sdes::Create(uint8_t* packet, size_t* length, size_t max_length) const { + assert(!chunks_.empty()); + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateSdes(chunks_, BlockToHeaderLength(BlockLength()), packet, length); +} + +void Sdes::WithCName(uint32_t ssrc, std::string cname) { + assert(cname.length() <= 0xff); + if (chunks_.size() >= kMaxNumberOfChunks) { + LOG(LS_WARNING) << "Max SDES chunks reached."; + return; + } + // In each chunk, the list of items must be terminated by one or more null + // octets. The next chunk must start on a 32-bit boundary. + // CNAME (1 byte) | length (1 byte) | name | padding. + int null_octets = 4 - ((2 + cname.length()) % 4); + Chunk chunk; + chunk.ssrc = ssrc; + chunk.name = cname; + chunk.null_octets = null_octets; + chunks_.push_back(chunk); +} + +size_t Sdes::BlockLength() const { + // Header (4 bytes). + // Chunk: + // SSRC/CSRC (4 bytes) | CNAME (1 byte) | length (1 byte) | name | padding. + size_t length = kHeaderLength; + for (std::vector<Chunk>::const_iterator it = chunks_.begin(); + it != chunks_.end(); ++it) { + length += 6 + (*it).name.length() + (*it).null_octets; + } + assert(length % 4 == 0); + return length; +} + +void Bye::Create(uint8_t* packet, size_t* length, size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateBye(bye_, csrcs_, BlockToHeaderLength(BlockLength()), packet, length); +} + +void Bye::WithCsrc(uint32_t csrc) { + if (csrcs_.size() >= kMaxNumberOfCsrcs) { + LOG(LS_WARNING) << "Max CSRC size reached."; + return; + } + csrcs_.push_back(csrc); +} + +void App::Create(uint8_t* packet, size_t* length, size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateApp(app_, ssrc_, BlockToHeaderLength(BlockLength()), packet, length); +} + +void Pli::Create(uint8_t* packet, size_t* length, size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreatePli(pli_, BlockToHeaderLength(BlockLength()), packet, length); +} + +void Sli::Create(uint8_t* packet, size_t* length, size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateSli(sli_, sli_item_, BlockToHeaderLength(BlockLength()), packet, + length); +} + +void Nack::Create(uint8_t* packet, size_t* length, size_t max_length) const { + assert(!nack_fields_.empty()); + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateNack(nack_, nack_fields_, BlockToHeaderLength(BlockLength()), packet, + length); +} + +void Nack::WithList(const uint16_t* nack_list, int length) { + assert(nack_list); + assert(nack_fields_.empty()); + int i = 0; + while (i < length) { + uint16_t pid = nack_list[i++]; + // Bitmask specifies losses in any of the 16 packets following the pid. + uint16_t bitmask = 0; + while (i < length) { + int shift = static_cast<uint16_t>(nack_list[i] - pid) - 1; + if (shift >= 0 && shift <= 15) { + bitmask |= (1 << shift); + ++i; + } else { + break; + } + } + RTCPUtility::RTCPPacketRTPFBNACKItem item; + item.PacketID = pid; + item.BitMask = bitmask; + nack_fields_.push_back(item); + } +} + +void Rpsi::Create(uint8_t* packet, size_t* length, size_t max_length) const { + assert(rpsi_.NumberOfValidBits > 0); + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateRpsi(rpsi_, padding_bytes_, BlockToHeaderLength(BlockLength()), packet, + length); +} + +void Rpsi::WithPictureId(uint64_t picture_id) { + const uint32_t kPidBits = 7; + const uint64_t k7MsbZeroMask = 0x1ffffffffffffff; + uint8_t required_bytes = 0; + uint64_t shifted_pid = picture_id; + do { + ++required_bytes; + shifted_pid = (shifted_pid >> kPidBits) & k7MsbZeroMask; + } while (shifted_pid > 0); + + // Convert picture id to native bit string (natively defined by the video + // codec). + int pos = 0; + for (int i = required_bytes - 1; i > 0; i--) { + rpsi_.NativeBitString[pos++] = + 0x80 | static_cast<uint8_t>(picture_id >> (i * kPidBits)); + } + rpsi_.NativeBitString[pos++] = static_cast<uint8_t>(picture_id & 0x7f); + rpsi_.NumberOfValidBits = pos * 8; + + // Calculate padding bytes (to reach next 32-bit boundary, 1, 2 or 3 bytes). + padding_bytes_ = 4 - ((2 + required_bytes) % 4); + if (padding_bytes_ == 4) { + padding_bytes_ = 0; + } +} + +void Fir::Create(uint8_t* packet, size_t* length, size_t max_length) const { + if (*length + BlockLength() > max_length) { + LOG(LS_WARNING) << "Max packet size reached."; + return; + } + CreateFir(fir_, fir_item_, BlockToHeaderLength(BlockLength()), packet, + length); +} +} // namespace rtcp +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.h new file mode 100644 index 00000000000..f60e848b50b --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.h @@ -0,0 +1,726 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + * + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_ +#define WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_ + +#include <map> +#include <string> +#include <vector> + +#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +namespace rtcp { + +enum { kCommonFbFmtLength = 12 }; +enum { kReportBlockLength = 24 }; + +class RawPacket; + +// Class for building RTCP packets. +// +// Example: +// ReportBlock report_block; +// report_block.To(234) +// report_block.FractionLost(10); +// +// ReceiverReport rr; +// rr.From(123); +// rr.WithReportBlock(&report_block) +// +// Fir fir; +// fir.From(123); +// fir.To(234) +// fir.WithCommandSeqNum(123); +// +// size_t length = 0; // Builds an intra frame request +// uint8_t packet[kPacketSize]; // with sequence number 123. +// fir.Build(packet, &length, kPacketSize); +// +// RawPacket packet = fir.Build(); // Returns a RawPacket holding +// // the built rtcp packet. +// +// rr.Append(&fir) // Builds a compound RTCP packet with +// RawPacket packet = rr.Build(); // a receiver report, report block +// // and fir message. + +class RtcpPacket { + public: + virtual ~RtcpPacket() {} + + void Append(RtcpPacket* packet); + + RawPacket Build() const; + + void Build(uint8_t* packet, size_t* length, size_t max_length) const; + + protected: + RtcpPacket() : kHeaderLength(4) {} + + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const = 0; + + const size_t kHeaderLength; + + private: + void CreateAndAddAppended( + uint8_t* packet, size_t* length, size_t max_length) const; + + std::vector<RtcpPacket*> appended_packets_; +}; + +class Empty : public RtcpPacket { + public: + Empty() {} + + virtual ~Empty() {} + + protected: + virtual void Create(uint8_t* packet, size_t* length, size_t max_length) const; +}; + +// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications. +// +// RTCP report block (RFC 3550). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// | SSRC_1 (SSRC of first source) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | fraction lost | cumulative number of packets lost | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | extended highest sequence number received | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | interarrival jitter | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | last SR (LSR) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | delay since last SR (DLSR) | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ + +class ReportBlock { + public: + ReportBlock() { + // TODO(asapersson): Consider adding a constructor to struct. + memset(&report_block_, 0, sizeof(report_block_)); + } + + ~ReportBlock() {} + + void To(uint32_t ssrc) { + report_block_.SSRC = ssrc; + } + void WithFractionLost(uint8_t fraction_lost) { + report_block_.FractionLost = fraction_lost; + } + void WithCumulativeLost(uint32_t cumulative_lost) { + report_block_.CumulativeNumOfPacketsLost = cumulative_lost; + } + void WithExtHighestSeqNum(uint32_t ext_highest_seq_num) { + report_block_.ExtendedHighestSequenceNumber = ext_highest_seq_num; + } + void WithJitter(uint32_t jitter) { + report_block_.Jitter = jitter; + } + void WithLastSr(uint32_t last_sr) { + report_block_.LastSR = last_sr; + } + void WithDelayLastSr(uint32_t delay_last_sr) { + report_block_.DelayLastSR = delay_last_sr; + } + + private: + friend class SenderReport; + friend class ReceiverReport; + RTCPUtility::RTCPPacketReportBlockItem report_block_; +}; + +// RTCP sender report (RFC 3550). +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| RC | PT=SR=200 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of sender | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// | NTP timestamp, most significant word | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | NTP timestamp, least significant word | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | RTP timestamp | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | sender's packet count | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | sender's octet count | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// | report block(s) | +// | .... | + +class SenderReport : public RtcpPacket { + public: + SenderReport() : RtcpPacket() { + memset(&sr_, 0, sizeof(sr_)); + } + + virtual ~SenderReport() {} + + void From(uint32_t ssrc) { + sr_.SenderSSRC = ssrc; + } + void WithNtpSec(uint32_t sec) { + sr_.NTPMostSignificant = sec; + } + void WithNtpFrac(uint32_t frac) { + sr_.NTPLeastSignificant = frac; + } + void WithRtpTimestamp(uint32_t rtp_timestamp) { + sr_.RTPTimestamp = rtp_timestamp; + } + void WithPacketCount(uint32_t packet_count) { + sr_.SenderPacketCount = packet_count; + } + void WithOctetCount(uint32_t octet_count) { + sr_.SenderOctetCount = octet_count; + } + void WithReportBlock(ReportBlock* block); + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + enum { kMaxNumberOfReportBlocks = 0x1f }; + + size_t BlockLength() const { + const size_t kSrHeaderLength = 8; + const size_t kSenderInfoLength = 20; + return kSrHeaderLength + kSenderInfoLength + + report_blocks_.size() * kReportBlockLength; + } + + RTCPUtility::RTCPPacketSR sr_; + std::vector<RTCPUtility::RTCPPacketReportBlockItem> report_blocks_; +}; + +// +// RTCP receiver report (RFC 3550). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| RC | PT=RR=201 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of packet sender | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// | report block(s) | +// | .... | + +class ReceiverReport : public RtcpPacket { + public: + ReceiverReport() : RtcpPacket() { + memset(&rr_, 0, sizeof(rr_)); + } + + virtual ~ReceiverReport() {} + + void From(uint32_t ssrc) { + rr_.SenderSSRC = ssrc; + } + void WithReportBlock(ReportBlock* block); + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + enum { kMaxNumberOfReportBlocks = 0x1f }; + + size_t BlockLength() const { + const size_t kRrHeaderLength = 8; + return kRrHeaderLength + report_blocks_.size() * kReportBlockLength; + } + + RTCPUtility::RTCPPacketRR rr_; + std::vector<RTCPUtility::RTCPPacketReportBlockItem> report_blocks_; +}; + +// Transmission Time Offsets in RTP Streams (RFC 5450). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// hdr |V=2|P| RC | PT=IJ=195 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | inter-arrival jitter | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// . . +// . . +// . . +// | inter-arrival jitter | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// +// If present, this RTCP packet must be placed after a receiver report +// (inside a compound RTCP packet), and MUST have the same value for RC +// (reception report count) as the receiver report. + +class Ij : public RtcpPacket { + public: + Ij() : RtcpPacket() {} + + virtual ~Ij() {} + + void WithJitterItem(uint32_t jitter); + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + enum { kMaxNumberOfIjItems = 0x1f }; + + size_t BlockLength() const { + return kHeaderLength + 4 * ij_items_.size(); + } + + std::vector<uint32_t> ij_items_; + + DISALLOW_COPY_AND_ASSIGN(Ij); +}; + +// Source Description (SDES) (RFC 3550). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// header |V=2|P| SC | PT=SDES=202 | length | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// chunk | SSRC/CSRC_1 | +// 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SDES items | +// | ... | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// chunk | SSRC/CSRC_2 | +// 2 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SDES items | +// | ... | +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// +// Canonical End-Point Identifier SDES Item (CNAME) +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | CNAME=1 | length | user and domain name ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class Sdes : public RtcpPacket { + public: + Sdes() : RtcpPacket() {} + + virtual ~Sdes() {} + + void WithCName(uint32_t ssrc, std::string cname); + + struct Chunk { + uint32_t ssrc; + std::string name; + int null_octets; + }; + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + enum { kMaxNumberOfChunks = 0x1f }; + + size_t BlockLength() const; + + std::vector<Chunk> chunks_; + + DISALLOW_COPY_AND_ASSIGN(Sdes); +}; + +// +// Bye packet (BYE) (RFC 3550). +// +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| SC | PT=BYE=203 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC/CSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// : ... : +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// (opt) | length | reason for leaving ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class Bye : public RtcpPacket { + public: + Bye() : RtcpPacket() { + memset(&bye_, 0, sizeof(bye_)); + } + + virtual ~Bye() {} + + void From(uint32_t ssrc) { + bye_.SenderSSRC = ssrc; + } + void WithCsrc(uint32_t csrc); + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + enum { kMaxNumberOfCsrcs = 0x1f - 1 }; + + size_t BlockLength() const { + size_t source_count = 1 + csrcs_.size(); + return kHeaderLength + 4 * source_count; + } + + RTCPUtility::RTCPPacketBYE bye_; + std::vector<uint32_t> csrcs_; +}; + +// Application-Defined packet (APP) (RFC 3550). +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| subtype | PT=APP=204 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC/CSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | name (ASCII) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | application-dependent data ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class App : public RtcpPacket { + public: + App() + : RtcpPacket(), + ssrc_(0) { + memset(&app_, 0, sizeof(app_)); + } + + virtual ~App() {} + + void From(uint32_t ssrc) { + ssrc_ = ssrc; + } + void WithSubType(uint8_t subtype) { + assert(subtype <= 0x1f); + app_.SubType = subtype; + } + void WithName(uint32_t name) { + app_.Name = name; + } + void WithData(const uint8_t* data, uint16_t data_length) { + assert(data); + assert(data_length <= kRtcpAppCode_DATA_SIZE); + assert(data_length % 4 == 0); + memcpy(app_.Data, data, data_length); + app_.Size = data_length; + } + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + size_t BlockLength() const { + return 12 + app_.Size; + } + + uint32_t ssrc_; + RTCPUtility::RTCPPacketAPP app_; + + DISALLOW_COPY_AND_ASSIGN(App); +}; + +// RFC 4585: Feedback format. +// +// Common packet format: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// |V=2|P| FMT | PT | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of packet sender | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC of media source | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// : Feedback Control Information (FCI) : +// : + +// Picture loss indication (PLI) (RFC 4585). +// +// FCI: no feedback control information. + +class Pli : public RtcpPacket { + public: + Pli() : RtcpPacket() { + memset(&pli_, 0, sizeof(pli_)); + } + + virtual ~Pli() {} + + void From(uint32_t ssrc) { + pli_.SenderSSRC = ssrc; + } + void To(uint32_t ssrc) { + pli_.MediaSSRC = ssrc; + } + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + size_t BlockLength() const { + return kCommonFbFmtLength; + } + + RTCPUtility::RTCPPacketPSFBPLI pli_; + + DISALLOW_COPY_AND_ASSIGN(Pli); +}; + +// Slice loss indication (SLI) (RFC 4585). +// +// FCI: +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | First | Number | PictureID | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class Sli : public RtcpPacket { + public: + Sli() : RtcpPacket() { + memset(&sli_, 0, sizeof(sli_)); + memset(&sli_item_, 0, sizeof(sli_item_)); + } + + virtual ~Sli() {} + + void From(uint32_t ssrc) { + sli_.SenderSSRC = ssrc; + } + void To(uint32_t ssrc) { + sli_.MediaSSRC = ssrc; + } + void WithFirstMb(uint16_t first_mb) { + assert(first_mb <= 0x1fff); + sli_item_.FirstMB = first_mb; + } + void WithNumberOfMb(uint16_t number_mb) { + assert(number_mb <= 0x1fff); + sli_item_.NumberOfMB = number_mb; + } + void WithPictureId(uint8_t picture_id) { + assert(picture_id <= 0x3f); + sli_item_.PictureId = picture_id; + } + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + size_t BlockLength() const { + const size_t kFciLength = 4; + return kCommonFbFmtLength + kFciLength; + } + + RTCPUtility::RTCPPacketPSFBSLI sli_; + RTCPUtility::RTCPPacketPSFBSLIItem sli_item_; + + DISALLOW_COPY_AND_ASSIGN(Sli); +}; + +// Generic NACK (RFC 4585). +// +// FCI: +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PID | BLP | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class Nack : public RtcpPacket { + public: + Nack() : RtcpPacket() { + memset(&nack_, 0, sizeof(nack_)); + } + + virtual ~Nack() {} + + void From(uint32_t ssrc) { + nack_.SenderSSRC = ssrc; + } + void To(uint32_t ssrc) { + nack_.MediaSSRC = ssrc; + } + void WithList(const uint16_t* nack_list, int length); + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + size_t BlockLength() const { + size_t fci_length = 4 * nack_fields_.size(); + return kCommonFbFmtLength + fci_length; + } + + RTCPUtility::RTCPPacketRTPFBNACK nack_; + std::vector<RTCPUtility::RTCPPacketRTPFBNACKItem> nack_fields_; + + DISALLOW_COPY_AND_ASSIGN(Nack); +}; + +// Reference picture selection indication (RPSI) (RFC 4585). +// +// FCI: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | PB |0| Payload Type| Native RPSI bit string | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | defined per codec ... | Padding (0) | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class Rpsi : public RtcpPacket { + public: + Rpsi() + : RtcpPacket(), + padding_bytes_(0) { + memset(&rpsi_, 0, sizeof(rpsi_)); + } + + virtual ~Rpsi() {} + + void From(uint32_t ssrc) { + rpsi_.SenderSSRC = ssrc; + } + void To(uint32_t ssrc) { + rpsi_.MediaSSRC = ssrc; + } + void WithPayloadType(uint8_t payload) { + assert(payload <= 0x7f); + rpsi_.PayloadType = payload; + } + void WithPictureId(uint64_t picture_id); + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + size_t BlockLength() const { + size_t fci_length = 2 + (rpsi_.NumberOfValidBits / 8) + padding_bytes_; + return kCommonFbFmtLength + fci_length; + } + + uint8_t padding_bytes_; + RTCPUtility::RTCPPacketPSFBRPSI rpsi_; + + DISALLOW_COPY_AND_ASSIGN(Rpsi); +}; + +// Full intra request (FIR) (RFC 5104). +// +// FCI: +// +// 0 1 2 3 +// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | Seq nr. | Reserved | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +class Fir : public RtcpPacket { + public: + Fir() + : RtcpPacket() { + memset(&fir_, 0, sizeof(fir_)); + memset(&fir_item_, 0, sizeof(fir_item_)); + } + + virtual ~Fir() {} + + void From(uint32_t ssrc) { + fir_.SenderSSRC = ssrc; + } + void To(uint32_t ssrc) { + fir_item_.SSRC = ssrc; + } + void WithCommandSeqNum(uint8_t seq_num) { + fir_item_.CommandSequenceNumber = seq_num; + } + + protected: + virtual void Create( + uint8_t* packet, size_t* length, size_t max_length) const OVERRIDE; + + private: + size_t BlockLength() const { + const size_t kFciLength = 8; + return kCommonFbFmtLength + kFciLength; + } + + RTCPUtility::RTCPPacketPSFBFIR fir_; + RTCPUtility::RTCPPacketPSFBFIRItem fir_item_; +}; + +// Class holding a RTCP packet. +// +// Takes a built rtcp packet. +// RawPacket raw_packet(buffer, length); +// +// To access the raw packet: +// raw_packet.buffer(); - pointer to the raw packet +// raw_packet.buffer_length(); - the length of the raw packet + +class RawPacket { + public: + RawPacket(const uint8_t* packet, size_t length) { + assert(length <= IP_PACKET_SIZE); + memcpy(buffer_, packet, length); + buffer_length_ = length; + } + + const uint8_t* buffer() { + return buffer_; + } + size_t buffer_length() const { + return buffer_length_; + } + + private: + size_t buffer_length_; + uint8_t buffer_[IP_PACKET_SIZE]; +}; + +} // namespace rtcp +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_ diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc new file mode 100644 index 00000000000..aa25c2e5f9d --- /dev/null +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc @@ -0,0 +1,592 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + * + * This file includes unit tests for the RtcpPacket. + */ + +#include "testing/gtest/include/gtest/gtest.h" + +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" +#include "webrtc/test/rtcp_packet_parser.h" + +using webrtc::rtcp::App; +using webrtc::rtcp::Bye; +using webrtc::rtcp::Empty; +using webrtc::rtcp::Fir; +using webrtc::rtcp::Ij; +using webrtc::rtcp::Nack; +using webrtc::rtcp::Pli; +using webrtc::rtcp::Sdes; +using webrtc::rtcp::SenderReport; +using webrtc::rtcp::Sli; +using webrtc::rtcp::RawPacket; +using webrtc::rtcp::ReceiverReport; +using webrtc::rtcp::ReportBlock; +using webrtc::rtcp::Rpsi; +using webrtc::test::RtcpPacketParser; + +namespace webrtc { + +const uint32_t kSenderSsrc = 0x12345678; +const uint32_t kRemoteSsrc = 0x23456789; + +TEST(RtcpPacketTest, Rr) { + ReceiverReport rr; + rr.From(kSenderSsrc); + + RawPacket packet = rr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); + EXPECT_EQ(0, parser.report_block()->num_packets()); +} + +TEST(RtcpPacketTest, RrWithOneReportBlock) { + ReportBlock rb; + rb.To(kRemoteSsrc); + rb.WithFractionLost(55); + rb.WithCumulativeLost(0x111111); + rb.WithExtHighestSeqNum(0x22222222); + rb.WithJitter(0x33333333); + rb.WithLastSr(0x44444444); + rb.WithDelayLastSr(0x55555555); + + ReceiverReport rr; + rr.From(kSenderSsrc); + rr.WithReportBlock(&rb); + + RawPacket packet = rr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); + EXPECT_EQ(1, parser.report_block()->num_packets()); + EXPECT_EQ(kRemoteSsrc, parser.report_block()->Ssrc()); + EXPECT_EQ(55U, parser.report_block()->FractionLost()); + EXPECT_EQ(0x111111U, parser.report_block()->CumPacketLost()); + EXPECT_EQ(0x22222222U, parser.report_block()->ExtHighestSeqNum()); + EXPECT_EQ(0x33333333U, parser.report_block()->Jitter()); + EXPECT_EQ(0x44444444U, parser.report_block()->LastSr()); + EXPECT_EQ(0x55555555U, parser.report_block()->DelayLastSr()); +} + +TEST(RtcpPacketTest, RrWithTwoReportBlocks) { + ReportBlock rb1; + rb1.To(kRemoteSsrc); + ReportBlock rb2; + rb2.To(kRemoteSsrc + 1); + + ReceiverReport rr; + rr.From(kSenderSsrc); + rr.WithReportBlock(&rb1); + rr.WithReportBlock(&rb2); + + RawPacket packet = rr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); + EXPECT_EQ(2, parser.report_block()->num_packets()); + EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc)); + EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc + 1)); +} + +TEST(RtcpPacketTest, Sr) { + SenderReport sr; + sr.From(kSenderSsrc); + sr.WithNtpSec(0x11111111); + sr.WithNtpFrac(0x22222222); + sr.WithRtpTimestamp(0x33333333); + sr.WithPacketCount(0x44444444); + sr.WithOctetCount(0x55555555); + + RawPacket packet = sr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + + EXPECT_EQ(1, parser.sender_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.sender_report()->Ssrc()); + EXPECT_EQ(0x11111111U, parser.sender_report()->NtpSec()); + EXPECT_EQ(0x22222222U, parser.sender_report()->NtpFrac()); + EXPECT_EQ(0x33333333U, parser.sender_report()->RtpTimestamp()); + EXPECT_EQ(0x44444444U, parser.sender_report()->PacketCount()); + EXPECT_EQ(0x55555555U, parser.sender_report()->OctetCount()); + EXPECT_EQ(0, parser.report_block()->num_packets()); +} + +TEST(RtcpPacketTest, SrWithOneReportBlock) { + ReportBlock rb; + rb.To(kRemoteSsrc); + + SenderReport sr; + sr.From(kSenderSsrc); + sr.WithReportBlock(&rb); + + RawPacket packet = sr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sender_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.sender_report()->Ssrc()); + EXPECT_EQ(1, parser.report_block()->num_packets()); + EXPECT_EQ(kRemoteSsrc, parser.report_block()->Ssrc()); +} + +TEST(RtcpPacketTest, SrWithTwoReportBlocks) { + ReportBlock rb1; + rb1.To(kRemoteSsrc); + ReportBlock rb2; + rb2.To(kRemoteSsrc + 1); + + SenderReport sr; + sr.From(kSenderSsrc); + sr.WithReportBlock(&rb1); + sr.WithReportBlock(&rb2); + + RawPacket packet = sr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sender_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.sender_report()->Ssrc()); + EXPECT_EQ(2, parser.report_block()->num_packets()); + EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc)); + EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc + 1)); +} + +TEST(RtcpPacketTest, IjNoItem) { + Ij ij; + + RawPacket packet = ij.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.ij()->num_packets()); + EXPECT_EQ(0, parser.ij_item()->num_packets()); +} + +TEST(RtcpPacketTest, IjOneItem) { + Ij ij; + ij.WithJitterItem(0x11111111); + + RawPacket packet = ij.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.ij()->num_packets()); + EXPECT_EQ(1, parser.ij_item()->num_packets()); + EXPECT_EQ(0x11111111U, parser.ij_item()->Jitter()); +} + +TEST(RtcpPacketTest, IjTwoItems) { + Ij ij; + ij.WithJitterItem(0x11111111); + ij.WithJitterItem(0x22222222); + + RawPacket packet = ij.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.ij()->num_packets()); + EXPECT_EQ(2, parser.ij_item()->num_packets()); + EXPECT_EQ(0x22222222U, parser.ij_item()->Jitter()); +} + +TEST(RtcpPacketTest, AppWithNoData) { + App app; + app.WithSubType(30); + uint32_t name = 'n' << 24; + name += 'a' << 16; + name += 'm' << 8; + name += 'e'; + app.WithName(name); + + RawPacket packet = app.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.app()->num_packets()); + EXPECT_EQ(30U, parser.app()->SubType()); + EXPECT_EQ(name, parser.app()->Name()); + EXPECT_EQ(0, parser.app_item()->num_packets()); +} + +TEST(RtcpPacketTest, App) { + App app; + app.From(kSenderSsrc); + app.WithSubType(30); + uint32_t name = 'n' << 24; + name += 'a' << 16; + name += 'm' << 8; + name += 'e'; + app.WithName(name); + const char kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'}; + const size_t kDataLength = sizeof(kData) / sizeof(kData[0]); + app.WithData((const uint8_t*)kData, kDataLength); + + RawPacket packet = app.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.app()->num_packets()); + EXPECT_EQ(30U, parser.app()->SubType()); + EXPECT_EQ(name, parser.app()->Name()); + EXPECT_EQ(1, parser.app_item()->num_packets()); + EXPECT_EQ(kDataLength, parser.app_item()->DataLength()); + EXPECT_EQ(0, strncmp(kData, (const char*)parser.app_item()->Data(), + parser.app_item()->DataLength())); +} + +TEST(RtcpPacketTest, SdesWithOneChunk) { + Sdes sdes; + sdes.WithCName(kSenderSsrc, "alice@host"); + + RawPacket packet = sdes.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sdes()->num_packets()); + EXPECT_EQ(1, parser.sdes_chunk()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.sdes_chunk()->Ssrc()); + EXPECT_EQ("alice@host", parser.sdes_chunk()->Cname()); +} + +TEST(RtcpPacketTest, SdesWithMultipleChunks) { + Sdes sdes; + sdes.WithCName(kSenderSsrc, "a"); + sdes.WithCName(kSenderSsrc + 1, "ab"); + sdes.WithCName(kSenderSsrc + 2, "abc"); + sdes.WithCName(kSenderSsrc + 3, "abcd"); + sdes.WithCName(kSenderSsrc + 4, "abcde"); + sdes.WithCName(kSenderSsrc + 5, "abcdef"); + + RawPacket packet = sdes.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sdes()->num_packets()); + EXPECT_EQ(6, parser.sdes_chunk()->num_packets()); + EXPECT_EQ(kSenderSsrc + 5, parser.sdes_chunk()->Ssrc()); + EXPECT_EQ("abcdef", parser.sdes_chunk()->Cname()); +} + +TEST(RtcpPacketTest, CnameItemWithEmptyString) { + Sdes sdes; + sdes.WithCName(kSenderSsrc, ""); + + RawPacket packet = sdes.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sdes()->num_packets()); + EXPECT_EQ(1, parser.sdes_chunk()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.sdes_chunk()->Ssrc()); + EXPECT_EQ("", parser.sdes_chunk()->Cname()); +} + +TEST(RtcpPacketTest, Pli) { + Pli pli; + pli.From(kSenderSsrc); + pli.To(kRemoteSsrc); + + RawPacket packet = pli.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.pli()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.pli()->Ssrc()); + EXPECT_EQ(kRemoteSsrc, parser.pli()->MediaSsrc()); +} + +TEST(RtcpPacketTest, Sli) { + const uint16_t kFirstMb = 7777; + const uint16_t kNumberOfMb = 6666; + const uint8_t kPictureId = 60; + Sli sli; + sli.From(kSenderSsrc); + sli.To(kRemoteSsrc); + sli.WithFirstMb(kFirstMb); + sli.WithNumberOfMb(kNumberOfMb); + sli.WithPictureId(kPictureId); + + RawPacket packet = sli.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sli()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.sli()->Ssrc()); + EXPECT_EQ(kRemoteSsrc, parser.sli()->MediaSsrc()); + EXPECT_EQ(1, parser.sli_item()->num_packets()); + EXPECT_EQ(kFirstMb, parser.sli_item()->FirstMb()); + EXPECT_EQ(kNumberOfMb, parser.sli_item()->NumberOfMb()); + EXPECT_EQ(kPictureId, parser.sli_item()->PictureId()); +} + +TEST(RtcpPacketTest, Nack) { + Nack nack; + const uint16_t kList[] = {0, 1, 3, 8, 16}; + const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]); + nack.From(kSenderSsrc); + nack.To(kRemoteSsrc); + nack.WithList(kList, kListLength); + RawPacket packet = nack.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.nack()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc()); + EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc()); + EXPECT_EQ(1, parser.nack_item()->num_packets()); + std::vector<uint16_t> seqs = parser.nack_item()->last_nack_list(); + EXPECT_EQ(kListLength, seqs.size()); + for (size_t i = 0; i < kListLength; ++i) { + EXPECT_EQ(kList[i], seqs[i]); + } +} + +TEST(RtcpPacketTest, NackWithWrap) { + Nack nack; + const uint16_t kList[] = {65500, 65516, 65534, 65535, 0, 1, 3, 20, 100}; + const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]); + nack.From(kSenderSsrc); + nack.To(kRemoteSsrc); + nack.WithList(kList, kListLength); + RawPacket packet = nack.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.nack()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc()); + EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc()); + EXPECT_EQ(4, parser.nack_item()->num_packets()); + std::vector<uint16_t> seqs = parser.nack_item()->last_nack_list(); + EXPECT_EQ(kListLength, seqs.size()); + for (size_t i = 0; i < kListLength; ++i) { + EXPECT_EQ(kList[i], seqs[i]); + } +} + +TEST(RtcpPacketTest, Rpsi) { + Rpsi rpsi; + // 1000001 (7 bits = 1 byte in native string). + const uint64_t kPictureId = 0x41; + const uint16_t kNumberOfValidBytes = 1; + rpsi.WithPayloadType(100); + rpsi.WithPictureId(kPictureId); + + RawPacket packet = rpsi.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(100, parser.rpsi()->PayloadType()); + EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); + EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); +} + +TEST(RtcpPacketTest, RpsiWithTwoByteNativeString) { + Rpsi rpsi; + // |1 0000001 (7 bits = 1 byte in native string). + const uint64_t kPictureId = 0x81; + const uint16_t kNumberOfValidBytes = 2; + rpsi.WithPictureId(kPictureId); + + RawPacket packet = rpsi.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); + EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); +} + +TEST(RtcpPacketTest, RpsiWithThreeByteNativeString) { + Rpsi rpsi; + // 10000|00 100000|0 1000000 (7 bits = 1 byte in native string). + const uint64_t kPictureId = 0x102040; + const uint16_t kNumberOfValidBytes = 3; + rpsi.WithPictureId(kPictureId); + + RawPacket packet = rpsi.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); + EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); +} + +TEST(RtcpPacketTest, RpsiWithFourByteNativeString) { + Rpsi rpsi; + // 1000|001 00001|01 100001|1 1000010 (7 bits = 1 byte in native string). + const uint64_t kPictureId = 0x84161C2; + const uint16_t kNumberOfValidBytes = 4; + rpsi.WithPictureId(kPictureId); + + RawPacket packet = rpsi.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); + EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); +} + +TEST(RtcpPacketTest, RpsiWithMaxPictureId) { + Rpsi rpsi; + // 1 1111111| 1111111 1|111111 11|11111 111|1111 1111|111 11111| + // 11 111111|1 1111111 (7 bits = 1 byte in native string). + const uint64_t kPictureId = 0xffffffffffffffff; + const uint16_t kNumberOfValidBytes = 10; + rpsi.WithPictureId(kPictureId); + + RawPacket packet = rpsi.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); + EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); +} + +TEST(RtcpPacketTest, Fir) { + Fir fir; + fir.From(kSenderSsrc); + fir.To(kRemoteSsrc); + fir.WithCommandSeqNum(123); + + RawPacket packet = fir.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.fir()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.fir()->Ssrc()); + EXPECT_EQ(1, parser.fir_item()->num_packets()); + EXPECT_EQ(kRemoteSsrc, parser.fir_item()->Ssrc()); + EXPECT_EQ(123U, parser.fir_item()->SeqNum()); +} + +TEST(RtcpPacketTest, AppendPacket) { + Fir fir; + ReportBlock rb; + ReceiverReport rr; + rr.From(kSenderSsrc); + rr.WithReportBlock(&rb); + rr.Append(&fir); + + RawPacket packet = rr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); + EXPECT_EQ(1, parser.report_block()->num_packets()); + EXPECT_EQ(1, parser.fir()->num_packets()); +} + +TEST(RtcpPacketTest, AppendPacketOnEmpty) { + Empty empty; + ReceiverReport rr; + rr.From(kSenderSsrc); + empty.Append(&rr); + + RawPacket packet = empty.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(0, parser.report_block()->num_packets()); +} + +TEST(RtcpPacketTest, AppendPacketWithOwnAppendedPacket) { + Fir fir; + Bye bye; + ReportBlock rb; + + ReceiverReport rr; + rr.WithReportBlock(&rb); + rr.Append(&fir); + + SenderReport sr; + sr.Append(&bye); + sr.Append(&rr); + + RawPacket packet = sr.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.sender_report()->num_packets()); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(1, parser.report_block()->num_packets()); + EXPECT_EQ(1, parser.bye()->num_packets()); + EXPECT_EQ(1, parser.fir()->num_packets()); +} + +TEST(RtcpPacketTest, Bye) { + Bye bye; + bye.From(kSenderSsrc); + + RawPacket packet = bye.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.bye()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc()); +} + +TEST(RtcpPacketTest, ByeWithCsrcs) { + Fir fir; + Bye bye; + bye.From(kSenderSsrc); + bye.WithCsrc(0x22222222); + bye.WithCsrc(0x33333333); + bye.Append(&fir); + + RawPacket packet = bye.Build(); + RtcpPacketParser parser; + parser.Parse(packet.buffer(), packet.buffer_length()); + EXPECT_EQ(1, parser.bye()->num_packets()); + EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc()); + EXPECT_EQ(1, parser.fir()->num_packets()); +} + +TEST(RtcpPacketTest, BuildWithInputBuffer) { + Fir fir; + ReportBlock rb; + ReceiverReport rr; + rr.From(kSenderSsrc); + rr.WithReportBlock(&rb); + rr.Append(&fir); + + const size_t kRrLength = 8; + const size_t kReportBlockLength = 24; + const size_t kFirLength = 20; + + size_t len = 0; + uint8_t packet[kRrLength + kReportBlockLength + kFirLength]; + rr.Build(packet, &len, kRrLength + kReportBlockLength + kFirLength); + + RtcpPacketParser parser; + parser.Parse(packet, len); + EXPECT_EQ(1, parser.receiver_report()->num_packets()); + EXPECT_EQ(1, parser.report_block()->num_packets()); + EXPECT_EQ(1, parser.fir()->num_packets()); +} + +TEST(RtcpPacketTest, BuildWithTooSmallBuffer) { + ReportBlock rb; + ReceiverReport rr; + rr.From(kSenderSsrc); + rr.WithReportBlock(&rb); + + const size_t kRrLength = 8; + const size_t kReportBlockLength = 24; + + // No packet. + size_t len = 0; + uint8_t packet[kRrLength + kReportBlockLength - 1]; + rr.Build(packet, &len, kRrLength + kReportBlockLength - 1); + RtcpPacketParser parser; + parser.Parse(packet, len); + EXPECT_EQ(0U, len); +} + +TEST(RtcpPacketTest, BuildWithTooSmallBuffer_LastBlockFits) { + Fir fir; + ReportBlock rb; + ReceiverReport rr; + rr.From(kSenderSsrc); + rr.WithReportBlock(&rb); + rr.Append(&fir); + + const size_t kRrLength = 8; + const size_t kReportBlockLength = 24; + + size_t len = 0; + uint8_t packet[kRrLength + kReportBlockLength - 1]; + rr.Build(packet, &len, kRrLength + kReportBlockLength - 1); + RtcpPacketParser parser; + parser.Parse(packet, len); + EXPECT_EQ(0, parser.receiver_report()->num_packets()); + EXPECT_EQ(0, parser.report_block()->num_packets()); + EXPECT_EQ(1, parser.fir()->num_packets()); +} +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc index a95fddede25..896bd5f4d3a 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -18,7 +18,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -57,7 +57,6 @@ RTCPReceiver::RTCPReceiver(const int32_t id, Clock* clock, _lastIncreasedSequenceNumberMs(0), stats_callback_(NULL) { memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo)); - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); } RTCPReceiver::~RTCPReceiver() { @@ -82,8 +81,6 @@ RTCPReceiver::~RTCPReceiver() { delete first->second; _receivedCnameMap.erase(first); } - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, - "%s deleted", __FUNCTION__); } void @@ -178,8 +175,7 @@ int32_t RTCPReceiver::ResetRTT(const uint32_t remoteSSRC) { RTCPReportBlockInformation* reportBlock = GetReportBlockInformation(remoteSSRC); if (reportBlock == NULL) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "\tfailed to GetReportBlockInformation(%u)", remoteSSRC); + LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC; return -1; } reportBlock->RTT = 0; @@ -282,22 +278,14 @@ bool RTCPReceiver::LastReceivedXrReferenceTimeInfo( return true; } -int32_t -RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const -{ - if(senderInfo == NULL) - { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); - return -1; - } - CriticalSectionScoped lock(_criticalSectionRTCPReceiver); - if(_lastReceivedSRNTPsecs == 0) - { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s No received SR", __FUNCTION__); - return -1; - } - memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo)); - return 0; +int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const { + assert(senderInfo); + CriticalSectionScoped lock(_criticalSectionRTCPReceiver); + if (_lastReceivedSRNTPsecs == 0) { + return -1; + } + memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo)); + return 0; } // statistics @@ -317,6 +305,12 @@ int32_t RTCPReceiver::StatisticsReceived( return 0; } +void RTCPReceiver::GetPacketTypeCounter( + RtcpPacketTypeCounter* packet_counter) const { + CriticalSectionScoped lock(_criticalSectionRTCPReceiver); + *packet_counter = packet_type_counter_; +} + int32_t RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation, RTCPUtility::RTCPParserV2* rtcpParser) @@ -480,11 +474,12 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser, } // no need for critsect we have _criticalSectionRTCPReceiver -void -RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket, - RTCPPacketInformation& rtcpPacketInformation, - const uint32_t remoteSSRC, - const uint8_t numberOfReportBlocks) { +void RTCPReceiver::HandleReportBlock( + const RTCPUtility::RTCPPacket& rtcpPacket, + RTCPPacketInformation& rtcpPacketInformation, + const uint32_t remoteSSRC, + const uint8_t numberOfReportBlocks) + EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver) { // This will be called once per report block in the RTCP packet. // We filter out all report blocks that are not for us. // Each packet has max 31 RR blocks. @@ -511,8 +506,8 @@ RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket, RTCPReportBlockInformation* reportBlock = CreateReportBlockInformation(remoteSSRC); if (reportBlock == NULL) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "\tfailed to CreateReportBlockInformation(%u)", remoteSSRC); + LOG(LS_WARNING) << "Failed to CreateReportBlockInformation(" + << remoteSSRC << ")"; return; } @@ -772,9 +767,6 @@ int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) { } RTCPReceiveInformation* receiveInfo = receiveInfoIt->second; if (receiveInfo == NULL) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s failed to get RTCPReceiveInformation", - __FUNCTION__); return -1; } if (receiveInfo->TmmbnBoundingSet.lengthOfSet() > 0) { @@ -838,6 +830,10 @@ RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser, HandleNACKItem(rtcpPacket, rtcpPacketInformation); pktType = rtcpParser.Iterate(); } + + if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) { + ++packet_type_counter_.nack_packets; + } } // no need for critsect we have _criticalSectionRTCPReceiver @@ -940,7 +936,8 @@ void RTCPReceiver::HandleXrDlrrReportBlock( void RTCPReceiver::HandleXrDlrrReportBlockItem( const RTCPUtility::RTCPPacket& packet, - RTCPPacketInformation& rtcpPacketInformation) { + RTCPPacketInformation& rtcpPacketInformation) + EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver) { if (registered_ssrcs_.find(packet.XRDLRRReportBlockItem.SSRC) == registered_ssrcs_.end()) { // Not to us. @@ -1026,6 +1023,7 @@ void RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser, if (main_ssrc_ == rtcpPacket.PLI.MediaSSRC) { TRACE_EVENT_INSTANT0("webrtc_rtp", "PLI"); + ++packet_type_counter_.pli_packets; // Received a signal that we need to send a new key frame. rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli; } @@ -1268,6 +1266,9 @@ void RTCPReceiver::HandleFIRItem(RTCPReceiveInformation* receiveInfo, if (main_ssrc_ != rtcpPacket.FIRItem.SSRC) { return; } + + ++packet_type_counter_.fir_packets; + // rtcpPacket.FIR.MediaSSRC SHOULD be 0 but we ignore to check it // we don't know who this originate from if (receiveInfo) { @@ -1332,8 +1333,7 @@ int32_t RTCPReceiver::UpdateTMMBR() { TMMBRSet* boundingSet = NULL; numBoundingSet = FindTMMBRBoundingSet(boundingSet); if (numBoundingSet == -1) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, - "Failed to find TMMBR bounding set."); + LOG(LS_WARNING) << "Failed to find TMMBR bounding set."; return -1; } // Set bounding set @@ -1353,8 +1353,6 @@ int32_t RTCPReceiver::UpdateTMMBR() { CriticalSectionScoped lock(_criticalSectionFeedbacks); if (_cbRtcpBandwidthObserver) { _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate * 1000); - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, - "Set TMMBR request:%d kbps", bitrate); } } return 0; @@ -1379,9 +1377,6 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket( // Process TMMBR and REMB first to avoid multiple callbacks // to OnNetworkChanged. if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr) { - WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, - "SIG [RTCP] Incoming TMMBR to id:%d", _id); - // Might trigger a OnReceivedBandwidthEstimateUpdate. UpdateTMMBR(); } @@ -1396,9 +1391,8 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket( } if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) { if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) { - WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, - "SIG [RTCP] Incoming NACK length:%d", - rtcpPacketInformation.nackSequenceNumbers.size()); + LOG(LS_VERBOSE) << "Incoming NACK length: " + << rtcpPacketInformation.nackSequenceNumbers.size(); _rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers); } } @@ -1413,13 +1407,11 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket( if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) || (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) { if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) { - WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, - "SIG [RTCP] Incoming PLI from SSRC:0x%x", - rtcpPacketInformation.remoteSSRC); + LOG(LS_VERBOSE) << "Incoming PLI from SSRC " + << rtcpPacketInformation.remoteSSRC; } else { - WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, - "SIG [RTCP] Incoming FIR from SSRC:0x%x", - rtcpPacketInformation.remoteSSRC); + LOG(LS_VERBOSE) << "Incoming FIR from SSRC " + << rtcpPacketInformation.remoteSSRC; } _cbRtcpIntraFrameObserver->OnReceivedIntraFrameRequest(local_ssrc); } @@ -1434,9 +1426,8 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket( } if (_cbRtcpBandwidthObserver) { if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) { - WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, - "SIG [RTCP] Incoming REMB:%d", - rtcpPacketInformation.receiverEstimatedMaxBitrate); + LOG(LS_VERBOSE) << "Incoming REMB: " + << rtcpPacketInformation.receiverEstimatedMaxBitrate; _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate( rtcpPacketInformation.receiverEstimatedMaxBitrate); } @@ -1532,9 +1523,6 @@ int32_t RTCPReceiver::TMMBRReceived(const uint32_t size, while (receiveInfoIt != _receivedInfoMap.end()) { RTCPReceiveInformation* receiveInfo = receiveInfoIt->second; if(receiveInfo == NULL) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s failed to get RTCPReceiveInformation", - __FUNCTION__); return -1; } num += receiveInfo->TmmbrSet.lengthOfSet(); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h index 637773dc744..ebffb7cfc9b 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h @@ -88,6 +88,8 @@ public: int32_t StatisticsReceived( std::vector<RTCPReportBlock>* receiveBlocks) const; + void GetPacketTypeCounter(RtcpPacketTypeCounter* packet_counter) const; + // Returns true if we haven't received an RTCP RR for several RTCP // intervals, but only triggers true once. bool RtcpRrTimeout(int64_t rtcp_interval_ms); @@ -266,6 +268,8 @@ protected: int64_t _lastIncreasedSequenceNumberMs; RtcpStatisticsCallback* stats_callback_; + + RtcpPacketTypeCounter packet_type_counter_; }; } // namespace webrtc #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_ diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h index 949beb9c7f0..0ca43fa53eb 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h @@ -12,10 +12,10 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_ +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" // RTCPReportBlock #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index b3f15bb9662..399c133cd10 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc @@ -261,6 +261,7 @@ class RtcpReceiverTest : public ::testing::Test { RemoteBitrateEstimatorFactory().Create( &remote_bitrate_observer_, &system_clock_, + kMimdControl, kRemoteBitrateEstimatorMinBitrateBps)) { test_transport_ = new TestTransport(); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index a407a42c913..d73de9c4243 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc @@ -19,7 +19,7 @@ #include "webrtc/common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -156,16 +156,11 @@ RTCPSender::RTCPSender(const int32_t id, xrSendReceiverReferenceTimeEnabled_(false), _xrSendVoIPMetric(false), - _xrVoIPMetric(), - _nackCount(0), - _pliCount(0), - _fullIntraRequestCount(0) + _xrVoIPMetric() { memset(_CNAME, 0, sizeof(_CNAME)); memset(_lastSendReport, 0, sizeof(_lastSendReport)); memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); - - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); } RTCPSender::~RTCPSender() { @@ -190,8 +185,6 @@ RTCPSender::~RTCPSender() { } delete _criticalSectionTransport; delete _criticalSectionRTCPSender; - - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__); } int32_t @@ -239,10 +232,7 @@ RTCPSender::Init() memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); last_xr_rr_.clear(); - _nackCount = 0; - _pliCount = 0; - _fullIntraRequestCount = 0; - + memset(&packet_type_counter_, 0, sizeof(packet_type_counter_)); return 0; } @@ -354,6 +344,9 @@ RTCPSender::SetREMBData(const uint32_t bitrate, _rembSSRC[i] = SSRC[i]; } _sendREMB = true; + // Send a REMB immediately if we have a new REMB. The frequency of REMBs is + // throttled by the caller. + _nextTimeToSendRTCP = _clock->TimeInMilliseconds(); return 0; } @@ -430,7 +423,8 @@ RTCPSender::SetCameraDelay(const int32_t delayMS) CriticalSectionScoped lock(_criticalSectionRTCPSender); if(delayMS > 1000 || delayMS < -1000) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__); + LOG(LS_WARNING) << "Delay can't be larger than 1 second: " + << delayMS << " ms"; return -1; } _cameraDelayMS = delayMS; @@ -489,14 +483,15 @@ RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const For audio we use a fix 5 sec interval For video we use 1 sec interval fo a BW smaller than 360 kbit/s, - technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare + technicaly we break the max 5% RTCP BW for video below 10 kbit/s but + that should be extremely rare From RFC 3550 MAX RTCP BW is 5% if the session BW A send report is approximately 65 bytes inc CNAME - A report report is approximately 28 bytes + A receiver report is approximately 28 bytes The RECOMMENDED value for the reduced minimum in seconds is 360 divided by the session bandwidth in kilobits/second. This minimum @@ -558,7 +553,7 @@ From RFC 3550 now += RTCP_SEND_BEFORE_KEY_FRAME_MS; } - if(now > _nextTimeToSendRTCP) + if(now >= _nextTimeToSendRTCP) { return true; @@ -616,6 +611,12 @@ bool RTCPSender::SendTimeOfXrRrReport(uint32_t mid_ntp, return true; } +void RTCPSender::GetPacketTypeCounter( + RtcpPacketTypeCounter* packet_counter) const { + CriticalSectionScoped lock(_criticalSectionRTCPSender); + *packet_counter = packet_type_counter_; +} + int32_t RTCPSender::AddExternalReportBlock( uint32_t SSRC, const RTCPReportBlock* reportBlock) { @@ -627,15 +628,10 @@ int32_t RTCPSender::AddReportBlock( uint32_t SSRC, std::map<uint32_t, RTCPReportBlock*>* report_blocks, const RTCPReportBlock* reportBlock) { - if (reportBlock == NULL) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s invalid argument", __FUNCTION__); - return -1; - } + assert(reportBlock); if (report_blocks->size() >= RTCP_MAX_REPORT_BLOCKS) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Too many report blocks."; return -1; } std::map<uint32_t, RTCPReportBlock*>::iterator it = @@ -673,7 +669,7 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state, // sanity if(pos + 52 >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Failed to build Sender Report."; return -2; } uint32_t RTPtime; @@ -756,8 +752,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { // sanity if(pos + 12 + lengthCname >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Failed to build SDEC."; return -2; } // SDEC Source Description @@ -909,7 +904,9 @@ RTCPSender::BuildExtendedJitterReport( { if (external_report_blocks_.size() > 0) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Not implemented."); + // TODO(andresp): Remove external report blocks since they are not + // supported. + LOG(LS_ERROR) << "Handling of external report blocks not implemented."; return 0; } @@ -1313,7 +1310,7 @@ RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos) // sanity if(pos + 12 + boundingSet->lengthOfSet()*8 >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Failed to build TMMBN."; return -2; } uint8_t FMT = 4; @@ -1380,12 +1377,12 @@ RTCPSender::BuildAPP(uint8_t* rtcpbuffer, int& pos) // sanity if(_appData == NULL) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__); + LOG(LS_WARNING) << "Failed to build app specific."; return -1; } if(pos + 12 + _appLength >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Failed to build app specific."; return -2; } rtcpbuffer[pos++]=(uint8_t)0x80 + _appSubType; @@ -1421,7 +1418,7 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer, // sanity if(pos + 16 >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Failed to build NACK."; return -2; } @@ -1474,8 +1471,7 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer, numOfNackFields++; } if (i != nackSize) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, - "Nack list to large for one packet."); + LOG(LS_WARNING) << "Nack list to large for one packet."; } rtcpbuffer[nackSizePos] = static_cast<uint8_t>(2 + numOfNackFields); *nackString = stringBuilder.GetResult(); @@ -1711,8 +1707,7 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state, CriticalSectionScoped lock(_criticalSectionRTCPSender); if(_method == kRtcpOff) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, - "%s invalid state", __FUNCTION__); + LOG(LS_WARNING) << "Can't send rtcp if it is disabled."; return -1; } } @@ -1778,10 +1773,9 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, rtcpPacketTypeFlags |= kRtcpTmmbn; _sendTMMBN = false; } - if (xrSendReceiverReferenceTimeEnabled_ && - (rtcpPacketTypeFlags & kRtcpReport)) + if (rtcpPacketTypeFlags & kRtcpReport) { - if (!_sending) + if (xrSendReceiverReferenceTimeEnabled_ && !_sending) { rtcpPacketTypeFlags |= kRtcpXrReceiverReferenceTime; } @@ -1920,8 +1914,9 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, return position; } TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::PLI"); - _pliCount++; - TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_PLICount", _SSRC, _pliCount); + ++packet_type_counter_.pli_packets; + TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_PLICount", _SSRC, + packet_type_counter_.pli_packets); } if(rtcpPacketTypeFlags & kRtcpFir) { @@ -1932,9 +1927,9 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, return position; } TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::FIR"); - _fullIntraRequestCount++; + ++packet_type_counter_.fir_packets; TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_FIRCount", _SSRC, - _fullIntraRequestCount); + packet_type_counter_.fir_packets); } if(rtcpPacketTypeFlags & kRtcpSli) { @@ -2017,8 +2012,9 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, } TRACE_EVENT_INSTANT1("webrtc_rtp", "RTCPSender::NACK", "nacks", TRACE_STR_COPY(nackString.c_str())); - _nackCount++; - TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_NACKCount", _SSRC, _nackCount); + ++packet_type_counter_.nack_packets; + TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_NACKCount", _SSRC, + packet_type_counter_.nack_packets); } if(rtcpPacketTypeFlags & kRtcpXrVoipMetric) { @@ -2065,7 +2061,7 @@ bool RTCPSender::PrepareReport(const FeedbackState& feedback_state, RTCPReportBlock* report_block, uint32_t* ntp_secs, uint32_t* ntp_frac) { // Do we have receive statistics to send? - StreamStatistician::Statistics stats; + RtcpStatistics stats; if (!statistician->GetStatistics(&stats, true)) return false; report_block->fractionLost = stats.fraction_lost; @@ -2123,13 +2119,7 @@ int32_t RTCPSender::SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize], const uint8_t arrLength) { - if(arrLength > kRtpCsrcSize) - { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); - assert(false); - return -1; - } - + assert(arrLength <= kRtpCsrcSize); CriticalSectionScoped lock(_criticalSectionRTCPSender); for(int i = 0; i < arrLength;i++) @@ -2148,7 +2138,7 @@ RTCPSender::SetApplicationSpecificData(const uint8_t subType, { if(length %4 != 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); + LOG(LS_ERROR) << "Failed to SetApplicationSpecificData."; return -1; } CriticalSectionScoped lock(_criticalSectionRTCPSender); @@ -2194,17 +2184,10 @@ int32_t RTCPSender::WriteAllReportBlocksToBuffer( uint8_t& numberOfReportBlocks, const uint32_t NTPsec, const uint32_t NTPfrac) { - // sanity one block - if(pos + 24 >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s invalid argument", __FUNCTION__); - return -1; - } numberOfReportBlocks = external_report_blocks_.size(); numberOfReportBlocks += internal_report_blocks_.size(); if ((pos + numberOfReportBlocks * 24) >= IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "%s invalid argument", __FUNCTION__); + LOG(LS_WARNING) << "Can't fit all report blocks."; return -1; } pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, internal_report_blocks_); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h index 9ed58244438..cbbc32aac6a 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h @@ -180,6 +180,8 @@ public: void SetTargetBitrate(unsigned int target_bitrate); + void GetPacketTypeCounter(RtcpPacketTypeCounter* packet_counter) const; + private: int32_t SendToNetwork(const uint8_t* dataBuffer, const uint16_t length); @@ -342,10 +344,7 @@ private: bool _xrSendVoIPMetric; RTCPVoIPMetric _xrVoIPMetric; - // Counters - uint32_t _nackCount; - uint32_t _pliCount; - uint32_t _fullIntraRequestCount; + RtcpPacketTypeCounter packet_type_counter_; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc index a8b5275fa20..dfb655c5167 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc @@ -278,12 +278,13 @@ class RtcpSenderTest : public ::testing::Test { : over_use_detector_options_(), clock_(1335900000), rtp_payload_registry_(new RTPPayloadRegistry( - 0, RTPPayloadStrategy::CreateStrategy(false))), + RTPPayloadStrategy::CreateStrategy(false))), remote_bitrate_observer_(), remote_bitrate_estimator_( RemoteBitrateEstimatorFactory().Create( &remote_bitrate_observer_, &clock_, + kMimdControl, kRemoteBitrateEstimatorMinBitrateBps)), receive_statistics_(ReceiveStatistics::Create(&clock_)) { test_transport_ = new TestTransport(); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc index 705a38b0161..9acab735e50 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc @@ -1266,31 +1266,27 @@ RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header) } } -bool -RTCPUtility::RTCPParserV2::ParseRPSIItem() -{ - // RFC 4585 6.3.3. Reference Picture Selection Indication (RPSI) - /* - 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | PB |0| Payload Type| Native RPSI bit string | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | defined per codec ... | Padding (0) | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - */ +bool RTCPUtility::RTCPParserV2::ParseRPSIItem() { + + // RFC 4585 6.3.3. Reference Picture Selection Indication (RPSI). + // + // 0 1 2 3 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // | PB |0| Payload Type| Native RPSI bit string | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // | defined per codec ... | Padding (0) | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData; - if (length < 4) - { + if (length < 4) { _state = State_TopLevel; EndCurrentBlock(); return false; } - if(length > 2+RTCP_RPSI_DATA_SIZE) - { + if (length > 2 + RTCP_RPSI_DATA_SIZE) { _state = State_TopLevel; EndCurrentBlock(); @@ -1299,12 +1295,14 @@ RTCPUtility::RTCPParserV2::ParseRPSIItem() _packetType = kRtcpPsfbRpsiCode; - uint8_t paddingBits = *_ptrRTCPData++; + uint8_t padding_bits = *_ptrRTCPData++; _packet.RPSI.PayloadType = *_ptrRTCPData++; - memcpy(_packet.RPSI.NativeBitString, _ptrRTCPData, length-2); + memcpy(_packet.RPSI.NativeBitString, _ptrRTCPData, length - 2); + _ptrRTCPData += length - 2; - _packet.RPSI.NumberOfValidBits = uint16_t(length-2)*8 - paddingBits; + _packet.RPSI.NumberOfValidBits = + static_cast<uint16_t>(length - 2) * 8 - padding_bits; return true; } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc index 904156e9445..fa847625ead 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc @@ -41,7 +41,7 @@ template <typename T> void ClearList(std::list<T*>* my_list) { class RtpFecTest : public ::testing::Test { protected: RtpFecTest() - : fec_(new ForwardErrorCorrection(0)), ssrc_(rand()), fec_seq_num_(0) {} + : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {} ForwardErrorCorrection* fec_; int ssrc_; @@ -86,43 +86,6 @@ class RtpFecTest : public ::testing::Test { void TearDown(); }; -// TODO(marpan): Consider adding table for input/output to simplify tests. - -TEST_F(RtpFecTest, HandleIncorrectInputs) { - int kNumImportantPackets = 0; - bool kUseUnequalProtection = false; - uint8_t kProtectionFactor = 60; - - // Media packet list is empty. - EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, - kNumImportantPackets, kUseUnequalProtection, - webrtc::kFecMaskBursty, &fec_packet_list_)); - - int num_media_packets = 10; - ConstructMediaPackets(num_media_packets); - - kNumImportantPackets = -1; - // Number of important packets below 0. - EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, - kNumImportantPackets, kUseUnequalProtection, - webrtc::kFecMaskBursty, &fec_packet_list_)); - - kNumImportantPackets = 12; - // Number of important packets greater than number of media packets. - EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, - kNumImportantPackets, kUseUnequalProtection, - webrtc::kFecMaskBursty, &fec_packet_list_)); - - num_media_packets = kMaxNumberMediaPackets + 1; - ConstructMediaPackets(num_media_packets); - - kNumImportantPackets = 0; - // Number of media packet is above maximum allowed (kMaxNumberMediaPackets). - EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, - kNumImportantPackets, kUseUnequalProtection, - webrtc::kFecMaskBursty, &fec_packet_list_)); -} - TEST_F(RtpFecTest, FecRecoveryNoLoss) { const int kNumImportantPackets = 0; const bool kUseUnequalProtection = false; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h index 650c0fad5e5..e4d3dc08362 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h @@ -28,8 +28,8 @@ #include <queue> #include <vector> +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h index 13eb0e101c7..e146492c72f 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h @@ -18,9 +18,9 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_ +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h index fcc7587c1fc..edffe8aecb6 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h @@ -22,7 +22,7 @@ const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; const size_t kRtpOneByteHeaderLength = 4; const size_t kTransmissionTimeOffsetLength = 4; -const size_t kAudioLevelLength = 2; +const size_t kAudioLevelLength = 4; const size_t kAbsoluteSendTimeLength = 4; struct HeaderExtension { @@ -37,11 +37,7 @@ struct HeaderExtension { length = kTransmissionTimeOffsetLength; break; case kRtpExtensionAudioLevel: - // TODO(solenberg): Because of how the audio level extension is handled - // in RTPSenderAudio::SendAudio(), we cannot set the actual length here - // but must leave it at zero. The consequence is that any other header - // extensions registered for an audio channel are effectively ignored. - // length = kAudioLevelLength; + length = kAudioLevelLength; break; case kRtpExtensionAbsoluteSendTime: length = kAbsoluteSendTimeLength; @@ -66,6 +62,8 @@ class RtpHeaderExtensionMap { int32_t Deregister(const RTPExtensionType type); + bool IsRegistered(RTPExtensionType type) const; + int32_t GetType(const uint8_t id, RTPExtensionType* type) const; int32_t GetId(const RTPExtensionType type, uint8_t* id) const; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc index d04872582b5..bb24d4dbfb7 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc @@ -13,7 +13,6 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { @@ -60,8 +59,6 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet, int length, const bool valid_rtpheader = rtp_parser.Parse(*header, &map); if (!valid_rtpheader) { - WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1, - "IncomingPacket invalid RTP header"); return false; } return true; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc index 14ca821f7e5..e3515f44543 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc @@ -18,7 +18,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -33,13 +33,21 @@ RTPPacketHistory::RTPPacketHistory(Clock* clock) } RTPPacketHistory::~RTPPacketHistory() { - Free(); + { + CriticalSectionScoped cs(critsect_); + Free(); + } delete critsect_; } -void RTPPacketHistory::SetStorePacketsStatus(bool enable, +void RTPPacketHistory::SetStorePacketsStatus(bool enable, uint16_t number_to_store) { + CriticalSectionScoped cs(critsect_); if (enable) { + if (store_) { + LOG(LS_WARNING) << "Purging packet history in order to re-set status."; + Free(); + } Allocate(number_to_store); } else { Free(); @@ -48,13 +56,7 @@ void RTPPacketHistory::SetStorePacketsStatus(bool enable, void RTPPacketHistory::Allocate(uint16_t number_to_store) { assert(number_to_store > 0); - CriticalSectionScoped cs(critsect_); - if (store_) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, - "SetStorePacketsStatus already set, number: %d", number_to_store); - return; - } - + assert(!store_); store_ = true; stored_packets_.resize(number_to_store); stored_seq_nums_.resize(number_to_store); @@ -65,13 +67,12 @@ void RTPPacketHistory::Allocate(uint16_t number_to_store) { } void RTPPacketHistory::Free() { - CriticalSectionScoped cs(critsect_); if (!store_) { return; } std::vector<std::vector<uint8_t> >::iterator it; - for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) { + for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) { it->clear(); } @@ -130,8 +131,8 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet, VerifyAndAllocatePacketLength(max_packet_length); if (packet_length > max_packet_length_) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, -1, - "Failed to store RTP packet, length: %d", packet_length); + LOG(LS_WARNING) << "Failed to store RTP packet with length: " + << packet_length; return -1; } @@ -156,46 +157,6 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet, return 0; } -int32_t RTPPacketHistory::ReplaceRTPHeader(const uint8_t* packet, - uint16_t sequence_number, - uint16_t rtp_header_length) { - CriticalSectionScoped cs(critsect_); - if (!store_) { - return 0; - } - - assert(packet); - assert(rtp_header_length > 3); - - if (rtp_header_length > max_packet_length_) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "Failed to replace RTP packet, length: %d", rtp_header_length); - return -1; - } - - int32_t index = 0; - bool found = FindSeqNum(sequence_number, &index); - if (!found) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "No match for getting seqNum %u", sequence_number); - return -1; - } - - uint16_t length = stored_lengths_.at(index); - if (length == 0 || length > max_packet_length_) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "No match for getting seqNum %u, len %d", sequence_number, length); - return -1; - } - assert(stored_seq_nums_[index] == sequence_number); - - // Update RTP header. - std::vector<std::vector<uint8_t> >::iterator it = - stored_packets_.begin() + index; - std::copy(packet, packet + rtp_header_length, it->begin()); - return 0; -} - bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const { CriticalSectionScoped cs(critsect_); if (!store_) { @@ -207,7 +168,7 @@ bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const { if (!found) { return false; } - + uint16_t length = stored_lengths_.at(index); if (length == 0 || length > max_packet_length_) { // Invalid length. @@ -222,6 +183,7 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number, uint8_t* packet, uint16_t* packet_length, int64_t* stored_time_ms) { + assert(*packet_length >= max_packet_length_); CriticalSectionScoped cs(critsect_); if (!store_) { return false; @@ -230,30 +192,22 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number, int32_t index = 0; bool found = FindSeqNum(sequence_number, &index); if (!found) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "No match for getting seqNum %u", sequence_number); + LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number; return false; } uint16_t length = stored_lengths_.at(index); - if (length == 0 || length > max_packet_length_) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "No match for getting seqNum %u, len %d", sequence_number, length); - return false; - } - - if (length > *packet_length) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, - "Input buffer too short for packet %u", sequence_number); + assert(length <= max_packet_length_); + if (length == 0) { + LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number + << ", len " << length; return false; } - // Verify elapsed time since last retrieve. + // Verify elapsed time since last retrieve. int64_t now = clock_->TimeInMilliseconds(); if (min_elapsed_time_ms > 0 && ((now - stored_send_times_.at(index)) < min_elapsed_time_ms)) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "Skip getting packet %u, packet recently resent.", sequence_number); return false; } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h index 785e4992b0a..190e5057bc9 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h @@ -18,6 +18,7 @@ #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { @@ -40,14 +41,6 @@ class RTPPacketHistory { int64_t capture_time_ms, StorageType type); - // Replaces the stored RTP packet with matching sequence number with the - // RTP header of the provided packet. - // Note: Calling this function assumes that the RTP header length should not - // have changed since the packet was stored. - int32_t ReplaceRTPHeader(const uint8_t* packet, - uint16_t sequence_number, - uint16_t rtp_header_length); - // Gets stored RTP packet corresponding to the input sequence number. // The packet is copied to the buffer pointed to by ptr_rtp_packet. // The rtp_packet_length should show the available buffer size. @@ -74,8 +67,8 @@ class RTPPacketHistory { private: void GetPacket(int index, uint8_t* packet, uint16_t* packet_length, int64_t* stored_time_ms) const; - void Allocate(uint16_t number_to_store); - void Free(); + void Allocate(uint16_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(*critsect_); + void Free() EXCLUSIVE_LOCKS_REQUIRED(*critsect_); void VerifyAndAllocatePacketLength(uint16_t packet_length); bool FindSeqNum(uint16_t sequence_number, int32_t* index) const; int FindBestFittingPacket(uint16_t size) const; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc index 1682b7c3387..7eb22ff69db 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc @@ -103,19 +103,6 @@ TEST_F(RtpPacketHistoryTest, PutRtpPacket_TooLargePacketLength) { kAllowRetransmission)); } -TEST_F(RtpPacketHistoryTest, GetRtpPacket_TooSmallBuffer) { - hist_->SetStorePacketsStatus(true, 10); - uint16_t len = 0; - int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); - CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len); - EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength, - capture_time_ms, kAllowRetransmission)); - uint16_t len_out = len - 1; - int64_t time; - EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_, - &len_out, &time)); -} - TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) { hist_->SetStorePacketsStatus(true, 10); uint16_t len = kMaxPacketLength; @@ -155,42 +142,6 @@ TEST_F(RtpPacketHistoryTest, GetRtpPacket) { } } -TEST_F(RtpPacketHistoryTest, ReplaceRtpHeader) { - hist_->SetStorePacketsStatus(true, 10); - - uint16_t len = 0; - int64_t capture_time_ms = 1; - CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len); - // Replace should fail, packet is not stored. - EXPECT_EQ(-1, hist_->ReplaceRTPHeader(packet_, kSeqNum, len)); - EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength, - capture_time_ms, kAllowRetransmission)); - - // Create modified packet and replace. - len = 0; - CreateRtpPacket(kSeqNum, kSsrc + 1, kPayload + 2, kTimestamp, packet_, &len); - EXPECT_EQ(0, hist_->ReplaceRTPHeader(packet_, kSeqNum, len)); - - uint16_t len_out = kMaxPacketLength; - int64_t time; - EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_out_, - &len_out, &time)); - EXPECT_EQ(len, len_out); - EXPECT_EQ(capture_time_ms, time); - for (int i = 0; i < len; i++) { - EXPECT_EQ(packet_[i], packet_out_[i]); - } - - // Replace should fail, too large length. - EXPECT_EQ(-1, hist_->ReplaceRTPHeader(packet_, kSeqNum, - kMaxPacketLength + 1)); - - // Replace should fail, packet is not stored. - len = 0; - CreateRtpPacket(kSeqNum + 1, kSsrc, kPayload, kTimestamp, packet_, &len); - EXPECT_EQ(-1, hist_->ReplaceRTPHeader(packet_, kSeqNum + 1, len)); -} - TEST_F(RtpPacketHistoryTest, NoCaptureTime) { hist_->SetStorePacketsStatus(true, 10); uint16_t len = 0; @@ -236,10 +187,10 @@ TEST_F(RtpPacketHistoryTest, MinResendTime) { capture_time_ms, kAllowRetransmission)); int64_t time; + len = kMaxPacketLength; EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 100, false, packet_, &len, &time)); fake_clock_.AdvanceTimeMilliseconds(100); - // Time has elapsed. len = kMaxPacketLength; EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 100, false, packet_, &len, diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc index 1c3b990c5de..db2e4cd31da 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -10,15 +10,13 @@ #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { RTPPayloadRegistry::RTPPayloadRegistry( - const int32_t id, RTPPayloadStrategy* rtp_payload_strategy) : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - id_(id), rtp_payload_strategy_(rtp_payload_strategy), red_payload_type_(-1), ulpfec_payload_type_(-1), @@ -60,9 +58,8 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( case 77: // 205 Transport layer FB message. case 78: // 206 Payload-specific FB message. case 79: // 207 Extended report. - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s invalid payloadtype:%d", - __FUNCTION__, payload_type); + LOG(LS_ERROR) << "Can't register invalid receiver payload type: " + << payload_type; return -1; default: break; @@ -94,9 +91,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( return 0; } } - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s invalid argument payload_type:%d already registered", - __FUNCTION__, payload_type); + LOG(LS_ERROR) << "Payload type already registered: " << payload_type; return -1; } @@ -138,14 +133,8 @@ int32_t RTPPayloadRegistry::DeRegisterReceivePayload( const int8_t payload_type) { CriticalSectionScoped cs(crit_sect_.get()); ModuleRTPUtility::PayloadTypeMap::iterator it = - payload_type_map_.find(payload_type); - - if (it == payload_type_map_.end()) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s failed to find payload_type:%d", - __FUNCTION__, payload_type); - return -1; - } + payload_type_map_.find(payload_type); + assert(it != payload_type_map_.end()); delete it->second; payload_type_map_.erase(it); return 0; @@ -194,11 +183,7 @@ int32_t RTPPayloadRegistry::ReceivePayloadType( const uint8_t channels, const uint32_t rate, int8_t* payload_type) const { - if (payload_type == NULL) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s invalid argument", __FUNCTION__); - return -1; - } + assert(payload_type); size_t payload_name_length = strlen(payload_name); CriticalSectionScoped cs(crit_sect_.get()); @@ -243,12 +228,6 @@ int32_t RTPPayloadRegistry::ReceivePayloadType( return -1; } -void RTPPayloadRegistry::SetRtxStatus(bool enable, uint32_t ssrc) { - CriticalSectionScoped cs(crit_sect_.get()); - rtx_ = enable; - ssrc_rtx_ = ssrc; -} - bool RTPPayloadRegistry::RtxEnabled() const { CriticalSectionScoped cs(crit_sect_.get()); return rtx_; @@ -296,17 +275,24 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet, (*restored_packet)[1] |= kRtpMarkerBitMask; // Marker bit is set. } } else { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "Incorrect RTX configuration, dropping packet."); + LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet."; return false; } } return true; } +void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) { + CriticalSectionScoped cs(crit_sect_.get()); + ssrc_rtx_ = ssrc; + rtx_ = true; +} + void RTPPayloadRegistry::SetRtxPayloadType(int payload_type) { CriticalSectionScoped cs(crit_sect_.get()); + assert(payload_type >= 0); payload_type_rtx_ = payload_type; + rtx_ = true; } bool RTPPayloadRegistry::IsRed(const RTPHeader& header) const { diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc index 96fa80ad842..c03ffcd1f3f 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc @@ -32,8 +32,7 @@ class RtpPayloadRegistryTest : public ::testing::Test { void SetUp() { // Note: the payload registry takes ownership of the strategy. mock_payload_strategy_ = new testing::NiceMock<MockRTPPayloadStrategy>(); - rtp_payload_registry_.reset( - new RTPPayloadRegistry(123, mock_payload_strategy_)); + rtp_payload_registry_.reset(new RTPPayloadRegistry(mock_payload_strategy_)); } protected: diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc index 134548518ea..c8104cc3731 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -15,7 +15,7 @@ #include <string.h> // memcpy() #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -277,11 +277,8 @@ int32_t RTPReceiverAudio::InvokeOnInitializeDecoder( specific_payload.Audio.frequency, specific_payload.Audio.channels, specific_payload.Audio.rate)) { - WEBRTC_TRACE(kTraceError, - kTraceRtpRtcp, - id, - "Failed to create video decoder for payload type:%d", - payload_type); + LOG(LS_ERROR) << "Failed to create decoder for payload type: " + << payload_name << "/" << payload_type; return -1; } return 0; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc index 9a276819ab8..d92618f2d57 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -18,7 +18,7 @@ #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -39,7 +39,7 @@ RtpReceiver* RtpReceiver::CreateVideoReceiver( return new RtpReceiverImpl( id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback, rtp_payload_registry, - RTPReceiverStrategy::CreateVideoStrategy(id, incoming_payload_callback)); + RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback)); } RtpReceiver* RtpReceiver::CreateAudioReceiver( @@ -87,8 +87,6 @@ RtpReceiverImpl::RtpReceiverImpl(int32_t id, assert(incoming_messages_callback); memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_)); - - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); } RtpReceiverImpl::~RtpReceiverImpl() { @@ -96,7 +94,6 @@ RtpReceiverImpl::~RtpReceiverImpl() { cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i], false); } - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__); } RTPReceiverStrategy* RtpReceiverImpl::GetMediaReceiver() const { @@ -127,9 +124,8 @@ int32_t RtpReceiverImpl::RegisterReceivePayload( if (created_new_payload) { if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type, frequency) != 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s failed to register payload", - __FUNCTION__); + LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/" + << payload_type; return -1; } } @@ -182,19 +178,12 @@ bool RtpReceiverImpl::IncomingRtpPacket( PayloadUnion payload_specific, bool in_order) { // Sanity check. - if (payload_length < 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s invalid argument", - __FUNCTION__); - return false; - } - int8_t first_payload_byte = 0; - if (payload_length > 0) { - first_payload_byte = payload[0]; - } + assert(payload_length >= 0); + // Trigger our callbacks. CheckSSRCChanged(rtp_header); + int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0; bool is_red = false; bool should_reset_statistics = false; @@ -205,14 +194,9 @@ bool RtpReceiverImpl::IncomingRtpPacket( &should_reset_statistics) == -1) { if (payload_length == 0) { // OK, keep-alive packet. - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "%s received keepalive", - __FUNCTION__); return true; } - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "%s received invalid payloadtype", - __FUNCTION__); + LOG(LS_WARNING) << "Receiving invalid payload type."; return false; } @@ -347,9 +331,8 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) { id_, rtp_header.payloadType, payload_name, rtp_header.payload_type_frequency, channels, rate)) { // New stream, same codec. - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "Failed to create decoder for payload type:%d", - rtp_header.payloadType); + LOG(LS_ERROR) << "Failed to create decoder for payload type: " + << rtp_header.payloadType; } } } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h index d8a22579621..09c9b6fc300 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -26,8 +26,7 @@ class TelephoneEventHandler; // This class is not thread-safe and must be protected by its caller. class RTPReceiverStrategy { public: - static RTPReceiverStrategy* CreateVideoStrategy(int32_t id, - RtpData* data_callback); + static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback); static RTPReceiverStrategy* CreateAudioStrategy( int32_t id, RtpData* data_callback, RtpAudioFeedback* incoming_messages_callback); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc index b733cdb4b05..5bb519f6220 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -17,19 +17,18 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy( - int32_t id, RtpData* data_callback) { - return new RTPReceiverVideo(id, data_callback); + RtpData* data_callback) { + return new RTPReceiverVideo(data_callback); } -RTPReceiverVideo::RTPReceiverVideo(int32_t id, RtpData* data_callback) - : RTPReceiverStrategy(data_callback), - id_(id) {} +RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback) + : RTPReceiverStrategy(data_callback) {} RTPReceiverVideo::~RTPReceiverVideo() { } @@ -93,11 +92,8 @@ int32_t RTPReceiverVideo::InvokeOnInitializeDecoder( // For video we just go with default values. if (-1 == callback->OnInitializeDecoder( id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) { - WEBRTC_TRACE(kTraceError, - kTraceRtpRtcp, - id, - "Failed to create video decoder for payload type:%d", - payload_type); + LOG(LS_ERROR) << "Failed to created decoder for payload type: " + << payload_type; return -1; } return 0; @@ -111,13 +107,6 @@ int32_t RTPReceiverVideo::ParseVideoCodecSpecific( RtpVideoCodecTypes video_type, int64_t now_ms, bool is_first_packet) { - WEBRTC_TRACE(kTraceStream, - kTraceRtpRtcp, - id_, - "%s(timestamp:%u)", - __FUNCTION__, - rtp_header->header.timestamp); - switch (rtp_header->type.Video.codec) { case kRtpVideoGeneric: rtp_header->type.Video.isFirstPacket = is_first_packet; @@ -170,13 +159,8 @@ int32_t RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_data_length) { ModuleRTPUtility::RTPPayload parsed_packet; - uint32_t id; - { - CriticalSectionScoped cs(crit_sect_.get()); - id = id_; - } ModuleRTPUtility::RTPPayloadParser rtp_payload_parser( - kRtpVideoVp8, payload_data, payload_data_length, id); + kRtpVideoVp8, payload_data, payload_data_length); if (!rtp_payload_parser.Parse(parsed_packet)) return -1; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h index ab69b40ee6f..4d81cb3972e 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h @@ -22,7 +22,7 @@ namespace webrtc { class RTPReceiverVideo : public RTPReceiverStrategy { public: - RTPReceiverVideo(const int32_t id, RtpData* data_callback); + RTPReceiverVideo(RtpData* data_callback); virtual ~RTPReceiverVideo(); @@ -80,8 +80,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy { RtpVideoCodecTypes video_type, int64_t now_ms, bool is_first_packet); - - int32_t id_; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi index 070845bc795..dcd65988058 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi @@ -20,6 +20,7 @@ # Common '../interface/fec_receiver.h', '../interface/receive_statistics.h', + '../interface/remote_ntp_time_estimator.h', '../interface/rtp_header_parser.h', '../interface/rtp_payload_registry.h', '../interface/rtp_receiver.h', @@ -32,10 +33,13 @@ 'fec_receiver_impl.h', 'receive_statistics_impl.cc', 'receive_statistics_impl.h', + 'remote_ntp_time_estimator.cc', 'rtp_header_parser.cc', 'rtp_rtcp_config.h', 'rtp_rtcp_impl.cc', 'rtp_rtcp_impl.h', + 'rtcp_packet.cc', + 'rtcp_packet.h', 'rtcp_receiver.cc', 'rtcp_receiver.h', 'rtcp_receiver_help.cc', diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index 89e9eb294f8..469a41e1d36 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -17,11 +17,6 @@ #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace.h" -#ifdef MATLAB -#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h" -extern MatlabEngine eng; // Global variable defined elsewhere. -#endif - #ifdef _WIN32 // Disable warning C4355: 'this' : used in base member initializer list. #pragma warning(disable : 4355) @@ -66,7 +61,9 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) configuration.outgoing_transport, configuration.audio_messages, configuration.paced_sender), - rtcp_sender_(configuration.id, configuration.audio, configuration.clock, + rtcp_sender_(configuration.id, + configuration.audio, + configuration.clock, configuration.receive_statistics), rtcp_receiver_(configuration.id, configuration.clock, this), clock_(configuration.clock), @@ -83,15 +80,13 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) CriticalSectionWrapper::CreateCriticalSection()), default_module_( static_cast<ModuleRtpRtcpImpl*>(configuration.default_module)), + padding_index_(-1), // Start padding at the first child module. nack_method_(kNackOff), nack_last_time_sent_full_(0), nack_last_seq_number_sent_(0), simulcast_(false), key_frame_req_method_(kKeyFrameReqFirRtp), remote_bitrate_(configuration.remote_bitrate_estimator), -#ifdef MATLAB - , plot1_(NULL), -#endif rtt_stats_(configuration.rtt_stats), critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()), rtt_ms_(0) { @@ -110,13 +105,9 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) uint32_t SSRC = rtp_sender_.SSRC(); rtcp_sender_.SetSSRC(SSRC); SetRtcpReceiverSsrcs(SSRC); - - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s created", __FUNCTION__); } ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() { - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__); - // All child modules MUST be deleted before deleting the default. assert(child_modules_.empty()); @@ -125,21 +116,9 @@ ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() { if (default_module_) { default_module_->DeRegisterChildModule(this); } -#ifdef MATLAB - if (plot1_) { - eng.DeletePlot(plot1_); - plot1_ = NULL; - } -#endif } void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "RegisterChildModule(module:0x%x)", - module); - CriticalSectionScoped lock( critical_section_module_ptrs_.get()); CriticalSectionScoped double_lock( @@ -153,17 +132,12 @@ void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) { } void ModuleRtpRtcpImpl::DeRegisterChildModule(RtpRtcp* remove_module) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "DeRegisterChildModule(module:0x%x)", remove_module); - CriticalSectionScoped lock( critical_section_module_ptrs_.get()); CriticalSectionScoped double_lock( critical_section_module_ptrs_feedback_.get()); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module == remove_module) { @@ -191,13 +165,7 @@ int32_t ModuleRtpRtcpImpl::Process() { last_bitrate_process_time_ = now; } - bool default_instance = false; - { - CriticalSectionScoped cs(critical_section_module_ptrs_.get()); - if (!child_modules_.empty()) - default_instance = true; - } - if (!default_instance) { + if (!IsDefaultModule()) { bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs; if (rtcp_sender_.Sending()) { // Process RTT if we have received a receiver report and we haven't @@ -269,16 +237,18 @@ int32_t ModuleRtpRtcpImpl::Process() { return 0; } -int32_t ModuleRtpRtcpImpl::SetRTXSendStatus(int mode, bool set_ssrc, - uint32_t ssrc) { - rtp_sender_.SetRTXStatus(mode, set_ssrc, ssrc); - return 0; +void ModuleRtpRtcpImpl::SetRTXSendStatus(int mode) { + rtp_sender_.SetRTXStatus(mode); } -int32_t ModuleRtpRtcpImpl::RTXSendStatus(int* mode, uint32_t* ssrc, - int* payload_type) const { +void ModuleRtpRtcpImpl::RTXSendStatus(int* mode, + uint32_t* ssrc, + int* payload_type) const { rtp_sender_.RTXStatus(mode, ssrc, payload_type); - return 0; +} + +void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) { + rtp_sender_.SetRtxSsrc(ssrc); } void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) { @@ -288,29 +258,12 @@ void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) { int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket( const uint8_t* rtcp_packet, const uint16_t length) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "IncomingRtcpPacket(packet_length:%u)", length); - // Minimum RTP is 12 bytes. - // Minimum RTCP is 8 bytes (RTCP BYE). - if (length == 8) { - WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1, - "IncomingRtcpPacket invalid length"); - return false; - } - // Check RTP version. - const uint8_t version = rtcp_packet[0] >> 6; - if (version != 2) { - WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1, - "IncomingRtcpPacket invalid RTP version"); - return false; - } // Allow receive of non-compound RTCP packets. RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true); const bool valid_rtcpheader = rtcp_parser.IsValid(); if (!valid_rtcpheader) { - WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, id_, - "IncomingRtcpPacket invalid RTCP packet"); + LOG(LS_WARNING) << "Incoming invalid RTCP packet"; return -1; } RTCPHelp::RTCPPacketInformation rtcp_packet_information; @@ -324,14 +277,6 @@ int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket( int32_t ModuleRtpRtcpImpl::RegisterSendPayload( const CodecInst& voice_codec) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "RegisterSendPayload(pl_name:%s pl_type:%d frequency:%u)", - voice_codec.plname, - voice_codec.pltype, - voice_codec.plfreq); - return rtp_sender_.RegisterPayload( voice_codec.plname, voice_codec.pltype, @@ -342,13 +287,6 @@ int32_t ModuleRtpRtcpImpl::RegisterSendPayload( int32_t ModuleRtpRtcpImpl::RegisterSendPayload( const VideoCodec& video_codec) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "RegisterSendPayload(pl_name:%s pl_type:%d)", - video_codec.plName, - video_codec.plType); - send_video_codec_ = video_codec; { // simulcast_ is accessed when accessing child_modules_, so this write needs @@ -365,11 +303,6 @@ int32_t ModuleRtpRtcpImpl::RegisterSendPayload( int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload( const int8_t payload_type) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "DeRegisterSendPayload(%d)", payload_type); - return rtp_sender_.DeRegisterSendPayload(payload_type); } @@ -378,58 +311,37 @@ int8_t ModuleRtpRtcpImpl::SendPayloadType() const { } uint32_t ModuleRtpRtcpImpl::StartTimestamp() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "StartTimestamp()"); - return rtp_sender_.StartTimestamp(); } // Configure start timestamp, default is a random number. int32_t ModuleRtpRtcpImpl::SetStartTimestamp( const uint32_t timestamp) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetStartTimestamp(%d)", - timestamp); rtcp_sender_.SetStartTimestamp(timestamp); rtp_sender_.SetStartTimestamp(timestamp, true); return 0; // TODO(pwestin): change to void. } uint16_t ModuleRtpRtcpImpl::SequenceNumber() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SequenceNumber()"); - return rtp_sender_.SequenceNumber(); } // Set SequenceNumber, default is a random number. int32_t ModuleRtpRtcpImpl::SetSequenceNumber( const uint16_t seq_num) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetSequenceNumber(%d)", - seq_num); - rtp_sender_.SetSequenceNumber(seq_num); return 0; // TODO(pwestin): change to void. } uint32_t ModuleRtpRtcpImpl::SSRC() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SSRC()"); - return rtp_sender_.SSRC(); } // Configure SSRC, default is a random number. -int32_t ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetSSRC(%d)", ssrc); - +void ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) { rtp_sender_.SetSSRC(ssrc); rtcp_sender_.SetSSRC(ssrc); SetRtcpReceiverSsrcs(ssrc); - - return 0; // TODO(pwestin): change to void. } int32_t ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) { @@ -440,27 +352,17 @@ int32_t ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) { int32_t ModuleRtpRtcpImpl::CSRCs( uint32_t arr_of_csrc[kRtpCsrcSize]) const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "CSRCs()"); - return rtp_sender_.CSRCs(arr_of_csrc); } int32_t ModuleRtpRtcpImpl::SetCSRCs( const uint32_t arr_of_csrc[kRtpCsrcSize], const uint8_t arr_length) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetCSRCs(arr_length:%d)", - arr_length); - - const bool default_instance(child_modules_.empty() ? false : true); - - if (default_instance) { + if (IsDefaultModule()) { // For default we need to update all child modules too. CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { @@ -469,10 +371,6 @@ int32_t ModuleRtpRtcpImpl::SetCSRCs( it++; } } else { - for (int i = 0; i < arr_length; ++i) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "\tidx:%d CSRC:%u", i, - arr_of_csrc[i]); - } rtcp_sender_.SetCSRCs(arr_of_csrc, arr_length); rtp_sender_.SetCSRCs(arr_of_csrc, arr_length); } @@ -480,38 +378,23 @@ int32_t ModuleRtpRtcpImpl::SetCSRCs( } uint32_t ModuleRtpRtcpImpl::PacketCountSent() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "PacketCountSent()"); - return rtp_sender_.Packets(); } uint32_t ModuleRtpRtcpImpl::ByteCountSent() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ByteCountSent()"); - return rtp_sender_.Bytes(); } int ModuleRtpRtcpImpl::CurrentSendFrequencyHz() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "CurrentSendFrequencyHz()"); - return rtp_sender_.SendPayloadFrequency(); } int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { - if (sending) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetSendingStatus(sending)"); - } else { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetSendingStatus(stopped)"); - } if (rtcp_sender_.Sending() != sending) { // Sends RTCP BYE when going from true to false RTCPSender::FeedbackState feedback_state(this); if (rtcp_sender_.SetSendingStatus(feedback_state, sending) != 0) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "Failed to send RTCP BYE"); + LOG(LS_WARNING) << "Failed to send RTCP BYE"; } collision_detected_ = false; @@ -536,33 +419,21 @@ int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { } bool ModuleRtpRtcpImpl::Sending() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "Sending()"); - return rtcp_sender_.Sending(); } int32_t ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { - if (sending) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetSendingMediaStatus(sending)"); - } else { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetSendingMediaStatus(stopped)"); - } rtp_sender_.SetSendingMediaStatus(sending); return 0; } bool ModuleRtpRtcpImpl::SendingMedia() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "Sending()"); - - const bool have_child_modules(child_modules_.empty() ? false : true); - if (!have_child_modules) { + if (!IsDefaultModule()) { return rtp_sender_.SendingMedia(); } CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::const_iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::const_iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RTPSender& rtp_sender = (*it)->rtp_sender_; if (rtp_sender.SendingMedia()) { @@ -582,17 +453,9 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData( uint32_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_hdr) { - WEBRTC_TRACE( - kTraceStream, - kTraceRtpRtcp, - id_, - "SendOutgoingData(frame_type:%d payload_type:%d time_stamp:%u size:%u)", - frame_type, payload_type, time_stamp, payload_size); - rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms); - const bool have_child_modules(child_modules_.empty() ? false : true); - if (!have_child_modules) { + if (!IsDefaultModule()) { // Don't send RTCP from default module. if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) { RTCPSender::FeedbackState feedback_state(this); @@ -615,7 +478,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData( return -1; } int idx = 0; - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); for (; idx < rtp_video_hdr->simulcastIdx; ++it) { if (it == child_modules_.end()) { return -1; @@ -633,11 +496,6 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData( if (it == child_modules_.end()) { return -1; } - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SendOutgoingData(SimulcastIdx:%u size:%u, ssrc:0x%x)", - idx, payload_size, (*it)->rtp_sender_.SSRC()); return (*it)->SendOutgoingData(frame_type, payload_type, time_stamp, @@ -647,7 +505,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData( fragmentation, rtp_video_hdr); } else { - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); // Send to all "child" modules while (it != child_modules_.end()) { if ((*it)->SendingMedia()) { @@ -670,19 +528,7 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission) { - WEBRTC_TRACE( - kTraceStream, - kTraceRtpRtcp, - id_, - "TimeToSendPacket(ssrc:0x%x sequence_number:%u capture_time_ms:%ll)", - ssrc, sequence_number, capture_time_ms); - - bool no_child_modules = false; - { - CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - no_child_modules = child_modules_.empty(); - } - if (no_child_modules) { + if (!IsDefaultModule()) { // Don't send from default module. if (SendingMedia() && ssrc == rtp_sender_.SSRC()) { return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms, @@ -690,7 +536,7 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, } } else { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); while (it != child_modules_.end()) { if ((*it)->SendingMedia() && ssrc == (*it)->rtp_sender_.SSRC()) { return (*it)->rtp_sender_.TimeToSendPacket(sequence_number, @@ -705,28 +551,18 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, } int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, "TimeToSendPadding(bytes: %d)", - bytes); - - bool no_child_modules = false; - { - CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - no_child_modules = child_modules_.empty(); - } - if (no_child_modules) { + if (!IsDefaultModule()) { // Don't send from default module. if (SendingMedia()) { return rtp_sender_.TimeToSendPadding(bytes); } } else { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); - while (it != child_modules_.end()) { + for (size_t i = 0; i < child_modules_.size(); ++i) { // Send padding on one of the modules sending media. - if ((*it)->SendingMedia()) { - return (*it)->rtp_sender_.TimeToSendPadding(bytes); + if (child_modules_[i]->SendingMedia()) { + return child_modules_[i]->rtp_sender_.TimeToSendPadding(bytes); } - ++it; } } return 0; @@ -737,7 +573,7 @@ bool ModuleRtpRtcpImpl::GetSendSideDelay(int* avg_send_delay_ms, assert(avg_send_delay_ms); assert(max_send_delay_ms); - if (!child_modules_.empty()) { + if (IsDefaultModule()) { // This API is only supported for child modules. return false; } @@ -745,26 +581,17 @@ bool ModuleRtpRtcpImpl::GetSendSideDelay(int* avg_send_delay_ms, } uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "MaxPayloadLength()"); - return rtp_sender_.MaxPayloadLength(); } uint16_t ModuleRtpRtcpImpl::MaxDataPayloadLength() const { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "MaxDataPayloadLength()"); - // Assuming IP/UDP. uint16_t min_data_payload_length = IP_PACKET_SIZE - 28; - const bool default_instance(child_modules_.empty() ? false : true); - if (default_instance) { + if (IsDefaultModule()) { // For default we need to update all child modules too. CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::const_iterator it = - child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::const_iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { @@ -789,13 +616,6 @@ int32_t ModuleRtpRtcpImpl::SetTransportOverhead( const bool tcp, const bool ipv6, const uint8_t authentication_overhead) { - WEBRTC_TRACE( - kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetTransportOverhead(TCP:%d, IPV6:%d authentication_overhead:%u)", - tcp, ipv6, authentication_overhead); - uint16_t packet_overhead = 0; if (ipv6) { packet_overhead = 40; @@ -827,12 +647,8 @@ int32_t ModuleRtpRtcpImpl::SetTransportOverhead( } int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetMaxTransferUnit(%u)", - mtu); - if (mtu > IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "Invalid in argument to SetMaxTransferUnit(%u)", mtu); + LOG(LS_ERROR) << "Invalid mtu: " << mtu; return -1; } return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_, @@ -840,8 +656,6 @@ int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) { } RTCPMethod ModuleRtpRtcpImpl::RTCP() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RTCP()"); - if (rtcp_sender_.Status() != kRtcpOff) { return rtcp_receiver_.Status(); } @@ -850,9 +664,6 @@ RTCPMethod ModuleRtpRtcpImpl::RTCP() const { // Configure RTCP status i.e on/off. int32_t ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPStatus(%d)", - method); - if (rtcp_sender_.SetRTCPStatus(method) == 0) { return rtcp_receiver_.SetRTCPStatus(method); } @@ -866,36 +677,26 @@ uint32_t ModuleRtpRtcpImpl::LastSendReport( } int32_t ModuleRtpRtcpImpl::SetCNAME(const char c_name[RTCP_CNAME_SIZE]) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetCNAME(%s)", c_name); return rtcp_sender_.SetCNAME(c_name); } int32_t ModuleRtpRtcpImpl::CNAME(char c_name[RTCP_CNAME_SIZE]) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "CNAME()"); return rtcp_sender_.CNAME(c_name); } int32_t ModuleRtpRtcpImpl::AddMixedCNAME( const uint32_t ssrc, const char c_name[RTCP_CNAME_SIZE]) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "AddMixedCNAME(SSRC:%u)", ssrc); - return rtcp_sender_.AddMixedCNAME(ssrc, c_name); } int32_t ModuleRtpRtcpImpl::RemoveMixedCNAME(const uint32_t ssrc) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "RemoveMixedCNAME(SSRC:%u)", ssrc); return rtcp_sender_.RemoveMixedCNAME(ssrc); } int32_t ModuleRtpRtcpImpl::RemoteCNAME( const uint32_t remote_ssrc, char c_name[RTCP_CNAME_SIZE]) const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "RemoteCNAME(SSRC:%u)", remote_ssrc); - return rtcp_receiver_.CNAME(remote_ssrc, c_name); } @@ -905,8 +706,6 @@ int32_t ModuleRtpRtcpImpl::RemoteNTP( uint32_t* rtcp_arrival_time_secs, uint32_t* rtcp_arrival_time_frac, uint32_t* rtcp_timestamp) const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteNTP()"); - return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac, rtcp_arrival_time_secs, @@ -920,24 +719,21 @@ int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc, uint16_t* avg_rtt, uint16_t* min_rtt, uint16_t* max_rtt) const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RTT()"); - - return rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt); + int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt); + if (rtt && *rtt == 0) { + // Try to get RTT from RtcpRttStats class. + *rtt = static_cast<uint16_t>(rtt_ms()); + } + return ret; } // Reset RoundTripTime statistics. int32_t ModuleRtpRtcpImpl::ResetRTT(const uint32_t remote_ssrc) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetRTT(SSRC:%u)", - remote_ssrc); - return rtcp_receiver_.ResetRTT(remote_ssrc); } // Reset RTP data counters for the sending side. int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "ResetSendDataCountersRTP()"); - rtp_sender_.ResetDataCounters(); return 0; // TODO(pwestin): change to void. } @@ -945,8 +741,6 @@ int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() { // Force a send of an RTCP packet. // Normal SR and RR are triggered via the process function. int32_t ModuleRtpRtcpImpl::SendRTCP(uint32_t rtcp_packet_type) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendRTCP(0x%x)", - rtcp_packet_type); RTCPSender::FeedbackState feedback_state(this); return rtcp_sender_.SendRTCP(feedback_state, rtcp_packet_type); } @@ -956,24 +750,16 @@ int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData( const uint32_t name, const uint8_t* data, const uint16_t length) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetRTCPApplicationSpecificData(sub_type:%d name:0x%x)", - sub_type, name); - return rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length); } // (XR) VOIP metric. int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics( const RTCPVoIPMetric* voip_metric) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPVoIPMetrics()"); - return rtcp_sender_.SetRTCPVoIPMetrics(voip_metric); } void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetRtcpXrRrtrStatus(%s)", enable ? "true" : "false"); return rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable); } @@ -984,8 +770,6 @@ bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const { int32_t ModuleRtpRtcpImpl::DataCountersRTP( uint32_t* bytes_sent, uint32_t* packets_sent) const { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, "DataCountersRTP()"); - if (bytes_sent) { *bytes_sent = rtp_sender_.Bytes(); } @@ -996,77 +780,54 @@ int32_t ModuleRtpRtcpImpl::DataCountersRTP( } int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteRTCPStat()"); - return rtcp_receiver_.SenderInfoReceived(sender_info); } // Received RTCP report. int32_t ModuleRtpRtcpImpl::RemoteRTCPStat( std::vector<RTCPReportBlock>* receive_blocks) const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteRTCPStat()"); - return rtcp_receiver_.StatisticsReceived(receive_blocks); } int32_t ModuleRtpRtcpImpl::AddRTCPReportBlock( const uint32_t ssrc, const RTCPReportBlock* report_block) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "AddRTCPReportBlock()"); - return rtcp_sender_.AddExternalReportBlock(ssrc, report_block); } int32_t ModuleRtpRtcpImpl::RemoveRTCPReportBlock( const uint32_t ssrc) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoveRTCPReportBlock()"); - return rtcp_sender_.RemoveExternalReportBlock(ssrc); } +void ModuleRtpRtcpImpl::GetRtcpPacketTypeCounters( + RtcpPacketTypeCounter* packets_sent, + RtcpPacketTypeCounter* packets_received) const { + rtcp_sender_.GetPacketTypeCounter(packets_sent); + rtcp_receiver_.GetPacketTypeCounter(packets_received); +} + // (REMB) Receiver Estimated Max Bitrate. bool ModuleRtpRtcpImpl::REMB() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "REMB()"); - return rtcp_sender_.REMB(); } int32_t ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) { - if (enable) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetREMBStatus(enable)"); - } else { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetREMBStatus(disable)"); - } return rtcp_sender_.SetREMBStatus(enable); } int32_t ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate, const uint8_t number_of_ssrc, const uint32_t* ssrc) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetREMBData(bitrate:%d,?,?)", bitrate); return rtcp_sender_.SetREMBData(bitrate, number_of_ssrc, ssrc); } // (IJ) Extended jitter report. bool ModuleRtpRtcpImpl::IJ() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "IJ()"); - return rtcp_sender_.IJ(); } int32_t ModuleRtpRtcpImpl::SetIJStatus(const bool enable) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetIJStatus(%s)", enable ? "true" : "false"); - return rtcp_sender_.SetIJStatus(enable); } @@ -1083,25 +844,14 @@ int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension( // (TMMBR) Temporary Max Media Bit Rate. bool ModuleRtpRtcpImpl::TMMBR() const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "TMMBR()"); - return rtcp_sender_.TMMBR(); } int32_t ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) { - if (enable) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetTMMBRStatus(enable)"); - } else { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetTMMBRStatus(disable)"); - } return rtcp_sender_.SetTMMBRStatus(enable); } int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetTMMBN()"); - uint32_t max_bitrate_kbit = rtp_sender_.MaxConfiguredBitrateVideo() / 1000; return rtcp_sender_.SetTMMBN(bounding_set, max_bitrate_kbit); @@ -1109,32 +859,18 @@ int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) { // Returns the currently configured retransmission mode. int ModuleRtpRtcpImpl::SelectiveRetransmissions() const { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SelectiveRetransmissions()"); return rtp_sender_.SelectiveRetransmissions(); } // Enable or disable a retransmission mode, which decides which packets will // be retransmitted if NACKed. int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetSelectiveRetransmissions(%u)", - settings); return rtp_sender_.SetSelectiveRetransmissions(settings); } // Send a Negative acknowledgment packet. int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list, const uint16_t size) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SendNACK(size:%u)", size); - // Use RTT from RtcpRttStats class if provided. uint16_t rtt = rtt_ms(); if (rtt == 0) { @@ -1188,14 +924,6 @@ int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list, int32_t ModuleRtpRtcpImpl::SetStorePacketsStatus( const bool enable, const uint16_t number_to_store) { - if (enable) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetStorePacketsStatus(enable, number_to_store:%d)", - number_to_store); - } else { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetStorePacketsStatus(disable)"); - } rtp_sender_.SetStorePacketsStatus(enable, number_to_store); return 0; // TODO(pwestin): change to void. } @@ -1219,21 +947,11 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband( const uint8_t key, const uint16_t time_ms, const uint8_t level) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SendTelephoneEventOutband(key:%u, time_ms:%u, level:%u)", key, - time_ms, level); - return rtp_sender_.SendTelephoneEvent(key, time_ms, level); } bool ModuleRtpRtcpImpl::SendTelephoneEventActive( int8_t& telephone_event) const { - - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SendTelephoneEventActive()"); - return rtp_sender_.SendTelephoneEventActive(&telephone_event); } @@ -1241,68 +959,23 @@ bool ModuleRtpRtcpImpl::SendTelephoneEventActive( // packet in silence (CNG). int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( const uint16_t packet_size_samples) { - - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetAudioPacketSize(%u)", - packet_size_samples); - return rtp_sender_.SetAudioPacketSize(packet_size_samples); } -int32_t ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus( - const bool enable, - const uint8_t id) { - - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetRTPAudioLevelIndicationStatus(enable=%d, ID=%u)", - enable, - id); - - return rtp_sender_.SetAudioLevelIndicationStatus(enable, id); -} - -int32_t ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus( - bool& enable, - uint8_t& id) const { - - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "GetRTPAudioLevelIndicationStatus()"); - return rtp_sender_.AudioLevelIndicationStatus(&enable, &id); -} - int32_t ModuleRtpRtcpImpl::SetAudioLevel( const uint8_t level_d_bov) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetAudioLevel(level_d_bov:%u)", - level_d_bov); return rtp_sender_.SetAudioLevel(level_d_bov); } // Set payload type for Redundant Audio Data RFC 2198. int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType( const int8_t payload_type) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetSendREDPayloadType(%d)", - payload_type); - return rtp_sender_.SetRED(payload_type); } // Get payload type for Redundant Audio Data RFC 2198. int32_t ModuleRtpRtcpImpl::SendREDPayloadType( int8_t& payload_type) const { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendREDPayloadType()"); - return rtp_sender_.RED(&payload_type); } @@ -1312,54 +985,41 @@ RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const { void ModuleRtpRtcpImpl::SetTargetSendBitrate( const std::vector<uint32_t>& stream_bitrates) { - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, - "SetTargetSendBitrate: %ld streams", stream_bitrates.size()); - - const bool have_child_modules(child_modules_.empty() ? false : true); - if (have_child_modules) { + if (IsDefaultModule()) { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); if (simulcast_) { - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); for (size_t i = 0; it != child_modules_.end() && i < stream_bitrates.size(); ++it) { if ((*it)->SendingMedia()) { RTPSender& rtp_sender = (*it)->rtp_sender_; - rtp_sender.SetTargetSendBitrate(stream_bitrates[i]); + rtp_sender.SetTargetBitrate(stream_bitrates[i]); ++i; } } } else { - assert(stream_bitrates.size() == 1); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + if (stream_bitrates.size() > 1) + return; + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); for (; it != child_modules_.end(); ++it) { RTPSender& rtp_sender = (*it)->rtp_sender_; - rtp_sender.SetTargetSendBitrate(stream_bitrates[0]); + rtp_sender.SetTargetBitrate(stream_bitrates[0]); } } } else { - assert(stream_bitrates.size() == 1); - rtp_sender_.SetTargetSendBitrate(stream_bitrates[0]); + if (stream_bitrates.size() > 1) + return; + rtp_sender_.SetTargetBitrate(stream_bitrates[0]); } } int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( const KeyFrameRequestMethod method) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetKeyFrameRequestMethod(method:%u)", - method); - key_frame_req_method_ = method; return 0; } int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "RequestKeyFrame"); - switch (key_frame_req_method_) { case kKeyFrameReqFirRtp: return rtp_sender_.SendRTPIntraRequest(); @@ -1373,29 +1033,15 @@ int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication( const uint8_t picture_id) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SendRTCPSliceLossIndication (picture_id:%d)", - picture_id); - RTCPSender::FeedbackState feedback_state(this); return rtcp_sender_.SendRTCP( feedback_state, kRtcpSli, 0, 0, false, picture_id); } int32_t ModuleRtpRtcpImpl::SetCameraDelay(const int32_t delay_ms) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetCameraDelay(%d)", - delay_ms); - const bool default_instance(child_modules_.empty() ? false : true); - - if (default_instance) { + if (IsDefaultModule()) { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { @@ -1412,18 +1058,6 @@ int32_t ModuleRtpRtcpImpl::SetGenericFECStatus( const bool enable, const uint8_t payload_type_red, const uint8_t payload_type_fec) { - if (enable) { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetGenericFECStatus(enable, %u)", - payload_type_red); - } else { - WEBRTC_TRACE(kTraceModuleCall, - kTraceRtpRtcp, - id_, - "SetGenericFECStatus(disable)"); - } return rtp_sender_.SetGenericFECStatus(enable, payload_type_red, payload_type_fec); @@ -1433,15 +1067,11 @@ int32_t ModuleRtpRtcpImpl::GenericFECStatus( bool& enable, uint8_t& payload_type_red, uint8_t& payload_type_fec) { - - WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "GenericFECStatus()"); - bool child_enabled = false; - const bool default_instance(child_modules_.empty() ? false : true); - if (default_instance) { + if (IsDefaultModule()) { // For default we need to check all child modules too. CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { @@ -1471,12 +1101,11 @@ int32_t ModuleRtpRtcpImpl::GenericFECStatus( int32_t ModuleRtpRtcpImpl::SetFecParameters( const FecProtectionParams* delta_params, const FecProtectionParams* key_params) { - const bool default_instance(child_modules_.empty() ? false : true); - if (default_instance) { + if (IsDefaultModule()) { // For default we need to update all child modules too. CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { @@ -1517,9 +1146,7 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, uint32_t* video_rate, uint32_t* fec_rate, uint32_t* nack_rate) const { - const bool default_instance(child_modules_.empty() ? false : true); - - if (default_instance) { + if (IsDefaultModule()) { // For default we need to update the send bitrate. CriticalSectionScoped lock(critical_section_module_ptrs_feedback_.get()); @@ -1532,8 +1159,7 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, if (nack_rate != NULL) *nack_rate = 0; - std::list<ModuleRtpRtcpImpl*>::const_iterator it = - child_modules_.begin(); + std::vector<ModuleRtpRtcpImpl*>::const_iterator it = child_modules_.begin(); while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { @@ -1570,21 +1196,7 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, void ModuleRtpRtcpImpl::RegisterVideoBitrateObserver( BitrateStatisticsObserver* observer) { - { - CriticalSectionScoped cs(critical_section_module_ptrs_.get()); - if (!child_modules_.empty()) { - for (std::list<ModuleRtpRtcpImpl*>::const_iterator it = - child_modules_.begin(); - it != child_modules_.end(); - ++it) { - RtpRtcp* module = *it; - if (module) - module->RegisterVideoBitrateObserver(observer); - } - return; - } - } - + assert(!IsDefaultModule()); rtp_sender_.RegisterBitrateObserver(observer); } @@ -1592,7 +1204,6 @@ BitrateStatisticsObserver* ModuleRtpRtcpImpl::GetVideoBitrateObserver() const { return rtp_sender_.GetBitrateObserver(); } -// Bad state of RTP receiver request a keyframe. void ModuleRtpRtcpImpl::OnRequestIntraFrame() { RequestKeyFrame(); } @@ -1716,4 +1327,9 @@ FrameCountObserver* ModuleRtpRtcpImpl::GetSendFrameCountObserver() const { return rtp_sender_.GetFrameCountObserver(); } +bool ModuleRtpRtcpImpl::IsDefaultModule() const { + CriticalSectionScoped cs(critical_section_module_ptrs_.get()); + return !child_modules_.empty(); +} + } // Namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h index 075770dd235..55826b6fe8b 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h @@ -21,10 +21,6 @@ #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/testsupport/gtest_prod_util.h" -#ifdef MATLAB -class MatlabPlot; -#endif - namespace webrtc { class ModuleRtpRtcpImpl : public RtpRtcp { @@ -80,7 +76,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp { virtual uint32_t SSRC() const OVERRIDE; // Configure SSRC, default is a random number. - virtual int32_t SetSSRC(const uint32_t ssrc) OVERRIDE; + virtual void SetSSRC(const uint32_t ssrc) OVERRIDE; virtual int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const OVERRIDE; @@ -95,13 +91,12 @@ class ModuleRtpRtcpImpl : public RtpRtcp { virtual uint32_t ByteCountSent() const; - virtual int32_t SetRTXSendStatus(const int mode, - const bool set_ssrc, - const uint32_t ssrc) OVERRIDE; + virtual void SetRTXSendStatus(const int mode) OVERRIDE; - virtual int32_t RTXSendStatus(int* mode, uint32_t* ssrc, - int* payloadType) const OVERRIDE; + virtual void RTXSendStatus(int* mode, uint32_t* ssrc, + int* payloadType) const OVERRIDE; + virtual void SetRtxSsrc(uint32_t ssrc) OVERRIDE; virtual void SetRtxSendPayloadType(int payload_type) OVERRIDE; @@ -197,10 +192,14 @@ class ModuleRtpRtcpImpl : public RtpRtcp { // Set received RTCP report block. virtual int32_t AddRTCPReportBlock( - const uint32_t ssrc, const RTCPReportBlock* receive_block) OVERRIDE; + const uint32_t ssrc, const RTCPReportBlock* receive_block) OVERRIDE; virtual int32_t RemoveRTCPReportBlock(const uint32_t ssrc) OVERRIDE; + virtual void GetRtcpPacketTypeCounters( + RtcpPacketTypeCounter* packets_sent, + RtcpPacketTypeCounter* packets_received) const OVERRIDE; + // (REMB) Receiver Estimated Max Bitrate. virtual bool REMB() const OVERRIDE; @@ -291,14 +290,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp { // Get payload type for Redundant Audio Data RFC 2198. virtual int32_t SendREDPayloadType(int8_t& payload_type) const OVERRIDE; - // Set status and id for header-extension-for-audio-level-indication. - virtual int32_t SetRTPAudioLevelIndicationStatus( - const bool enable, const uint8_t id) OVERRIDE; - - // Get status and id for header-extension-for-audio-level-indication. - virtual int32_t GetRTPAudioLevelIndicationStatus( - bool& enable, uint8_t& id) const OVERRIDE; - // Store the audio level in d_bov for header-extension-for-audio-level- // indication. virtual int32_t SetAudioLevel(const uint8_t level_d_bov) OVERRIDE; @@ -418,6 +409,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp { void set_rtt_ms(uint32_t rtt_ms); uint32_t rtt_ms() const; + bool IsDefaultModule() const; + int32_t id_; const bool audio_; bool collision_detected_; @@ -429,7 +422,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp { scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_; scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_; ModuleRtpRtcpImpl* default_module_; - std::list<ModuleRtpRtcpImpl*> child_modules_; + std::vector<ModuleRtpRtcpImpl*> child_modules_; + size_t padding_index_; // Send side NACKMethod nack_method_; @@ -442,10 +436,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp { RemoteBitrateEstimator* remote_bitrate_; -#ifdef MATLAB - MatlabPlot* plot1_; -#endif - RtcpRttStats* rtt_stats_; // The processed RTT from RtcpRttStats. diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index 50f7f2e1cb3..eba4e010b59 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -12,11 +12,23 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" +#include "webrtc/modules/pacing/include/mock/mock_paced_sender.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" +#include "webrtc/system_wrappers/interface/scoped_vector.h" + +using ::testing::_; +using ::testing::NiceMock; +using ::testing::Return; +using ::testing::SaveArg; namespace webrtc { namespace { +const uint32_t kSenderSsrc = 0x12345; +const uint32_t kReceiverSsrc = 0x23456; +const uint32_t kSenderRtxSsrc = 0x32345; +const uint32_t kOneWayNetworkDelayMs = 100; class RtcpRttStatsTestImpl : public RtcpRttStats { public: @@ -35,12 +47,12 @@ class RtcpRttStatsTestImpl : public RtcpRttStats { class SendTransport : public Transport, public NullRtpData { public: - SendTransport() : rtp_rtcp_impl_(NULL), clock_(NULL), delay_ms_(0) {} + SendTransport() : receiver_(NULL), clock_(NULL), delay_ms_(0) {} - void SetRtpRtcpModule(ModuleRtpRtcpImpl* rtp_rtcp_impl) { - rtp_rtcp_impl_ = rtp_rtcp_impl; + void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) { + receiver_ = receiver; } - void SimulateNetworkDelay(int delay_ms, SimulatedClock* clock) { + void SimulateNetworkDelay(uint32_t delay_ms, SimulatedClock* clock) { clock_ = clock; delay_ms_ = delay_ms; } @@ -51,14 +63,48 @@ class SendTransport : public Transport, if (clock_) { clock_->AdvanceTimeMilliseconds(delay_ms_); } - EXPECT_TRUE(rtp_rtcp_impl_ != NULL); - EXPECT_EQ(0, rtp_rtcp_impl_->IncomingRtcpPacket( + EXPECT_TRUE(receiver_ != NULL); + EXPECT_EQ(0, receiver_->IncomingRtcpPacket( static_cast<const uint8_t*>(data), len)); return len; } - ModuleRtpRtcpImpl* rtp_rtcp_impl_; + ModuleRtpRtcpImpl* receiver_; SimulatedClock* clock_; - int delay_ms_; + uint32_t delay_ms_; +}; + +class RtpRtcpModule { + public: + RtpRtcpModule(SimulatedClock* clock) + : receive_statistics_(ReceiveStatistics::Create(clock)) { + RtpRtcp::Configuration config; + config.audio = false; + config.clock = clock; + config.outgoing_transport = &transport_; + config.receive_statistics = receive_statistics_.get(); + config.rtt_stats = &rtt_stats_; + + impl_.reset(new ModuleRtpRtcpImpl(config)); + EXPECT_EQ(0, impl_->SetRTCPStatus(kRtcpCompound)); + + transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, clock); + } + + RtcpPacketTypeCounter packets_sent_; + RtcpPacketTypeCounter packets_received_; + scoped_ptr<ReceiveStatistics> receive_statistics_; + SendTransport transport_; + RtcpRttStatsTestImpl rtt_stats_; + scoped_ptr<ModuleRtpRtcpImpl> impl_; + + RtcpPacketTypeCounter RtcpSent() { + impl_->GetRtcpPacketTypeCounters(&packets_sent_, &packets_received_); + return packets_sent_; + } + RtcpPacketTypeCounter RtcpReceived() { + impl_->GetRtcpPacketTypeCounters(&packets_sent_, &packets_received_); + return packets_received_; + } }; } // namespace @@ -66,97 +112,357 @@ class RtpRtcpImplTest : public ::testing::Test { protected: RtpRtcpImplTest() : clock_(1335900000), - receive_statistics_(ReceiveStatistics::Create(&clock_)) { - RtpRtcp::Configuration configuration; - configuration.id = 0; - configuration.audio = false; - configuration.clock = &clock_; - configuration.outgoing_transport = &transport_; - configuration.receive_statistics = receive_statistics_.get(); - configuration.rtt_stats = &rtt_stats_; - - rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration)); - transport_.SetRtpRtcpModule(rtp_rtcp_impl_.get()); + sender_(&clock_), + receiver_(&clock_) { + // Send module. + EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true)); + sender_.impl_->SetSSRC(kSenderSsrc); + sender_.impl_->SetRemoteSSRC(kReceiverSsrc); + // Receive module. + EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false)); + receiver_.impl_->SetSSRC(kReceiverSsrc); + receiver_.impl_->SetRemoteSSRC(kSenderSsrc); + // Transport settings. + sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get()); + receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get()); } - SimulatedClock clock_; - scoped_ptr<ReceiveStatistics> receive_statistics_; - scoped_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_; - SendTransport transport_; - RtcpRttStatsTestImpl rtt_stats_; + RtpRtcpModule sender_; + RtpRtcpModule receiver_; }; TEST_F(RtpRtcpImplTest, Rtt) { - const uint32_t kSsrc = 0x12345; - RTPHeader header = {}; + RTPHeader header; header.timestamp = 1; header.sequenceNumber = 123; - header.ssrc = kSsrc; + header.ssrc = kSenderSsrc; header.headerLength = 12; - receive_statistics_->IncomingPacket(header, 100, false); - - rtp_rtcp_impl_->SetRemoteSSRC(kSsrc); - EXPECT_EQ(0, rtp_rtcp_impl_->SetSendingStatus(true)); - EXPECT_EQ(0, rtp_rtcp_impl_->SetRTCPStatus(kRtcpCompound)); - EXPECT_EQ(0, rtp_rtcp_impl_->SetSSRC(kSsrc)); + receiver_.receive_statistics_->IncomingPacket(header, 100, false); - // A SR should have been sent and received. - EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport)); + // Sender module should send a SR. + EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport)); - // Send new SR. A response to the last SR should be sent. + // Receiver module should send a RR with a response to the last received SR. clock_.AdvanceTimeMilliseconds(1000); - transport_.SimulateNetworkDelay(100, &clock_); - EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport)); + EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport)); // Verify RTT. uint16_t rtt; uint16_t avg_rtt; uint16_t min_rtt; uint16_t max_rtt; - EXPECT_EQ(0, rtp_rtcp_impl_->RTT(kSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt)); - EXPECT_EQ(100, rtt); - EXPECT_EQ(100, avg_rtt); - EXPECT_EQ(100, min_rtt); - EXPECT_EQ(100, max_rtt); + EXPECT_EQ(0, + sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt)); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, rtt); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, avg_rtt); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, min_rtt); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, max_rtt); // No RTT from other ssrc. EXPECT_EQ(-1, - rtp_rtcp_impl_->RTT(kSsrc + 1, &rtt, &avg_rtt, &min_rtt, &max_rtt)); + sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt)); // Verify RTT from rtt_stats config. - EXPECT_EQ(0U, rtt_stats_.LastProcessedRtt()); - EXPECT_EQ(0U, rtp_rtcp_impl_->rtt_ms()); - rtp_rtcp_impl_->Process(); - EXPECT_EQ(100U, rtt_stats_.LastProcessedRtt()); - EXPECT_EQ(100U, rtp_rtcp_impl_->rtt_ms()); + EXPECT_EQ(0U, sender_.rtt_stats_.LastProcessedRtt()); + EXPECT_EQ(0U, sender_.impl_->rtt_ms()); + sender_.impl_->Process(); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.rtt_stats_.LastProcessedRtt()); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms()); } TEST_F(RtpRtcpImplTest, SetRtcpXrRrtrStatus) { - EXPECT_FALSE(rtp_rtcp_impl_->RtcpXrRrtrStatus()); - rtp_rtcp_impl_->SetRtcpXrRrtrStatus(true); - EXPECT_TRUE(rtp_rtcp_impl_->RtcpXrRrtrStatus()); + EXPECT_FALSE(receiver_.impl_->RtcpXrRrtrStatus()); + receiver_.impl_->SetRtcpXrRrtrStatus(true); + EXPECT_TRUE(receiver_.impl_->RtcpXrRrtrStatus()); } TEST_F(RtpRtcpImplTest, RttForReceiverOnly) { - rtp_rtcp_impl_->SetRtcpXrRrtrStatus(true); - EXPECT_EQ(0, rtp_rtcp_impl_->SetSendingStatus(false)); - EXPECT_EQ(0, rtp_rtcp_impl_->SetRTCPStatus(kRtcpCompound)); - EXPECT_EQ(0, rtp_rtcp_impl_->SetSSRC(0x12345)); + receiver_.impl_->SetRtcpXrRrtrStatus(true); - // A Receiver time reference report (RTRR) should be sent and received. - EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport)); + // Receiver module should send a Receiver time reference report (RTRR). + EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport)); - // Send new RTRR. A response to the last RTRR should be sent. + // Sender module should send a response to the last received RTRR (DLRR). clock_.AdvanceTimeMilliseconds(1000); - transport_.SimulateNetworkDelay(100, &clock_); - EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport)); + EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport)); // Verify RTT. - EXPECT_EQ(0U, rtt_stats_.LastProcessedRtt()); - EXPECT_EQ(0U, rtp_rtcp_impl_->rtt_ms()); - rtp_rtcp_impl_->Process(); - EXPECT_EQ(100U, rtt_stats_.LastProcessedRtt()); - EXPECT_EQ(100U, rtp_rtcp_impl_->rtt_ms()); + EXPECT_EQ(0U, receiver_.rtt_stats_.LastProcessedRtt()); + EXPECT_EQ(0U, receiver_.impl_->rtt_ms()); + receiver_.impl_->Process(); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.rtt_stats_.LastProcessedRtt()); + EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms()); +} + +TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) { + EXPECT_EQ(0U, sender_.RtcpReceived().nack_packets); + EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets); + // Receive module sends a NACK. + const uint16_t kNackLength = 1; + uint16_t nack_list[kNackLength] = {123}; + EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength)); + EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets); + + // Send module receives the NACK. + EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets); +} + +TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_FirAndPli) { + EXPECT_EQ(0U, sender_.RtcpReceived().fir_packets); + EXPECT_EQ(0U, receiver_.RtcpSent().fir_packets); + // Receive module sends a FIR. + EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpFir)); + EXPECT_EQ(1U, receiver_.RtcpSent().fir_packets); + // Send module receives the FIR. + EXPECT_EQ(1U, sender_.RtcpReceived().fir_packets); + + // Receive module sends a FIR and PLI. + EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpFir | kRtcpPli)); + EXPECT_EQ(2U, receiver_.RtcpSent().fir_packets); + EXPECT_EQ(1U, receiver_.RtcpSent().pli_packets); + // Send module receives the FIR and PLI. + EXPECT_EQ(2U, sender_.RtcpReceived().fir_packets); + EXPECT_EQ(1U, sender_.RtcpReceived().pli_packets); +} + +class RtpSendingTestTransport : public Transport { + public: + void ResetCounters() { bytes_received_.clear(); } + + virtual int SendPacket(int channel, const void* data, int length) { + RTPHeader header; + scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + EXPECT_TRUE( + parser->Parse(static_cast<const uint8_t*>(data), length, &header)); + bytes_received_[header.ssrc] += length; + ++packets_received_[header.ssrc]; + return length; + } + + virtual int SendRTCPPacket(int channel, const void* data, int length) { + return length; + } + + int GetPacketsReceived(uint32_t ssrc) const { + std::map<uint32_t, int>::const_iterator it = packets_received_.find(ssrc); + if (it == packets_received_.end()) + return 0; + return it->second; + } + + int GetBytesReceived(uint32_t ssrc) const { + std::map<uint32_t, int>::const_iterator it = bytes_received_.find(ssrc); + if (it == bytes_received_.end()) + return 0; + return it->second; + } + + int GetTotalBytesReceived() const { + int sum = 0; + for (std::map<uint32_t, int>::const_iterator it = bytes_received_.begin(); + it != bytes_received_.end(); + ++it) { + sum += it->second; + } + return sum; + } + + private: + std::map<uint32_t, int> bytes_received_; + std::map<uint32_t, int> packets_received_; +}; + +class RtpSendingTest : public ::testing::Test { + protected: + // Map from SSRC to number of received packets and bytes. + typedef std::map<uint32_t, std::pair<int, int> > PaddingMap; + + RtpSendingTest() { + // Send module. + RtpRtcp::Configuration config; + config.audio = false; + config.clock = Clock::GetRealTimeClock(); + config.outgoing_transport = &transport_; + config.receive_statistics = receive_statistics_.get(); + config.rtt_stats = &rtt_stats_; + config.paced_sender = &pacer_; + memset(&codec_, 0, sizeof(VideoCodec)); + codec_.plType = 100; + strncpy(codec_.plName, "VP8", 3); + codec_.numberOfSimulcastStreams = 3; + codec_.simulcastStream[0].width = 320; + codec_.simulcastStream[0].height = 180; + codec_.simulcastStream[0].maxBitrate = 300; + codec_.simulcastStream[1].width = 640; + codec_.simulcastStream[1].height = 360; + codec_.simulcastStream[1].maxBitrate = 600; + codec_.simulcastStream[2].width = 1280; + codec_.simulcastStream[2].height = 720; + codec_.simulcastStream[2].maxBitrate = 1200; + // We need numberOfSimulcastStreams + 1 RTP modules since we need one + // default module. + for (int i = 0; i < codec_.numberOfSimulcastStreams + 1; ++i) { + RtpRtcp* sender = RtpRtcp::CreateRtpRtcp(config); + EXPECT_EQ(0, sender->RegisterSendPayload(codec_)); + EXPECT_EQ(0, sender->SetSendingStatus(true)); + EXPECT_EQ(0, sender->SetSendingMediaStatus(true)); + sender->SetSSRC(kSenderSsrc + i); + sender->SetRemoteSSRC(kReceiverSsrc + i); + senders_.push_back(sender); + config.default_module = senders_[0]; + } + std::vector<uint32_t> bitrates; + bitrates.push_back(codec_.simulcastStream[0].maxBitrate); + bitrates.push_back(codec_.simulcastStream[1].maxBitrate); + bitrates.push_back(codec_.simulcastStream[2].maxBitrate); + senders_[0]->SetTargetSendBitrate(bitrates); + } + + ~RtpSendingTest() { + for (int i = senders_.size() - 1; i >= 0; --i) { + delete senders_[i]; + } + } + + void SendFrameOnSender(int sender_index, + const uint8_t* payload, + size_t length) { + RTPVideoHeader rtp_video_header = { + codec_.simulcastStream[sender_index].width, + codec_.simulcastStream[sender_index].height, + true, + 0, + kRtpVideoVp8, + {}}; + uint32_t seq_num = 0; + uint32_t ssrc = 0; + int64_t capture_time_ms = 0; + bool retransmission = false; + EXPECT_CALL(pacer_, SendPacket(_, _, _, _, _, _)) + .WillRepeatedly(DoAll(SaveArg<1>(&ssrc), + SaveArg<2>(&seq_num), + SaveArg<3>(&capture_time_ms), + SaveArg<5>(&retransmission), + Return(true))); + EXPECT_EQ(0, + senders_[sender_index]->SendOutgoingData(kVideoFrameKey, + codec_.plType, + 0, + 0, + payload, + length, + NULL, + &rtp_video_header)); + EXPECT_TRUE(senders_[sender_index]->TimeToSendPacket( + ssrc, seq_num, capture_time_ms, retransmission)); + } + + void ExpectPadding(const PaddingMap& expected_padding) { + int expected_total_bytes = 0; + for (PaddingMap::const_iterator it = expected_padding.begin(); + it != expected_padding.end(); + ++it) { + int packets_received = transport_.GetBytesReceived(it->first); + if (it->second.first > 0) { + EXPECT_GE(packets_received, it->second.first) + << "On SSRC: " << it->first; + } + int bytes_received = transport_.GetBytesReceived(it->first); + expected_total_bytes += bytes_received; + if (it->second.second > 0) { + EXPECT_GE(bytes_received, it->second.second) + << "On SSRC: " << it->first; + } else { + EXPECT_EQ(0, bytes_received) << "On SSRC: " << it->first; + } + } + EXPECT_EQ(expected_total_bytes, transport_.GetTotalBytesReceived()); + } + + scoped_ptr<ReceiveStatistics> receive_statistics_; + RtcpRttStatsTestImpl rtt_stats_; + std::vector<RtpRtcp*> senders_; + RtpSendingTestTransport transport_; + NiceMock<MockPacedSender> pacer_; + VideoCodec codec_; +}; + +TEST_F(RtpSendingTest, DISABLED_RoundRobinPadding) { + // We have to send on an SSRC to be allowed to pad, since a marker bit must + // be sent prior to padding packets. + const uint8_t payload[200] = {0}; + for (int i = 0; i < codec_.numberOfSimulcastStreams; ++i) { + SendFrameOnSender(i + 1, payload, sizeof(payload)); + } + transport_.ResetCounters(); + senders_[0]->TimeToSendPadding(500); + PaddingMap expected_padding; + expected_padding[kSenderSsrc + 1] = std::make_pair(2, 500); + expected_padding[kSenderSsrc + 2] = std::make_pair(0, 0); + expected_padding[kSenderSsrc + 3] = std::make_pair(0, 0); + ExpectPadding(expected_padding); + senders_[0]->TimeToSendPadding(1000); + expected_padding[kSenderSsrc + 2] = std::make_pair(4, 1000); + ExpectPadding(expected_padding); + senders_[0]->TimeToSendPadding(1500); + expected_padding[kSenderSsrc + 3] = std::make_pair(6, 1500); + ExpectPadding(expected_padding); +} + +TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtx) { + // Enable RTX to allow padding to be sent prior to media. + for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) { + senders_[i]->SetRtxSendPayloadType(96); + senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i); + senders_[i]->SetRTXSendStatus(kRtxRetransmitted); + } + transport_.ResetCounters(); + senders_[0]->TimeToSendPadding(500); + PaddingMap expected_padding; + expected_padding[kSenderSsrc + 1] = std::make_pair(0, 0); + expected_padding[kSenderSsrc + 2] = std::make_pair(0, 0); + expected_padding[kSenderSsrc + 3] = std::make_pair(0, 0); + expected_padding[kSenderRtxSsrc + 1] = std::make_pair(2, 500); + expected_padding[kSenderRtxSsrc + 2] = std::make_pair(0, 0); + expected_padding[kSenderRtxSsrc + 3] = std::make_pair(0, 0); + ExpectPadding(expected_padding); + senders_[0]->TimeToSendPadding(1000); + expected_padding[kSenderRtxSsrc + 2] = std::make_pair(4, 500); + ExpectPadding(expected_padding); + senders_[0]->TimeToSendPadding(1500); + + expected_padding[kSenderRtxSsrc + 3] = std::make_pair(6, 500); + ExpectPadding(expected_padding); } +TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtxRedundantPayloads) { + for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) { + senders_[i]->SetRtxSendPayloadType(96); + senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i); + senders_[i]->SetRTXSendStatus(kRtxRetransmitted | kRtxRedundantPayloads); + senders_[i]->SetStorePacketsStatus(true, 100); + } + // First send payloads so that we have something to retransmit. + const size_t kPayloadSize = 500; + const uint8_t payload[kPayloadSize] = {0}; + for (int i = 0; i < codec_.numberOfSimulcastStreams; ++i) { + SendFrameOnSender(i + 1, payload, sizeof(payload)); + } + transport_.ResetCounters(); + senders_[0]->TimeToSendPadding(500); + PaddingMap expected_padding; + expected_padding[kSenderSsrc + 1] = std::make_pair<int, int>(0, 0); + expected_padding[kSenderSsrc + 2] = std::make_pair<int, int>(0, 0); + expected_padding[kSenderSsrc + 3] = std::make_pair<int, int>(0, 0); + expected_padding[kSenderRtxSsrc + 1] = std::make_pair<int, int>(1, 500); + expected_padding[kSenderRtxSsrc + 2] = std::make_pair<int, int>(0, 0); + expected_padding[kSenderRtxSsrc + 3] = std::make_pair<int, int>(0, 0); + ExpectPadding(expected_padding); + senders_[0]->TimeToSendPadding(1000); + expected_padding[kSenderRtxSsrc + 2] = std::make_pair<int, int>(2, 1000); + ExpectPadding(expected_padding); + senders_[0]->TimeToSendPadding(1500); + expected_padding[kSenderRtxSsrc + 3] = std::make_pair<int, int>(3, 1500); + ExpectPadding(expected_padding); +} } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc index fd320324b8a..7cfcd7222c8 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc @@ -15,7 +15,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -56,7 +56,6 @@ RTPSender::RTPSender(const int32_t id, transport_(transport), sending_media_(true), // Default to sending media. max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP. - target_send_bitrate_(0), packet_over_head_(28), payload_type_(-1), payload_type_map_(), @@ -88,7 +87,9 @@ RTPSender::RTPSender(const int32_t id, csrcs_(), include_csrcs_(true), rtx_(kRtxOff), - payload_type_rtx_(-1) { + payload_type_rtx_(-1), + target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()), + target_bitrate_(0) { memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_)); memset(nack_byte_count_, 0, sizeof(nack_byte_count_)); memset(csrcs_, 0, sizeof(csrcs_)); @@ -104,9 +105,8 @@ RTPSender::RTPSender(const int32_t id, audio_ = new RTPSenderAudio(id, clock_, this); audio_->RegisterAudioCallback(audio_feedback); } else { - video_ = new RTPSenderVideo(id, clock_, this); + video_ = new RTPSenderVideo(clock_, this); } - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); } RTPSender::~RTPSender() { @@ -125,12 +125,16 @@ RTPSender::~RTPSender() { } delete audio_; delete video_; +} - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__); +void RTPSender::SetTargetBitrate(uint32_t bitrate) { + CriticalSectionScoped cs(target_bitrate_critsect_.get()); + target_bitrate_ = bitrate; } -void RTPSender::SetTargetSendBitrate(const uint32_t bits) { - target_send_bitrate_ = static_cast<uint16_t>(bits / 1000); +uint32_t RTPSender::GetTargetBitrate() { + CriticalSectionScoped cs(target_bitrate_critsect_.get()); + return target_bitrate_; } uint16_t RTPSender::ActualSendBitrateKbit() const { @@ -157,10 +161,12 @@ uint32_t RTPSender::NackOverheadRate() const { bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const { + if (!SendingMedia()) + return false; CriticalSectionScoped cs(statistics_crit_.get()); SendDelayMap::const_iterator it = send_delays_.upper_bound( clock_->TimeInMilliseconds() - kSendSideDelayWindowMs); - if (!sending_media_ || it == send_delays_.end()) + if (it == send_delays_.end()) return false; int num_delays = 0; for (; it != send_delays_.end(); ++it) { @@ -273,7 +279,10 @@ int32_t RTPSender::DeRegisterSendPayload( return 0; } -int8_t RTPSender::SendPayloadType() const { return payload_type_; } +int8_t RTPSender::SendPayloadType() const { + CriticalSectionScoped cs(send_critsect_); + return payload_type_; +} int RTPSender::SendPayloadFrequency() const { return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency; @@ -284,16 +293,12 @@ int32_t RTPSender::SetMaxPayloadLength( const uint16_t packet_over_head) { // Sanity check. if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "%s invalid argument", - __FUNCTION__); + LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length; return -1; } CriticalSectionScoped cs(send_critsect_); max_payload_length_ = max_payload_length; packet_over_head_ = packet_over_head; - - WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, id_, "SetMaxPayloadLength to %d.", - max_payload_length); return 0; } @@ -301,9 +306,9 @@ uint16_t RTPSender::MaxDataPayloadLength() const { if (audio_configured_) { return max_payload_length_ - RTPHeaderLength(); } else { - return max_payload_length_ - RTPHeaderLength() - - video_->FECPacketOverhead() - ((rtx_) ? 2 : 0); - // Include the FEC/ULP/RED overhead. + return max_payload_length_ - RTPHeaderLength() // RTP overhead. + - video_->FECPacketOverhead() // FEC/ULP/RED overhead. + - ((rtx_) ? 2 : 0); // RTX overhead. } } @@ -313,16 +318,14 @@ uint16_t RTPSender::MaxPayloadLength() const { uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; } -void RTPSender::SetRTXStatus(int mode, bool set_ssrc, uint32_t ssrc) { +void RTPSender::SetRTXStatus(int mode) { CriticalSectionScoped cs(send_critsect_); rtx_ = mode; - if (rtx_ != kRtxOff) { - if (set_ssrc) { - ssrc_rtx_ = ssrc; - } else { - ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0. - } - } +} + +void RTPSender::SetRtxSsrc(uint32_t ssrc) { + CriticalSectionScoped cs(send_critsect_); + ssrc_rtx_ = ssrc; } void RTPSender::RTXStatus(int* mode, uint32_t* ssrc, @@ -333,7 +336,6 @@ void RTPSender::RTXStatus(int* mode, uint32_t* ssrc, *payload_type = payload_type_rtx_; } - void RTPSender::SetRtxPayloadType(int payload_type) { CriticalSectionScoped cs(send_critsect_); payload_type_rtx_ = payload_type; @@ -344,8 +346,7 @@ int32_t RTPSender::CheckPayloadType(const int8_t payload_type, CriticalSectionScoped cs(send_critsect_); if (payload_type < 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "\tinvalid payload_type (%d)", - payload_type); + LOG(LS_ERROR) << "Invalid payload_type " << payload_type; return -1; } if (audio_configured_) { @@ -367,8 +368,7 @@ int32_t RTPSender::CheckPayloadType(const int8_t payload_type, std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it = payload_type_map_.find(payload_type); if (it == payload_type_map_.end()) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "\tpayloadType:%d not registered", payload_type); + LOG(LS_WARNING) << "Payload type " << payload_type << " not registered."; return -1; } payload_type_ = payload_type; @@ -397,9 +397,7 @@ int32_t RTPSender::SendOutgoingData( } RtpVideoCodecTypes video_type = kRtpVideoGeneric; if (CheckPayloadType(payload_type, &video_type) != 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "%s invalid argument failed to find payload_type:%d", - __FUNCTION__, payload_type); + LOG(LS_ERROR) << "Don't send data with unknown payload type."; return -1; } @@ -456,7 +454,7 @@ int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) { &capture_time_ms)) { break; } - if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true)) + if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false)) return -1; ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length); RTPHeader rtp_header; @@ -472,7 +470,8 @@ bool RTPSender::SendPaddingAccordingToBitrate( // Current bitrate since last estimate(1 second) averaged with the // estimate since then, to get the most up to date bitrate. uint32_t current_bitrate = bitrate_sent_.BitrateNow(); - int bitrate_diff = target_send_bitrate_ * 1000 - current_bitrate; + uint32_t target_bitrate = GetTargetBitrate(); + int bitrate_diff = target_bitrate - current_bitrate; if (bitrate_diff <= 0) { return true; } @@ -483,7 +482,7 @@ bool RTPSender::SendPaddingAccordingToBitrate( } else { bytes = (bitrate_diff / 8); // Cap at 200 ms of target send data. - int bytes_cap = target_send_bitrate_ * 25; // 1000 / 8 / 5. + int bytes_cap = target_bitrate / 1000 * 25; // 1000 / 8 / 5. if (bytes > bytes_cap) { bytes = bytes_cap; } @@ -528,7 +527,7 @@ int RTPSender::SendPadData(int payload_type, uint32_t timestamp, StorageType store, bool force_full_size_packets, bool only_pad_after_markerbit) { // Drop this packet if we're not sending media packets. - if (!sending_media_) { + if (!SendingMedia()) { return bytes; } int padding_bytes_in_packet = 0; @@ -596,7 +595,6 @@ bool RTPSender::StorePackets() const { int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { uint16_t length = IP_PACKET_SIZE; uint8_t data_buffer[IP_PACKET_SIZE]; - uint8_t *buffer_to_send_ptr = data_buffer; int64_t capture_time_ms; if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true, data_buffer, &length, @@ -605,19 +603,13 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { return 0; } - ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length); - RTPHeader header; - if (!rtp_parser.Parse(header)) { - assert(false); - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, - "Failed to parse RTP header of packet to be retransmitted."); - return -1; - } - TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::ReSendPacket", - "timestamp", header.timestamp, - "seqnum", header.sequenceNumber); - if (paced_sender_) { + ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length); + RTPHeader header; + if (!rtp_parser.Parse(header)) { + assert(false); + return -1; + } if (!paced_sender_->SendPacket(PacedSender::kHighPriority, header.ssrc, header.sequenceNumber, @@ -630,17 +622,9 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { } } - uint8_t data_buffer_rtx[IP_PACKET_SIZE]; - if ((rtx_ & kRtxRetransmitted) > 0) { - BuildRtxPacket(data_buffer, &length, data_buffer_rtx); - buffer_to_send_ptr = data_buffer_rtx; - } - - if (SendPacketToNetwork(buffer_to_send_ptr, length)) { - UpdateRtpStats(buffer_to_send_ptr, length, header, rtx_ != kRtxOff, true); - return length; - } - return -1; + return PrepareAndSendPacket(data_buffer, length, capture_time_ms, + (rtx_ & kRtxRetransmitted) > 0, true) ? + length : -1; } bool RTPSender::SendPacketToNetwork(const uint8_t *packet, uint32_t size) { @@ -650,10 +634,9 @@ bool RTPSender::SendPacketToNetwork(const uint8_t *packet, uint32_t size) { } TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::SendPacketToNetwork", "size", size, "sent", bytes_sent); - // TODO(pwesin): Add a separate bitrate for sent bitrate after pacer. + // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer. if (bytes_sent <= 0) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "Transport failed to send packet"); + LOG(LS_WARNING) << "Transport failed to send packet"; return false; } return true; @@ -678,12 +661,12 @@ void RTPSender::OnReceivedNACK( "num_seqnum", nack_sequence_numbers.size(), "avg_rtt", avg_rtt); const int64_t now = clock_->TimeInMilliseconds(); uint32_t bytes_re_sent = 0; + uint32_t target_bitrate = GetTargetBitrate(); // Enough bandwidth to send NACK? if (!ProcessNACKBitRate(now)) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "NACK bitrate reached. Skip sending NACK response. Target %d", - target_send_bitrate_); + LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target " + << target_bitrate; return; } @@ -698,16 +681,15 @@ void RTPSender::OnReceivedNACK( continue; } else if (bytes_sent < 0) { // Failed to send one Sequence number. Give up the rest in this nack. - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_, - "Failed resending RTP packet %d, Discard rest of packets", - *it); + LOG(LS_WARNING) << "Failed resending RTP packet " << *it + << ", Discard rest of packets"; break; } // Delay bandwidth estimate (RTT * BW). - if (target_send_bitrate_ != 0 && avg_rtt) { + if (target_bitrate != 0 && avg_rtt) { // kbits/s * ms = bits => bits/8 = bytes uint32_t target_bytes = - (static_cast<uint32_t>(target_send_bitrate_) * avg_rtt) >> 3; + (static_cast<uint32_t>(target_bitrate / 1000) * avg_rtt) >> 3; if (bytes_re_sent > target_bytes) { break; // Ignore the rest of the packets in the list. } @@ -722,32 +704,34 @@ void RTPSender::OnReceivedNACK( bool RTPSender::ProcessNACKBitRate(const uint32_t now) { uint32_t num = 0; - int32_t byte_count = 0; - const uint32_t avg_interval = 1000; + int byte_count = 0; + const int kAvgIntervalMs = 1000; + uint32_t target_bitrate = GetTargetBitrate(); CriticalSectionScoped cs(send_critsect_); - if (target_send_bitrate_ == 0) { + if (target_bitrate == 0) { return true; } for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) { - if ((now - nack_byte_count_times_[num]) > avg_interval) { + if ((now - nack_byte_count_times_[num]) > kAvgIntervalMs) { // Don't use data older than 1sec. break; } else { byte_count += nack_byte_count_[num]; } } - int32_t time_interval = avg_interval; + int time_interval = kAvgIntervalMs; if (num == NACK_BYTECOUNT_SIZE) { // More than NACK_BYTECOUNT_SIZE nack messages has been received // during the last msg_interval. time_interval = now - nack_byte_count_times_[num - 1]; if (time_interval < 0) { - time_interval = avg_interval; + time_interval = kAvgIntervalMs; } } - return (byte_count * 8) < (target_send_bitrate_ * time_interval); + return (byte_count * 8) < + static_cast<int>(target_bitrate / 1000 * time_interval); } void RTPSender::UpdateNACKBitRate(const uint32_t bytes, @@ -796,19 +780,21 @@ bool RTPSender::TimeToSendPacket(uint16_t sequence_number, UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds()); } return PrepareAndSendPacket(data_buffer, length, capture_time_ms, - retransmission && (rtx_ & kRtxRetransmitted) > 0); + retransmission && (rtx_ & kRtxRetransmitted) > 0, + retransmission); } bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, uint16_t length, int64_t capture_time_ms, - bool send_over_rtx) { + bool send_over_rtx, + bool is_retransmit) { uint8_t *buffer_to_send_ptr = buffer; ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); - TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::TimeToSendPacket", + TRACE_EVENT_INSTANT2("webrtc_rtp", "PrepareAndSendPacket", "timestamp", rtp_header.timestamp, "seqnum", rtp_header.sequenceNumber); @@ -820,20 +806,12 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, int64_t now_ms = clock_->TimeInMilliseconds(); int64_t diff_ms = now_ms - capture_time_ms; - bool updated_transmission_time_offset = - UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header, - diff_ms); - bool updated_abs_send_time = - UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms); - if (updated_transmission_time_offset || updated_abs_send_time) { - // Update stored packet in case of receiving a re-transmission request. - packet_history_.ReplaceRTPHeader(buffer_to_send_ptr, - rtp_header.sequenceNumber, - rtp_header.headerLength); - } - + UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header, + diff_ms); + UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms); bool ret = SendPacketToNetwork(buffer_to_send_ptr, length); - UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, false, false); + UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx, + is_retransmit); return ret; } @@ -842,15 +820,16 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer, const RTPHeader& header, bool is_rtx, bool is_retransmit) { - CriticalSectionScoped lock(statistics_crit_.get()); StreamDataCounters* counters; - uint32_t ssrc; + // Get ssrc before taking statistics_crit_ to avoid possible deadlock. + uint32_t ssrc = SSRC(); + + CriticalSectionScoped lock(statistics_crit_.get()); if (is_rtx) { counters = &rtx_rtp_stats_; ssrc = ssrc_rtx_; } else { counters = &rtp_stats_; - ssrc = ssrc_; } bitrate_sent_.Update(size); @@ -887,14 +866,14 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer, } int RTPSender::TimeToSendPadding(int bytes) { - if (!sending_media_) { - return 0; - } int payload_type; int64_t capture_time_ms; uint32_t timestamp; { CriticalSectionScoped cs(send_critsect_); + if (!sending_media_) { + return 0; + } payload_type = ((rtx_ & kRtxRedundantPayloads) > 0) ? payload_type_rtx_ : payload_type_; timestamp = timestamp_; @@ -909,8 +888,13 @@ int RTPSender::TimeToSendPadding(int bytes) { int bytes_sent = SendRedundantPayloads(payload_type, bytes); bytes -= bytes_sent; if (bytes > 0) { - int padding_sent = SendPadData(payload_type, timestamp, capture_time_ms, - bytes, kDontStore, true, true); + int padding_sent = SendPadData(payload_type, + timestamp, + capture_time_ms, + bytes, + kDontStore, + true, + rtx_ == kRtxOff); bytes_sent += padding_sent; } return bytes_sent; @@ -1116,9 +1100,8 @@ uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const { data_buffer + kHeaderLength + total_block_length); break; case kRtpExtensionAudioLevel: - // Because AudioLevel is handled specially by RTPSenderAudio, we pretend - // we don't have to care about it here, which is true until we wan't to - // use it together with any of the other extensions we support. + block_length = BuildAudioLevelExtension( + data_buffer + kHeaderLength + total_block_length); break; case kRtpExtensionAbsoluteSendTime: block_length = BuildAbsoluteSendTimeExtension( @@ -1178,8 +1161,42 @@ uint8_t RTPSender::BuildTransmissionTimeOffsetExtension( return kTransmissionTimeOffsetLength; } -uint8_t RTPSender::BuildAbsoluteSendTimeExtension( - uint8_t* data_buffer) const { +uint8_t RTPSender::BuildAudioLevelExtension(uint8_t* data_buffer) const { + // An RTP Header Extension for Client-to-Mixer Audio Level Indication + // + // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/ + // + // The form of the audio level extension block: + // + // 0 1 2 3 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // | ID | len=0 |V| level | 0x00 | 0x00 | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // + // Note that we always include 2 pad bytes, which will result in legal and + // correctly parsed RTP, but may be a bit wasteful if more short extensions + // are implemented. Right now the pad bytes would anyway be required at end + // of the extension block, so it makes no difference. + + // Get id defined by user. + uint8_t id; + if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) { + // Not registered. + return 0; + } + size_t pos = 0; + const uint8_t len = 0; + data_buffer[pos++] = (id << 4) + len; + data_buffer[pos++] = (1 << 7) + 0; // Voice, 0 dBov. + data_buffer[pos++] = 0; // Padding. + data_buffer[pos++] = 0; // Padding. + // kAudioLevelLength is including pad bytes. + assert(pos == kAudioLevelLength); + return kAudioLevelLength; +} + +uint8_t RTPSender::BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const { // Absolute send time in RTP streams. // // The absolute send time is signaled to the receiver in-band using the @@ -1213,106 +1230,138 @@ uint8_t RTPSender::BuildAbsoluteSendTimeExtension( return kAbsoluteSendTimeLength; } -bool RTPSender::UpdateTransmissionTimeOffset( +void RTPSender::UpdateTransmissionTimeOffset( uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const int64_t time_diff_ms) const { CriticalSectionScoped cs(send_critsect_); - + // Get id. + uint8_t id = 0; + if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset, + &id) != 0) { + // Not registered. + return; + } // Get length until start of header extension block. int extension_block_pos = rtp_header_extension_map_.GetLengthUntilBlockStartInBytes( kRtpExtensionTransmissionTimeOffset); if (extension_block_pos < 0) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update transmission time offset, not registered."); - return false; + LOG(LS_WARNING) + << "Failed to update transmission time offset, not registered."; + return; } int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos; if (rtp_packet_length < block_pos + kTransmissionTimeOffsetLength || rtp_header.headerLength < block_pos + kTransmissionTimeOffsetLength) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update transmission time offset, invalid length."); - return false; + LOG(LS_WARNING) + << "Failed to update transmission time offset, invalid length."; + return; } // Verify that header contains extension. if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) && (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) { - WEBRTC_TRACE( - kTraceStream, kTraceRtpRtcp, id_, - "Failed to update transmission time offset, hdr extension not found."); - return false; - } - // Get id. - uint8_t id = 0; - if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset, - &id) != 0) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update transmission time offset, no id."); - return false; + LOG(LS_WARNING) << "Failed to update transmission time offset, hdr " + "extension not found."; + return; } // Verify first byte in block. const uint8_t first_block_byte = (id << 4) + 2; if (rtp_packet[block_pos] != first_block_byte) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update transmission time offset."); - return false; + LOG(LS_WARNING) << "Failed to update transmission time offset."; + return; } // Update transmission offset field (converting to a 90 kHz timestamp). ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, time_diff_ms * 90); // RTP timestamp. - return true; } -bool RTPSender::UpdateAbsoluteSendTime( - uint8_t *rtp_packet, const uint16_t rtp_packet_length, - const RTPHeader &rtp_header, const int64_t now_ms) const { +bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet, + const uint16_t rtp_packet_length, + const RTPHeader &rtp_header, + const bool is_voiced, + const uint8_t dBov) const { CriticalSectionScoped cs(send_critsect_); + // Get id. + uint8_t id = 0; + if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) { + // Not registered. + return false; + } // Get length until start of header extension block. int extension_block_pos = rtp_header_extension_map_.GetLengthUntilBlockStartInBytes( - kRtpExtensionAbsoluteSendTime); + kRtpExtensionAudioLevel); if (extension_block_pos < 0) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update absolute send time, not registered."); + // The feature is not enabled. return false; } int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos; - if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength || - rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update absolute send time, invalid length."); + if (rtp_packet_length < block_pos + kAudioLevelLength || + rtp_header.headerLength < block_pos + kAudioLevelLength) { + LOG(LS_WARNING) << "Failed to update audio level, invalid length."; return false; } // Verify that header contains extension. if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) && (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) { - WEBRTC_TRACE( - kTraceStream, kTraceRtpRtcp, id_, - "Failed to update absolute send time, hdr extension not found."); + LOG(LS_WARNING) << "Failed to update audio level, hdr extension not found."; + return false; + } + // Verify first byte in block. + const uint8_t first_block_byte = (id << 4) + 0; + if (rtp_packet[block_pos] != first_block_byte) { + LOG(LS_WARNING) << "Failed to update audio level."; return false; } + rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f); + return true; +} + +void RTPSender::UpdateAbsoluteSendTime( + uint8_t *rtp_packet, const uint16_t rtp_packet_length, + const RTPHeader &rtp_header, const int64_t now_ms) const { + CriticalSectionScoped cs(send_critsect_); + // Get id. uint8_t id = 0; if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime, &id) != 0) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update absolute send time, no id."); - return false; + // Not registered. + return; + } + // Get length until start of header extension block. + int extension_block_pos = + rtp_header_extension_map_.GetLengthUntilBlockStartInBytes( + kRtpExtensionAbsoluteSendTime); + if (extension_block_pos < 0) { + // The feature is not enabled. + return; + } + int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos; + if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength || + rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) { + LOG(LS_WARNING) << "Failed to update absolute send time, invalid length."; + return; + } + // Verify that header contains extension. + if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) && + (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) { + LOG(LS_WARNING) + << "Failed to update absolute send time, hdr extension not found."; + return; } // Verify first byte in block. const uint8_t first_block_byte = (id << 4) + 2; if (rtp_packet[block_pos] != first_block_byte) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, - "Failed to update absolute send time."); - return false; + LOG(LS_WARNING) << "Failed to update absolute send time."; + return; } // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit // fractional part). ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, ((now_ms << 18) / 1000) & 0x00ffffff); - return true; } void RTPSender::SetSendingStatus(bool enabled) { @@ -1462,19 +1511,6 @@ int32_t RTPSender::SetAudioPacketSize( return audio_->SetAudioPacketSize(packet_size_samples); } -int32_t RTPSender::SetAudioLevelIndicationStatus(const bool enable, - const uint8_t ID) { - if (!audio_configured_) { - return -1; - } - return audio_->SetAudioLevelIndicationStatus(enable, ID); -} - -int32_t RTPSender::AudioLevelIndicationStatus(bool *enable, - uint8_t* id) const { - return audio_->AudioLevelIndicationStatus(*enable, *id); -} - int32_t RTPSender::SetAudioLevel(const uint8_t level_d_bov) { return audio_->SetAudioLevel(level_d_bov); } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h index e1cc3a182a7..291e619b43a 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h @@ -25,6 +25,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1. @@ -82,7 +83,8 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { // was sent within the statistics window. bool GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const; - void SetTargetSendBitrate(const uint32_t bits); + void SetTargetBitrate(uint32_t bitrate); + uint32_t GetTargetBitrate(); virtual uint16_t MaxDataPayloadLength() const OVERRIDE; // with RTP and FEC headers. @@ -153,19 +155,15 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer) const; - uint8_t BuildTransmissionTimeOffsetExtension( - uint8_t *data_buffer) const; - uint8_t BuildAbsoluteSendTimeExtension( - uint8_t* data_buffer) const; + uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const; + uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const; + uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const; - bool UpdateTransmissionTimeOffset(uint8_t *rtp_packet, - const uint16_t rtp_packet_length, - const RTPHeader &rtp_header, - const int64_t time_diff_ms) const; - bool UpdateAbsoluteSendTime(uint8_t *rtp_packet, - const uint16_t rtp_packet_length, - const RTPHeader &rtp_header, - const int64_t now_ms) const; + bool UpdateAudioLevel(uint8_t *rtp_packet, + const uint16_t rtp_packet_length, + const RTPHeader &rtp_header, + const bool is_voiced, + const uint8_t dBov) const; bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms, bool retransmission); @@ -187,10 +185,12 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { bool ProcessNACKBitRate(const uint32_t now); // RTX. - void SetRTXStatus(int mode, bool set_ssrc, uint32_t ssrc); + void SetRTXStatus(int mode); void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const; + void SetRtxSsrc(uint32_t ssrc); + void SetRtxPayloadType(int payloadType); // Functions wrapping RTPSenderInterface. @@ -228,12 +228,6 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { // packet in silence (CNG). int32_t SetAudioPacketSize(const uint16_t packet_size_samples); - // Set status and ID for header-extension-for-audio-level-indication. - int32_t SetAudioLevelIndicationStatus(const bool enable, const uint8_t ID); - - // Get status and ID for header-extension-for-audio-level-indication. - int32_t AudioLevelIndicationStatus(bool *enable, uint8_t *id) const; - // Store the audio level in d_bov for // header-extension-for-audio-level-indication. int32_t SetAudioLevel(const uint8_t level_d_bov); @@ -303,7 +297,8 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { bool PrepareAndSendPacket(uint8_t* buffer, uint16_t length, int64_t capture_time_ms, - bool send_over_rtx); + bool send_over_rtx, + bool is_retransmit); int SendRedundantPayloads(int payload_type, int bytes); @@ -319,6 +314,15 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms); + void UpdateTransmissionTimeOffset(uint8_t *rtp_packet, + const uint16_t rtp_packet_length, + const RTPHeader &rtp_header, + const int64_t time_diff_ms) const; + void UpdateAbsoluteSendTime(uint8_t *rtp_packet, + const uint16_t rtp_packet_length, + const RTPHeader &rtp_header, + const int64_t now_ms) const; + void UpdateRtpStats(const uint8_t* buffer, uint32_t size, const RTPHeader& header, @@ -338,13 +342,12 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { CriticalSectionWrapper *send_critsect_; Transport *transport_; - bool sending_media_; + bool sending_media_ GUARDED_BY(send_critsect_); uint16_t max_payload_length_; - uint16_t target_send_bitrate_; uint16_t packet_over_head_; - int8_t payload_type_; + int8_t payload_type_ GUARDED_BY(send_critsect_); std::map<int8_t, ModuleRTPUtility::Payload *> payload_type_map_; RtpHeaderExtensionMap rtp_header_extension_map_; @@ -388,6 +391,13 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { int rtx_; uint32_t ssrc_rtx_; int payload_type_rtx_; + + // Note: Don't access this variable directly, always go through + // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember + // that by the time the function returns there is no guarantee + // that the target bitrate is still valid. + scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; + uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc index f800142886d..6b3e2276ee6 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -17,7 +17,7 @@ namespace webrtc { RTPSenderAudio::RTPSenderAudio(const int32_t id, Clock* clock, - RTPSenderInterface* rtpSender) : + RTPSender* rtpSender) : _id(id), _clock(clock), _rtpSender(rtpSender), @@ -42,8 +42,6 @@ RTPSenderAudio::RTPSenderAudio(const int32_t id, Clock* clock, _cngSWBPayloadType(-1), _cngFBPayloadType(-1), _lastPayloadType(-1), - _includeAudioLevelIndication(false), // @TODO - reset at Init()? - _audioLevelIndicationID(0), _audioLevel_dBov(0) { }; @@ -365,52 +363,12 @@ int32_t RTPSenderAudio::SendAudio( if (rtpHeaderLength <= 0) { return -1; } + if (maxPayloadLength < (rtpHeaderLength + payloadSize)) { + // Too large payload buffer. + return -1; + } { CriticalSectionScoped cs(_sendAudioCritsect); - - // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/ - if (_includeAudioLevelIndication) { - dataBuffer[0] |= 0x10; // set eXtension bit - /* - 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | 0xBE | 0xDE | length=1 | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | ID | len=0 |V| level | 0x00 | 0x00 | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - */ - // add our ID (0xBEDE) - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, - RTP_AUDIO_LEVEL_UNIQUE_ID); - rtpHeaderLength += 2; - - // add the length (length=1) in number of word32 - const uint8_t length = 1; - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, - length); - rtpHeaderLength += 2; - - // add ID (defined by the user) and len(=0) byte - const uint8_t id = _audioLevelIndicationID; - const uint8_t len = 0; - dataBuffer[rtpHeaderLength++] = (id << 4) + len; - - // add voice-activity flag (V) bit and the audio level (in dBov) - const uint8_t V = (frameType == kAudioFrameSpeech); - uint8_t level = _audioLevel_dBov; - dataBuffer[rtpHeaderLength++] = (V << 7) + level; - - // add two bytes zero padding - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, 0); - rtpHeaderLength += 2; - } - - if(maxPayloadLength < rtpHeaderLength + payloadSize ) { - // too large payload buffer - return -1; - } - if (_REDPayloadType >= 0 && // Have we configured RED? fragmentation && fragmentation->fragmentationVectorSize > 1 && @@ -474,6 +432,17 @@ int32_t RTPSenderAudio::SendAudio( } } _lastPayloadType = payloadType; + + // Update audio level extension, if included. + { + uint16_t packetSize = payloadSize + rtpHeaderLength; + ModuleRTPUtility::RTPHeaderParser rtp_parser(dataBuffer, packetSize); + RTPHeader rtp_header; + rtp_parser.Parse(rtp_header); + _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header, + (frameType == kAudioFrameSpeech), + _audioLevel_dBov); + } } // end critical section TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp", _rtpSender->Timestamp(), @@ -486,32 +455,6 @@ int32_t RTPSenderAudio::SendAudio( PacedSender::kHighPriority); } -int32_t -RTPSenderAudio::SetAudioLevelIndicationStatus(const bool enable, - const uint8_t ID) -{ - if(enable && (ID < 1 || ID > 14)) - { - return -1; - } - CriticalSectionScoped cs(_sendAudioCritsect); - - _includeAudioLevelIndication = enable; - _audioLevelIndicationID = ID; - - return 0; -} - -int32_t -RTPSenderAudio::AudioLevelIndicationStatus(bool& enable, - uint8_t& ID) const -{ - CriticalSectionScoped cs(_sendAudioCritsect); - enable = _includeAudioLevelIndication; - ID = _audioLevelIndicationID; - return 0; -} - // Audio level magnitude and voice activity flag are set for each RTP packet int32_t RTPSenderAudio::SetAudioLevel(const uint8_t level_dBov) diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h index 7074e7b29ad..732199c17a1 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h @@ -23,7 +23,7 @@ class RTPSenderAudio: public DTMFqueue { public: RTPSenderAudio(const int32_t id, Clock* clock, - RTPSenderInterface* rtpSender); + RTPSender* rtpSender); virtual ~RTPSenderAudio(); int32_t RegisterAudioPayload( @@ -44,13 +44,6 @@ public: // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG) int32_t SetAudioPacketSize(const uint16_t packetSizeSamples); - // Set status and ID for header-extension-for-audio-level-indication. - // Valid ID range is [1,14]. - int32_t SetAudioLevelIndicationStatus(const bool enable, const uint8_t ID); - - // Get status and ID for header-extension-for-audio-level-indication. - int32_t AudioLevelIndicationStatus(bool& enable, uint8_t& ID) const; - // Store the audio level in dBov for header-extension-for-audio-level-indication. // Valid range is [0,100]. Actual value is negative. int32_t SetAudioLevel(const uint8_t level_dBov); @@ -86,7 +79,7 @@ protected: private: int32_t _id; Clock* _clock; - RTPSenderInterface* _rtpSender; + RTPSender* _rtpSender; CriticalSectionWrapper* _audioFeedbackCritsect; RtpAudioFeedback* _audioFeedback; @@ -117,8 +110,6 @@ private: int8_t _lastPayloadType; // Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) - bool _includeAudioLevelIndication; - uint8_t _audioLevelIndicationID; uint8_t _audioLevel_dBov; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc index ce615be0494..18482890f7a 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -160,11 +160,8 @@ TEST_F(RtpSenderTest, RegisterRtpAudioLevelHeaderExtension) { EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAudioLevel, kAudioLevelExtensionId)); - // Accounted size for audio level is zero because it is currently specially - // treated by RTPSenderAudio. - EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); - // EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength, - // rtp_sender_->RtpHeaderExtensionTotalLength()); + EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength, + rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( kRtpExtensionAudioLevel)); EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); @@ -183,14 +180,16 @@ TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) { EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAudioLevel, kAudioLevelExtensionId)); EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength + - kAbsoluteSendTimeLength, rtp_sender_->RtpHeaderExtensionTotalLength()); + kAbsoluteSendTimeLength + kAudioLevelLength, + rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( kRtpExtensionTransmissionTimeOffset)); - EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength, - rtp_sender_->RtpHeaderExtensionTotalLength()); + EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength + + kAudioLevelLength, rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( kRtpExtensionAbsoluteSendTime)); - EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); + EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength, + rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( kRtpExtensionAudioLevel)); EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); @@ -202,23 +201,24 @@ TEST_F(RtpSenderTest, BuildRTPPacket) { kMarkerBit, kTimestamp, 0); - EXPECT_EQ(12, length); + EXPECT_EQ(kRtpHeaderSize, length); // Verify webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; - RtpHeaderExtensionMap map; - map.Register(kRtpExtensionTransmissionTimeOffset, - kTransmissionTimeOffsetExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(rtp_header, NULL); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header); EXPECT_EQ(length, rtp_header.headerLength); + EXPECT_FALSE(rtp_header.extension.hasTransmissionTimeOffset); + EXPECT_FALSE(rtp_header.extension.hasAbsoluteSendTime); + EXPECT_FALSE(rtp_header.extension.hasAudioLevel); EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset); EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime); + EXPECT_EQ(0u, rtp_header.extension.audioLevel); } TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { @@ -231,7 +231,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { kMarkerBit, kTimestamp, 0); - EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length); + EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), + length); // Verify webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); @@ -246,6 +247,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header); EXPECT_EQ(length, rtp_header.headerLength); + EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset); EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset); // Parse without map extension @@ -255,6 +257,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); EXPECT_EQ(length, rtp_header2.headerLength); + EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset); EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset); } @@ -269,7 +272,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) { kMarkerBit, kTimestamp, 0); - EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length); + EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), + length); // Verify webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); @@ -284,6 +288,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) { ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header); EXPECT_EQ(length, rtp_header.headerLength); + EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset); EXPECT_EQ(kNegTimeOffset, rtp_header.extension.transmissionTimeOffset); } @@ -297,7 +302,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) { kMarkerBit, kTimestamp, 0); - EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length); + EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), + length); // Verify webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); @@ -311,6 +317,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) { ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header); EXPECT_EQ(length, rtp_header.headerLength); + EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime); EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime); // Parse without map extension @@ -320,9 +327,54 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) { ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); EXPECT_EQ(length, rtp_header2.headerLength); + EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime); EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime); } +TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) { + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( + kRtpExtensionAudioLevel, kAudioLevelExtensionId)); + + int32_t length = rtp_sender_->BuildRTPheader(packet_, + kPayload, + kMarkerBit, + kTimestamp, + 0); + EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), + length); + + // Verify + webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RTPHeader rtp_header; + + // Updating audio level is done in RTPSenderAudio, so simulate it here. + rtp_parser.Parse(rtp_header); + rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel); + + RtpHeaderExtensionMap map; + map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId); + const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + + ASSERT_TRUE(valid_rtp_header); + ASSERT_FALSE(rtp_parser.RTCP()); + VerifyRTPHeaderCommon(rtp_header); + EXPECT_EQ(length, rtp_header.headerLength); + EXPECT_TRUE(rtp_header.extension.hasAudioLevel); + // Expect kAudioLevel + 0x80 because we set "voiced" to true in the call to + // UpdateAudioLevel(), above. + EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel); + + // Parse without map extension + webrtc::RTPHeader rtp_header2; + const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL); + + ASSERT_TRUE(valid_rtp_header2); + VerifyRTPHeaderCommon(rtp_header2); + EXPECT_EQ(length, rtp_header2.headerLength); + EXPECT_FALSE(rtp_header2.extension.hasAudioLevel); + EXPECT_EQ(0u, rtp_header2.extension.audioLevel); +} + TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset)); EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime)); @@ -330,30 +382,42 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( + kRtpExtensionAudioLevel, kAudioLevelExtensionId)); int32_t length = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit, kTimestamp, 0); - EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length); + EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), + length); // Verify webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; + // Updating audio level is done in RTPSenderAudio, so simulate it here. + rtp_parser.Parse(rtp_header); + rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel); + RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId); map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); + map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId); const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header); EXPECT_EQ(length, rtp_header.headerLength); + EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset); + EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime); + EXPECT_TRUE(rtp_header.extension.hasAudioLevel); EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset); EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime); + EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel); // Parse without map extension webrtc::RTPHeader rtp_header2; @@ -362,8 +426,12 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); EXPECT_EQ(length, rtp_header2.headerLength); + EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset); + EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime); + EXPECT_FALSE(rtp_header2.extension.hasAudioLevel); EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset); EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime); + EXPECT_EQ(0u, rtp_header2.extension.audioLevel); } TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { @@ -376,7 +444,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); - rtp_sender_->SetTargetSendBitrate(300000); + rtp_sender_->SetTargetBitrate(300000); int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, kPayload, @@ -430,7 +498,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) { kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); - rtp_sender_->SetTargetSendBitrate(300000); + rtp_sender_->SetTargetBitrate(300000); int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, kPayload, @@ -493,7 +561,7 @@ TEST_F(RtpSenderTest, SendPadding) { uint16_t seq_num = kSeqNum; uint32_t timestamp = kTimestamp; rtp_sender_->SetStorePacketsStatus(true, 10); - int rtp_header_len = 12; + int32_t rtp_header_len = kRtpHeaderSize; EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); rtp_header_len += 4; // 4 bytes extension. @@ -512,7 +580,7 @@ TEST_F(RtpSenderTest, SendPadding) { kAbsoluteSendTimeExtensionId); webrtc::RTPHeader rtp_header; - rtp_sender_->SetTargetSendBitrate(300000); + rtp_sender_->SetTargetBitrate(300000); int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, kPayload, @@ -613,14 +681,14 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) { uint16_t seq_num = kSeqNum; rtp_sender_->SetStorePacketsStatus(true, 10); - int rtp_header_len = 12; + int32_t rtp_header_len = kRtpHeaderSize; EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); rtp_header_len += 4; // 4 bytes extension. rtp_header_len += 4; // 4 extra bytes common to all extension headers. - rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads, true, - 1234); + rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads); + rtp_sender_->SetRtxSsrc(1234); // Create and set up parser. scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( @@ -630,7 +698,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) { kTransmissionTimeOffsetExtensionId); rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); - rtp_sender_->SetTargetSendBitrate(300000); + rtp_sender_->SetTargetBitrate(300000); const size_t kNumPayloadSizes = 10; const int kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750, 800, 850, 900, 950}; @@ -939,48 +1007,6 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) { rtp_sender_->RegisterRtpStatisticsCallback(NULL); } -TEST_F(RtpSenderAudioTest, BuildRTPPacketWithAudioLevelExtension) { - EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(true, - kAudioLevelExtensionId)); - EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, kAudioLevelExtensionId)); - - int32_t length = rtp_sender_->BuildRTPheader(packet_, - kAudioPayload, - kMarkerBit, - kTimestamp, - 0); - EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length); - - // Currently, no space is added by for header extension by BuildRTPHeader(). - EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); - - // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); - webrtc::RTPHeader rtp_header; - - RtpHeaderExtensionMap map; - map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); - - ASSERT_TRUE(valid_rtp_header); - ASSERT_FALSE(rtp_parser.RTCP()); - VerifyRTPHeaderCommon(rtp_header); - EXPECT_EQ(length, rtp_header.headerLength); - // TODO(solenberg): Should verify that we got audio level in header extension. - - // Parse without map extension - webrtc::RTPHeader rtp_header2; - const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL); - - ASSERT_TRUE(valid_rtp_header2); - VerifyRTPHeaderCommon(rtp_header2); - EXPECT_EQ(length, rtp_header2.headerLength); - // TODO(solenberg): Should verify that we didn't get audio level. - EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(false, 0)); -} - TEST_F(RtpSenderAudioTest, SendAudio) { char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; const uint8_t payload_type = 127; @@ -1007,8 +1033,6 @@ TEST_F(RtpSenderAudioTest, SendAudio) { } TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { - EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(true, - kAudioLevelExtensionId)); EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionAudioLevel, kAudioLevelExtensionId)); @@ -1044,7 +1068,6 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension), sizeof(extension))); - EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(false, 0)); } } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc index 7b36f7cced8..5d8ae166565 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -14,12 +14,13 @@ #include <stdlib.h> #include <string.h> +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -30,11 +31,9 @@ struct RtpPacket { ForwardErrorCorrection::Packet* pkt; }; -RTPSenderVideo::RTPSenderVideo(const int32_t id, - Clock* clock, +RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender) - : _id(id), - _rtpSender(*rtpSender), + : _rtpSender(*rtpSender), _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()), _videoType(kRtpVideoGeneric), _videoCodecInformation(NULL), @@ -42,7 +41,7 @@ RTPSenderVideo::RTPSenderVideo(const int32_t id, _retransmissionSettings(kRetransmitBaseLayer), // Generic FEC - _fec(id), + _fec(), _fecEnabled(false), _payloadTypeRED(-1), _payloadTypeFEC(-1), @@ -253,8 +252,13 @@ RTPSenderVideo::FECPacketOverhead() const { if (_fecEnabled) { - return ForwardErrorCorrection::PacketOverhead() + - REDForFECHeaderLength; + // Overhead is FEC headers plus RED for FEC header plus anything in RTP + // header beyond the 12 bytes base header (CSRC list, extensions...) + // This reason for the header extensions to be included here is that + // from an FEC viewpoint, they are part of the payload to be protected. + // (The base RTP header is already protected by the FEC header.) + return ForwardErrorCorrection::PacketOverhead() + REDForFECHeaderLength + + (_rtpSender.RTPHeaderLength() - kRtpHeaderSize); } return 0; } @@ -323,8 +327,6 @@ RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType, { return retVal; } - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)", - __FUNCTION__, captureTimeStamp); return 0; } @@ -470,9 +472,9 @@ RTPSenderVideo::SendVP8(const FrameType frameType, rtpHeaderLength, captureTimeStamp, capture_time_ms, storage, protect)) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "RTPSenderVideo::SendVP8 failed to send packet number" - " %d", _rtpSender.SequenceNumber()); + LOG(LS_WARNING) + << "RTPSenderVideo::SendVP8 failed to send packet number " + << _rtpSender.SequenceNumber(); } } TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h index 4c406d75014..daa730e8c24 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h @@ -31,7 +31,7 @@ struct RtpPacket; class RTPSenderVideo { public: - RTPSenderVideo(const int32_t id, Clock* clock, + RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender); virtual ~RTPSenderVideo(); @@ -112,7 +112,6 @@ private: const RTPVideoTypeHeader* rtpTypeHdr); private: - int32_t _id; RTPSenderInterface& _rtpSender; CriticalSectionWrapper* _sendVideoCritsect; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc index 102ebecb02e..c1f3c642749 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc @@ -30,7 +30,7 @@ #endif #include "webrtc/system_wrappers/interface/tick_util.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400)) #define DEBUG_PRINT(...) \ @@ -398,6 +398,10 @@ bool RTPHeaderParser::Parse(RTPHeader& header, header.extension.hasAbsoluteSendTime = false; header.extension.absoluteSendTime = 0; + // May not be present in packet. + header.extension.hasAudioLevel = false; + header.extension.audioLevel = 0; + if (X) { /* RTP header extension, RFC 3550. 0 1 2 3 @@ -453,89 +457,96 @@ void RTPHeaderParser::ParseOneByteExtensionHeader( // | ID | len | // +-+-+-+-+-+-+-+-+ + // Note that 'len' is the header extension element length, which is the + // number of bytes - 1. const uint8_t id = (*ptr & 0xf0) >> 4; const uint8_t len = (*ptr & 0x0f); ptr++; if (id == 15) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, - "Ext id: 15 encountered, parsing terminated."); + LOG(LS_WARNING) + << "RTP extension header 15 encountered. Terminate parsing."; return; } RTPExtensionType type; if (ptrExtensionMap->GetType(id, &type) != 0) { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "Failed to find extension id: %d", id); - return; - } - - switch (type) { - case kRtpExtensionTransmissionTimeOffset: { - if (len != 2) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, - "Incorrect transmission time offset len: %d", len); - return; + // If we encounter an unknown extension, just skip over it. + LOG(LS_WARNING) << "Failed to find extension id: " + << static_cast<int>(id); + } else { + switch (type) { + case kRtpExtensionTransmissionTimeOffset: { + if (len != 2) { + LOG(LS_WARNING) << "Incorrect transmission time offset len: " + << len; + return; + } + // 0 1 2 3 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // | ID | len=2 | transmission offset | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + + int32_t transmissionTimeOffset = ptr[0] << 16; + transmissionTimeOffset += ptr[1] << 8; + transmissionTimeOffset += ptr[2]; + header.extension.transmissionTimeOffset = + transmissionTimeOffset; + if (transmissionTimeOffset & 0x800000) { + // Negative offset, correct sign for Word24 to Word32. + header.extension.transmissionTimeOffset |= 0xFF000000; + } + header.extension.hasTransmissionTimeOffset = true; + break; } - // 0 1 2 3 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // | ID | len=2 | transmission offset | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - - int32_t transmissionTimeOffset = *ptr++ << 16; - transmissionTimeOffset += *ptr++ << 8; - transmissionTimeOffset += *ptr++; - header.extension.transmissionTimeOffset = - transmissionTimeOffset; - if (transmissionTimeOffset & 0x800000) { - // Negative offset, correct sign for Word24 to Word32. - header.extension.transmissionTimeOffset |= 0xFF000000; + case kRtpExtensionAudioLevel: { + if (len != 0) { + LOG(LS_WARNING) << "Incorrect audio level len: " << len; + return; + } + // 0 1 2 3 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // | ID | len=0 |V| level | 0x00 | 0x00 | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // + + // Parse out the fields but only use it for debugging for now. + // const uint8_t V = (*ptr & 0x80) >> 7; + // const uint8_t level = (*ptr & 0x7f); + // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u, + // level=%u", ID, len, V, level); + + header.extension.audioLevel = ptr[0]; + header.extension.hasAudioLevel = true; + break; } - header.extension.hasTransmissionTimeOffset = true; - break; - } - case kRtpExtensionAudioLevel: { - // --- Only used for debugging --- - // 0 1 2 3 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // | ID | len=0 |V| level | 0x00 | 0x00 | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // - - // Parse out the fields but only use it for debugging for now. - // const uint8_t V = (*ptr & 0x80) >> 7; - // const uint8_t level = (*ptr & 0x7f); - // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u, - // level=%u", ID, len, V, level); - break; - } - case kRtpExtensionAbsoluteSendTime: { - if (len != 2) { - WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, - "Incorrect absolute send time len: %d", len); + case kRtpExtensionAbsoluteSendTime: { + if (len != 2) { + LOG(LS_WARNING) << "Incorrect absolute send time len: " << len; + return; + } + // 0 1 2 3 + // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + // | ID | len=2 | absolute send time | + // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + + uint32_t absoluteSendTime = ptr[0] << 16; + absoluteSendTime += ptr[1] << 8; + absoluteSendTime += ptr[2]; + header.extension.absoluteSendTime = absoluteSendTime; + header.extension.hasAbsoluteSendTime = true; + break; + } + default: { + LOG(LS_WARNING) << "Extension type not implemented: " << type; return; } - // 0 1 2 3 - // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - // | ID | len=2 | absolute send time | - // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - - uint32_t absoluteSendTime = *ptr++ << 16; - absoluteSendTime += *ptr++ << 8; - absoluteSendTime += *ptr++; - header.extension.absoluteSendTime = absoluteSendTime; - header.extension.hasAbsoluteSendTime = true; - break; - } - default: { - WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, - "Extension type not implemented."); - return; } } + ptr += (len + 1); uint8_t num_bytes = ParsePaddingBytes(ptrRTPDataExtensionEnd, ptr); ptr += num_bytes; } @@ -556,17 +567,12 @@ uint8_t RTPHeaderParser::ParsePaddingBytes( return num_zero_bytes; } -// RTP payload parser RTPPayloadParser::RTPPayloadParser(const RtpVideoCodecTypes videoType, const uint8_t* payloadData, - uint16_t payloadDataLength, - int32_t id) - : - _id(id), - _dataPtr(payloadData), - _dataLength(payloadDataLength), - _videoType(videoType) { -} + uint16_t payloadDataLength) + : _dataPtr(payloadData), + _dataLength(payloadDataLength), + _videoType(videoType) {} RTPPayloadParser::~RTPPayloadParser() { } @@ -641,8 +647,7 @@ bool RTPPayloadParser::ParseVP8(RTPPayload& parsedPacket) const { } if (dataLength <= 0) { - WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, - "Error parsing VP8 payload descriptor; payload too short"); + LOG(LS_ERROR) << "Error parsing VP8 payload descriptor!"; return false; } diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h index 8002273c374..732301f6fbe 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h @@ -166,8 +166,8 @@ namespace ModuleRTPUtility public: RTPPayloadParser(const RtpVideoCodecTypes payloadType, const uint8_t* payloadData, - const uint16_t payloadDataLength, // Length w/o padding. - const int32_t id); + // Length w/o padding. + const uint16_t payloadDataLength); ~RTPPayloadParser(); @@ -202,7 +202,6 @@ namespace ModuleRTPUtility int dataLength) const; private: - int32_t _id; const uint8_t* _dataPtr; const uint16_t _dataLength; const RtpVideoCodecTypes _videoType; diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc index 02a89fc4fd0..d33eaf4c849 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc @@ -76,7 +76,7 @@ TEST(ParseVP8Test, BasicHeader) { payload[0] = 0x14; // Binary 0001 0100; S = 1, PartID = 4. payload[1] = 0x01; // P frame. - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); @@ -97,7 +97,7 @@ TEST(ParseVP8Test, PictureID) { payload[1] = 0x80; payload[2] = 17; - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); @@ -117,7 +117,7 @@ TEST(ParseVP8Test, PictureID) { // Re-use payload, but change to long PictureID. payload[2] = 0x80 | 17; payload[3] = 17; - RTPPayloadParser rtpPayloadParser2(kRtpVideoVp8, payload, 10, 0); + RTPPayloadParser rtpPayloadParser2(kRtpVideoVp8, payload, 10); ASSERT_TRUE(rtpPayloadParser2.Parse(parsedPacket)); @@ -136,7 +136,7 @@ TEST(ParseVP8Test, Tl0PicIdx) { payload[1] = 0x40; payload[2] = 17; - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 13, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 13); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); @@ -159,7 +159,7 @@ TEST(ParseVP8Test, TIDAndLayerSync) { payload[1] = 0x20; payload[2] = 0x80; // TID(2) + LayerSync(false) - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); @@ -183,7 +183,7 @@ TEST(ParseVP8Test, KeyIdx) { payload[1] = 0x10; // K = 1. payload[2] = 0x11; // KEYIDX = 17 decimal. - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); @@ -209,7 +209,7 @@ TEST(ParseVP8Test, MultipleExtensions) { payload[4] = 42; // Tl0PicIdx. payload[5] = 0x40 | 0x20 | 0x11; // TID(1) + LayerSync(true) + KEYIDX(17). - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); @@ -236,7 +236,7 @@ TEST(ParseVP8Test, TooShortHeader) { payload[2] = 0x80 | 17; // ... but only 2 bytes PictureID is provided. payload[3] = 17; // PictureID, low 8 bits. - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4); RTPPayload parsedPacket; EXPECT_FALSE(rtpPayloadParser.Parse(parsedPacket)); @@ -258,7 +258,7 @@ TEST(ParseVP8Test, TestWithPacketizer) { ASSERT_EQ(0, packetizer.NextPacket(packet, &send_bytes, &last)); ASSERT_TRUE(last); - RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, packet, send_bytes, 0); + RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, packet, send_bytes); RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc index 1e57970c0d0..df09b01bdf2 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc @@ -14,7 +14,6 @@ #include <stdlib.h> #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" #ifdef _WIN32 #include <windows.h> @@ -185,8 +184,6 @@ SSRCDatabase::SSRCDatabase() _ssrcVector = new uint32_t[10]; #endif _critSect = CriticalSectionWrapper::CreateCriticalSection(); - - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s created", __FUNCTION__); } SSRCDatabase::~SSRCDatabase() @@ -197,8 +194,6 @@ SSRCDatabase::~SSRCDatabase() _ssrcMap.clear(); #endif delete _critSect; - - WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s deleted", __FUNCTION__); } uint32_t SSRCDatabase::GenerateRandom() diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h index 2ca5f287afa..cbb1207b8ec 100644 --- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h +++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h @@ -13,8 +13,8 @@ #include <vector> +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/utility/interface/mock/mock_process_thread.h b/chromium/third_party/webrtc/modules/utility/interface/mock/mock_process_thread.h new file mode 100644 index 00000000000..fc0c1fb1ce1 --- /dev/null +++ b/chromium/third_party/webrtc/modules/utility/interface/mock/mock_process_thread.h @@ -0,0 +1,29 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_ +#define WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_ + +#include "webrtc/modules/utility/interface/process_thread.h" + +#include "testing/gmock/include/gmock/gmock.h" + +namespace webrtc { + +class MockProcessThread : public ProcessThread { + public: + MOCK_METHOD0(Start, int32_t()); + MOCK_METHOD0(Stop, int32_t()); + MOCK_METHOD1(RegisterModule, int32_t(Module* module)); + MOCK_METHOD1(DeRegisterModule, int32_t(const Module* module)); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_ diff --git a/chromium/third_party/webrtc/modules/utility/interface/process_thread.h b/chromium/third_party/webrtc/modules/utility/interface/process_thread.h index cdbb4d39127..4db92a308a8 100644 --- a/chromium/third_party/webrtc/modules/utility/interface/process_thread.h +++ b/chromium/third_party/webrtc/modules/utility/interface/process_thread.h @@ -25,7 +25,7 @@ public: virtual int32_t Start() = 0; virtual int32_t Stop() = 0; - virtual int32_t RegisterModule(const Module* module) = 0; + virtual int32_t RegisterModule(Module* module) = 0; virtual int32_t DeRegisterModule(const Module* module) = 0; protected: virtual ~ProcessThread(); diff --git a/chromium/third_party/webrtc/modules/utility/source/OWNERS b/chromium/third_party/webrtc/modules/utility/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/utility/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations.cc b/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations.cc index 18dba52b651..e3b00104761 100644 --- a/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations.cc +++ b/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations.cc @@ -72,7 +72,6 @@ void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) { void AudioFrameOperations::Mute(AudioFrame& frame) { memset(frame.data_, 0, sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_); - frame.energy_ = 0; } int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) { diff --git a/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations_unittest.cc index 34c08a89e6b..f4d881cf871 100644 --- a/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations_unittest.cc +++ b/chromium/third_party/webrtc/modules/utility/source/audio_frame_operations_unittest.cc @@ -142,17 +142,13 @@ TEST_F(AudioFrameOperationsTest, SwapStereoChannelsFailsOnMono) { TEST_F(AudioFrameOperationsTest, MuteSucceeds) { SetFrameData(&frame_, 1000, 1000); - frame_.energy_ = 1000 * 1000 * frame_.samples_per_channel_ * - frame_.num_channels_; AudioFrameOperations::Mute(frame_); AudioFrame muted_frame; muted_frame.samples_per_channel_ = 320; muted_frame.num_channels_ = 2; SetFrameData(&muted_frame, 0, 0); - muted_frame.energy_ = 0; VerifyFramesAreEqual(muted_frame, frame_); - EXPECT_EQ(muted_frame.energy_, frame_.energy_); } // TODO(andrew): should not allow negative scales. diff --git a/chromium/third_party/webrtc/modules/utility/source/file_player_impl.cc b/chromium/third_party/webrtc/modules/utility/source/file_player_impl.cc index 9240e64691e..8049245fb06 100644 --- a/chromium/third_party/webrtc/modules/utility/source/file_player_impl.cc +++ b/chromium/third_party/webrtc/modules/utility/source/file_player_impl.cc @@ -9,7 +9,7 @@ */ #include "webrtc/modules/utility/source/file_player_impl.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #ifdef WEBRTC_MODULE_UTILITY_VIDEO #include "frame_scaler.h" @@ -35,8 +35,6 @@ FilePlayer* FilePlayer::CreateFilePlayer(uint32_t instanceID, #ifdef WEBRTC_MODULE_UTILITY_VIDEO return new VideoFilePlayerImpl(instanceID, fileFormat); #else - WEBRTC_TRACE(kTraceError, kTraceFile, -1, - "Invalid file format: %d", kFileFormatAviFile); assert(false); return NULL; #endif @@ -114,10 +112,9 @@ int32_t FilePlayerImpl::Get10msAudioFromFile( { if(_codec.plfreq == 0) { - WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID, - "FilePlayerImpl::Get10msAudioFromFile() playing not started!\ - codecFreq = %d, wantedFreq = %d", - _codec.plfreq, frequencyInHz); + LOG(LS_WARNING) << "Get10msAudioFromFile() playing not started!" + << " codec freq = " << _codec.plfreq + << ", wanted freq = " << frequencyInHz; return -1; } @@ -175,8 +172,7 @@ int32_t FilePlayerImpl::Get10msAudioFromFile( if(_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_, frequencyInHz, kResamplerSynchronous)) { - WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID, - "FilePlayerImpl::Get10msAudioFromFile() unexpected codec"); + LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec."; // New sampling frequency. Update state. outLen = frequencyInHz / 100; @@ -214,8 +210,7 @@ int32_t FilePlayerImpl::SetAudioScaling(float scaleFactor) _scaling = scaleFactor; return 0; } - WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID, - "FilePlayerImpl::SetAudioScaling() not allowed scale factor"); + LOG(LS_WARNING) << "SetAudioScaling() non-allowed scale factor."; return -1; } @@ -255,9 +250,8 @@ int32_t FilePlayerImpl::StartPlayingFile(const char* fileName, codecInstL16.pacsize = 160; } else { - WEBRTC_TRACE(kTraceError, kTraceVoice, _instanceID, - "FilePlayerImpl::StartPlayingFile() sample frequency\ - specifed not supported for PCM format."); + LOG(LS_ERROR) << "StartPlayingFile() sample frequency not " + << "supported for PCM format."; return -1; } @@ -266,12 +260,8 @@ int32_t FilePlayerImpl::StartPlayingFile(const char* fileName, startPosition, stopPosition) == -1) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() failed to initialize file\ - %s playout.", fileName); + LOG(LS_WARNING) << "StartPlayingFile() failed to initialize " + << "pcm file " << fileName; return -1; } SetAudioScaling(volumeScaling); @@ -280,13 +270,8 @@ int32_t FilePlayerImpl::StartPlayingFile(const char* fileName, if (_fileModule.StartPlayingAudioFile(fileName, notification, loop, _fileFormat, codecInst) == -1) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingPreEncodedFile() failed to\ - initialize pre-encoded file %s playout.", - fileName); + LOG(LS_WARNING) << "StartPlayingFile() failed to initialize " + << "pre-encoded file " << fileName; return -1; } } else @@ -297,12 +282,8 @@ int32_t FilePlayerImpl::StartPlayingFile(const char* fileName, startPosition, stopPosition) == -1) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() failed to initialize file\ - %s playout.", fileName); + LOG(LS_WARNING) << "StartPlayingFile() failed to initialize file " + << fileName; return -1; } SetAudioScaling(volumeScaling); @@ -350,12 +331,8 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream, codecInstL16.pacsize = 160; }else { - WEBRTC_TRACE( - kTraceError, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() sample frequency specifed\ - not supported for PCM format."); + LOG(LS_ERROR) << "StartPlayingFile() sample frequency not " + << "supported for PCM format."; return -1; } if (_fileModule.StartPlayingAudioStream(sourceStream, notification, @@ -363,12 +340,8 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream, startPosition, stopPosition) == -1) { - WEBRTC_TRACE( - kTraceError, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() failed to initialize stream\ - playout."); + LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream " + << "playout."; return -1; } @@ -377,12 +350,8 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream, if (_fileModule.StartPlayingAudioStream(sourceStream, notification, _fileFormat, codecInst) == -1) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() failed to initialize stream\ - playout."); + LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream " + << "playout."; return -1; } } else { @@ -392,9 +361,8 @@ int32_t FilePlayerImpl::StartPlayingFile(InStream& sourceStream, startPosition, stopPosition) == -1) { - WEBRTC_TRACE(kTraceError, kTraceVoice, _instanceID, - "FilePlayerImpl::StartPlayingFile() failed to initialize\ - stream playout."); + LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream " + << "playout."; return -1; } } @@ -430,23 +398,14 @@ int32_t FilePlayerImpl::SetUpAudioDecoder() { if ((_fileModule.codec_info(_codec) == -1)) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() failed to retrieve Codec info\ - of file data."); + LOG(LS_WARNING) << "Failed to retrieve codec info of file data."; return -1; } if( STR_CASE_CMP(_codec.plname, "L16") != 0 && _audioDecoder.SetDecodeCodec(_codec,AMRFileStorage) == -1) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FilePlayerImpl::StartPlayingFile() codec %s not supported", - _codec.plname); + LOG(LS_WARNING) << "SetUpAudioDecoder() codec " << _codec.plname + << " not supported."; return -1; } _numberOf10MsPerFrame = _codec.pacsize / (_codec.plfreq / 100); @@ -458,7 +417,7 @@ int32_t FilePlayerImpl::SetUpAudioDecoder() VideoFilePlayerImpl::VideoFilePlayerImpl(uint32_t instanceID, FileFormats fileFormat) : FilePlayerImpl(instanceID, fileFormat), - video_decoder_(new VideoCoder(instanceID)), + video_decoder_(new VideoCoder()), video_codec_info_(), _decodedVideoFrames(0), _encodedData(*new EncodedVideoData()), @@ -522,7 +481,7 @@ int32_t VideoFilePlayerImpl::StopPlayingFile() CriticalSectionScoped lock( _critSec); _decodedVideoFrames = 0; - video_decoder_.reset(new VideoCoder(_instanceID)); + video_decoder_.reset(new VideoCoder()); return FilePlayerImpl::StopPlayingFile(); } @@ -627,12 +586,7 @@ int32_t VideoFilePlayerImpl::TimeUntilNextVideoFrame() reinterpret_cast< int8_t*>(_encodedData.payloadData), encodedBufferLengthInBytes) != 0) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVideo, - _instanceID, - "FilePlayerImpl::TimeUntilNextVideoFrame() error reading\ - video data"); + LOG(LS_WARNING) << "Error reading video data."; return -1; } _encodedData.payloadSize = encodedBufferLengthInBytes; @@ -685,23 +639,16 @@ int32_t VideoFilePlayerImpl::SetUpVideoDecoder() { if (_fileModule.VideoCodecInst(video_codec_info_) != 0) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVideo, - _instanceID, - "FilePlayerImpl::SetVideoDecoder() failed to retrieve Codec info of\ - file data."); + LOG(LS_WARNING) << "SetVideoDecoder() failed to retrieve codec info of " + << "file data."; return -1; } int32_t useNumberOfCores = 1; if (video_decoder_->SetDecodeCodec(video_codec_info_, useNumberOfCores) != 0) { - WEBRTC_TRACE(kTraceWarning, - kTraceVideo, - _instanceID, - "FilePlayerImpl::SetUpVideoDecoder() codec %s not supported", - video_codec_info_.plName); + LOG(LS_WARNING) << "SetUpVideoDecoder() codec " + << video_codec_info_.plName << " not supported."; return -1; } diff --git a/chromium/third_party/webrtc/modules/utility/source/file_player_unittests.cc b/chromium/third_party/webrtc/modules/utility/source/file_player_unittests.cc new file mode 100644 index 00000000000..d430d9f59ad --- /dev/null +++ b/chromium/third_party/webrtc/modules/utility/source/file_player_unittests.cc @@ -0,0 +1,106 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Unit tests for FilePlayer. + +#include "webrtc/modules/utility/interface/file_player.h" + +#include <stdio.h> +#include <string> + +#include "gflags/gflags.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/md5digest.h" +#include "webrtc/base/stringencode.h" +#include "webrtc/test/testsupport/fileutils.h" + +DEFINE_bool(file_player_output, false, "Generate reference files."); + +namespace webrtc { + +class FilePlayerTest : public ::testing::Test { + protected: + static const uint32_t kId = 0; + static const FileFormats kFileFormat = kFileFormatWavFile; + static const int kSampleRateHz = 8000; + + FilePlayerTest() + : player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)), + output_file_(NULL) {} + + virtual void SetUp() OVERRIDE { + if (FLAGS_file_player_output) { + std::string output_file = + webrtc::test::OutputPath() + "file_player_unittest_out.pcm"; + output_file_ = fopen(output_file.c_str(), "wb"); + ASSERT_TRUE(output_file_ != NULL); + } + } + + virtual void TearDown() OVERRIDE { + if (output_file_) + fclose(output_file_); + } + + ~FilePlayerTest() { FilePlayer::DestroyFilePlayer(player_); } + + void PlayFileAndCheck(const std::string& input_file, + const std::string& ref_checksum, + int output_length_ms) { + const float kScaling = 1; + ASSERT_EQ(0, + player_->StartPlayingFile( + input_file.c_str(), false, 0, kScaling, 0, 0, NULL)); + rtc::Md5Digest checksum; + for (int i = 0; i < output_length_ms / 10; ++i) { + int16_t out[10 * kSampleRateHz / 1000] = {0}; + int num_samples; + EXPECT_EQ(0, + player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz)); + checksum.Update(out, num_samples * sizeof(out[0])); + if (FLAGS_file_player_output) { + ASSERT_EQ(static_cast<size_t>(num_samples), + fwrite(out, sizeof(out[0]), num_samples, output_file_)); + } + } + char checksum_result[rtc::Md5Digest::kSize]; + EXPECT_EQ(rtc::Md5Digest::kSize, + checksum.Finish(checksum_result, rtc::Md5Digest::kSize)); + EXPECT_EQ(ref_checksum, + rtc::hex_encode(checksum_result, sizeof(checksum_result))); + } + + FilePlayer* player_; + FILE* output_file_; +}; + +TEST_F(FilePlayerTest, PlayWavPcmuFile) { + const std::string kFileName = + test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav"); + // The file is longer than this, but keeping the output shorter limits the + // runtime for the test. + const int kOutputLengthMs = 10000; + const std::string kRefChecksum = "c74e7fd432d439b1311e1c16815b3e9a"; + + PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs); +} + +TEST_F(FilePlayerTest, PlayWavPcm16File) { + const std::string kFileName = + test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav"); + // The file is longer than this, but keeping the output shorter limits the + // runtime for the test. + const int kOutputLengthMs = 10000; + const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71"; + + PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc index 16faa58d3fc..264b867a259 100644 --- a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc +++ b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc @@ -12,7 +12,7 @@ #include "webrtc/engine_configurations.h" #include "webrtc/modules/media_file/interface/media_file.h" #include "webrtc/modules/utility/source/file_recorder_impl.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #ifdef WEBRTC_MODULE_UTILITY_VIDEO #include "critical_section_wrapper.h" @@ -38,8 +38,6 @@ FileRecorder* FileRecorder::CreateFileRecorder(uint32_t instanceID, #ifdef WEBRTC_MODULE_UTILITY_VIDEO return new AviRecorder(instanceID, fileFormat); #else - WEBRTC_TRACE(kTraceError, kTraceFile, -1, - "Invalid file format: %d", kFileFormatAviFile); assert(false); return NULL; #endif @@ -115,13 +113,8 @@ int32_t FileRecorderImpl::StartRecordingAudioFile( } if( retVal != 0) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FileRecorder::StartRecording() failed to initialize file %s for\ - recording.", - fileName); + LOG(LS_WARNING) << "Failed to initialize file " << fileName + << " for recording."; if(IsRecording()) { @@ -152,12 +145,7 @@ int32_t FileRecorderImpl::StartRecordingAudioFile( } if( retVal != 0) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FileRecorder::StartRecording() failed to initialize outStream for\ - recording."); + LOG(LS_WARNING) << "Failed to initialize outStream for recording."; if(IsRecording()) { @@ -184,12 +172,8 @@ int32_t FileRecorderImpl::RecordAudioToFile( { if (codec_info_.plfreq == 0) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FileRecorder::RecordAudioToFile() recording audio is not turned\ - on"); + LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not " + << "turned on."; return -1; } AudioFrame tempAudioFrame; @@ -250,13 +234,9 @@ int32_t FileRecorderImpl::RecordAudioToFile( if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, encodedLenInBytes) == -1) { - WEBRTC_TRACE( - kTraceWarning, - kTraceVoice, - _instanceID, - "FileRecorder::RecordAudioToFile() codec %s not supported or\ - failed to encode stream", - codec_info_.plname); + LOG(LS_WARNING) << "RecordAudioToFile() codec " + << codec_info_.plname + << " not supported or failed to encode stream."; return -1; } } else { @@ -309,12 +289,8 @@ int32_t FileRecorderImpl::SetUpAudioEncoder() { if(_audioEncoder.SetEncodeCodec(codec_info_,_amrFormat) == -1) { - WEBRTC_TRACE( - kTraceError, - kTraceVoice, - _instanceID, - "FileRecorder::StartRecording() codec %s not supported", - codec_info_.plname); + LOG(LS_ERROR) << "SetUpAudioEncoder() codec " + << codec_info_.plname << " not supported."; return -1; } } @@ -342,31 +318,6 @@ int32_t FileRecorderImpl::WriteEncodedAudioData( #ifdef WEBRTC_MODULE_UTILITY_VIDEO -class AudioFrameFileInfo -{ - public: - AudioFrameFileInfo(const int8_t* audioData, - const uint16_t audioSize, - const uint16_t audioMS, - const TickTime& playoutTS) - : _audioData(), _audioSize(audioSize), _audioMS(audioMS), - _playoutTS(playoutTS) - { - if(audioSize > MAX_AUDIO_BUFFER_IN_BYTES) - { - assert(false); - _audioSize = 0; - return; - } - memcpy(_audioData, audioData, audioSize); - }; - // TODO (hellner): either turn into a struct or provide get/set functions. - int8_t _audioData[MAX_AUDIO_BUFFER_IN_BYTES]; - uint16_t _audioSize; - uint16_t _audioMS; - TickTime _playoutTS; -}; - AviRecorder::AviRecorder(uint32_t instanceID, FileFormats fileFormat) : FileRecorderImpl(instanceID, fileFormat), _videoOnly(false), @@ -377,7 +328,7 @@ AviRecorder::AviRecorder(uint32_t instanceID, FileFormats fileFormat) _writtenAudioMS(0), _writtenVideoMS(0) { - _videoEncoder = new VideoCoder(instanceID); + _videoEncoder = new VideoCoder(); _frameScaler = new FrameScaler(); _videoFramesQueue = new VideoFramesQueue(); _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority, @@ -545,49 +496,39 @@ int32_t AviRecorder::ProcessAudio() { // Syncronize audio to the current frame to process by throwing away // audio samples with older timestamp than the video frame. - uint32_t numberOfAudioElements = - _audioFramesToWrite.GetSize(); - for (uint32_t i = 0; i < numberOfAudioElements; ++i) + size_t numberOfAudioElements = + _audioFramesToWrite.size(); + for (size_t i = 0; i < numberOfAudioElements; ++i) { - AudioFrameFileInfo* frameInfo = - (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem(); - if(frameInfo) + AudioFrameFileInfo* frameInfo = _audioFramesToWrite.front(); + if(TickTime::TicksToMilliseconds( + frameInfo->_playoutTS.Ticks()) < + frameToProcess->render_time_ms()) + { + delete frameInfo; + _audioFramesToWrite.pop_front(); + } else { - if(TickTime::TicksToMilliseconds( - frameInfo->_playoutTS.Ticks()) < - frameToProcess->render_time_ms()) - { - delete frameInfo; - _audioFramesToWrite.PopFront(); - } else - { - break; - } + break; } } } } // Write all audio up to current timestamp. int32_t error = 0; - uint32_t numberOfAudioElements = _audioFramesToWrite.GetSize(); - for (uint32_t i = 0; i < numberOfAudioElements; ++i) + size_t numberOfAudioElements = _audioFramesToWrite.size(); + for (size_t i = 0; i < numberOfAudioElements; ++i) { - AudioFrameFileInfo* frameInfo = - (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem(); - if(frameInfo) + AudioFrameFileInfo* frameInfo = _audioFramesToWrite.front(); + if((TickTime::Now() - frameInfo->_playoutTS).Milliseconds() > 0) { - if((TickTime::Now() - frameInfo->_playoutTS).Milliseconds() > 0) - { - _moduleFile->IncomingAudioData(frameInfo->_audioData, - frameInfo->_audioSize); - _writtenAudioMS += frameInfo->_audioMS; - delete frameInfo; - _audioFramesToWrite.PopFront(); - } else { - break; - } + _moduleFile->IncomingAudioData(frameInfo->_audioData, + frameInfo->_audioSize); + _writtenAudioMS += frameInfo->_audioMS; + delete frameInfo; + _audioFramesToWrite.pop_front(); } else { - _audioFramesToWrite.PopFront(); + break; } } return error; @@ -635,8 +576,8 @@ bool AviRecorder::Process() error = EncodeAndWriteVideoToFile( *frameToProcess); if( error != 0) { - WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, - "AviRecorder::Process() error writing to file."); + LOG(LS_ERROR) << "AviRecorder::Process() error writing to " + << "file."; break; } else { uint32_t frameLengthMS = 1000 / @@ -675,8 +616,7 @@ bool AviRecorder::Process() error = EncodeAndWriteVideoToFile( *frameToProcess); if(error != 0) { - WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, - "AviRecorder::Process() error writing to file."); + LOG(LS_ERROR) << "AviRecorder::Process() error writing to file."; } else { _writtenVideoMS += frameLengthMS; } @@ -727,17 +667,12 @@ int32_t AviRecorder::EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame) (int8_t*)(_videoEncodedData.payloadData), _videoEncodedData.payloadSize)) { - WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, - "Error writing AVI file"); + LOG(LS_ERROR) << "Error writing AVI file."; return -1; } } else { - WEBRTC_TRACE( - kTraceError, - kTraceVideo, - _instanceID, - "FileRecorder::RecordVideoToFile() frame dropped by encoder bitrate\ - likely to low."); + LOG(LS_ERROR) << "FileRecorder::RecordVideoToFile() frame dropped by " + << "encoder, bitrate likely too low."; } return 0; } @@ -750,6 +685,8 @@ int32_t AviRecorder::WriteEncodedAudioData( uint16_t millisecondsOfData, const TickTime* playoutTS) { + CriticalSectionScoped lock(_critSec); + if (!IsRecording()) { return -1; @@ -762,7 +699,7 @@ int32_t AviRecorder::WriteEncodedAudioData( { return -1; } - if (_audioFramesToWrite.GetSize() > kMaxAudioBufferQueueLength) + if (_audioFramesToWrite.size() > kMaxAudioBufferQueueLength) { StopRecording(); return -1; @@ -771,15 +708,15 @@ int32_t AviRecorder::WriteEncodedAudioData( if(playoutTS) { - _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer, - bufferLength, - millisecondsOfData, - *playoutTS)); + _audioFramesToWrite.push_back(new AudioFrameFileInfo(audioBuffer, + bufferLength, + millisecondsOfData, + *playoutTS)); } else { - _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer, - bufferLength, - millisecondsOfData, - TickTime::Now())); + _audioFramesToWrite.push_back(new AudioFrameFileInfo(audioBuffer, + bufferLength, + millisecondsOfData, + TickTime::Now())); } _timeEvent.Set(); return 0; diff --git a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h index 0b7290eddce..53fd26bc25d 100644 --- a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h +++ b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h @@ -15,6 +15,8 @@ #ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ #define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ +#include <list> + #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" @@ -40,6 +42,8 @@ enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32}; enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2}; enum { kMaxAudioBufferQueueLength = 100 }; +class CriticalSectionWrapper; + class FileRecorderImpl : public FileRecorder { public: @@ -103,6 +107,31 @@ private: #ifdef WEBRTC_MODULE_UTILITY_VIDEO +class AudioFrameFileInfo +{ + public: + AudioFrameFileInfo(const int8_t* audioData, + const uint16_t audioSize, + const uint16_t audioMS, + const TickTime& playoutTS) + : _audioData(), _audioSize(audioSize), _audioMS(audioMS), + _playoutTS(playoutTS) + { + if(audioSize > MAX_AUDIO_BUFFER_IN_BYTES) + { + assert(false); + _audioSize = 0; + return; + } + memcpy(_audioData, audioData, audioSize); + }; + // TODO (hellner): either turn into a struct or provide get/set functions. + int8_t _audioData[MAX_AUDIO_BUFFER_IN_BYTES]; + uint16_t _audioSize; + uint16_t _audioMS; + TickTime _playoutTS; +}; + class AviRecorder : public FileRecorderImpl { public: @@ -126,6 +155,7 @@ protected: uint16_t millisecondsOfData, const TickTime* playoutTS); private: + typedef std::list<AudioFrameFileInfo*> AudioInfoList; static bool Run(ThreadObj threadObj); bool Process(); @@ -141,7 +171,7 @@ private: VideoCodec _videoCodecInst; bool _videoOnly; - ListWrapper _audioFramesToWrite; + AudioInfoList _audioFramesToWrite; bool _firstAudioFrameReceived; VideoFramesQueue* _videoFramesQueue; diff --git a/chromium/third_party/webrtc/modules/utility/source/frame_scaler.cc b/chromium/third_party/webrtc/modules/utility/source/frame_scaler.cc index ed127a6715a..50ccf8adc67 100644 --- a/chromium/third_party/webrtc/modules/utility/source/frame_scaler.cc +++ b/chromium/third_party/webrtc/modules/utility/source/frame_scaler.cc @@ -13,7 +13,6 @@ #ifdef WEBRTC_MODULE_UTILITY_VIDEO #include "webrtc/common_video/libyuv/include/scaler.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { diff --git a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc index 08979d24935..bf7db3bc846 100644 --- a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc +++ b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc @@ -10,7 +10,7 @@ #include "webrtc/modules/interface/module.h" #include "webrtc/modules/utility/source/process_thread_impl.h" -#include "webrtc/system_wrappers/interface/trace.h" + namespace webrtc { ProcessThread::~ProcessThread() @@ -32,14 +32,12 @@ ProcessThreadImpl::ProcessThreadImpl() _critSectModules(CriticalSectionWrapper::CreateCriticalSection()), _thread(NULL) { - WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s created", __FUNCTION__); } ProcessThreadImpl::~ProcessThreadImpl() { delete _critSectModules; delete &_timeEvent; - WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__); } int32_t ProcessThreadImpl::Start() @@ -87,25 +85,21 @@ int32_t ProcessThreadImpl::Stop() return 0; } -int32_t ProcessThreadImpl::RegisterModule(const Module* module) +int32_t ProcessThreadImpl::RegisterModule(Module* module) { CriticalSectionScoped lock(_critSectModules); // Only allow module to be registered once. - ListItem* item = _modules.First(); - for(uint32_t i = 0; i < _modules.GetSize() && item; i++) - { - if(module == item->GetItem()) + for (ModuleList::iterator iter = _modules.begin(); + iter != _modules.end(); ++iter) { + if(module == *iter) { return -1; } - item = _modules.Next(item); } - _modules.PushFront(module); - WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, - "number of registered modules has increased to %d", - _modules.GetSize()); + _modules.push_front(module); + // Wake the thread calling ProcessThreadImpl::Process() to update the // waiting time. The waiting time for the just registered module may be // shorter than all other registered modules. @@ -116,19 +110,13 @@ int32_t ProcessThreadImpl::RegisterModule(const Module* module) int32_t ProcessThreadImpl::DeRegisterModule(const Module* module) { CriticalSectionScoped lock(_critSectModules); - - ListItem* item = _modules.First(); - for(uint32_t i = 0; i < _modules.GetSize() && item; i++) - { - if(module == item->GetItem()) + for (ModuleList::iterator iter = _modules.begin(); + iter != _modules.end(); ++iter) { + if(module == *iter) { - int res = _modules.Erase(item); - WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, - "number of registered modules has decreased to %d", - _modules.GetSize()); - return res; + _modules.erase(iter); + return 0; } - item = _modules.Next(item); } return -1; } @@ -145,16 +133,13 @@ bool ProcessThreadImpl::Process() int32_t minTimeToNext = 100; { CriticalSectionScoped lock(_critSectModules); - ListItem* item = _modules.First(); - for(uint32_t i = 0; i < _modules.GetSize() && item; i++) - { - int32_t timeToNext = - static_cast<Module*>(item->GetItem())->TimeUntilNextProcess(); + for (ModuleList::iterator iter = _modules.begin(); + iter != _modules.end(); ++iter) { + int32_t timeToNext = (*iter)->TimeUntilNextProcess(); if(minTimeToNext > timeToNext) { minTimeToNext = timeToNext; } - item = _modules.Next(item); } } @@ -172,16 +157,13 @@ bool ProcessThreadImpl::Process() } { CriticalSectionScoped lock(_critSectModules); - ListItem* item = _modules.First(); - for(uint32_t i = 0; i < _modules.GetSize() && item; i++) - { - int32_t timeToNext = - static_cast<Module*>(item->GetItem())->TimeUntilNextProcess(); + for (ModuleList::iterator iter = _modules.begin(); + iter != _modules.end(); ++iter) { + int32_t timeToNext = (*iter)->TimeUntilNextProcess(); if(timeToNext < 1) { - static_cast<Module*>(item->GetItem())->Process(); + (*iter)->Process(); } - item = _modules.Next(item); } } return true; diff --git a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h index d1913c47c49..14fbc18a2a5 100644 --- a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h +++ b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h @@ -11,10 +11,11 @@ #ifndef WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_ #define WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_ +#include <list> + #include "webrtc/modules/utility/interface/process_thread.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h" -#include "webrtc/system_wrappers/interface/list_wrapper.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h" #include "webrtc/typedefs.h" @@ -28,7 +29,7 @@ public: virtual int32_t Start(); virtual int32_t Stop(); - virtual int32_t RegisterModule(const Module* module); + virtual int32_t RegisterModule(Module* module); virtual int32_t DeRegisterModule(const Module* module); protected: @@ -37,9 +38,10 @@ protected: bool Process(); private: + typedef std::list<Module*> ModuleList; EventWrapper& _timeEvent; CriticalSectionWrapper* _critSectModules; - ListWrapper _modules; + ModuleList _modules; ThreadWrapper* _thread; }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/utility/source/rtp_dump_impl.cc b/chromium/third_party/webrtc/modules/utility/source/rtp_dump_impl.cc index 39316f47858..547df332f9f 100644 --- a/chromium/third_party/webrtc/modules/utility/source/rtp_dump_impl.cc +++ b/chromium/third_party/webrtc/modules/utility/source/rtp_dump_impl.cc @@ -14,7 +14,7 @@ #include <stdio.h> #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #if defined(_WIN32) #include <Windows.h> @@ -71,7 +71,6 @@ RtpDumpImpl::RtpDumpImpl() _file(*FileWrapper::Create()), _startTime(0) { - WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s created", __FUNCTION__); } RtpDump::~RtpDump() @@ -84,7 +83,6 @@ RtpDumpImpl::~RtpDumpImpl() _file.CloseFile(); delete &_file; delete _critSect; - WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__); } int32_t RtpDumpImpl::Start(const char* fileNameUTF8) @@ -100,8 +98,7 @@ int32_t RtpDumpImpl::Start(const char* fileNameUTF8) _file.CloseFile(); if (_file.OpenFile(fileNameUTF8, false, false, false) == -1) { - WEBRTC_TRACE(kTraceError, kTraceUtility, -1, - "failed to open the specified file"); + LOG(LS_ERROR) << "Failed to open file."; return -1; } @@ -113,8 +110,7 @@ int32_t RtpDumpImpl::Start(const char* fileNameUTF8) sprintf(magic, "#!rtpplay%s \n", RTPFILE_VERSION); if (_file.WriteText(magic) == -1) { - WEBRTC_TRACE(kTraceError, kTraceUtility, -1, - "error writing to file"); + LOG(LS_ERROR) << "Error writing to file."; return -1; } @@ -129,8 +125,7 @@ int32_t RtpDumpImpl::Start(const char* fileNameUTF8) memset(dummyHdr, 0, 16); if (!_file.Write(dummyHdr, sizeof(dummyHdr))) { - WEBRTC_TRACE(kTraceError, kTraceUtility, -1, - "error writing to file"); + LOG(LS_ERROR) << "Error writing to file."; return -1; } return 0; @@ -198,14 +193,12 @@ int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength) if (!_file.Write(&hdr, sizeof(hdr))) { - WEBRTC_TRACE(kTraceError, kTraceUtility, -1, - "error writing to file"); + LOG(LS_ERROR) << "Error writing to file."; return -1; } if (!_file.Write(packet, packetLength)) { - WEBRTC_TRACE(kTraceError, kTraceUtility, -1, - "error writing to file"); + LOG(LS_ERROR) << "Error writing to file."; return -1; } diff --git a/chromium/third_party/webrtc/modules/utility/source/video_coder.cc b/chromium/third_party/webrtc/modules/utility/source/video_coder.cc index 267ed810489..5096acecd25 100644 --- a/chromium/third_party/webrtc/modules/utility/source/video_coder.cc +++ b/chromium/third_party/webrtc/modules/utility/source/video_coder.cc @@ -13,10 +13,7 @@ #include "webrtc/modules/utility/source/video_coder.h" namespace webrtc { -VideoCoder::VideoCoder(uint32_t instanceID) - : _vcm(VideoCodingModule::Create(instanceID)), - _decodedVideo(0) -{ +VideoCoder::VideoCoder() : _vcm(VideoCodingModule::Create()), _decodedVideo(0) { _vcm->InitializeSender(); _vcm->InitializeReceiver(); diff --git a/chromium/third_party/webrtc/modules/utility/source/video_coder.h b/chromium/third_party/webrtc/modules/utility/source/video_coder.h index cb8bfa5a182..8e4344be16b 100644 --- a/chromium/third_party/webrtc/modules/utility/source/video_coder.h +++ b/chromium/third_party/webrtc/modules/utility/source/video_coder.h @@ -20,7 +20,7 @@ namespace webrtc { class VideoCoder : public VCMPacketizationCallback, public VCMReceiveCallback { public: - VideoCoder(uint32_t instanceID); + VideoCoder(); ~VideoCoder(); int32_t SetEncodeCodec(VideoCodec& videoCodecInst, diff --git a/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.cc b/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.cc index d3d37bec2d0..9ade8b51a49 100644 --- a/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.cc +++ b/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.cc @@ -16,80 +16,48 @@ #include "webrtc/common_video/interface/texture_video_frame.h" #include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/tick_util.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { VideoFramesQueue::VideoFramesQueue() - : _incomingFrames(), - _renderDelayMs(10) + : _renderDelayMs(10) { } VideoFramesQueue::~VideoFramesQueue() { - while (!_incomingFrames.Empty()) { - ListItem* item = _incomingFrames.First(); - if (item) { - I420VideoFrame* ptrFrame = static_cast<I420VideoFrame*>(item->GetItem()); - assert(ptrFrame != NULL); - delete ptrFrame; - } - _incomingFrames.Erase(item); + for (FrameList::iterator iter = _incomingFrames.begin(); + iter != _incomingFrames.end(); ++iter) { + delete *iter; } - while (!_emptyFrames.Empty()) { - ListItem* item = _emptyFrames.First(); - if (item) { - I420VideoFrame* ptrFrame = - static_cast<I420VideoFrame*>(item->GetItem()); - assert(ptrFrame != NULL); - delete ptrFrame; - } - _emptyFrames.Erase(item); + for (FrameList::iterator iter = _emptyFrames.begin(); + iter != _emptyFrames.end(); ++iter) { + delete *iter; } } int32_t VideoFramesQueue::AddFrame(const I420VideoFrame& newFrame) { if (newFrame.native_handle() != NULL) { - _incomingFrames.PushBack(new TextureVideoFrame( - static_cast<NativeHandle*>(newFrame.native_handle()), - newFrame.width(), - newFrame.height(), - newFrame.timestamp(), - newFrame.render_time_ms())); + _incomingFrames.push_back(newFrame.CloneFrame()); return 0; } I420VideoFrame* ptrFrameToAdd = NULL; // Try to re-use a VideoFrame. Only allocate new memory if it is necessary. - if (!_emptyFrames.Empty()) { - ListItem* item = _emptyFrames.First(); - if (item) { - ptrFrameToAdd = static_cast<I420VideoFrame*>(item->GetItem()); - _emptyFrames.Erase(item); - } + if (!_emptyFrames.empty()) { + ptrFrameToAdd = _emptyFrames.front(); + _emptyFrames.pop_front(); } if (!ptrFrameToAdd) { - if (_emptyFrames.GetSize() + _incomingFrames.GetSize() > + if (_emptyFrames.size() + _incomingFrames.size() > KMaxNumberOfFrames) { - WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1, - "%s: too many frames, limit: %d", __FUNCTION__, - KMaxNumberOfFrames); + LOG(LS_WARNING) << "Too many frames, limit: " << KMaxNumberOfFrames; return -1; } - - WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, -1, - "%s: allocating buffer %d", __FUNCTION__, - _emptyFrames.GetSize() + _incomingFrames.GetSize()); - ptrFrameToAdd = new I420VideoFrame(); - if (!ptrFrameToAdd) { - WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1, - "%s: could not create new frame for", __FUNCTION__); - return -1; - } } ptrFrameToAdd->CopyFrame(newFrame); - _incomingFrames.PushBack(ptrFrameToAdd); + _incomingFrames.push_back(ptrFrameToAdd); return 0; } @@ -99,20 +67,18 @@ int32_t VideoFramesQueue::AddFrame(const I420VideoFrame& newFrame) { // Recycle all frames that are older than the most recent frame. I420VideoFrame* VideoFramesQueue::FrameToRecord() { I420VideoFrame* ptrRenderFrame = NULL; - ListItem* item = _incomingFrames.First(); - while(item) { - I420VideoFrame* ptrOldestFrameInList = - static_cast<I420VideoFrame*>(item->GetItem()); + for (FrameList::iterator iter = _incomingFrames.begin(); + iter != _incomingFrames.end(); ++iter) { + I420VideoFrame* ptrOldestFrameInList = *iter; if (ptrOldestFrameInList->render_time_ms() <= TickTime::MillisecondTimestamp() + _renderDelayMs) { + // List is traversed beginning to end. If ptrRenderFrame is not + // NULL it must be the first, and thus oldest, VideoFrame in the + // queue. It can be recycled. if (ptrRenderFrame) { - // List is traversed beginning to end. If ptrRenderFrame is not - // NULL it must be the first, and thus oldest, VideoFrame in the - // queue. It can be recycled. ReturnFrame(ptrRenderFrame); - _incomingFrames.PopFront(); + _incomingFrames.pop_front(); } - item = _incomingFrames.Next(item); ptrRenderFrame = ptrOldestFrameInList; } else { // All VideoFrames following this one will be even newer. No match @@ -131,7 +97,7 @@ int32_t VideoFramesQueue::ReturnFrame(I420VideoFrame* ptrOldFrame) { ptrOldFrame->set_height(0); ptrOldFrame->set_render_time_ms(0); ptrOldFrame->ResetSize(); - _emptyFrames.PushBack(ptrOldFrame); + _emptyFrames.push_back(ptrOldFrame); } else { delete ptrOldFrame; } diff --git a/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.h b/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.h index 4316bf7c047..afc64d9b71e 100644 --- a/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.h +++ b/chromium/third_party/webrtc/modules/utility/source/video_frames_queue.h @@ -13,9 +13,10 @@ #ifdef WEBRTC_MODULE_UTILITY_VIDEO +#include <list> + #include "webrtc/common_video/interface/i420_video_frame.h" #include "webrtc/engine_configurations.h" -#include "webrtc/system_wrappers/interface/list_wrapper.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -42,6 +43,7 @@ class VideoFramesQueue { int32_t ReturnFrame(I420VideoFrame* ptrOldFrame); private: + typedef std::list<I420VideoFrame*> FrameList; // Don't allow the buffer to expand beyond KMaxNumberOfFrames VideoFrames. // 300 frames correspond to 10 seconds worth of frames at 30 fps. enum {KMaxNumberOfFrames = 300}; @@ -49,9 +51,9 @@ class VideoFramesQueue { // List of VideoFrame pointers. The list is sorted in the order of when the // VideoFrame was inserted into the list. The first VideoFrame in the list // was inserted first. - ListWrapper _incomingFrames; + FrameList _incomingFrames; // A list of frames that are free to be re-used. - ListWrapper _emptyFrames; + FrameList _emptyFrames; // Estimated render delay. uint32_t _renderDelayMs; diff --git a/chromium/third_party/webrtc/modules/video_capture/OWNERS b/chromium/third_party/webrtc/modules/video_capture/OWNERS index 3b02126feaa..fdc2a3ff702 100644 --- a/chromium/third_party/webrtc/modules/video_capture/OWNERS +++ b/chromium/third_party/webrtc/modules/video_capture/OWNERS @@ -1,5 +1,13 @@ fischman@webrtc.org +glaznev@webrtc.org mallinath@webrtc.org mflodman@webrtc.org perkj@webrtc.org wu@webrtc.org + +per-file *.isolate=kjellander@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.cc b/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.cc index 10c277eeb7a..4a80fe27229 100644 --- a/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.cc +++ b/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.cc @@ -21,36 +21,40 @@ #include "webrtc/system_wrappers/interface/ref_count.h" #include "webrtc/system_wrappers/interface/trace.h" -namespace webrtc -{ +namespace webrtc { -namespace videocapturemodule -{ +namespace videocapturemodule { -static std::string ResolutionsToString( - const std::vector<std::pair<int, int> >& pairs) { +// Helper for storing lists of pairs of ints. Used e.g. for resolutions & FPS +// ranges. +typedef std::pair<int, int> IntPair; +typedef std::vector<IntPair> IntPairs; + +static std::string IntPairsToString(const IntPairs& pairs, char separator) { std::stringstream stream; for (size_t i = 0; i < pairs.size(); ++i) { if (i > 0) stream << ", "; - stream << "(" << pairs[i].first << "x" << pairs[i].second << ")"; + stream << "(" << pairs[i].first << separator << pairs[i].second << ")"; } return stream.str(); } struct AndroidCameraInfo { std::string name; - int min_mfps, max_mfps; // FPS*1000. bool front_facing; int orientation; - std::vector<std::pair<int, int> > resolutions; // Pairs are: (width,height). + IntPairs resolutions; // Pairs are: (width,height). + // Pairs are (min,max) in units of FPS*1000 ("milli-frame-per-second"). + IntPairs mfpsRanges; std::string ToString() { std::stringstream stream; - stream << "Name: [" << name << "], mfps: [" << min_mfps << ":" << max_mfps + stream << "Name: [" << name << "], MFPS ranges: [" + << IntPairsToString(mfpsRanges, ':') << "], front_facing: " << front_facing << ", orientation: " << orientation << ", resolutions: [" - << ResolutionsToString(resolutions) << "]"; + << IntPairsToString(resolutions, 'x') << "]"; return stream.str(); } }; @@ -120,8 +124,6 @@ void DeviceInfoAndroid::Initialize(JNIEnv* jni) { const Json::Value& camera = cameras[i]; AndroidCameraInfo info; info.name = camera["name"].asString(); - info.min_mfps = camera["min_mfps"].asInt(); - info.max_mfps = camera["max_mfps"].asInt(); info.front_facing = camera["front_facing"].asBool(); info.orientation = camera["orientation"].asInt(); Json::Value sizes = camera["sizes"]; @@ -130,10 +132,23 @@ void DeviceInfoAndroid::Initialize(JNIEnv* jni) { info.resolutions.push_back(std::make_pair( size["width"].asInt(), size["height"].asInt())); } + Json::Value mfpsRanges = camera["mfpsRanges"]; + for (Json::ArrayIndex j = 0; j < mfpsRanges.size(); ++j) { + const Json::Value& mfpsRange = mfpsRanges[j]; + info.mfpsRanges.push_back(std::make_pair(mfpsRange["min_mfps"].asInt(), + mfpsRange["max_mfps"].asInt())); + } g_camera_info->push_back(info); } } +void DeviceInfoAndroid::DeInitialize() { + if (g_camera_info) { + delete g_camera_info; + g_camera_info = NULL; + } +} + VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo( const int32_t id) { return new videocapturemodule::DeviceInfoAndroid(id); @@ -187,14 +202,17 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap( return -1; for (size_t i = 0; i < info->resolutions.size(); ++i) { - const std::pair<int, int>& size = info->resolutions[i]; - VideoCaptureCapability cap; - cap.width = size.first; - cap.height = size.second; - cap.maxFPS = info->max_mfps / 1000; - cap.expectedCaptureDelay = kExpectedCaptureDelay; - cap.rawType = kVideoNV21; - _captureCapabilities.push_back(cap); + for (size_t j = 0; j < info->mfpsRanges.size(); ++j) { + const IntPair& size = info->resolutions[i]; + const IntPair& mfpsRange = info->mfpsRanges[j]; + VideoCaptureCapability cap; + cap.width = size.first; + cap.height = size.second; + cap.maxFPS = mfpsRange.second / 1000; + cap.expectedCaptureDelay = kExpectedCaptureDelay; + cap.rawType = kVideoNV21; + _captureCapabilities.push_back(cap); + } } return _captureCapabilities.size(); } @@ -210,13 +228,22 @@ int32_t DeviceInfoAndroid::GetOrientation( return 0; } -void DeviceInfoAndroid::GetFpsRange(const char* deviceUniqueIdUTF8, - int* min_mfps, int* max_mfps) { +void DeviceInfoAndroid::GetMFpsRange(const char* deviceUniqueIdUTF8, + int max_fps_to_match, + int* min_mfps, int* max_mfps) { const AndroidCameraInfo* info = FindCameraInfoByName(deviceUniqueIdUTF8); if (info == NULL) return; - *min_mfps = info->min_mfps; - *max_mfps = info->max_mfps; + // Rely on CameraParameters.getSupportedPreviewFpsRange() to sort its return + // value (per its documentation) and return the first (most flexible) range + // whose high end is at least as high as that requested. + for (size_t i = 0; i < info->mfpsRanges.size(); ++i) { + if (info->mfpsRanges[i].second / 1000 >= max_fps_to_match) { + *min_mfps = info->mfpsRanges[i].first; + *max_mfps = info->mfpsRanges[i].second; + return; + } + } } } // namespace videocapturemodule diff --git a/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.h b/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.h index d277113e5e1..542cbba0881 100644 --- a/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.h +++ b/chromium/third_party/webrtc/modules/video_capture/android/device_info_android.h @@ -24,6 +24,7 @@ namespace videocapturemodule class DeviceInfoAndroid : public DeviceInfoImpl { public: static void Initialize(JNIEnv* env); + static void DeInitialize(); DeviceInfoAndroid(int32_t id); virtual ~DeviceInfoAndroid(); @@ -53,10 +54,12 @@ class DeviceInfoAndroid : public DeviceInfoImpl { virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8, VideoCaptureRotation& orientation); - // Populate |min_mfps| and |max_mfps| with the supported range of the device. - void GetFpsRange(const char* deviceUniqueIdUTF8, - int* min_mfps, - int* max_mfps); + // Populate |min_mfps| and |max_mfps| with the closest supported range of the + // device to |max_fps_to_match|. + void GetMFpsRange(const char* deviceUniqueIdUTF8, + int max_fps_to_match, + int* min_mfps, + int* max_mfps); private: enum { kExpectedCaptureDelay = 190}; diff --git a/chromium/third_party/webrtc/modules/video_capture/android/video_capture_android.cc b/chromium/third_party/webrtc/modules/video_capture/android/video_capture_android.cc index 2b6d60644f5..c9aa52ce7f1 100644 --- a/chromium/third_party/webrtc/modules/video_capture/android/video_capture_android.cc +++ b/chromium/third_party/webrtc/modules/video_capture/android/video_capture_android.cc @@ -10,6 +10,7 @@ #include "webrtc/modules/video_capture/android/video_capture_android.h" +#include "webrtc/base/common.h" #include "webrtc/modules/utility/interface/helpers_android.h" #include "webrtc/modules/video_capture/android/device_info_android.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" @@ -19,9 +20,16 @@ static JavaVM* g_jvm = NULL; static jclass g_java_capturer_class = NULL; // VideoCaptureAndroid.class. +static jobject g_context = NULL; // Owned android.content.Context. namespace webrtc { +// Called by Java to get the global application context. +jobject JNICALL GetContext(JNIEnv* env, jclass) { + assert(g_context); + return g_context; +} + // Called by Java when the camera has a new frame to deliver. void JNICALL ProvideCameraFrame( JNIEnv* env, @@ -38,25 +46,67 @@ void JNICALL ProvideCameraFrame( env->ReleaseByteArrayElements(javaCameraFrame, cameraFrame, JNI_ABORT); } -int32_t SetCaptureAndroidVM(JavaVM* javaVM) { - g_jvm = javaVM; - AttachThreadScoped ats(g_jvm); - - videocapturemodule::DeviceInfoAndroid::Initialize(ats.env()); - - jclass j_capture_class = - ats.env()->FindClass("org/webrtc/videoengine/VideoCaptureAndroid"); - assert(j_capture_class); - g_java_capturer_class = - reinterpret_cast<jclass>(ats.env()->NewGlobalRef(j_capture_class)); - assert(g_java_capturer_class); +// Called by Java when the device orientation has changed. +void JNICALL OnOrientationChanged( + JNIEnv* env, jobject, jlong context, jint degrees) { + webrtc::videocapturemodule::VideoCaptureAndroid* captureModule = + reinterpret_cast<webrtc::videocapturemodule::VideoCaptureAndroid*>( + context); + degrees = (360 + degrees) % 360; + assert(degrees >= 0 && degrees < 360); + VideoCaptureRotation rotation = + (degrees <= 45 || degrees > 315) ? kCameraRotate0 : + (degrees > 45 && degrees <= 135) ? kCameraRotate90 : + (degrees > 135 && degrees <= 225) ? kCameraRotate180 : + (degrees > 225 && degrees <= 315) ? kCameraRotate270 : + kCameraRotate0; // Impossible. + int32_t status = + captureModule->VideoCaptureImpl::SetCaptureRotation(rotation); + RTC_UNUSED(status); + assert(status == 0); +} - JNINativeMethod native_method = { - "ProvideCameraFrame", "([BIJ)V", - reinterpret_cast<void*>(&ProvideCameraFrame) - }; - if (ats.env()->RegisterNatives(g_java_capturer_class, &native_method, 1) != 0) - assert(false); +int32_t SetCaptureAndroidVM(JavaVM* javaVM, jobject context) { + if (javaVM) { + assert(!g_jvm); + g_jvm = javaVM; + AttachThreadScoped ats(g_jvm); + g_context = ats.env()->NewGlobalRef(context); + + videocapturemodule::DeviceInfoAndroid::Initialize(ats.env()); + + jclass j_capture_class = + ats.env()->FindClass("org/webrtc/videoengine/VideoCaptureAndroid"); + assert(j_capture_class); + g_java_capturer_class = + reinterpret_cast<jclass>(ats.env()->NewGlobalRef(j_capture_class)); + assert(g_java_capturer_class); + + JNINativeMethod native_methods[] = { + {"GetContext", + "()Landroid/content/Context;", + reinterpret_cast<void*>(&GetContext)}, + {"OnOrientationChanged", + "(JI)V", + reinterpret_cast<void*>(&OnOrientationChanged)}, + {"ProvideCameraFrame", + "([BIJ)V", + reinterpret_cast<void*>(&ProvideCameraFrame)}}; + if (ats.env()->RegisterNatives(g_java_capturer_class, + native_methods, 3) != 0) + assert(false); + } else { + if (g_jvm) { + AttachThreadScoped ats(g_jvm); + ats.env()->UnregisterNatives(g_java_capturer_class); + ats.env()->DeleteGlobalRef(g_java_capturer_class); + g_java_capturer_class = NULL; + ats.env()->DeleteGlobalRef(g_context); + g_context = NULL; + videocapturemodule::DeviceInfoAndroid::DeInitialize(); + g_jvm = NULL; + } + } return 0; } @@ -143,7 +193,8 @@ int32_t VideoCaptureAndroid::StartCapture( assert(j_start); int min_mfps = 0; int max_mfps = 0; - _deviceInfo.GetFpsRange(_deviceUniqueId, &min_mfps, &max_mfps); + _deviceInfo.GetMFpsRange(_deviceUniqueId, _captureCapability.maxFPS, + &min_mfps, &max_mfps); bool started = env->CallBooleanMethod(_jCapturer, j_start, _captureCapability.width, _captureCapability.height, @@ -184,8 +235,9 @@ int32_t VideoCaptureAndroid::CaptureSettings( int32_t VideoCaptureAndroid::SetCaptureRotation( VideoCaptureRotation rotation) { CriticalSectionScoped cs(&_apiCs); - if (VideoCaptureImpl::SetCaptureRotation(rotation) != 0) - return 0; + int32_t status = VideoCaptureImpl::SetCaptureRotation(rotation); + if (status != 0) + return status; AttachThreadScoped ats(g_jvm); JNIEnv* env = ats.env(); diff --git a/chromium/third_party/webrtc/modules/video_capture/device_info_impl.cc b/chromium/third_party/webrtc/modules/video_capture/device_info_impl.cc index 2d2bc7fb547..7db6103fecb 100644 --- a/chromium/third_party/webrtc/modules/video_capture/device_info_impl.cc +++ b/chromium/third_party/webrtc/modules/video_capture/device_info_impl.cc @@ -8,11 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <assert.h> #include <stdlib.h> #include "webrtc/modules/video_capture/device_info_impl.h" #include "webrtc/modules/video_capture/video_capture_config.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #ifndef abs #define abs(a) (a>=0?a:-a) @@ -75,13 +76,8 @@ int32_t DeviceInfoImpl::GetCapability(const char* deviceUniqueIdUTF8, const uint32_t deviceCapabilityNumber, VideoCaptureCapability& capability) { + assert(deviceUniqueIdUTF8 != NULL); - if (!deviceUniqueIdUTF8) - { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, - "deviceUniqueIdUTF8 parameter not set in call to GetCapability"); - return -1; - } ReadLockScoped cs(_apiLock); if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8)) @@ -111,9 +107,9 @@ int32_t DeviceInfoImpl::GetCapability(const char* deviceUniqueIdUTF8, // Make sure the number is valid if (deviceCapabilityNumber >= (unsigned int) _captureCapabilities.size()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, - "deviceCapabilityNumber %d is invalid in call to GetCapability", - deviceCapabilityNumber); + LOG(LS_ERROR) << "Invalid deviceCapabilityNumber " + << deviceCapabilityNumber << ">= number of capabilities (" + << _captureCapabilities.size() << ")."; return -1; } @@ -266,9 +262,9 @@ int32_t DeviceInfoImpl::GetBestMatchedCapability( }// else height not good }//end for - WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, - "Best camera format: Width %d, Height %d, Frame rate %d, Color format %d", - bestWidth, bestHeight, bestFrameRate, bestRawType); + LOG(LS_VERBOSE) << "Best camera format: " << bestWidth << "x" << bestHeight + << "@" << bestFrameRate + << "fps, color format: " << bestRawType; // Copy the capability if (bestformatIndex < 0) @@ -343,11 +339,10 @@ int32_t DeviceInfoImpl::GetExpectedCaptureDelay( } if (bestDelay > kMaxCaptureDelay) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id, - "Expected capture delay too high. %dms, will use %d", bestDelay, - kMaxCaptureDelay); + LOG(LS_WARNING) << "Expected capture delay (" << bestDelay + << " ms) too high, using " << kMaxCaptureDelay + << " ms."; bestDelay = kMaxCaptureDelay; - } return bestDelay; diff --git a/chromium/third_party/webrtc/modules/video_capture/ensure_initialized.cc b/chromium/third_party/webrtc/modules/video_capture/ensure_initialized.cc new file mode 100644 index 00000000000..65c9a8dbe7a --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_capture/ensure_initialized.cc @@ -0,0 +1,63 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Platform-specific initialization bits, if any, go here. + +#if !defined(ANDROID) || !defined(WEBRTC_CHROMIUM_BUILD) + +namespace webrtc { +namespace videocapturemodule { +void EnsureInitialized() {} +} // namespace videocapturemodule +} // namespace webrtc + +#else // !defined(ANDROID) || !defined(WEBRTC_CHROMIUM_BUILD) + +#include <assert.h> +#include <pthread.h> + +#include "base/android/jni_android.h" + +// Handy alternative to assert() which suppresses unused-variable warnings when +// assert() is a no-op (i.e. in Release builds). +#ifdef NDEBUG +#define ASSERT(x) if (false && (x)); else +#else +#define ASSERT(x) assert(x) +#endif + +namespace webrtc { + +// Declared in webrtc/modules/video_capture/include/video_capture.h. +int32_t SetCaptureAndroidVM(JavaVM* javaVM, jobject g_context); + +namespace videocapturemodule { + +static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT; + +void EnsureInitializedOnce() { + JNIEnv* jni = ::base::android::AttachCurrentThread(); + jobject context = ::base::android::GetApplicationContext(); + JavaVM* jvm = NULL; + int status = jni->GetJavaVM(&jvm); + ASSERT(status == 0); + status = webrtc::SetCaptureAndroidVM(jvm, context) == 0; + ASSERT(status); +} + +void EnsureInitialized() { + int ret = pthread_once(&g_initialize_once, &EnsureInitializedOnce); + ASSERT(ret == 0); +} + +} // namespace videocapturemodule +} // namespace webrtc + +#endif // ANDROID & WEBRTC_CHROMIUM_BUILD diff --git a/chromium/third_party/webrtc/modules/video_capture/ensure_initialized.h b/chromium/third_party/webrtc/modules/video_capture/ensure_initialized.h new file mode 100644 index 00000000000..429879537cd --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_capture/ensure_initialized.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +namespace webrtc { +namespace videocapturemodule { + +// Ensure any necessary initialization of webrtc::videocapturemodule has +// completed. +void EnsureInitialized(); + +} // namespace videocapturemodule. +} // namespace webrtc. diff --git a/chromium/third_party/webrtc/modules/video_capture/include/mock/mock_video_capture.h b/chromium/third_party/webrtc/modules/video_capture/include/mock/mock_video_capture.h new file mode 100644 index 00000000000..8ad74a23886 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_capture/include/mock/mock_video_capture.h @@ -0,0 +1,50 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_MOCK_MOCK_VIDEO_CAPTURE_H_ +#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_MOCK_MOCK_VIDEO_CAPTURE_H_ + +#include "webrtc/modules/video_capture/include/video_capture.h" +#include "testing/gmock/include/gmock/gmock.h" + +namespace webrtc { + +class MockVideoCaptureModule : public VideoCaptureModule { + public: + // from Module + MOCK_METHOD0(TimeUntilNextProcess, int32_t()); + MOCK_METHOD0(Process, int32_t()); + + // from RefCountedModule + MOCK_METHOD0(AddRef, int32_t()); + MOCK_METHOD0(Release, int32_t()); + + // from VideoCaptureModule + MOCK_METHOD1(RegisterCaptureDataCallback, + void(VideoCaptureDataCallback& dataCallback)); + MOCK_METHOD0(DeRegisterCaptureDataCallback, void()); + MOCK_METHOD1(RegisterCaptureCallback, void(VideoCaptureFeedBack& callBack)); + MOCK_METHOD0(DeRegisterCaptureCallback, void()); + MOCK_METHOD1(StartCapture, int32_t(const VideoCaptureCapability& capability)); + MOCK_METHOD0(StopCapture, int32_t()); + MOCK_CONST_METHOD0(CurrentDeviceName, const char*()); + MOCK_METHOD0(CaptureStarted, bool()); + MOCK_METHOD1(CaptureSettings, int32_t(VideoCaptureCapability& settings)); + MOCK_METHOD1(SetCaptureDelay, void(int32_t delayMS)); + MOCK_METHOD0(CaptureDelay, int32_t()); + MOCK_METHOD1(SetCaptureRotation, int32_t(VideoCaptureRotation rotation)); + MOCK_METHOD1(GetEncodeInterface, + VideoCaptureEncodeInterface*(const VideoCodec& codec)); + MOCK_METHOD1(EnableFrameRateCallback, void(const bool enable)); + MOCK_METHOD1(EnableNoPictureAlarm, void(const bool enable)); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_MOCK_MOCK_VIDEO_CAPTURE_H_ diff --git a/chromium/third_party/webrtc/modules/video_capture/include/video_capture.h b/chromium/third_party/webrtc/modules/video_capture/include/video_capture.h index 6b6247c2777..7398af60448 100644 --- a/chromium/third_party/webrtc/modules/video_capture/include/video_capture.h +++ b/chromium/third_party/webrtc/modules/video_capture/include/video_capture.h @@ -20,8 +20,8 @@ namespace webrtc { -#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) -int32_t SetCaptureAndroidVM(JavaVM* javaVM); +#if defined(ANDROID) +int32_t SetCaptureAndroidVM(JavaVM* javaVM, jobject context); #endif class VideoCaptureModule: public RefCountedModule { @@ -105,18 +105,17 @@ class VideoCaptureModule: public RefCountedModule { }; // Register capture data callback - virtual int32_t RegisterCaptureDataCallback( + virtual void RegisterCaptureDataCallback( VideoCaptureDataCallback& dataCallback) = 0; // Remove capture data callback - virtual int32_t DeRegisterCaptureDataCallback() = 0; + virtual void DeRegisterCaptureDataCallback() = 0; // Register capture callback. - virtual int32_t RegisterCaptureCallback( - VideoCaptureFeedBack& callBack) = 0; + virtual void RegisterCaptureCallback(VideoCaptureFeedBack& callBack) = 0; // Remove capture callback. - virtual int32_t DeRegisterCaptureCallback() = 0; + virtual void DeRegisterCaptureCallback() = 0; // Start capture device virtual int32_t StartCapture( @@ -133,7 +132,7 @@ class VideoCaptureModule: public RefCountedModule { // Gets the current configuration. virtual int32_t CaptureSettings(VideoCaptureCapability& settings) = 0; - virtual int32_t SetCaptureDelay(int32_t delayMS) = 0; + virtual void SetCaptureDelay(int32_t delayMS) = 0; // Returns the current CaptureDelay. Only valid when the camera is running. virtual int32_t CaptureDelay() = 0; @@ -149,8 +148,8 @@ class VideoCaptureModule: public RefCountedModule { virtual VideoCaptureEncodeInterface* GetEncodeInterface( const VideoCodec& codec) = 0; - virtual int32_t EnableFrameRateCallback(const bool enable) = 0; - virtual int32_t EnableNoPictureAlarm(const bool enable) = 0; + virtual void EnableFrameRateCallback(const bool enable) = 0; + virtual void EnableNoPictureAlarm(const bool enable) = 0; protected: virtual ~VideoCaptureModule() {}; diff --git a/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios.mm b/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios.mm index c51a53a3372..dea9fc34a41 100644 --- a/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios.mm +++ b/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios.mm @@ -8,6 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #include "webrtc/modules/video_capture/ios/device_info_ios.h" #include "webrtc/modules/video_capture/ios/device_info_ios_objc.h" #include "webrtc/modules/video_capture/video_capture_impl.h" diff --git a/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios_objc.mm b/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios_objc.mm index 2d11a2043fb..d06d3361f45 100644 --- a/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios_objc.mm +++ b/chromium/third_party/webrtc/modules/video_capture/ios/device_info_ios_objc.mm @@ -8,6 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #import <AVFoundation/AVFoundation.h> #import "webrtc/modules/video_capture/ios/device_info_ios_objc.h" diff --git a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios_objc.h b/chromium/third_party/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h index 8e50facba19..7d4147b430d 100644 --- a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios_objc.h +++ b/chromium/third_party/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h @@ -11,18 +11,17 @@ #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_IOS_VIDEO_CAPTURE_IOS_OBJC_H_ #define WEBRTC_MODULES_VIDEO_CAPTURE_IOS_VIDEO_CAPTURE_IOS_OBJC_H_ +#import <Foundation/Foundation.h> #import <UIKit/UIKit.h> #include "webrtc/modules/video_capture/ios/video_capture_ios.h" -@interface VideoCaptureIosObjC - : UIViewController<AVCaptureVideoDataOutputSampleBufferDelegate> { - @private - webrtc::videocapturemodule::VideoCaptureIos* _owner; - webrtc::VideoCaptureCapability _capability; - AVCaptureSession* _captureSession; - int _captureId; -} +// The following class listens to a notification with name: +// 'StatusBarOrientationDidChange'. +// This notification must be posted in order for the capturer to reflect the +// orientation change in video w.r.t. the application orientation. +@interface RTCVideoCaptureIosObjC + : NSObject<AVCaptureVideoDataOutputSampleBufferDelegate> @property webrtc::VideoCaptureRotation frameRotation; @@ -31,7 +30,7 @@ // default init methods have been overridden to return nil. - (id)initWithOwner:(webrtc::videocapturemodule::VideoCaptureIos*)owner captureId:(int)captureId; -- (BOOL)setCaptureDeviceByUniqueId:(NSString*)uniequeId; +- (BOOL)setCaptureDeviceByUniqueId:(NSString*)uniqueId; - (BOOL)startCaptureWithCapability: (const webrtc::VideoCaptureCapability&)capability; - (BOOL)stopCapture; diff --git a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios_objc.mm b/chromium/third_party/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm index 5b8d69786e8..641ca2416b2 100644 --- a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios_objc.mm +++ b/chromium/third_party/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm @@ -8,20 +8,33 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + +#import <UIKit/UIKit.h> + #import "webrtc/modules/video_capture/ios/device_info_ios_objc.h" -#import "webrtc/modules/video_capture/ios/video_capture_ios_objc.h" +#import "webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h" #include "webrtc/system_wrappers/interface/trace.h" using namespace webrtc; using namespace webrtc::videocapturemodule; -@interface VideoCaptureIosObjC (hidden) +@interface RTCVideoCaptureIosObjC (hidden) - (int)changeCaptureInputWithName:(NSString*)captureDeviceName; - @end -@implementation VideoCaptureIosObjC +@implementation RTCVideoCaptureIosObjC { + webrtc::videocapturemodule::VideoCaptureIos* _owner; + webrtc::VideoCaptureCapability _capability; + AVCaptureSession* _captureSession; + int _captureId; + AVCaptureConnection* _connection; + BOOL _captureChanging; // Guarded by _captureChangingCondition. + NSCondition* _captureChangingCondition; +} @synthesize frameRotation = _framRotation; @@ -30,16 +43,16 @@ using namespace webrtc::videocapturemodule; _owner = owner; _captureId = captureId; _captureSession = [[AVCaptureSession alloc] init]; + _captureChanging = NO; + _captureChangingCondition = [[NSCondition alloc] init]; - if (!_captureSession) { + if (!_captureSession || !_captureChangingCondition) { return nil; } // create and configure a new output (using callbacks) AVCaptureVideoDataOutput* captureOutput = [[AVCaptureVideoDataOutput alloc] init]; - [captureOutput setSampleBufferDelegate:self - queue:dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0)]; NSString* key = (NSString*)kCVPixelBufferPixelFormatTypeKey; NSNumber* val = [NSNumber @@ -66,12 +79,36 @@ using namespace webrtc::videocapturemodule; selector:@selector(onVideoError:) name:AVCaptureSessionRuntimeErrorNotification object:_captureSession]; + [notify addObserver:self + selector:@selector(statusBarOrientationDidChange:) + name:@"StatusBarOrientationDidChange" + object:nil]; } return self; } +- (void)directOutputToSelf { + [[self currentOutput] + setSampleBufferDelegate:self + queue:dispatch_get_global_queue( + DISPATCH_QUEUE_PRIORITY_DEFAULT, 0)]; +} + +- (void)directOutputToNil { + [[self currentOutput] setSampleBufferDelegate:nil queue:NULL]; +} + +- (void)statusBarOrientationDidChange:(NSNotification*)notification { + [self setRelativeVideoOrientation]; +} + +- (void)dealloc { + [[NSNotificationCenter defaultCenter] removeObserver:self]; +} + - (BOOL)setCaptureDeviceByUniqueId:(NSString*)uniqueId { + [self waitForCaptureChangeToFinish]; // check to see if the camera is already set if (_captureSession) { NSArray* currentInputs = [NSArray arrayWithArray:[_captureSession inputs]]; @@ -87,6 +124,7 @@ using namespace webrtc::videocapturemodule; } - (BOOL)startCaptureWithCapability:(const VideoCaptureCapability&)capability { + [self waitForCaptureChangeToFinish]; if (!_captureSession) { return NO; } @@ -121,11 +159,25 @@ using namespace webrtc::videocapturemodule; _capability = capability; - NSArray* currentOutputs = [_captureSession outputs]; - if ([currentOutputs count] == 0) { + AVCaptureVideoDataOutput* currentOutput = [self currentOutput]; + if (!currentOutput) return NO; - } + [self directOutputToSelf]; + + _captureChanging = YES; + dispatch_async( + dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), + ^(void) { [self startCaptureInBackgroundWithOutput:currentOutput]; }); + return YES; +} + +- (AVCaptureVideoDataOutput*)currentOutput { + return [[_captureSession outputs] firstObject]; +} + +- (void)startCaptureInBackgroundWithOutput: + (AVCaptureVideoDataOutput*)currentOutput { NSString* captureQuality = [NSString stringWithString:AVCaptureSessionPresetLow]; if (_capability.width >= 1920 || _capability.height >= 1080) { @@ -139,9 +191,6 @@ using namespace webrtc::videocapturemodule; captureQuality = [NSString stringWithString:AVCaptureSessionPreset352x288]; } - AVCaptureVideoDataOutput* currentOutput = - (AVCaptureVideoDataOutput*)[currentOutputs objectAtIndex:0]; - // begin configuration for the AVCaptureSession [_captureSession beginConfiguration]; @@ -149,25 +198,42 @@ using namespace webrtc::videocapturemodule; [_captureSession setSessionPreset:captureQuality]; // take care of capture framerate now - AVCaptureConnection* connection = - [currentOutput connectionWithMediaType:AVMediaTypeVideo]; - + _connection = [currentOutput connectionWithMediaType:AVMediaTypeVideo]; + [self setRelativeVideoOrientation]; CMTime cm_time = {1, _capability.maxFPS, kCMTimeFlags_Valid, 0}; - [connection setVideoMinFrameDuration:cm_time]; - [connection setVideoMaxFrameDuration:cm_time]; + [_connection setVideoMinFrameDuration:cm_time]; + [_connection setVideoMaxFrameDuration:cm_time]; // finished configuring, commit settings to AVCaptureSession. [_captureSession commitConfiguration]; [_captureSession startRunning]; + [self signalCaptureChangeEnd]; +} - [captureQuality release]; - - return YES; +- (void)setRelativeVideoOrientation { + if (!_connection.supportsVideoOrientation) + return; + switch ([UIApplication sharedApplication].statusBarOrientation) { + case UIInterfaceOrientationPortrait: + _connection.videoOrientation = AVCaptureVideoOrientationPortrait; + break; + case UIInterfaceOrientationPortraitUpsideDown: + _connection.videoOrientation = + AVCaptureVideoOrientationPortraitUpsideDown; + break; + case UIInterfaceOrientationLandscapeLeft: + _connection.videoOrientation = AVCaptureVideoOrientationLandscapeLeft; + break; + case UIInterfaceOrientationLandscapeRight: + _connection.videoOrientation = AVCaptureVideoOrientationLandscapeRight; + break; + } } -- (void)onVideoError { +- (void)onVideoError:(NSNotification*)notification { + NSLog(@"onVideoError: %@", notification); // TODO(sjlee): make the specific error handling with this notification. WEBRTC_TRACE(kTraceError, kTraceVideoCapture, @@ -179,16 +245,26 @@ using namespace webrtc::videocapturemodule; } - (BOOL)stopCapture { + [self waitForCaptureChangeToFinish]; + [self directOutputToNil]; + if (!_captureSession) { return NO; } - [_captureSession stopRunning]; - + _captureChanging = YES; + dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), + ^(void) { [self stopCaptureInBackground]; }); return YES; } +- (void)stopCaptureInBackground { + [_captureSession stopRunning]; + [self signalCaptureChangeEnd]; +} + - (BOOL)changeCaptureInputByUniqueId:(NSString*)uniqueId { + [self waitForCaptureChangeToFinish]; NSArray* currentInputs = [_captureSession inputs]; // remove current input if ([currentInputs count] > 0) { @@ -284,4 +360,18 @@ using namespace webrtc::videocapturemodule; CVPixelBufferUnlockBaseAddress(videoFrame, kFlags); } +- (void)signalCaptureChangeEnd { + [_captureChangingCondition lock]; + _captureChanging = NO; + [_captureChangingCondition signal]; + [_captureChangingCondition unlock]; +} + +- (void)waitForCaptureChangeToFinish { + [_captureChangingCondition lock]; + while (_captureChanging) { + [_captureChangingCondition wait]; + } + [_captureChangingCondition unlock]; +} @end diff --git a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.h b/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.h index 5d7e4b35725..ff8345f26ee 100644 --- a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.h +++ b/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.h @@ -13,7 +13,7 @@ #include "webrtc/modules/video_capture/video_capture_impl.h" -@class VideoCaptureIosObjC; +@class RTCVideoCaptureIosObjC; namespace webrtc { namespace videocapturemodule { @@ -33,7 +33,7 @@ class VideoCaptureIos : public VideoCaptureImpl { virtual int32_t CaptureSettings(VideoCaptureCapability& settings) OVERRIDE; private: - VideoCaptureIosObjC* capture_device_; + RTCVideoCaptureIosObjC* capture_device_; bool is_capturing_; int32_t id_; VideoCaptureCapability capability_; diff --git a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.mm b/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.mm index bb576c3fcdf..2010f03080c 100644 --- a/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.mm +++ b/chromium/third_party/webrtc/modules/video_capture/ios/video_capture_ios.mm @@ -8,8 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #include "webrtc/modules/video_capture/ios/device_info_ios_objc.h" -#include "webrtc/modules/video_capture/ios/video_capture_ios_objc.h" +#include "webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h" #include "webrtc/system_wrappers/interface/ref_count.h" #include "webrtc/system_wrappers/interface/scoped_refptr.h" #include "webrtc/system_wrappers/interface/trace.h" @@ -30,7 +34,7 @@ VideoCaptureIos::VideoCaptureIos(const int32_t capture_id) } VideoCaptureIos::~VideoCaptureIos() { - if (capture_device_) { + if (is_capturing_) { [capture_device_ stopCapture]; } } @@ -53,8 +57,8 @@ VideoCaptureModule* VideoCaptureIos::Create(const int32_t capture_id, capture_module->_deviceUniqueId[name_length] = '\0'; capture_module->capture_device_ = - [[VideoCaptureIosObjC alloc] initWithOwner:capture_module - captureId:capture_module->id_]; + [[RTCVideoCaptureIosObjC alloc] initWithOwner:capture_module + captureId:capture_module->id_]; if (!capture_module->capture_device_) { return NULL; } @@ -86,7 +90,6 @@ int32_t VideoCaptureIos::StopCapture() { } is_capturing_ = false; - return 0; } diff --git a/chromium/third_party/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.mm b/chromium/third_party/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.mm index 7b46aec1925..2b18e1eca17 100644 --- a/chromium/third_party/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.mm +++ b/chromium/third_party/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.mm @@ -155,11 +155,14 @@ using namespace webrtc; - (void)checkOSSupported { Class osSupportedTest = NSClassFromString(@"QTCaptureSession"); - _OSSupportedInfo = NO; if(nil == osSupportedTest) { + _OSSupportedInfo = NO; + } + else + { + _OSSupportedInfo = YES; } - _OSSupportedInfo = YES; } /// ***** Retrieves the number of capture devices currently available diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi b/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi index b8a63c9c4a6..dced22ce08d 100644 --- a/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi +++ b/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi @@ -60,6 +60,7 @@ 'link_settings': { 'xcode_settings': { 'OTHER_LDFLAGS': [ + '-framework CoreVideo', '-framework QTKit', ], }, @@ -108,11 +109,14 @@ 'ios/device_info_ios.mm', 'ios/device_info_ios_objc.h', 'ios/device_info_ios_objc.mm', + 'ios/rtc_video_capture_ios_objc.h', + 'ios/rtc_video_capture_ios_objc.mm', 'ios/video_capture_ios.h', 'ios/video_capture_ios.mm', - 'ios/video_capture_ios_objc.h', - 'ios/video_capture_ios_objc.mm', ], + 'xcode_settings': { + 'CLANG_ENABLE_OBJC_ARC': 'YES', + }, 'all_dependent_settings': { 'xcode_settings': { 'OTHER_LDFLAGS': [ @@ -130,11 +134,20 @@ }, ], 'conditions': [ + ['include_tests==1 and build_with_chromium==1 and OS=="android"', { + # Use WebRTC capture code for Android APK tests that are built from a + # Chromium checkout. Normally when built as a part of Chromium the + # Chromium video capture code is used. This overrides the default in + # webrtc/build/common.gypi. + 'variables': { + 'include_internal_video_capture': 1, + }, + }], ['include_tests==1', { 'targets': [ { 'target_name': 'video_capture_tests', - 'type': 'executable', + 'type': '<(gtest_target_type)', 'dependencies': [ 'video_capture_module', 'webrtc_utility', @@ -142,6 +155,8 @@ '<(DEPTH)/testing/gtest.gyp:gtest', ], 'sources': [ + 'ensure_initialized.cc', + 'ensure_initialized.h', 'test/video_capture_unittest.cc', 'test/video_capture_main_mac.mm', ], @@ -161,6 +176,13 @@ '-lX11', ], }], + # TODO(henrike): remove build_with_chromium==1 when the bots are + # using Chromium's buildbots. + ['build_with_chromium==1 and OS=="android"', { + 'dependencies': [ + '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', + ], + }], ['OS=="mac"', { 'dependencies': [ # Link with a special main for mac so we can use the webcam. diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc index 6689fd132c7..6f179e2da66 100644 --- a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc +++ b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc @@ -17,9 +17,9 @@ #include "webrtc/modules/video_capture/video_capture_config.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/ref_count.h" #include "webrtc/system_wrappers/interface/tick_util.h" -#include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc @@ -187,45 +187,33 @@ VideoCaptureImpl::~VideoCaptureImpl() delete[] _deviceUniqueId; } -int32_t VideoCaptureImpl::RegisterCaptureDataCallback( - VideoCaptureDataCallback& dataCallBack) -{ +void VideoCaptureImpl::RegisterCaptureDataCallback( + VideoCaptureDataCallback& dataCallBack) { CriticalSectionScoped cs(&_apiCs); CriticalSectionScoped cs2(&_callBackCs); _dataCallBack = &dataCallBack; - - return 0; } -int32_t VideoCaptureImpl::DeRegisterCaptureDataCallback() -{ +void VideoCaptureImpl::DeRegisterCaptureDataCallback() { CriticalSectionScoped cs(&_apiCs); CriticalSectionScoped cs2(&_callBackCs); _dataCallBack = NULL; - return 0; } -int32_t VideoCaptureImpl::RegisterCaptureCallback(VideoCaptureFeedBack& callBack) -{ +void VideoCaptureImpl::RegisterCaptureCallback(VideoCaptureFeedBack& callBack) { CriticalSectionScoped cs(&_apiCs); CriticalSectionScoped cs2(&_callBackCs); _captureCallBack = &callBack; - return 0; } -int32_t VideoCaptureImpl::DeRegisterCaptureCallback() -{ +void VideoCaptureImpl::DeRegisterCaptureCallback() { CriticalSectionScoped cs(&_apiCs); CriticalSectionScoped cs2(&_callBackCs); _captureCallBack = NULL; - return 0; - } -int32_t VideoCaptureImpl::SetCaptureDelay(int32_t delayMS) -{ +void VideoCaptureImpl::SetCaptureDelay(int32_t delayMS) { CriticalSectionScoped cs(&_apiCs); _captureDelay = delayMS; - return 0; } int32_t VideoCaptureImpl::CaptureDelay() { @@ -272,13 +260,8 @@ int32_t VideoCaptureImpl::IncomingFrame( const VideoCaptureCapability& frameInfo, int64_t captureTime/*=0*/) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideoCapture, _id, - "IncomingFrame width %d, height %d", (int) frameInfo.width, - (int) frameInfo.height); - - TickTime startProcessTime = TickTime::Now(); - - CriticalSectionScoped cs(&_callBackCs); + CriticalSectionScoped cs(&_apiCs); + CriticalSectionScoped cs2(&_callBackCs); const int32_t width = frameInfo.width; const int32_t height = frameInfo.height; @@ -295,8 +278,7 @@ int32_t VideoCaptureImpl::IncomingFrame( CalcBufferSize(commonVideoType, width, abs(height)) != videoFrameLength) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, - "Wrong incoming frame length."); + LOG(LS_ERROR) << "Wrong incoming frame length."; return -1; } @@ -320,8 +302,8 @@ int32_t VideoCaptureImpl::IncomingFrame( stride_uv, stride_uv); if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, - "Failed to allocate I420 frame."); + LOG(LS_ERROR) << "Failed to create empty frame, this should only " + "happen due to bad parameters."; return -1; } const int conversionResult = ConvertToI420(commonVideoType, @@ -333,9 +315,8 @@ int32_t VideoCaptureImpl::IncomingFrame( &_captureFrame); if (conversionResult < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, - "Failed to convert capture frame from type %d to I420", - frameInfo.rawType); + LOG(LS_ERROR) << "Failed to convert capture frame from type " + << frameInfo.rawType << "to I420."; return -1; } DeliverCapturedFrame(_captureFrame, captureTime); @@ -346,22 +327,14 @@ int32_t VideoCaptureImpl::IncomingFrame( return -1; } - const uint32_t processTime = - (uint32_t)(TickTime::Now() - startProcessTime).Milliseconds(); - if (processTime > 10) // If the process time is too long MJPG will not work well. - { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id, - "Too long processing time of Incoming frame: %ums", - (unsigned int) processTime); - } - return 0; } int32_t VideoCaptureImpl::IncomingI420VideoFrame(I420VideoFrame* video_frame, int64_t captureTime) { - CriticalSectionScoped cs(&_callBackCs); + CriticalSectionScoped cs(&_apiCs); + CriticalSectionScoped cs2(&_callBackCs); DeliverCapturedFrame(*video_frame, captureTime); return 0; @@ -389,8 +362,7 @@ int32_t VideoCaptureImpl::SetCaptureRotation(VideoCaptureRotation rotation) { return 0; } -int32_t VideoCaptureImpl::EnableFrameRateCallback(const bool enable) -{ +void VideoCaptureImpl::EnableFrameRateCallback(const bool enable) { CriticalSectionScoped cs(&_apiCs); CriticalSectionScoped cs2(&_callBackCs); _frameRateCallBack = enable; @@ -398,15 +370,12 @@ int32_t VideoCaptureImpl::EnableFrameRateCallback(const bool enable) { _lastFrameRateCallbackTime = TickTime::Now(); } - return 0; } -int32_t VideoCaptureImpl::EnableNoPictureAlarm(const bool enable) -{ +void VideoCaptureImpl::EnableNoPictureAlarm(const bool enable) { CriticalSectionScoped cs(&_apiCs); CriticalSectionScoped cs2(&_callBackCs); _noPictureAlarmCallBack = enable; - return 0; } void VideoCaptureImpl::UpdateFrameCount() diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h index 80d5e67862a..f3a4c64cbd6 100644 --- a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h +++ b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h @@ -62,17 +62,18 @@ public: virtual int32_t ChangeUniqueId(const int32_t id); //Call backs - virtual int32_t RegisterCaptureDataCallback(VideoCaptureDataCallback& dataCallback); - virtual int32_t DeRegisterCaptureDataCallback(); - virtual int32_t RegisterCaptureCallback(VideoCaptureFeedBack& callBack); - virtual int32_t DeRegisterCaptureCallback(); + virtual void RegisterCaptureDataCallback( + VideoCaptureDataCallback& dataCallback); + virtual void DeRegisterCaptureDataCallback(); + virtual void RegisterCaptureCallback(VideoCaptureFeedBack& callBack); + virtual void DeRegisterCaptureCallback(); - virtual int32_t SetCaptureDelay(int32_t delayMS); + virtual void SetCaptureDelay(int32_t delayMS); virtual int32_t CaptureDelay(); virtual int32_t SetCaptureRotation(VideoCaptureRotation rotation); - virtual int32_t EnableFrameRateCallback(const bool enable); - virtual int32_t EnableNoPictureAlarm(const bool enable); + virtual void EnableFrameRateCallback(const bool enable); + virtual void EnableNoPictureAlarm(const bool enable); virtual const char* CurrentDeviceName() const; diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture_tests.isolate b/chromium/third_party/webrtc/modules/video_capture/video_capture_tests.isolate index 30374ce4e4a..57dd66739fd 100644 --- a/chromium/third_party/webrtc/modules/video_capture/video_capture_tests.isolate +++ b/chromium/third_party/webrtc/modules/video_capture/video_capture_tests.isolate @@ -8,27 +8,25 @@ { 'conditions': [ ['OS=="android"', { - # When doing Android builds, the WebRTC code is put in third_party/webrtc - # of a Chromium checkout, this is one level above the standalone build. 'variables': { 'isolate_dependency_untracked': [ - '../../../../data/', - '../../../../resources/', + '<(DEPTH)/data/', + '<(DEPTH)/resources/', ], }, }], ['OS=="linux" or OS=="mac" or OS=="win"', { 'variables': { 'command': [ - '../../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/video_capture_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_tracked': [ - '../../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/video_capture_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_untracked': [ - '../../../tools/swarming_client/', + '<(DEPTH)/tools/swarming_client/', ], }, }], diff --git a/chromium/third_party/webrtc/modules/video_capture/windows/sink_filter_ds.cc b/chromium/third_party/webrtc/modules/video_capture/windows/sink_filter_ds.cc index 144f8833b75..2edbe59c4aa 100644 --- a/chromium/third_party/webrtc/modules/video_capture/windows/sink_filter_ds.cc +++ b/chromium/third_party/webrtc/modules/video_capture/windows/sink_filter_ds.cc @@ -437,6 +437,7 @@ CaptureSinkFilter::GetPin(IN int Index) STDMETHODIMP CaptureSinkFilter::Pause() { + LockReceive(); LockFilter(); if (m_State == State_Stopped) { @@ -456,6 +457,7 @@ STDMETHODIMP CaptureSinkFilter::Pause() m_State = State_Paused; } UnlockFilter(); + UnlockReceive(); return S_OK; } diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/i420/main/source/OWNERS b/chromium/third_party/webrtc/modules/video_coding/codecs/i420/main/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/i420/main/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/OWNERS b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.cc index 9eda36e8439..dcd74790c29 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.cc +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.cc @@ -26,14 +26,12 @@ using namespace webrtc; NormalAsyncTest::NormalAsyncTest() : -NormalTest("Async Normal Test 1", "A test of normal execution of the codec", - _testNo), +NormalTest("Async Normal Test 1", "A test of normal execution of the codec", 1), _decodeCompleteTime(0), _encodeCompleteTime(0), _encFrameCnt(0), _decFrameCnt(0), _requestKeyFrame(false), -_testNo(1), _appendNext(false), _missingFrames(false), _rttFrames(0), @@ -47,13 +45,13 @@ _waitForKey(false) NormalAsyncTest::NormalAsyncTest(uint32_t bitRate) : NormalTest("Async Normal Test 1", "A test of normal execution of the codec", - bitRate, _testNo), + bitRate, + 1), _decodeCompleteTime(0), _encodeCompleteTime(0), _encFrameCnt(0), _decFrameCnt(0), _requestKeyFrame(false), -_testNo(1), _appendNext(false), _missingFrames(false), _rttFrames(0), @@ -67,13 +65,12 @@ _waitForKey(false) NormalAsyncTest::NormalAsyncTest(std::string name, std::string description, unsigned int testNo) : -NormalTest(name, description, _testNo), +NormalTest(name, description, testNo), _decodeCompleteTime(0), _encodeCompleteTime(0), _encFrameCnt(0), _decFrameCnt(0), _requestKeyFrame(false), -_testNo(testNo), _lengthEncFrame(0), _appendNext(false), _missingFrames(false), @@ -88,13 +85,12 @@ _waitForKey(false) NormalAsyncTest::NormalAsyncTest(std::string name, std::string description, uint32_t bitRate, unsigned int testNo) : -NormalTest(name, description, bitRate, _testNo), +NormalTest(name, description, bitRate, testNo), _decodeCompleteTime(0), _encodeCompleteTime(0), _encFrameCnt(0), _decFrameCnt(0), _requestKeyFrame(false), -_testNo(testNo), _lengthEncFrame(0), _appendNext(false), _missingFrames(false), @@ -110,13 +106,12 @@ NormalAsyncTest::NormalAsyncTest(std::string name, std::string description, uint32_t bitRate, unsigned int testNo, unsigned int rttFrames) : -NormalTest(name, description, bitRate, _testNo), +NormalTest(name, description, bitRate, testNo), _decodeCompleteTime(0), _encodeCompleteTime(0), _encFrameCnt(0), _decFrameCnt(0), _requestKeyFrame(false), -_testNo(testNo), _lengthEncFrame(0), _appendNext(false), _missingFrames(false), diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.h index d2d17eebc7f..1e62534acab 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.h +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/normal_async_test.h @@ -118,7 +118,6 @@ protected: int _encFrameCnt; int _decFrameCnt; bool _requestKeyFrame; - unsigned int _testNo; unsigned int _lengthEncFrame; FrameQueueTuple* _frameToDecode; bool _appendNext; diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/test.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/test.h index 890d0cb50cb..7558abe6bbf 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/test.h +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/test.h @@ -41,7 +41,7 @@ protected: virtual void Teardown(); double ActualBitRate(int nFrames); virtual bool PacketLoss(double lossRate, int /*thrown*/); - static double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); } + static double RandUniform() { return (rand() + 1.0)/(RAND_MAX + 1.0); } static void VideoEncodedBufferToEncodedImage( webrtc::VideoFrame& videoBuffer, webrtc::EncodedImage &image); diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/unit_test.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/unit_test.cc index 3b034e01c60..ec12a51693c 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/unit_test.cc +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test_framework/unit_test.cc @@ -565,7 +565,7 @@ UnitTest::Perform() frameLength = WaitForDecodedFrame(); } unsigned int length = CalcBufferSize(kI420, width, height); - scoped_array<uint8_t> decoded_buffer(new uint8_t[length]); + scoped_ptr<uint8_t[]> decoded_buffer(new uint8_t[length]); ExtractBuffer(_decodedVideoBuffer, _lengthSourceFrame, decoded_buffer.get()); EXPECT_TRUE(CheckIfBitExact(decoded_buffer.get(), frameLength, _refDecFrame, @@ -645,7 +645,7 @@ UnitTest::Perform() // check that decoded frame matches with reference unsigned int length = CalcBufferSize(kI420, width, height); - scoped_array<uint8_t> decoded_buffer(new uint8_t[length]); + scoped_ptr<uint8_t[]> decoded_buffer(new uint8_t[length]); ExtractBuffer(_decodedVideoBuffer, length, decoded_buffer.get()); EXPECT_TRUE(CheckIfBitExact(decoded_buffer.get(), length, _refDecFrame, _lengthSourceFrame) == true); diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/tools/OWNERS b/chromium/third_party/webrtc/modules/video_coding/codecs/tools/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/tools/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/tools/video_codecs_tools.gypi b/chromium/third_party/webrtc/modules/video_coding/codecs/tools/video_codecs_tools.gypi index cdae0afeaad..8f15b28504d 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/tools/video_codecs_tools.gypi +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/tools/video_codecs_tools.gypi @@ -17,6 +17,7 @@ 'video_codecs_test_framework', 'webrtc_video_coding', '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', '<(webrtc_root)/test/metrics.gyp:metrics', '<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8', ], diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/OWNERS b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8.gyp b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8.gyp index b3859a5978d..621c244cdfe 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8.gyp +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8.gyp @@ -31,6 +31,7 @@ 'reference_picture_selection.cc', 'include/vp8.h', 'include/vp8_common_types.h', + 'vp8_factory.cc', 'vp8_impl.cc', 'default_temporal_layers.cc', 'default_temporal_layers.h', diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_factory.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_factory.cc new file mode 100644 index 00000000000..995191e3c04 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_factory.cc @@ -0,0 +1,24 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + * + */ + +#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h" + +namespace webrtc { + +VP8Encoder* VP8Encoder::Create() { + return new VP8EncoderImpl(); +} + +VP8Decoder* VP8Decoder::Create() { + return new VP8DecoderImpl(); +} + +} // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc index 910a5f3f8bb..4901edff3d5 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc @@ -35,10 +35,6 @@ enum { kVp8ErrorPropagationTh = 30 }; namespace webrtc { -VP8Encoder* VP8Encoder::Create() { - return new VP8EncoderImpl(); -} - VP8EncoderImpl::VP8EncoderImpl() : encoded_image_(), encoded_complete_callback_(NULL), @@ -218,7 +214,10 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst, } config_->g_lag_in_frames = 0; // 0- no frame lagging - if (codec_.width * codec_.height > 1280 * 960 && number_of_cores >= 6) { + if (codec_.width * codec_.height >= 1920 * 1080 && number_of_cores > 8) { + config_->g_threads = 8; // 8 threads for 1080p on high perf machines. + } else if (codec_.width * codec_.height > 1280 * 960 && + number_of_cores >= 6) { config_->g_threads = 3; // 3 threads for 1080p. } else if (codec_.width * codec_.height > 640 * 480 && number_of_cores >= 3) { config_->g_threads = 2; // 2 threads for qHD/HD. @@ -477,8 +476,8 @@ int VP8EncoderImpl::GetEncodedPartitions(const I420VideoFrame& input_image) { TRACE_COUNTER1("webrtc", "EncodedFrameSize", encoded_image_._length); encoded_image_._timeStamp = input_image.timestamp(); encoded_image_.capture_time_ms_ = input_image.render_time_ms(); - encoded_image_._encodedHeight = raw_->h; - encoded_image_._encodedWidth = raw_->w; + encoded_image_._encodedHeight = codec_.height; + encoded_image_._encodedWidth = codec_.width; encoded_complete_callback_->Encoded(encoded_image_, &codec_specific, &frag_info); } @@ -496,10 +495,6 @@ int VP8EncoderImpl::RegisterEncodeCompleteCallback( return WEBRTC_VIDEO_CODEC_OK; } -VP8Decoder* VP8Decoder::Create() { - return new VP8DecoderImpl(); -} - VP8DecoderImpl::VP8DecoderImpl() : decode_complete_callback_(NULL), inited_(false), @@ -718,7 +713,7 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image, } img = vpx_codec_get_frame(decoder_, &iter); - ret = ReturnFrame(img, input_image._timeStamp); + ret = ReturnFrame(img, input_image._timeStamp, input_image.ntp_time_ms_); if (ret != 0) { // Reset to avoid requesting key frames too often. if (ret < 0 && propagation_cnt_ > 0) @@ -798,7 +793,9 @@ int VP8DecoderImpl::DecodePartitions( return WEBRTC_VIDEO_CODEC_OK; } -int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img, uint32_t timestamp) { +int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img, + uint32_t timestamp, + int64_t ntp_time_ms) { if (img == NULL) { // Decoder OK and NULL image => No show frame return WEBRTC_VIDEO_CODEC_NO_OUTPUT; @@ -816,6 +813,7 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img, uint32_t timestamp) { img->stride[VPX_PLANE_U], img->stride[VPX_PLANE_V]); decoded_image_.set_timestamp(timestamp); + decoded_image_.set_ntp_time_ms(ntp_time_ms); int ret = decode_complete_callback_->Decoded(decoded_image_); if (ret != 0) return ret; diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h index 26dd52e6a93..56f7219fc1a 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h @@ -214,7 +214,9 @@ class VP8DecoderImpl : public VP8Decoder { int DecodePartitions(const EncodedImage& input_image, const RTPFragmentationHeader* fragmentation); - int ReturnFrame(const vpx_image_t* img, uint32_t timeStamp); + int ReturnFrame(const vpx_image_t* img, + uint32_t timeStamp, + int64_t ntp_time_ms); I420VideoFrame decoded_image_; DecodedImageCallback* decode_complete_callback_; diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc index 1bd3e1a6238..ffa0bcc681f 100644 --- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc +++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc @@ -142,7 +142,7 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) { EXPECT_EQ(0, decoder->InitDecode(&inst, 1)); webrtc::I420VideoFrame input_frame; unsigned int length = webrtc::CalcBufferSize(webrtc::kI420, width, height); - webrtc::scoped_array<uint8_t> frame_buffer(new uint8_t[length]); + webrtc::scoped_ptr<uint8_t[]> frame_buffer(new uint8_t[length]); int half_width = (width + 1) / 2; // Set and register callbacks. diff --git a/chromium/third_party/webrtc/modules/video_coding/main/interface/video_coding.h b/chromium/third_party/webrtc/modules/video_coding/main/interface/video_coding.h index c0166761e83..cad0e5ab879 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/interface/video_coding.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/interface/video_coding.h @@ -72,11 +72,9 @@ public: kReferenceSelection }; - static VideoCodingModule* Create(const int32_t id); + static VideoCodingModule* Create(); - static VideoCodingModule* Create(const int32_t id, - Clock* clock, - EventFactory* event_factory); + static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory); static void Destroy(VideoCodingModule* module); diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/Android.mk b/chromium/third_party/webrtc/modules/video_coding/main/source/Android.mk index 9ebdbed9a66..a8cf2d0e64c 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/Android.mk +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/Android.mk @@ -37,7 +37,6 @@ LOCAL_SRC_FILES := \ receiver.cc \ rtt_filter.cc \ session_info.cc \ - timestamp_extrapolator.cc \ timestamp_map.cc \ timing.cc \ video_coding_impl.cc @@ -56,7 +55,7 @@ LOCAL_C_INCLUDES := \ $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \ $(LOCAL_PATH)/../../../../common_video/interface \ $(LOCAL_PATH)/../../utility/include \ - $(LOCAL_PATH)/../../../../system_wrappers/interface + $(LOCAL_PATH)/../../../../system_wrappers/interface LOCAL_SHARED_LIBRARIES := \ libcutils \ diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/OWNERS b/chromium/third_party/webrtc/modules/video_coding/main/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.cc index e41c6b42534..e7a9d91b138 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.cc @@ -20,7 +20,7 @@ #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" #endif #include "webrtc/modules/video_coding/main/source/internal_defines.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -42,9 +42,8 @@ VCMExtDecoderMapItem::VCMExtDecoderMapItem( internal_render_timing(internal_render_timing) { } -VCMCodecDataBase::VCMCodecDataBase(int id) - : id_(id), - number_of_cores_(0), +VCMCodecDataBase::VCMCodecDataBase() + : number_of_cores_(0), max_payload_size_(kDefaultPayloadSize), periodic_key_frames_(false), pending_encoder_reset_(true), @@ -58,8 +57,7 @@ VCMCodecDataBase::VCMCodecDataBase(int id) ptr_decoder_(NULL), current_dec_is_external_(false), dec_map_(), - dec_external_map_() { -} + dec_external_map_() {} VCMCodecDataBase::~VCMCodecDataBase() { ResetSender(); @@ -160,7 +158,7 @@ bool VCMCodecDataBase::SetSendCodec( if (max_payload_size <= 0) { max_payload_size = kDefaultPayloadSize; } - if (number_of_cores <= 0 || number_of_cores > 32) { + if (number_of_cores <= 0) { return false; } if (send_codec->plType <= 0) { @@ -221,24 +219,14 @@ bool VCMCodecDataBase::SetSendCodec( } else { ptr_encoder_ = CreateEncoder(send_codec->codecType); current_enc_is_external_ = false; + if (!ptr_encoder_) { + return false; + } } encoded_frame_callback->SetPayloadType(send_codec->plType); - if (!ptr_encoder_) { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(id_), - "Failed to create encoder: %s.", - send_codec->plName); - return false; - } if (ptr_encoder_->InitEncode(send_codec, number_of_cores_, max_payload_size_) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(id_), - "Failed to initialize encoder: %s.", - send_codec->plName); DeleteEncoder(); return false; } else if (ptr_encoder_->RegisterEncodeCallback(encoded_frame_callback) < 0) { @@ -257,8 +245,6 @@ bool VCMCodecDataBase::SetSendCodec( } bool VCMCodecDataBase::SendCodec(VideoCodec* current_send_codec) const { - WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(id_), - "SendCodec"); if (!ptr_encoder_) { return false; } @@ -267,8 +253,6 @@ bool VCMCodecDataBase::SendCodec(VideoCodec* current_send_codec) const { } VideoCodecType VCMCodecDataBase::SendCodec() const { - WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(id_), - "SendCodec type"); if (!ptr_encoder_) { return kVideoCodecUnknown; } @@ -338,12 +322,6 @@ bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) { } break; case kVideoCodecGeneric: - if (memcmp(&new_send_codec.codecSpecific.Generic, - &send_codec_.codecSpecific.Generic, - sizeof(new_send_codec.codecSpecific.Generic)) != - 0) { - return true; - } break; // Known codecs without payload-specifics case kVideoCodecI420: @@ -404,7 +382,11 @@ bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) { // Not found return false; } - if (receive_codec_.plType == payload_type) { + // We can't use payload_type to check if the decoder is currently in use, + // because payload type may be out of date (e.g. before we decode the first + // frame after RegisterReceiveCodec) + if (ptr_decoder_ != NULL && + &ptr_decoder_->_decoder == (*it).second->external_decoder_instance) { // Release it if it was registered and in use. ReleaseDecoder(ptr_decoder_); ptr_decoder_ = NULL; @@ -443,12 +425,6 @@ bool VCMCodecDataBase::RegisterReceiveCodec( if (number_of_cores < 0) { return false; } - WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCoding, VCMId(id_), - "Codec: %s, Payload type %d, Height %d, Width %d, Bitrate %d," - "Framerate %d.", - receive_codec->plName, receive_codec->plType, - receive_codec->height, receive_codec->width, - receive_codec->startBitrate, receive_codec->maxFramerate); // Check if payload value already exists, if so - erase old and insert new. DeregisterReceiveCodec(receive_codec->plType); if (receive_codec->codecType == kVideoCodecUnknown) { @@ -530,7 +506,7 @@ VCMGenericDecoder* VCMCodecDataBase::CreateDecoderCopy() const { if (!decoder_copy) { return NULL; } - return new VCMGenericDecoder(*decoder_copy, id_, ptr_decoder_->External()); + return new VCMGenericDecoder(*decoder_copy, ptr_decoder_->External()); } void VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const { @@ -549,8 +525,7 @@ void VCMCodecDataBase::CopyDecoder(const VCMGenericDecoder& decoder) { if (decoder_copy) { VCMDecodedFrameCallback* cb = ptr_decoder_->_callback; ReleaseDecoder(ptr_decoder_); - ptr_decoder_ = new VCMGenericDecoder(*decoder_copy, id_, - decoder.External()); + ptr_decoder_ = new VCMGenericDecoder(*decoder_copy, decoder.External()); if (cb && ptr_decoder_->RegisterDecodeCompleteCallback(cb)) { assert(false); } @@ -575,8 +550,8 @@ VCMGenericDecoder* VCMCodecDataBase::CreateAndInitDecoder( assert(new_codec); const VCMDecoderMapItem* decoder_item = FindDecoderItem(payload_type); if (!decoder_item) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(id_), - "Unknown payload type: %u", payload_type); + LOG(LS_ERROR) << "Can't find a decoder associated with payload type: " + << payload_type; return NULL; } VCMGenericDecoder* ptr_decoder = NULL; @@ -585,7 +560,7 @@ VCMGenericDecoder* VCMCodecDataBase::CreateAndInitDecoder( if (external_dec_item) { // External codec. ptr_decoder = new VCMGenericDecoder( - *external_dec_item->external_decoder_instance, id_, true); + *external_dec_item->external_decoder_instance, true); *external = true; } else { // Create decoder. @@ -617,6 +592,7 @@ VCMGenericEncoder* VCMCodecDataBase::CreateEncoder( return new VCMGenericEncoder(*(new I420Encoder)); #endif default: + LOG(LS_WARNING) << "No internal encoder of this type exists."; return NULL; } } @@ -636,11 +612,11 @@ VCMGenericDecoder* VCMCodecDataBase::CreateDecoder(VideoCodecType type) const { switch (type) { #ifdef VIDEOCODEC_VP8 case kVideoCodecVP8: - return new VCMGenericDecoder(*(VP8Decoder::Create()), id_); + return new VCMGenericDecoder(*(VP8Decoder::Create())); #endif #ifdef VIDEOCODEC_I420 case kVideoCodecI420: - return new VCMGenericDecoder(*(new I420Decoder), id_); + return new VCMGenericDecoder(*(new I420Decoder)); #endif default: return NULL; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.h b/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.h index 2a28ed4fb87..f27218f61cf 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/codec_database.h @@ -50,7 +50,7 @@ struct VCMExtDecoderMapItem { class VCMCodecDataBase { public: - explicit VCMCodecDataBase(int id); + VCMCodecDataBase(); ~VCMCodecDataBase(); // Sender Side @@ -174,7 +174,6 @@ class VCMCodecDataBase { const VCMExtDecoderMapItem* FindExternalDecoderItem( uint8_t payload_type) const; - int id_; int number_of_cores_; int max_payload_size_; bool periodic_key_frames_; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.cc index 6760762c9a5..3ccf0b0fd99 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.cc @@ -149,17 +149,12 @@ const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const { return &_fragmentation; } -int32_t -VCMEncodedFrame::VerifyAndAllocate(const uint32_t minimumSize) +void VCMEncodedFrame::VerifyAndAllocate(const uint32_t minimumSize) { if(minimumSize > _size) { // create buffer of sufficient size uint8_t* newBuffer = new uint8_t[minimumSize]; - if (newBuffer == NULL) - { - return -1; - } if(_buffer) { // copy old data @@ -169,7 +164,6 @@ VCMEncodedFrame::VerifyAndAllocate(const uint32_t minimumSize) _buffer = newBuffer; _size = minimumSize; } - return 0; } webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType) diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.h b/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.h index 3e73be51803..dd0f843d267 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/encoded_frame.h @@ -104,7 +104,7 @@ protected: * is copied to the new buffer. * Buffer size is updated to minimumSize. */ - int32_t VerifyAndAllocate(const uint32_t minimumSize); + void VerifyAndAllocate(const uint32_t minimumSize); void Reset(); diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/frame_buffer.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/frame_buffer.cc index 531c7ac112d..fce68fb32d5 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/frame_buffer.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/frame_buffer.cc @@ -14,6 +14,7 @@ #include <string.h> #include "webrtc/modules/video_coding/main/source/packet.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -86,20 +87,7 @@ VCMFrameBuffer::InsertPacket(const VCMPacket& packet, int64_t timeInMs, VCMDecodeErrorMode decode_error_mode, const FrameData& frame_data) { - // Is this packet part of this frame? - if (TimeStamp() && (TimeStamp() != packet.timestamp)) { - return kTimeStampError; - } - - // sanity checks - if (_size + packet.sizeBytes + - (packet.insertStartCode ? kH264StartCodeLengthBytes : 0 ) - > kMaxJBFrameSizeBytes) { - return kSizeError; - } - if (NULL == packet.dataPtr && packet.sizeBytes > 0) { - return kSizeError; - } + assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0)); if (packet.dataPtr != NULL) { _payloadType = packet.payloadType; } @@ -108,6 +96,8 @@ VCMFrameBuffer::InsertPacket(const VCMPacket& packet, // First packet (empty and/or media) inserted into this frame. // store some info and set some initial values. _timeStamp = packet.timestamp; + // We only take the ntp timestamp of the first packet of a frame. + ntp_time_ms_ = packet.ntp_time_ms_; _codec = packet.codec; if (packet.frameType != kFrameEmpty) { // first media packet @@ -126,11 +116,11 @@ VCMFrameBuffer::InsertPacket(const VCMPacket& packet, const uint32_t newSize = _size + increments * kBufferIncStepSizeBytes; if (newSize > kMaxJBFrameSizeBytes) { + LOG(LS_ERROR) << "Failed to insert packet due to frame being too " + "big."; return kSizeError; } - if (VerifyAndAllocate(newSize) == -1) { - return kSizeError; - } + VerifyAndAllocate(newSize); _sessionInfo.UpdateDataPointers(prevBuffer, _buffer); } diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.cc index 50b1eda70fc..cb0faf9a901 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.cc @@ -12,8 +12,7 @@ #include "webrtc/modules/video_coding/main/source/generic_decoder.h" #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/system_wrappers/interface/trace_event.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -59,10 +58,11 @@ int32_t VCMDecodedFrameCallback::Decoded(I420VideoFrame& decodedImage) _timestampMap.Pop(decodedImage.timestamp())); callback = _receiveCallback; } - if (frameInfo == NULL) - { - // The map should never be empty or full if this callback is called. - return WEBRTC_VIDEO_CODEC_ERROR; + + if (frameInfo == NULL) { + LOG(LS_WARNING) << "Too many frames backed up in the decoder, dropping " + "this one."; + return WEBRTC_VIDEO_CODEC_OK; } _timing.StopDecodeTimer( @@ -73,14 +73,7 @@ int32_t VCMDecodedFrameCallback::Decoded(I420VideoFrame& decodedImage) if (callback != NULL) { decodedImage.set_render_time_ms(frameInfo->renderTimeMs); - int32_t callbackReturn = callback->FrameToRender(decodedImage); - if (callbackReturn < 0) - { - WEBRTC_TRACE(webrtc::kTraceDebug, - webrtc::kTraceVideoCoding, - -1, - "Render callback returned error: %d", callbackReturn); - } + callback->FrameToRender(decodedImage); } return WEBRTC_VIDEO_CODEC_OK; } @@ -125,15 +118,15 @@ int32_t VCMDecodedFrameCallback::Pop(uint32_t timestamp) return VCM_OK; } -VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, int32_t id, bool isExternal) +VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, bool isExternal) : -_id(id), _callback(NULL), _frameInfos(), _nextFrameInfoIdx(0), _decoder(decoder), _codecType(kVideoCodecUnknown), -_isExternal(isExternal) +_isExternal(isExternal), +_keyFrameDecoded(false) { } @@ -156,11 +149,6 @@ int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame, _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs(); _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]); - WEBRTC_TRACE(webrtc::kTraceDebug, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Decoding timestamp %u", frame.TimeStamp()); - _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength; int32_t ret = _decoder.Decode(frame.EncodedImage(), frame.MissingFrame(), @@ -170,7 +158,8 @@ int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame, if (ret < WEBRTC_VIDEO_CODEC_OK) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id), "Decoder error: %d\n", ret); + LOG(LS_WARNING) << "Failed to decode frame with timestamp " + << frame.TimeStamp() << ", error code: " << ret; _callback->Pop(frame.TimeStamp()); return ret; } diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.h b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.h index e1993fbb906..846d4d3e111 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_decoder.h @@ -63,7 +63,7 @@ class VCMGenericDecoder { friend class VCMCodecDataBase; public: - VCMGenericDecoder(VideoDecoder& decoder, int32_t id = 0, bool isExternal = false); + VCMGenericDecoder(VideoDecoder& decoder, bool isExternal = false); ~VCMGenericDecoder(); /** @@ -105,17 +105,14 @@ public: bool External() const; -protected: - - int32_t _id; +private: VCMDecodedFrameCallback* _callback; VCMFrameInformation _frameInfos[kDecoderFrameMemoryLength]; - uint32_t _nextFrameInfoIdx; + uint32_t _nextFrameInfoIdx; VideoDecoder& _decoder; VideoCodecType _codecType; bool _isExternal; bool _keyFrameDecoded; - }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.cc index 064470b1667..6fb2c9f81b7 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.cc @@ -13,8 +13,42 @@ #include "webrtc/modules/video_coding/main/source/generic_encoder.h" #include "webrtc/modules/video_coding/main/source/media_optimization.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { +namespace { +// Map information from info into rtp. If no relevant information is found +// in info, rtp is set to NULL. +void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader** rtp) { + if (!info) { + *rtp = NULL; + return; + } + switch (info->codecType) { + case kVideoCodecVP8: { + (*rtp)->codec = kRtpVideoVp8; + (*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8(); + (*rtp)->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId; + (*rtp)->codecHeader.VP8.nonReference = + info->codecSpecific.VP8.nonReference; + (*rtp)->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx; + (*rtp)->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync; + (*rtp)->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx; + (*rtp)->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx; + (*rtp)->simulcastIdx = info->codecSpecific.VP8.simulcastIdx; + return; + } + case kVideoCodecGeneric: + (*rtp)->codec = kRtpVideoGeneric; + (*rtp)->simulcastIdx = info->codecSpecific.generic.simulcast_idx; + return; + default: + // No codec specific info. Change RTP header pointer to NULL. + *rtp = NULL; + return; + } +} +} // namespace //#define DEBUG_ENCODER_BIT_STREAM @@ -50,11 +84,12 @@ VCMGenericEncoder::InitEncode(const VideoCodec* settings, _bitRate = settings->startBitrate * 1000; _frameRate = settings->maxFramerate; _codecType = settings->codecType; - if (_VCMencodedFrameCallback != NULL) - { - _VCMencodedFrameCallback->SetCodecType(_codecType); + if (_encoder.InitEncode(settings, numberOfCores, maxPayloadSize) != 0) { + LOG(LS_ERROR) << "Failed to initialize the encoder associated with " + "payload name: " << settings->plName; + return -1; } - return _encoder.InitEncode(settings, numberOfCores, maxPayloadSize); + return 0; } int32_t @@ -127,8 +162,6 @@ int32_t VCMGenericEncoder::RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback) { _VCMencodedFrameCallback = VCMencodedFrameCallback; - - _VCMencodedFrameCallback->SetCodecType(_codecType); _VCMencodedFrameCallback->SetInternalSource(_internalSource); return _encoder.RegisterEncodeCompleteCallback(_VCMencodedFrameCallback); } @@ -142,15 +175,13 @@ VCMGenericEncoder::InternalSource() const /*************************** * Callback Implementation ***************************/ -VCMEncodedFrameCallback::VCMEncodedFrameCallback(): +VCMEncodedFrameCallback::VCMEncodedFrameCallback( + EncodedImageCallback* post_encode_callback): _sendCallback(), _mediaOpt(NULL), -_encodedBytes(0), _payloadType(0), -_codecType(kVideoCodecUnknown), _internalSource(false), -post_encode_callback_lock_(CriticalSectionWrapper::CreateCriticalSection()), -post_encode_callback_(NULL) +post_encode_callback_(post_encode_callback) #ifdef DEBUG_ENCODER_BIT_STREAM , _bitStreamAfterEncoder(NULL) #endif @@ -180,12 +211,8 @@ VCMEncodedFrameCallback::Encoded( const CodecSpecificInfo* codecSpecificInfo, const RTPFragmentationHeader* fragmentationHeader) { - { - CriticalSectionScoped cs(post_encode_callback_lock_.get()); - if (post_encode_callback_) { - post_encode_callback_->Encoded(encodedImage); - } - } + post_encode_callback_->Encoded(encodedImage); + FrameType frameType = VCMEncodedFrame::ConvertFrameType(encodedImage._frameType); uint32_t encodedBytes = 0; @@ -202,14 +229,7 @@ VCMEncodedFrameCallback::Encoded( RTPVideoHeader rtpVideoHeader; RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader; - if (codecSpecificInfo) - { - CopyCodecSpecific(*codecSpecificInfo, &rtpVideoHeaderPtr); - } - else - { - rtpVideoHeaderPtr = NULL; - } + CopyCodecSpecific(codecSpecificInfo, &rtpVideoHeaderPtr); int32_t callbackReturn = _sendCallback->SendData( frameType, @@ -229,9 +249,8 @@ VCMEncodedFrameCallback::Encoded( { return VCM_UNINITIALIZED; } - _encodedBytes = encodedBytes; if (_mediaOpt != NULL) { - _mediaOpt->UpdateWithEncodedData(_encodedBytes, encodedImage._timeStamp, + _mediaOpt->UpdateWithEncodedData(encodedBytes, encodedImage._timeStamp, frameType); if (_internalSource) { @@ -241,12 +260,6 @@ VCMEncodedFrameCallback::Encoded( return VCM_OK; } -uint32_t -VCMEncodedFrameCallback::EncodedBytes() -{ - return _encodedBytes; -} - void VCMEncodedFrameCallback::SetMediaOpt( media_optimization::MediaOptimization *mediaOpt) @@ -254,36 +267,4 @@ VCMEncodedFrameCallback::SetMediaOpt( _mediaOpt = mediaOpt; } -void VCMEncodedFrameCallback::CopyCodecSpecific(const CodecSpecificInfo& info, - RTPVideoHeader** rtp) { - switch (info.codecType) { - case kVideoCodecVP8: { - (*rtp)->codec = kRtpVideoVp8; - (*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8(); - (*rtp)->codecHeader.VP8.pictureId = info.codecSpecific.VP8.pictureId; - (*rtp)->codecHeader.VP8.nonReference = - info.codecSpecific.VP8.nonReference; - (*rtp)->codecHeader.VP8.temporalIdx = info.codecSpecific.VP8.temporalIdx; - (*rtp)->codecHeader.VP8.layerSync = info.codecSpecific.VP8.layerSync; - (*rtp)->codecHeader.VP8.tl0PicIdx = info.codecSpecific.VP8.tl0PicIdx; - (*rtp)->codecHeader.VP8.keyIdx = info.codecSpecific.VP8.keyIdx; - (*rtp)->simulcastIdx = info.codecSpecific.VP8.simulcastIdx; - return; - } - case kVideoCodecGeneric: - (*rtp)->codec = kRtpVideoGeneric; - (*rtp)->simulcastIdx = info.codecSpecific.generic.simulcast_idx; - return; - default: - // No codec specific info. Change RTP header pointer to NULL. - *rtp = NULL; - return; - } -} - -void VCMEncodedFrameCallback::RegisterPostEncodeImageCallback( - EncodedImageCallback* callback) { - CriticalSectionScoped cs(post_encode_callback_lock_.get()); - post_encode_callback_ = callback; -} } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.h b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.h index c5cfeabc26f..9277260af34 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/generic_encoder.h @@ -17,9 +17,7 @@ #include "webrtc/system_wrappers/interface/scoped_ptr.h" -namespace webrtc -{ - +namespace webrtc { class CriticalSectionWrapper; namespace media_optimization { @@ -32,7 +30,7 @@ class MediaOptimization; class VCMEncodedFrameCallback : public EncodedImageCallback { public: - VCMEncodedFrameCallback(); + VCMEncodedFrameCallback(EncodedImageCallback* post_encode_callback); virtual ~VCMEncodedFrameCallback(); /* @@ -43,10 +41,6 @@ public: const CodecSpecificInfo* codecSpecificInfo = NULL, const RTPFragmentationHeader* fragmentationHeader = NULL); /* - * Get number of encoded bytes - */ - uint32_t EncodedBytes(); - /* * Callback implementation - generic encoder encode complete */ int32_t SetTransportCallback(VCMPacketizationCallback* transport); @@ -56,27 +50,14 @@ public: void SetMediaOpt (media_optimization::MediaOptimization* mediaOpt); void SetPayloadType(uint8_t payloadType) { _payloadType = payloadType; }; - void SetCodecType(VideoCodecType codecType) {_codecType = codecType;}; void SetInternalSource(bool internalSource) { _internalSource = internalSource; }; - void RegisterPostEncodeImageCallback(EncodedImageCallback* callback); - private: - /* - * Map information from info into rtp. If no relevant information is found - * in info, rtp is set to NULL. - */ - static void CopyCodecSpecific(const CodecSpecificInfo& info, - RTPVideoHeader** rtp); - VCMPacketizationCallback* _sendCallback; media_optimization::MediaOptimization* _mediaOpt; - uint32_t _encodedBytes; uint8_t _payloadType; - VideoCodecType _codecType; bool _internalSource; - scoped_ptr<CriticalSectionWrapper> post_encode_callback_lock_; EncodedImageCallback* post_encode_callback_; #ifdef DEBUG_ENCODER_BIT_STREAM diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.cc index f11f81b46f2..d8792f21afb 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.cc @@ -25,7 +25,6 @@ #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" -#include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -77,10 +76,6 @@ int FrameList::RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it, FrameList::iterator it = begin(); while (!empty()) { // Throw at least one frame. - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, -1, - "Recycling: type=%s, low seqnum=%u", - it->second->FrameType() == kVideoFrameKey ? - "key" : "delta", it->second->GetLowSeqNum()); it->second->Reset(); free_frames->push_back(it->second); erase(it++); @@ -128,16 +123,10 @@ void FrameList::Reset(UnorderedFrameList* free_frames) { } VCMJitterBuffer::VCMJitterBuffer(Clock* clock, - EventFactory* event_factory, - int vcm_id, - int receiver_id, - bool master) - : vcm_id_(vcm_id), - receiver_id_(receiver_id), - clock_(clock), + EventFactory* event_factory) + : clock_(clock), running_(false), crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - master_(master), frame_event_(event_factory->CreateEvent()), packet_event_(event_factory->CreateEvent()), max_number_of_frames_(kStartNumberOfFrames), @@ -156,7 +145,7 @@ VCMJitterBuffer::VCMJitterBuffer(Clock* clock, num_consecutive_old_frames_(0), num_consecutive_old_packets_(0), num_discarded_packets_(0), - jitter_estimate_(vcm_id, receiver_id), + jitter_estimate_(), inter_frame_delay_(clock_->TimeInMilliseconds()), rtt_ms_(kDefaultRtt), nack_mode_(kNoNack), @@ -192,10 +181,7 @@ void VCMJitterBuffer::CopyFrom(const VCMJitterBuffer& rhs) { if (this != &rhs) { crit_sect_->Enter(); rhs.crit_sect_->Enter(); - vcm_id_ = rhs.vcm_id_; - receiver_id_ = rhs.receiver_id_; running_ = rhs.running_; - master_ = !rhs.master_; max_number_of_frames_ = rhs.max_number_of_frames_; incoming_frame_rate_ = rhs.incoming_frame_rate_; incoming_frame_count_ = rhs.incoming_frame_count_; @@ -276,10 +262,6 @@ void VCMJitterBuffer::Start() { first_packet_since_reset_ = true; rtt_ms_ = kDefaultRtt; last_decoded_state_.Reset(); - - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), "JB(0x%x): Jitter buffer: start", - this); } void VCMJitterBuffer::Stop() { @@ -300,9 +282,6 @@ void VCMJitterBuffer::Stop() { // Make sure we wake up any threads waiting on these events. frame_event_->Set(); packet_event_->Set(); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), "JB(0x%x): Jitter buffer: stop", - this); } bool VCMJitterBuffer::Running() const { @@ -327,9 +306,6 @@ void VCMJitterBuffer::Flush() { waiting_for_completion_.latest_packet_time = -1; first_packet_since_reset_ = true; missing_sequence_numbers_.clear(); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), "JB(0x%x): Jitter buffer: flush", - this); } // Get received key and delta frames @@ -583,6 +559,8 @@ VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet, DropPacketsFromNackList(last_decoded_state_.sequence_num()); if (num_consecutive_old_packets_ > kMaxConsecutiveOldPackets) { + LOG(LS_WARNING) << num_consecutive_old_packets_ << " consecutive old " + "packets received. Flushing the jitter buffer."; Flush(); return kFlushIndicator; } @@ -602,13 +580,13 @@ VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet, VCMFrameBufferEnum ret = kNoError; if (!*frame) { // No free frame! Try to reclaim some... - LOG_F(LS_INFO) << "Unable to get empty frame; Recycling."; + LOG(LS_WARNING) << "Unable to get empty frame; Recycling."; bool found_key_frame = RecycleFramesUntilKeyFrame(); *frame = GetEmptyFrame(); - if (!*frame) - return kGeneralError; - else if (!found_key_frame) + assert(*frame); + if (!found_key_frame) { ret = kFlushIndicator; + } } (*frame)->Reset(); return ret; @@ -650,6 +628,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet, // Flush if this happens consistently. num_consecutive_old_frames_++; if (num_consecutive_old_frames_ > kMaxConsecutiveOldFrames) { + LOG(LS_WARNING) << num_consecutive_old_packets_ << " consecutive old " + "frames received. Flushing the jitter buffer."; Flush(); return kFlushIndicator; } @@ -702,8 +682,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet, if (IsPacketRetransmitted(packet)) { frame->IncrementNackCount(); } - if (!UpdateNackList(packet.seqNum)) { - LOG_F(LS_INFO) << "Requesting key frame due to flushed NACK list."; + if (!UpdateNackList(packet.seqNum) && + packet.frameType != kVideoFrameKey) { buffer_return = kFlushIndicator; } latest_received_sequence_number_ = LatestSequenceNumber( @@ -725,15 +705,6 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet, } case kCompleteSession: { if (update_decodable_list) { - if (master_) { - // Only trace the primary jitter buffer to make it possible to parse - // and plot the trace file. - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "JB(0x%x) FB(0x%x): Complete frame added to jitter" - "buffer, size:%d type %d", - this, frame, frame->Length(), frame->FrameType()); - } CountFrame(*frame); frame->SetCountedFrame(true); if (continuous) { @@ -960,8 +931,6 @@ uint16_t* VCMJitterBuffer::GetNackList(uint16_t* nack_list_size, incomplete_frames_.begin(), incomplete_frames_.end(), HasNonEmptyState); } - if (have_non_empty_frame) - LOG_F(LS_INFO) << "First frame is not key; Recycling."; bool found_key_frame = RecycleFramesUntilKeyFrame(); if (!found_key_frame) { *request_key_frame = have_non_empty_frame; @@ -977,9 +946,9 @@ uint16_t* VCMJitterBuffer::GetNackList(uint16_t* nack_list_size, int non_continuous_incomplete_duration = NonContinuousOrIncompleteDuration(); if (non_continuous_incomplete_duration > 90 * max_incomplete_time_ms_) { - LOG_F(LS_INFO) << "Too long non-decodable duration: " << - non_continuous_incomplete_duration << " > " << - 90 * max_incomplete_time_ms_; + LOG_F(LS_WARNING) << "Too long non-decodable duration: " + << non_continuous_incomplete_duration << " > " + << 90 * max_incomplete_time_ms_; FrameList::reverse_iterator rit = find_if(incomplete_frames_.rbegin(), incomplete_frames_.rend(), IsKeyFrame); if (rit == incomplete_frames_.rend()) { @@ -1038,10 +1007,12 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) { TRACE_EVENT_INSTANT1("webrtc", "AddNack", "seqnum", i); } if (TooLargeNackList() && !HandleTooLargeNackList()) { + LOG(LS_WARNING) << "Requesting key frame due to too large NACK list."; return false; } if (MissingTooOldPacket(sequence_number) && !HandleTooOldPackets(sequence_number)) { + LOG(LS_WARNING) << "Requesting key frame due to missing too old packets"; return false; } } else { @@ -1058,8 +1029,9 @@ bool VCMJitterBuffer::TooLargeNackList() const { bool VCMJitterBuffer::HandleTooLargeNackList() { // Recycle frames until the NACK list is small enough. It is likely cheaper to // request a key frame than to retransmit this many missing packets. - LOG_F(LS_INFO) << "NACK list has grown too large: " << - missing_sequence_numbers_.size() << " > " << max_nack_list_size_; + LOG_F(LS_WARNING) << "NACK list has grown too large: " + << missing_sequence_numbers_.size() << " > " + << max_nack_list_size_; bool key_frame_found = false; while (TooLargeNackList()) { key_frame_found = RecycleFramesUntilKeyFrame(); @@ -1083,8 +1055,9 @@ bool VCMJitterBuffer::HandleTooOldPackets(uint16_t latest_sequence_number) { bool key_frame_found = false; const uint16_t age_of_oldest_missing_packet = latest_sequence_number - *missing_sequence_numbers_.begin(); - LOG_F(LS_INFO) << "NACK list contains too old sequence numbers: " << - age_of_oldest_missing_packet << " > " << max_packet_age_to_nack_; + LOG_F(LS_WARNING) << "NACK list contains too old sequence numbers: " + << age_of_oldest_missing_packet << " > " + << max_packet_age_to_nack_; while (MissingTooOldPacket(latest_sequence_number)) { key_frame_found = RecycleFramesUntilKeyFrame(); } @@ -1136,10 +1109,6 @@ bool VCMJitterBuffer::TryToIncreaseJitterBufferSize() { frame_buffers_[max_number_of_frames_] = new_frame; free_frames_.push_back(new_frame); ++max_number_of_frames_; - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "JB(0x%x) FB(0x%x): Jitter buffer increased to:%d frames", - this, new_frame, max_number_of_frames_); TRACE_COUNTER1("webrtc", "JBMaxFrames", max_number_of_frames_); return true; } @@ -1161,13 +1130,9 @@ bool VCMJitterBuffer::RecycleFramesUntilKeyFrame() { key_frame_found = key_frame_it != decodable_frames_.end(); } drop_count_ += dropped_frames; - if (dropped_frames) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "Jitter buffer drop count:%u", drop_count_); - } TRACE_EVENT_INSTANT0("webrtc", "JB::RecycleFramesUntilKeyFrame"); if (key_frame_found) { + LOG(LS_INFO) << "Found key frame while dropping frames."; // Reset last decoded state to make sure the next frame decoded is a key // frame, and start NACKing from here. last_decoded_state_.Reset(); @@ -1246,19 +1211,6 @@ void VCMJitterBuffer::UpdateJitterEstimate(const VCMJitterSample& sample, if (sample.latest_packet_time == -1) { return; } - if (incomplete_frame) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), "Received incomplete frame " - "timestamp %u frame size %u at time %u", - sample.timestamp, sample.frame_size, - MaskWord64ToUWord32(sample.latest_packet_time)); - } else { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), "Received complete frame " - "timestamp %u frame size %u at time %u", - sample.timestamp, sample.frame_size, - MaskWord64ToUWord32(sample.latest_packet_time)); - } UpdateJitterEstimate(sample.latest_packet_time, sample.timestamp, sample.frame_size, incomplete_frame); } @@ -1273,23 +1225,6 @@ void VCMJitterBuffer::UpdateJitterEstimate(const VCMFrameBuffer& frame, } // No retransmitted frames should be a part of the jitter // estimate. - if (incomplete_frame) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "Received incomplete frame timestamp %u frame type %d " - "frame size %u at time %u, jitter estimate was %u", - frame.TimeStamp(), frame.FrameType(), frame.Length(), - MaskWord64ToUWord32(frame.LatestPacketTimeMs()), - EstimatedJitterMs()); - } else { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), "Received complete frame " - "timestamp %u frame type %d frame size %u at time %u, " - "jitter estimate was %u", - frame.TimeStamp(), frame.FrameType(), frame.Length(), - MaskWord64ToUWord32(frame.LatestPacketTimeMs()), - EstimatedJitterMs()); - } UpdateJitterEstimate(frame.LatestPacketTimeMs(), frame.TimeStamp(), frame.Length(), incomplete_frame); } @@ -1306,12 +1241,6 @@ void VCMJitterBuffer::UpdateJitterEstimate( return; } int64_t frame_delay; - // Calculate the delay estimate - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "Packet received and sent to jitter estimate with: " - "timestamp=%u wall_clock=%u", timestamp, - MaskWord64ToUWord32(latest_packet_time_ms)); bool not_reordered = inter_frame_delay_.CalculateDelay(timestamp, &frame_delay, latest_packet_time_ms); diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.h b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.h index 8586f115f88..6ed9cfb85c6 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer.h @@ -16,6 +16,7 @@ #include <set> #include <vector> +#include "webrtc/base/constructormagic.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" @@ -23,7 +24,6 @@ #include "webrtc/modules/video_coding/main/source/inter_frame_delay.h" #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" #include "webrtc/modules/video_coding/main/source/jitter_estimator.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/typedefs.h" @@ -77,10 +77,7 @@ class FrameList class VCMJitterBuffer { public: VCMJitterBuffer(Clock* clock, - EventFactory* event_factory, - int vcm_id, - int receiver_id, - bool master); + EventFactory* event_factory); virtual ~VCMJitterBuffer(); // Makes |this| a deep copy of |rhs|. @@ -274,13 +271,10 @@ class VCMJitterBuffer { uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const; - int vcm_id_; - int receiver_id_; Clock* clock_; // If we are running (have started) or not. bool running_; CriticalSectionWrapper* crit_sect_; - bool master_; // Event to signal when we have a frame ready for decoder. scoped_ptr<EventWrapper> frame_event_; // Event to signal when we have received a packet. diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc index e535a8a4043..0490658b420 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc @@ -27,8 +27,8 @@ class TestBasicJitterBuffer : public ::testing::Test { protected: virtual void SetUp() { clock_.reset(new SimulatedClock(0)); - jitter_buffer_.reset(new VCMJitterBuffer(clock_.get(), - &event_factory_, -1, -1, true)); + jitter_buffer_.reset( + new VCMJitterBuffer(clock_.get(), &event_factory_)); jitter_buffer_->Start(); seq_num_ = 1234; timestamp_ = 0; @@ -126,8 +126,7 @@ class TestRunningJitterBuffer : public ::testing::Test { clock_.reset(new SimulatedClock(0)); max_nack_list_size_ = 150; oldest_packet_to_nack_ = 250; - jitter_buffer_ = new VCMJitterBuffer(clock_.get(), &event_factory_, -1, -1, - true); + jitter_buffer_ = new VCMJitterBuffer(clock_.get(), &event_factory_); stream_generator_ = new StreamGenerator(0, 0, clock_->TimeInMilliseconds()); jitter_buffer_->Start(); jitter_buffer_->SetNackSettings(max_nack_list_size_, @@ -2034,4 +2033,31 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap2) { EXPECT_EQ(65535, list[0]); } +TEST_F(TestJitterBufferNack, ResetByFutureKeyFrameDoesntError) { + stream_generator_->Init(0, 0, clock_->TimeInMilliseconds()); + InsertFrame(kVideoFrameKey); + EXPECT_TRUE(DecodeCompleteFrame()); + uint16_t nack_list_size = 0; + bool extended = false; + jitter_buffer_->GetNackList(&nack_list_size, &extended); + EXPECT_EQ(0, nack_list_size); + + // Far-into-the-future video frame, could be caused by resetting the encoder + // or otherwise restarting. This should not fail when error when the packet is + // a keyframe, even if all of the nack list needs to be flushed. + stream_generator_->Init(10000, 0, clock_->TimeInMilliseconds()); + clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs); + InsertFrame(kVideoFrameKey); + EXPECT_TRUE(DecodeCompleteFrame()); + jitter_buffer_->GetNackList(&nack_list_size, &extended); + EXPECT_EQ(0, nack_list_size); + + // Stream should be decodable from this point. + clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs); + InsertFrame(kVideoFrameDelta); + EXPECT_TRUE(DecodeCompleteFrame()); + jitter_buffer_->GetNackList(&nack_list_size, &extended); + EXPECT_EQ(0, nack_list_size); +} + } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_estimator.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_estimator.cc index deb036300ee..71c54a00cd2 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_estimator.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/jitter_estimator.cc @@ -11,7 +11,6 @@ #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/source/jitter_estimator.h" #include "webrtc/modules/video_coding/main/source/rtt_filter.h" -#include "webrtc/system_wrappers/interface/trace.h" #include <assert.h> #include <math.h> @@ -20,20 +19,20 @@ namespace webrtc { -VCMJitterEstimator::VCMJitterEstimator(int32_t vcmId, int32_t receiverId) : -_vcmId(vcmId), -_receiverId(receiverId), -_phi(0.97), -_psi(0.9999), -_alphaCountMax(400), -_thetaLow(0.000001), -_nackLimit(3), -_numStdDevDelayOutlier(15), -_numStdDevFrameSizeOutlier(3), -_noiseStdDevs(2.33), // ~Less than 1% chance - // (look up in normal distribution table)... -_noiseStdDevOffset(30.0), // ...of getting 30 ms freezes -_rttFilter(vcmId, receiverId) { +VCMJitterEstimator::VCMJitterEstimator(int32_t vcmId, int32_t receiverId) + : _vcmId(vcmId), + _receiverId(receiverId), + _phi(0.97), + _psi(0.9999), + _alphaCountMax(400), + _thetaLow(0.000001), + _nackLimit(3), + _numStdDevDelayOutlier(15), + _numStdDevFrameSizeOutlier(3), + _noiseStdDevs(2.33), // ~Less than 1% chance + // (look up in normal distribution table)... + _noiseStdDevOffset(30.0), // ...of getting 30 ms freezes + _rttFilter() { Reset(); } @@ -108,10 +107,6 @@ void VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS, uint32_t frameSizeBytes, bool incompleteFrame /* = false */) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(_vcmId, _receiverId), - "Jitter estimate updated with: frameSize=%d frameDelayMS=%d", - frameSizeBytes, frameDelayMS); if (frameSizeBytes == 0) { return; @@ -162,7 +157,7 @@ VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS, uint32_t frameSizeBytes // deviation is probably due to an incorrect line slope. double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS); - if (abs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) || + if (fabs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) || frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize)) { // Update the variance of the deviation from the @@ -195,16 +190,6 @@ VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS, uint32_t frameSizeBytes { _startupCount++; } - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "Framesize statistics: max=%f average=%f", _maxFrameSize, _avgFrameSize); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "The estimated slope is: theta=(%f, %f)", _theta[0], _theta[1]); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "Random jitter: mean=%f variance=%f", _avgNoise, _varNoise); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "Current jitter estimate: %f", _filterJitterEstimate); - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "Current max RTT: %u", _rttFilter.RttMs()); } // Updates the nack/packet ratio @@ -257,7 +242,7 @@ VCMJitterEstimator::KalmanEstimateChannel(int64_t frameDelayMS, { return; } - double sigma = (300.0 * exp(-abs(static_cast<double>(deltaFSBytes)) / + double sigma = (300.0 * exp(-fabs(static_cast<double>(deltaFSBytes)) / (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise); if (sigma < 1.0) { diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.cc index 27fa6819338..4dc72253be3 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.cc @@ -14,13 +14,68 @@ #include "webrtc/modules/video_coding/main/source/qm_select.h" #include "webrtc/modules/video_coding/utility/include/frame_dropper.h" #include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { namespace media_optimization { +namespace { +void UpdateProtectionCallback( + VCMProtectionMethod* selected_method, + uint32_t* video_rate_bps, + uint32_t* nack_overhead_rate_bps, + uint32_t* fec_overhead_rate_bps, + VCMProtectionCallback* video_protection_callback) { + FecProtectionParams delta_fec_params; + FecProtectionParams key_fec_params; + // Get the FEC code rate for Key frames (set to 0 when NA). + key_fec_params.fec_rate = selected_method->RequiredProtectionFactorK(); + + // Get the FEC code rate for Delta frames (set to 0 when NA). + delta_fec_params.fec_rate = selected_method->RequiredProtectionFactorD(); + + // Get the FEC-UEP protection status for Key frames: UEP on/off. + key_fec_params.use_uep_protection = selected_method->RequiredUepProtectionK(); + + // Get the FEC-UEP protection status for Delta frames: UEP on/off. + delta_fec_params.use_uep_protection = + selected_method->RequiredUepProtectionD(); -MediaOptimization::MediaOptimization(int32_t id, Clock* clock) - : id_(id), - clock_(clock), + // The RTP module currently requires the same |max_fec_frames| for both + // key and delta frames. + delta_fec_params.max_fec_frames = selected_method->MaxFramesFec(); + key_fec_params.max_fec_frames = selected_method->MaxFramesFec(); + + // Set the FEC packet mask type. |kFecMaskBursty| is more effective for + // consecutive losses and little/no packet re-ordering. As we currently + // do not have feedback data on the degree of correlated losses and packet + // re-ordering, we keep default setting to |kFecMaskRandom| for now. + delta_fec_params.fec_mask_type = kFecMaskRandom; + key_fec_params.fec_mask_type = kFecMaskRandom; + + // TODO(Marco): Pass FEC protection values per layer. + video_protection_callback->ProtectionRequest(&delta_fec_params, + &key_fec_params, + video_rate_bps, + nack_overhead_rate_bps, + fec_overhead_rate_bps); +} +} // namespace + +struct MediaOptimization::EncodedFrameSample { + EncodedFrameSample(int size_bytes, + uint32_t timestamp, + int64_t time_complete_ms) + : size_bytes(size_bytes), + timestamp(timestamp), + time_complete_ms(time_complete_ms) {} + + uint32_t size_bytes; + uint32_t timestamp; + int64_t time_complete_ms; +}; + +MediaOptimization::MediaOptimization(Clock* clock) + : clock_(clock), max_bit_rate_(0), send_codec_type_(kVideoCodecUnknown), codec_width_(0), @@ -35,8 +90,6 @@ MediaOptimization::MediaOptimization(int32_t id, Clock* clock) target_bit_rate_(0), incoming_frame_rate_(0), enable_qm_(false), - video_protection_callback_(NULL), - video_qmsettings_callback_(NULL), encoded_frame_samples_(), avg_sent_bit_rate_bps_(0), avg_sent_framerate_(0), @@ -59,7 +112,8 @@ MediaOptimization::~MediaOptimization(void) { loss_prot_logic_->Release(); } -int32_t MediaOptimization::Reset() { +void MediaOptimization::Reset() { + SetEncodingData(kVideoCodecUnknown, 0, 0, 0, 0, 0, 0, max_payload_size_); memset(incoming_frame_times_, -1, sizeof(incoming_frame_times_)); incoming_frame_rate_ = 0.0; frame_dropper_->Reset(); @@ -81,12 +135,52 @@ int32_t MediaOptimization::Reset() { encoded_frame_samples_.clear(); avg_sent_bit_rate_bps_ = 0; num_layers_ = 1; - return VCM_OK; } -uint32_t MediaOptimization::SetTargetRates(uint32_t target_bitrate, - uint8_t fraction_lost, - uint32_t round_trip_time_ms) { +void MediaOptimization::SetEncodingData(VideoCodecType send_codec_type, + int32_t max_bit_rate, + uint32_t frame_rate, + uint32_t target_bitrate, + uint16_t width, + uint16_t height, + int num_layers, + int32_t mtu) { + // Everything codec specific should be reset here since this means the codec + // has changed. If native dimension values have changed, then either user + // initiated change, or QM initiated change. Will be able to determine only + // after the processing of the first frame. + last_change_time_ = clock_->TimeInMilliseconds(); + content_->Reset(); + content_->UpdateFrameRate(frame_rate); + + max_bit_rate_ = max_bit_rate; + send_codec_type_ = send_codec_type; + target_bit_rate_ = target_bitrate; + float target_bitrate_kbps = static_cast<float>(target_bitrate) / 1000.0f; + loss_prot_logic_->UpdateBitRate(target_bitrate_kbps); + loss_prot_logic_->UpdateFrameRate(static_cast<float>(frame_rate)); + loss_prot_logic_->UpdateFrameSize(width, height); + loss_prot_logic_->UpdateNumLayers(num_layers); + frame_dropper_->Reset(); + frame_dropper_->SetRates(target_bitrate_kbps, static_cast<float>(frame_rate)); + user_frame_rate_ = static_cast<float>(frame_rate); + codec_width_ = width; + codec_height_ = height; + num_layers_ = (num_layers <= 1) ? 1 : num_layers; // Can also be zero. + max_payload_size_ = mtu; + qm_resolution_->Initialize(target_bitrate_kbps, + user_frame_rate_, + codec_width_, + codec_height_, + num_layers_); +} + +uint32_t MediaOptimization::SetTargetRates( + uint32_t target_bitrate, + uint8_t fraction_lost, + uint32_t round_trip_time_ms, + VCMProtectionCallback* protection_callback, + VCMQMSettingsCallback* qmsettings_callback) { // TODO(holmer): Consider putting this threshold only on the video bitrate, // and not on protection. if (max_bit_rate_ > 0 && @@ -145,10 +239,13 @@ uint32_t MediaOptimization::SetTargetRates(uint32_t target_bitrate, // Get the bit cost of protection method, based on the amount of // overhead data actually transmitted (including headers) the last // second. - UpdateProtectionCallback(selected_method, - &sent_video_rate_bps, - &sent_nack_rate_bps, - &sent_fec_rate_bps); + if (protection_callback) { + UpdateProtectionCallback(selected_method, + &sent_video_rate_bps, + &sent_nack_rate_bps, + &sent_fec_rate_bps, + protection_callback); + } uint32_t sent_total_rate_bps = sent_video_rate_bps + sent_nack_rate_bps + sent_fec_rate_bps; // Estimate the overhead costs of the next second as staying the same @@ -178,7 +275,7 @@ uint32_t MediaOptimization::SetTargetRates(uint32_t target_bitrate, static_cast<float>(target_bit_rate_) / 1000.0f; frame_dropper_->SetRates(target_video_bitrate_kbps, incoming_frame_rate_); - if (enable_qm_) { + if (enable_qm_ && qmsettings_callback) { // Update QM with rates. qm_resolution_->UpdateRates(target_video_bitrate_kbps, sent_video_rate_kbps, @@ -187,7 +284,7 @@ uint32_t MediaOptimization::SetTargetRates(uint32_t target_bitrate, // Check for QM selection. bool select_qm = CheckStatusForQMchange(); if (select_qm) { - SelectQuality(); + SelectQuality(qmsettings_callback); } // Reset the short-term averaged content data. content_->ResetShortTermAvgData(); @@ -198,44 +295,6 @@ uint32_t MediaOptimization::SetTargetRates(uint32_t target_bitrate, return target_bit_rate_; } -int32_t MediaOptimization::SetEncodingData(VideoCodecType send_codec_type, - int32_t max_bit_rate, - uint32_t frame_rate, - uint32_t target_bitrate, - uint16_t width, - uint16_t height, - int num_layers) { - // Everything codec specific should be reset here since this means the codec - // has changed. If native dimension values have changed, then either user - // initiated change, or QM initiated change. Will be able to determine only - // after the processing of the first frame. - last_change_time_ = clock_->TimeInMilliseconds(); - content_->Reset(); - content_->UpdateFrameRate(frame_rate); - - max_bit_rate_ = max_bit_rate; - send_codec_type_ = send_codec_type; - target_bit_rate_ = target_bitrate; - float target_bitrate_kbps = static_cast<float>(target_bitrate) / 1000.0f; - loss_prot_logic_->UpdateBitRate(target_bitrate_kbps); - loss_prot_logic_->UpdateFrameRate(static_cast<float>(frame_rate)); - loss_prot_logic_->UpdateFrameSize(width, height); - loss_prot_logic_->UpdateNumLayers(num_layers); - frame_dropper_->Reset(); - frame_dropper_->SetRates(target_bitrate_kbps, static_cast<float>(frame_rate)); - user_frame_rate_ = static_cast<float>(frame_rate); - codec_width_ = width; - codec_height_ = height; - num_layers_ = (num_layers <= 1) ? 1 : num_layers; // Can also be zero. - int32_t ret = VCM_OK; - ret = qm_resolution_->Initialize(target_bitrate_kbps, - user_frame_rate_, - codec_width_, - codec_height_, - num_layers_); - return ret; -} - void MediaOptimization::EnableProtectionMethod(bool enable, VCMProtectionMethodEnum method) { bool updated = false; @@ -249,11 +308,6 @@ void MediaOptimization::EnableProtectionMethod(bool enable, } } -bool MediaOptimization::IsProtectionMethodEnabled( - VCMProtectionMethodEnum method) { - return (loss_prot_logic_->SelectedType() == method); -} - uint32_t MediaOptimization::InputFrameRate() { ProcessIncomingFrameRate(clock_->TimeInMilliseconds()); return uint32_t(incoming_frame_rate_ + 0.5f); @@ -272,6 +326,13 @@ uint32_t MediaOptimization::SentBitRate() { return avg_sent_bit_rate_bps_; } +VCMFrameCount MediaOptimization::SentFrameCount() { + VCMFrameCount count; + count.numDeltaFrames = delta_frame_cnt_; + count.numKeyFrames = key_frame_cnt_; + return count; +} + int32_t MediaOptimization::UpdateWithEncodedData(int encoded_length, uint32_t timestamp, FrameType encoded_frame_type) { @@ -325,29 +386,14 @@ int32_t MediaOptimization::UpdateWithEncodedData(int encoded_length, return VCM_OK; } -int32_t MediaOptimization::RegisterProtectionCallback( - VCMProtectionCallback* protection_callback) { - video_protection_callback_ = protection_callback; - return VCM_OK; -} - -int32_t MediaOptimization::RegisterVideoQMCallback( - VCMQMSettingsCallback* video_qmsettings) { - video_qmsettings_callback_ = video_qmsettings; - // Callback setting controls QM. - if (video_qmsettings_callback_ != NULL) { - enable_qm_ = true; - } else { - enable_qm_ = false; - } - return VCM_OK; -} +void MediaOptimization::EnableQM(bool enable) { enable_qm_ = enable; } void MediaOptimization::EnableFrameDropper(bool enable) { frame_dropper_->Enable(enable); } bool MediaOptimization::DropFrame() { + UpdateIncomingFrameRate(); // Leak appropriate number of bytes. frame_dropper_->Leak((uint32_t)(InputFrameRate() + 0.5f)); if (video_suspended_) { @@ -356,12 +402,6 @@ bool MediaOptimization::DropFrame() { return frame_dropper_->DropFrame(); } -int32_t MediaOptimization::SentFrameCount(VCMFrameCount* frame_count) const { - frame_count->numDeltaFrames = delta_frame_cnt_; - frame_count->numKeyFrames = key_frame_cnt_; - return VCM_OK; -} - void MediaOptimization::UpdateIncomingFrameRate() { int64_t now = clock_->TimeInMilliseconds(); if (incoming_frame_times_[0] == 0) { @@ -388,7 +428,8 @@ void MediaOptimization::UpdateContentData( } } -int32_t MediaOptimization::SelectQuality() { +int32_t MediaOptimization::SelectQuality( + VCMQMSettingsCallback* video_qmsettings_callback) { // Reset quantities for QM select. qm_resolution_->ResetQM(); @@ -403,7 +444,7 @@ int32_t MediaOptimization::SelectQuality() { } // Check for updates to spatial/temporal modes. - QMUpdate(qm); + QMUpdate(qm, video_qmsettings_callback); // Reset all the rate and related frame counters quantities. qm_resolution_->ResetRates(); @@ -426,50 +467,7 @@ void MediaOptimization::SuspendBelowMinBitrate(int threshold_bps, video_suspended_ = false; } -// Private methods below this line. - -int MediaOptimization::UpdateProtectionCallback( - VCMProtectionMethod* selected_method, - uint32_t* video_rate_bps, - uint32_t* nack_overhead_rate_bps, - uint32_t* fec_overhead_rate_bps) { - if (!video_protection_callback_) { - return VCM_OK; - } - FecProtectionParams delta_fec_params; - FecProtectionParams key_fec_params; - // Get the FEC code rate for Key frames (set to 0 when NA). - key_fec_params.fec_rate = selected_method->RequiredProtectionFactorK(); - - // Get the FEC code rate for Delta frames (set to 0 when NA). - delta_fec_params.fec_rate = selected_method->RequiredProtectionFactorD(); - - // Get the FEC-UEP protection status for Key frames: UEP on/off. - key_fec_params.use_uep_protection = selected_method->RequiredUepProtectionK(); - - // Get the FEC-UEP protection status for Delta frames: UEP on/off. - delta_fec_params.use_uep_protection = - selected_method->RequiredUepProtectionD(); - - // The RTP module currently requires the same |max_fec_frames| for both - // key and delta frames. - delta_fec_params.max_fec_frames = selected_method->MaxFramesFec(); - key_fec_params.max_fec_frames = selected_method->MaxFramesFec(); - - // Set the FEC packet mask type. |kFecMaskBursty| is more effective for - // consecutive losses and little/no packet re-ordering. As we currently - // do not have feedback data on the degree of correlated losses and packet - // re-ordering, we keep default setting to |kFecMaskRandom| for now. - delta_fec_params.fec_mask_type = kFecMaskRandom; - key_fec_params.fec_mask_type = kFecMaskRandom; - - // TODO(Marco): Pass FEC protection values per layer. - return video_protection_callback_->ProtectionRequest(&delta_fec_params, - &key_fec_params, - video_rate_bps, - nack_overhead_rate_bps, - fec_overhead_rate_bps); -} +bool MediaOptimization::IsVideoSuspended() const { return video_suspended_; } void MediaOptimization::PurgeOldFrameSamples(int64_t now_ms) { while (!encoded_frame_samples_.empty()) { @@ -518,7 +516,9 @@ void MediaOptimization::UpdateSentFramerate() { } } -bool MediaOptimization::QMUpdate(VCMResolutionScale* qm) { +bool MediaOptimization::QMUpdate( + VCMResolutionScale* qm, + VCMQMSettingsCallback* video_qmsettings_callback) { // Check for no change. if (!qm->change_resolution_spatial && !qm->change_resolution_temporal) { return false; @@ -537,13 +537,9 @@ bool MediaOptimization::QMUpdate(VCMResolutionScale* qm) { codec_height_ = qm->codec_height; } - WEBRTC_TRACE(webrtc::kTraceDebug, - webrtc::kTraceVideoCoding, - id_, - "Resolution change from QM select: W = %d, H = %d, FR = %f", - qm->codec_width, - qm->codec_height, - qm->frame_rate); + LOG(LS_INFO) << "Media optimizer requests the video resolution to be changed " + "to " << qm->codec_width << "x" << qm->codec_height << "@" + << qm->frame_rate; // Update VPM with new target frame rate and frame size. // Note: use |qm->frame_rate| instead of |_incoming_frame_rate| for updating @@ -551,7 +547,7 @@ bool MediaOptimization::QMUpdate(VCMResolutionScale* qm) { // will vary/fluctuate, and since we don't want to change the state of the // VPM frame dropper, unless a temporal action was selected, we use the // quantity |qm->frame_rate| for updating. - video_qmsettings_callback_->SetVideoQMSettings( + video_qmsettings_callback->SetVideoQMSettings( qm->frame_rate, codec_width_, codec_height_); content_->UpdateFrameRate(qm->frame_rate); qm_resolution_->UpdateCodecParameters( diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.h b/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.h index cde28d23288..35a49712504 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization.h @@ -18,7 +18,6 @@ #include "webrtc/modules/video_coding/main/source/media_opt_util.h" #include "webrtc/modules/video_coding/main/source/qm_select.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { @@ -29,33 +28,25 @@ class VCMContentMetricsProcessing; namespace media_optimization { -enum { - kBitrateMaxFrameSamples = 60 -}; -enum { - kBitrateAverageWinMs = 1000 -}; - -struct EncodedFrameSample { - EncodedFrameSample(int size_bytes, - uint32_t timestamp, - int64_t time_complete_ms) - : size_bytes(size_bytes), - timestamp(timestamp), - time_complete_ms(time_complete_ms) {} - - uint32_t size_bytes; - uint32_t timestamp; - int64_t time_complete_ms; -}; - +// TODO(andresp): Make thread safe. class MediaOptimization { public: - MediaOptimization(int32_t id, Clock* clock); - ~MediaOptimization(void); + explicit MediaOptimization(Clock* clock); + ~MediaOptimization(); + + // TODO(andresp): Can Reset and SetEncodingData be done at construction time + // only? + void Reset(); - // Resets the Media Optimization module. - int32_t Reset(); + // Informs media optimization of initial encoding state. + void SetEncodingData(VideoCodecType send_codec_type, + int32_t max_bit_rate, + uint32_t frame_rate, + uint32_t bit_rate, + uint16_t width, + uint16_t height, + int num_temporal_layers, + int32_t mtu); // Sets target rates for the encoder given the channel parameters. // Inputs: target bitrate - the encoder target bitrate in bits/s. @@ -63,95 +54,64 @@ class MediaOptimization { // round_trip_time_ms - round trip time in milliseconds. // min_bit_rate - the bit rate of the end-point with lowest rate. // max_bit_rate - the bit rate of the end-point with highest rate. + // TODO(andresp): Find if the callbacks can be triggered only after releasing + // an internal critical section. uint32_t SetTargetRates(uint32_t target_bitrate, uint8_t fraction_lost, - uint32_t round_trip_time_ms); + uint32_t round_trip_time_ms, + VCMProtectionCallback* protection_callback, + VCMQMSettingsCallback* qmsettings_callback); - // Informs media optimization of initial encoding state. - int32_t SetEncodingData(VideoCodecType send_codec_type, - int32_t max_bit_rate, - uint32_t frame_rate, - uint32_t bit_rate, - uint16_t width, - uint16_t height, - int num_temporal_layers); - - // Enables protection method. void EnableProtectionMethod(bool enable, VCMProtectionMethodEnum method); + void EnableQM(bool enable); + void EnableFrameDropper(bool enable); - // Returns weather or not protection method is enabled. - bool IsProtectionMethodEnabled(VCMProtectionMethodEnum method); - - // Returns the actual input frame rate. - uint32_t InputFrameRate(); + // Lets the sender suspend video when the rate drops below + // |threshold_bps|, and turns back on when the rate goes back up above + // |threshold_bps| + |window_bps|. + void SuspendBelowMinBitrate(int threshold_bps, int window_bps); + bool IsVideoSuspended() const; - // Returns the actual sent frame rate. - uint32_t SentFrameRate(); + bool DropFrame(); - // Returns the actual sent bit rate. - uint32_t SentBitRate(); + void UpdateContentData(const VideoContentMetrics* content_metrics); // Informs Media Optimization of encoding output: Length and frame type. int32_t UpdateWithEncodedData(int encoded_length, uint32_t timestamp, FrameType encoded_frame_type); - // Registers a protection callback to be used to inform the user about the - // protection methods used. - int32_t RegisterProtectionCallback( - VCMProtectionCallback* protection_callback); - - // Registers a quality settings callback to be used to inform VPM/user. - int32_t RegisterVideoQMCallback(VCMQMSettingsCallback* video_qmsettings); - - void EnableFrameDropper(bool enable); - - bool DropFrame(); - - // Returns the number of key/delta frames encoded. - int32_t SentFrameCount(VCMFrameCount* frame_count) const; - - // Updates incoming frame rate value. - void UpdateIncomingFrameRate(); - - // Update content metric data. - void UpdateContentData(const VideoContentMetrics* content_metrics); - - // Computes new Quality Mode. - int32_t SelectQuality(); - - // Lets the sender suspend video when the rate drops below - // |threshold_bps|, and turns back on when the rate goes back up above - // |threshold_bps| + |window_bps|. - void SuspendBelowMinBitrate(int threshold_bps, int window_bps); - - // Accessors and mutators. - int32_t max_bit_rate() const { return max_bit_rate_; } - void set_max_payload_size(int32_t mtu) { max_payload_size_ = mtu; } - bool video_suspended() const { return video_suspended_; } + uint32_t InputFrameRate(); + uint32_t SentFrameRate(); + uint32_t SentBitRate(); + VCMFrameCount SentFrameCount(); private: - typedef std::list<EncodedFrameSample> FrameSampleList; enum { kFrameCountHistorySize = 90 }; enum { kFrameHistoryWinMs = 2000 }; + enum { + kBitrateAverageWinMs = 1000 + }; - // Updates protection callback with protection settings. - int UpdateProtectionCallback(VCMProtectionMethod* selected_method, - uint32_t* total_video_rate_bps, - uint32_t* nack_overhead_rate_bps, - uint32_t* fec_overhead_rate_bps); + struct EncodedFrameSample; + typedef std::list<EncodedFrameSample> FrameSampleList; + void UpdateIncomingFrameRate(); void PurgeOldFrameSamples(int64_t now_ms); void UpdateSentBitrate(int64_t now_ms); void UpdateSentFramerate(); + // Computes new Quality Mode. + int32_t SelectQuality(VCMQMSettingsCallback* qmsettings_callback); + // Verifies if QM settings differ from default, i.e. if an update is required. // Computes actual values, as will be sent to the encoder. - bool QMUpdate(VCMResolutionScale* qm); + bool QMUpdate(VCMResolutionScale* qm, + VCMQMSettingsCallback* qmsettings_callback); // Checks if we should make a QM change. Return true if yes, false otherwise. bool CheckStatusForQMchange(); @@ -163,7 +123,6 @@ class MediaOptimization { // the state of |video_suspended_| accordingly. void CheckSuspendConditions(); - int32_t id_; Clock* clock_; int32_t max_bit_rate_; VideoCodecType send_codec_type_; @@ -180,8 +139,6 @@ class MediaOptimization { float incoming_frame_rate_; int64_t incoming_frame_times_[kFrameCountHistorySize]; bool enable_qm_; - VCMProtectionCallback* video_protection_callback_; - VCMQMSettingsCallback* video_qmsettings_callback_; std::list<EncodedFrameSample> encoded_frame_samples_; uint32_t avg_sent_bit_rate_bps_; uint32_t avg_sent_framerate_; @@ -196,8 +153,7 @@ class MediaOptimization { bool video_suspended_; int suspension_threshold_bps_; int suspension_window_bps_; -}; // End of MediaOptimization class declaration. - +}; } // namespace media_optimization } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc index 1425dad837b..bacfdc604ca 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc @@ -18,9 +18,6 @@ namespace media_optimization { class TestMediaOptimization : public ::testing::Test { protected: enum { - kId = 4711 // Id number for the MediaOptimization class. - }; - enum { kSampleRate = 90000 // RTP timestamps per second. }; @@ -28,14 +25,13 @@ class TestMediaOptimization : public ::testing::Test { // a special case (e.g. frame rate in media optimization). TestMediaOptimization() : clock_(1000), - media_opt_(kId, &clock_), + media_opt_(&clock_), frame_time_ms_(33), next_timestamp_(0) {} // This method mimics what happens in VideoSender::AddVideoFrame. void AddFrameAndAdvanceTime(int bitrate_bps, bool expect_frame_drop) { ASSERT_GE(bitrate_bps, 0); - media_opt_.UpdateIncomingFrameRate(); bool frame_dropped = media_opt_.DropFrame(); EXPECT_EQ(expect_frame_drop, frame_dropped); if (!frame_dropped) { @@ -63,12 +59,14 @@ TEST_F(TestMediaOptimization, VerifyMuting) { media_opt_.SuspendBelowMinBitrate(kThresholdBps, kWindowBps); // The video should not be suspended from the start. - EXPECT_FALSE(media_opt_.video_suspended()); + EXPECT_FALSE(media_opt_.IsVideoSuspended()); int target_bitrate_kbps = 100; media_opt_.SetTargetRates(target_bitrate_kbps * 1000, 0, // Lossrate. - 100); // RTT in ms. + 100, + NULL, + NULL); // RTT in ms. media_opt_.EnableFrameDropper(true); for (int time = 0; time < 2000; time += frame_time_ms_) { ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, false)); @@ -77,11 +75,13 @@ TEST_F(TestMediaOptimization, VerifyMuting) { // Set the target rate below the limit for muting. media_opt_.SetTargetRates(kThresholdBps - 1000, 0, // Lossrate. - 100); // RTT in ms. + 100, + NULL, + NULL); // RTT in ms. // Expect the muter to engage immediately and stay muted. // Test during 2 seconds. for (int time = 0; time < 2000; time += frame_time_ms_) { - EXPECT_TRUE(media_opt_.video_suspended()); + EXPECT_TRUE(media_opt_.IsVideoSuspended()); ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, true)); } @@ -89,22 +89,26 @@ TEST_F(TestMediaOptimization, VerifyMuting) { // limit + window. media_opt_.SetTargetRates(kThresholdBps + 1000, 0, // Lossrate. - 100); // RTT in ms. - // Expect the muter to stay muted. + 100, + NULL, + NULL); // RTT in ms. + // Expect the muter to stay muted. // Test during 2 seconds. for (int time = 0; time < 2000; time += frame_time_ms_) { - EXPECT_TRUE(media_opt_.video_suspended()); + EXPECT_TRUE(media_opt_.IsVideoSuspended()); ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, true)); } // Set the target above limit + window. media_opt_.SetTargetRates(kThresholdBps + kWindowBps + 1000, 0, // Lossrate. - 100); // RTT in ms. + 100, + NULL, + NULL); // RTT in ms. // Expect the muter to disengage immediately. // Test during 2 seconds. for (int time = 0; time < 2000; time += frame_time_ms_) { - EXPECT_FALSE(media_opt_.video_suspended()); + EXPECT_FALSE(media_opt_.IsVideoSuspended()); ASSERT_NO_FATAL_FAILURE( AddFrameAndAdvanceTime((kThresholdBps + kWindowBps) / 1000, false)); } diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/packet.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/packet.cc index 61ef2ee859a..c1f1a048e8f 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/packet.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/packet.cc @@ -19,6 +19,7 @@ VCMPacket::VCMPacket() : payloadType(0), timestamp(0), + ntp_time_ms_(0), seqNum(0), dataPtr(NULL), sizeBytes(0), @@ -38,6 +39,7 @@ VCMPacket::VCMPacket(const uint8_t* ptr, const WebRtcRTPHeader& rtpHeader) : payloadType(rtpHeader.header.payloadType), timestamp(rtpHeader.header.timestamp), + ntp_time_ms_(rtpHeader.ntp_time_ms), seqNum(rtpHeader.header.sequenceNumber), dataPtr(ptr), sizeBytes(size), @@ -58,6 +60,7 @@ VCMPacket::VCMPacket(const uint8_t* ptr, VCMPacket::VCMPacket(const uint8_t* ptr, uint32_t size, uint16_t seq, uint32_t ts, bool mBit) : payloadType(0), timestamp(ts), + ntp_time_ms_(0), seqNum(seq), dataPtr(ptr), sizeBytes(size), @@ -76,6 +79,7 @@ VCMPacket::VCMPacket(const uint8_t* ptr, uint32_t size, uint16_t seq, uint32_t t void VCMPacket::Reset() { payloadType = 0; timestamp = 0; + ntp_time_ms_ = 0; seqNum = 0; dataPtr = NULL; sizeBytes = 0; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/packet.h b/chromium/third_party/webrtc/modules/video_coding/main/source/packet.h index e9a81bb1fd2..242d3a43142 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/packet.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/packet.h @@ -33,6 +33,8 @@ public: uint8_t payloadType; uint32_t timestamp; + // NTP time of the capture time in local timebase in milliseconds. + int64_t ntp_time_ms_; uint16_t seqNum; const uint8_t* dataPtr; uint32_t sizeBytes; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.cc index ae13ddd4211..e179423a77a 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.cc @@ -12,11 +12,13 @@ #include <assert.h> +#include <cstdlib> + #include "webrtc/modules/video_coding/main/source/encoded_frame.h" #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/source/media_opt_util.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -26,15 +28,11 @@ enum { kMaxReceiverDelayMs = 10000 }; VCMReceiver::VCMReceiver(VCMTiming* timing, Clock* clock, EventFactory* event_factory, - int32_t vcm_id, - int32_t receiver_id, bool master) : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vcm_id_(vcm_id), clock_(clock), - receiver_id_(receiver_id), master_(master), - jitter_buffer_(clock_, event_factory, vcm_id, receiver_id, master), + jitter_buffer_(clock_, event_factory), timing_(timing), render_wait_event_(event_factory->CreateEvent()), state_(kPassive), @@ -76,13 +74,6 @@ void VCMReceiver::UpdateRtt(uint32_t rtt) { int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, uint16_t frame_width, uint16_t frame_height) { - if (packet.frameType == kVideoFrameKey) { - WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "Inserting key frame packet seqnum=%u, timestamp=%u", - packet.seqNum, packet.timestamp); - } - // Insert the packet into the jitter buffer. The packet can either be empty or // contain media at this point. bool retransmitted = false; @@ -93,10 +84,6 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, } else if (ret == kFlushIndicator) { return VCM_FLUSH_INDICATOR; } else if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "Error inserting packet seqnum=%u, timestamp=%u", - packet.seqNum, packet.timestamp); return VCM_JITTER_BUFFER_ERROR; } if (ret == kCompleteSession && !retransmitted) { @@ -105,15 +92,6 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, // delay within the jitter estimate. timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds()); } - if (master_) { - // Only trace the primary receiver to make it possible to parse and plot - // the trace file. - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "Packet seqnum=%u timestamp=%u inserted at %u", - packet.seqNum, packet.timestamp, - MaskWord64ToUWord32(clock_->TimeInMilliseconds())); - } return VCM_OK; } @@ -156,20 +134,17 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding( // Assume that render timing errors are due to changes in the video stream. if (next_render_time_ms < 0) { timing_error = true; - } else if (abs(next_render_time_ms - now_ms) > max_video_delay_ms_) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "This frame is out of our delay bounds, resetting jitter " - "buffer: %d > %d", - static_cast<int>(abs(next_render_time_ms - now_ms)), - max_video_delay_ms_); + } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) { + int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms)); + LOG(LS_WARNING) << "A frame about to be decoded is out of the configured " + << "delay bounds (" << frame_delay << " > " + << max_video_delay_ms_ + << "). Resetting the video jitter buffer."; timing_error = true; } else if (static_cast<int>(timing_->TargetVideoDelay()) > max_video_delay_ms_) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, receiver_id_), - "More than %u ms target delay. Flushing jitter buffer and" - "resetting timing.", max_video_delay_ms_); + LOG(LS_WARNING) << "The video target delay has grown larger than " + << max_video_delay_ms_ << " ms. Resetting jitter buffer."; timing_error = true; } @@ -278,10 +253,7 @@ VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list, bool request_key_frame = false; uint16_t* internal_nack_list = jitter_buffer_.GetNackList( nack_list_length, &request_key_frame); - if (*nack_list_length > size) { - *nack_list_length = 0; - return kNackNeedMoreMemory; - } + assert(*nack_list_length <= size); if (internal_nack_list != NULL && *nack_list_length > 0) { memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t)); } diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.h b/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.h index ac510ea22b3..c037303978f 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/receiver.h @@ -25,7 +25,6 @@ class VCMEncodedFrame; enum VCMNackStatus { kNackOk, - kNackNeedMoreMemory, kNackKeyFrameRequest }; @@ -40,8 +39,6 @@ class VCMReceiver { VCMReceiver(VCMTiming* timing, Clock* clock, EventFactory* event_factory, - int32_t vcm_id, - int32_t receiver_id, bool master); ~VCMReceiver(); @@ -95,9 +92,7 @@ class VCMReceiver { static int32_t GenerateReceiverId(); CriticalSectionWrapper* crit_sect_; - int32_t vcm_id_; Clock* clock_; - int32_t receiver_id_; bool master_; VCMJitterBuffer jitter_buffer_; VCMTiming* timing_; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/receiver_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/receiver_unittest.cc index 33a3d95f969..e80b9cc9e9e 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/receiver_unittest.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/receiver_unittest.cc @@ -31,7 +31,7 @@ class TestVCMReceiver : public ::testing::Test { TestVCMReceiver() : clock_(new SimulatedClock(0)), timing_(clock_.get()), - receiver_(&timing_, clock_.get(), &event_factory_, 1, 1, true) { + receiver_(&timing_, clock_.get(), &event_factory_, true) { stream_generator_.reset(new StreamGenerator(0, 0, clock_->TimeInMilliseconds())); memset(data_buffer_, 0, kDataBufferSize); diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.cc index 25d89e54c08..739cc824c37 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.cc @@ -10,7 +10,6 @@ #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/source/rtt_filter.h" -#include "webrtc/system_wrappers/interface/trace.h" #include <math.h> #include <stdlib.h> @@ -18,15 +17,11 @@ namespace webrtc { -VCMRttFilter::VCMRttFilter(int32_t vcmId, int32_t receiverId) -: -_vcmId(vcmId), -_receiverId(receiverId), -_filtFactMax(35), -_jumpStdDevs(2.5), -_driftStdDevs(3.5), -_detectThreshold(kMaxDriftJumpCount) -{ +VCMRttFilter::VCMRttFilter() + : _filtFactMax(35), + _jumpStdDevs(2.5), + _driftStdDevs(3.5), + _detectThreshold(kMaxDriftJumpCount) { Reset(); } @@ -105,16 +100,13 @@ VCMRttFilter::Update(uint32_t rttMs) _avgRtt = oldAvg; _varRtt = oldVar; } - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "RttFilter Update: sample=%u avgRtt=%f varRtt=%f maxRtt=%u", - rttMs, _avgRtt, _varRtt, _maxRtt); } bool VCMRttFilter::JumpDetection(uint32_t rttMs) { double diffFromAvg = _avgRtt - rttMs; - if (abs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt)) + if (fabs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt)) { int diffSign = (diffFromAvg >= 0) ? 1 : -1; int jumpCountSign = (_jumpCount >= 0) ? 1 : -1; @@ -141,8 +133,6 @@ VCMRttFilter::JumpDetection(uint32_t rttMs) ShortRttFilter(_jumpBuf, abs(_jumpCount)); _filtFactCount = _detectThreshold + 1; _jumpCount = 0; - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "Detected an RTT jump"); } else { @@ -174,8 +164,6 @@ VCMRttFilter::DriftDetection(uint32_t rttMs) ShortRttFilter(_driftBuf, _driftCount); _filtFactCount = _detectThreshold + 1; _driftCount = 0; - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), - "Detected an RTT drift"); } } else diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.h b/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.h index 9ce3798b053..8b816a0b46c 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/rtt_filter.h @@ -19,7 +19,7 @@ namespace webrtc class VCMRttFilter { public: - VCMRttFilter(int32_t vcmId = 0, int32_t receiverId = 0); + VCMRttFilter(); VCMRttFilter& operator=(const VCMRttFilter& rhs); @@ -48,8 +48,6 @@ private: // Computes the short time average and maximum of the vector buf. void ShortRttFilter(uint32_t* buf, uint32_t length); - int32_t _vcmId; - int32_t _receiverId; bool _gotNonZeroUpdate; double _avgRtt; double _varRtt; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.cc index 1cb60d39b58..dab3da1383d 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.cc @@ -11,6 +11,7 @@ #include "webrtc/modules/video_coding/main/source/session_info.h" #include "webrtc/modules/video_coding/main/source/packet.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { @@ -403,6 +404,7 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet, } if (packets_.size() == kMaxPacketsInSession) { + LOG(LS_ERROR) << "Max number of packets per frame has been reached."; return -1; } @@ -429,6 +431,8 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet, first_packet_seq_num_ = static_cast<int>(packet.seqNum); } else if (first_packet_seq_num_ != -1 && !IsNewerSequenceNumber(packet.seqNum, first_packet_seq_num_)) { + LOG(LS_WARNING) << "Received packet with a sequence number which is out of" + "frame boundaries"; return -3; } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) { // Update the frame type with the type of the first media packet. @@ -441,6 +445,8 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet, last_packet_seq_num_ = static_cast<int>(packet.seqNum); } else if (last_packet_seq_num_ != -1 && IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_)) { + LOG(LS_WARNING) << "Received packet with a sequence number which is out of" + "frame boundaries"; return -3; } diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.h b/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.h index 039f09763a2..cae3ee13892 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/session_info.h @@ -147,7 +147,6 @@ class VCMSessionInfo { bool complete_; bool decodable_; webrtc::FrameType frame_type_; - bool previous_frame_loss_; // Packets in this frame. PacketList packets_; int empty_seq_num_low_; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc deleted file mode 100644 index 1d911a54e44..00000000000 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc +++ /dev/null @@ -1,248 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/video_coding/main/source/internal_defines.h" -#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -VCMTimestampExtrapolator::VCMTimestampExtrapolator(Clock* clock, - int32_t vcmId, - int32_t id) -: -_rwLock(RWLockWrapper::CreateRWLock()), -_vcmId(vcmId), -_id(id), -_clock(clock), -_startMs(0), -_firstTimestamp(0), -_wrapArounds(0), -_prevUnwrappedTimestamp(-1), -_prevWrapTimestamp(-1), -_lambda(1), -_firstAfterReset(true), -_packetCount(0), -_startUpFilterDelayInPackets(2), -_detectorAccumulatorPos(0), -_detectorAccumulatorNeg(0), -_alarmThreshold(60e3), -_accDrift(6600), // in timestamp ticks, i.e. 15 ms -_accMaxError(7000), -_P11(1e10) -{ - Reset(); -} - -VCMTimestampExtrapolator::~VCMTimestampExtrapolator() -{ - delete _rwLock; -} - -void -VCMTimestampExtrapolator::Reset() -{ - WriteLockScoped wl(*_rwLock); - _startMs = _clock->TimeInMilliseconds(); - _prevMs = _startMs; - _firstTimestamp = 0; - _w[0] = 90.0; - _w[1] = 0; - _P[0][0] = 1; - _P[1][1] = _P11; - _P[0][1] = _P[1][0] = 0; - _firstAfterReset = true; - _prevUnwrappedTimestamp = -1; - _prevWrapTimestamp = -1; - _wrapArounds = 0; - _packetCount = 0; - _detectorAccumulatorPos = 0; - _detectorAccumulatorNeg = 0; -} - -void -VCMTimestampExtrapolator::Update(int64_t tMs, uint32_t ts90khz, bool trace) -{ - - _rwLock->AcquireLockExclusive(); - if (tMs - _prevMs > 10e3) - { - // Ten seconds without a complete frame. - // Reset the extrapolator - _rwLock->ReleaseLockExclusive(); - Reset(); - _rwLock->AcquireLockExclusive(); - } - else - { - _prevMs = tMs; - } - - // Remove offset to prevent badly scaled matrices - tMs -= _startMs; - - CheckForWrapArounds(ts90khz); - - int64_t unwrapped_ts90khz = static_cast<int64_t>(ts90khz) + - _wrapArounds * ((static_cast<int64_t>(1) << 32) - 1); - - if (_prevUnwrappedTimestamp >= 0 && - unwrapped_ts90khz < _prevUnwrappedTimestamp) - { - // Drop reordered frames. - _rwLock->ReleaseLockExclusive(); - return; - } - - if (_firstAfterReset) - { - // Make an initial guess of the offset, - // should be almost correct since tMs - _startMs - // should about zero at this time. - _w[1] = -_w[0] * tMs; - _firstTimestamp = unwrapped_ts90khz; - _firstAfterReset = false; - } - - double residual = - (static_cast<double>(unwrapped_ts90khz) - _firstTimestamp) - - static_cast<double>(tMs) * _w[0] - _w[1]; - if (DelayChangeDetection(residual, trace) && - _packetCount >= _startUpFilterDelayInPackets) - { - // A sudden change of average network delay has been detected. - // Force the filter to adjust its offset parameter by changing - // the offset uncertainty. Don't do this during startup. - _P[1][1] = _P11; - } - //T = [t(k) 1]'; - //that = T'*w; - //K = P*T/(lambda + T'*P*T); - double K[2]; - K[0] = _P[0][0] * tMs + _P[0][1]; - K[1] = _P[1][0] * tMs + _P[1][1]; - double TPT = _lambda + tMs * K[0] + K[1]; - K[0] /= TPT; - K[1] /= TPT; - //w = w + K*(ts(k) - that); - _w[0] = _w[0] + K[0] * residual; - _w[1] = _w[1] + K[1] * residual; - //P = 1/lambda*(P - K*T'*P); - double p00 = 1 / _lambda * (_P[0][0] - (K[0] * tMs * _P[0][0] + K[0] * _P[1][0])); - double p01 = 1 / _lambda * (_P[0][1] - (K[0] * tMs * _P[0][1] + K[0] * _P[1][1])); - _P[1][0] = 1 / _lambda * (_P[1][0] - (K[1] * tMs * _P[0][0] + K[1] * _P[1][0])); - _P[1][1] = 1 / _lambda * (_P[1][1] - (K[1] * tMs * _P[0][1] + K[1] * _P[1][1])); - _P[0][0] = p00; - _P[0][1] = p01; - _prevUnwrappedTimestamp = unwrapped_ts90khz; - if (_packetCount < _startUpFilterDelayInPackets) - { - _packetCount++; - } - if (trace) - { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "w[0]=%f w[1]=%f ts=%u tMs=%u", _w[0], _w[1], ts90khz, tMs); - } - _rwLock->ReleaseLockExclusive(); -} - -int64_t -VCMTimestampExtrapolator::ExtrapolateLocalTime(uint32_t timestamp90khz) -{ - ReadLockScoped rl(*_rwLock); - int64_t localTimeMs = 0; - CheckForWrapArounds(timestamp90khz); - double unwrapped_ts90khz = static_cast<double>(timestamp90khz) + - _wrapArounds * ((static_cast<int64_t>(1) << 32) - 1); - if (_packetCount == 0) - { - localTimeMs = -1; - } - else if (_packetCount < _startUpFilterDelayInPackets) - { - localTimeMs = _prevMs + static_cast<int64_t>( - static_cast<double>(unwrapped_ts90khz - _prevUnwrappedTimestamp) / - 90.0 + 0.5); - } - else - { - if (_w[0] < 1e-3) - { - localTimeMs = _startMs; - } - else - { - double timestampDiff = unwrapped_ts90khz - - static_cast<double>(_firstTimestamp); - localTimeMs = static_cast<int64_t>( - static_cast<double>(_startMs) + (timestampDiff - _w[1]) / - _w[0] + 0.5); - } - } - return localTimeMs; -} - -// Investigates if the timestamp clock has overflowed since the last timestamp and -// keeps track of the number of wrap arounds since reset. -void -VCMTimestampExtrapolator::CheckForWrapArounds(uint32_t ts90khz) -{ - if (_prevWrapTimestamp == -1) - { - _prevWrapTimestamp = ts90khz; - return; - } - if (ts90khz < _prevWrapTimestamp) - { - // This difference will probably be less than -2^31 if we have had a wrap around - // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is casted to a Word32, - // it should be positive. - if (static_cast<int32_t>(ts90khz - _prevWrapTimestamp) > 0) - { - // Forward wrap around - _wrapArounds++; - } - } - // This difference will probably be less than -2^31 if we have had a backward wrap around. - // Since it is casted to a Word32, it should be positive. - else if (static_cast<int32_t>(_prevWrapTimestamp - ts90khz) > 0) - { - // Backward wrap around - _wrapArounds--; - } - _prevWrapTimestamp = ts90khz; -} - -bool -VCMTimestampExtrapolator::DelayChangeDetection(double error, bool trace) -{ - // CUSUM detection of sudden delay changes - error = (error > 0) ? VCM_MIN(error, _accMaxError) : VCM_MAX(error, -_accMaxError); - _detectorAccumulatorPos = VCM_MAX(_detectorAccumulatorPos + error - _accDrift, (double)0); - _detectorAccumulatorNeg = VCM_MIN(_detectorAccumulatorNeg + error + _accDrift, (double)0); - if (_detectorAccumulatorPos > _alarmThreshold || _detectorAccumulatorNeg < -_alarmThreshold) - { - // Alarm - if (trace) - { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "g1=%f g2=%f alarm=1", _detectorAccumulatorPos, _detectorAccumulatorNeg); - } - _detectorAccumulatorPos = _detectorAccumulatorNeg = 0; - return true; - } - if (trace) - { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "g1=%f g2=%f alarm=0", _detectorAccumulatorPos, _detectorAccumulatorNeg); - } - return false; -} - -} diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h b/chromium/third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h deleted file mode 100644 index 4565186a353..00000000000 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h +++ /dev/null @@ -1,63 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_ -#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_ - -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/typedefs.h" - -namespace webrtc -{ - -class Clock; - -class VCMTimestampExtrapolator -{ -public: - VCMTimestampExtrapolator(Clock* clock, - int32_t vcmId = 0, - int32_t receiverId = 0); - ~VCMTimestampExtrapolator(); - void Update(int64_t tMs, uint32_t ts90khz, bool trace = true); - int64_t ExtrapolateLocalTime(uint32_t timestamp90khz); - void Reset(); - -private: - void CheckForWrapArounds(uint32_t ts90khz); - bool DelayChangeDetection(double error, bool trace = true); - RWLockWrapper* _rwLock; - int32_t _vcmId; - int32_t _id; - Clock* _clock; - double _w[2]; - double _P[2][2]; - int64_t _startMs; - int64_t _prevMs; - uint32_t _firstTimestamp; - int32_t _wrapArounds; - int64_t _prevUnwrappedTimestamp; - int64_t _prevWrapTimestamp; - const double _lambda; - bool _firstAfterReset; - uint32_t _packetCount; - const uint32_t _startUpFilterDelayInPackets; - - double _detectorAccumulatorPos; - double _detectorAccumulatorNeg; - const double _alarmThreshold; - const double _accDrift; - const double _accMaxError; - const double _P11; -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_ diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/timing.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/timing.cc index 98a69e962fd..af0e35c4e1f 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/timing.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/timing.cc @@ -10,25 +10,18 @@ #include "webrtc/modules/video_coding/main/source/timing.h" - #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" -#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace.h" - +#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h" namespace webrtc { VCMTiming::VCMTiming(Clock* clock, - int32_t vcm_id, - int32_t timing_id, VCMTiming* master_timing) : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vcm_id_(vcm_id), clock_(clock), - timing_id_(timing_id), master_(false), ts_extrapolator_(), codec_timer_(), @@ -40,7 +33,7 @@ VCMTiming::VCMTiming(Clock* clock, prev_frame_timestamp_(0) { if (master_timing == NULL) { master_ = true; - ts_extrapolator_ = new VCMTimestampExtrapolator(clock_, vcm_id, timing_id); + ts_extrapolator_ = new TimestampExtrapolator(clock_->TimeInMilliseconds()); } else { ts_extrapolator_ = master_timing->ts_extrapolator_; } @@ -55,7 +48,7 @@ VCMTiming::~VCMTiming() { void VCMTiming::Reset() { CriticalSectionScoped cs(crit_sect_); - ts_extrapolator_->Reset(); + ts_extrapolator_->Reset(clock_->TimeInMilliseconds()); codec_timer_.Reset(); render_delay_ms_ = kDefaultRenderDelayMs; min_playout_delay_ms_ = 0; @@ -81,11 +74,6 @@ void VCMTiming::set_min_playout_delay(uint32_t min_playout_delay_ms) { void VCMTiming::SetJitterDelay(uint32_t jitter_delay_ms) { CriticalSectionScoped cs(crit_sect_); if (jitter_delay_ms != jitter_delay_ms_) { - if (master_) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, timing_id_), - "Desired jitter buffer level: %u ms", jitter_delay_ms); - } jitter_delay_ms_ = jitter_delay_ms; // When in initial state, set current delay to minimum delay. if (current_delay_ms_ == 0) { @@ -152,39 +140,21 @@ int32_t VCMTiming::StopDecodeTimer(uint32_t time_stamp, int64_t start_time_ms, int64_t now_ms) { CriticalSectionScoped cs(crit_sect_); - const int32_t max_dec_time = MaxDecodeTimeMs(); int32_t time_diff_ms = codec_timer_.StopTimer(start_time_ms, now_ms); - if (time_diff_ms < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(vcm_id_, - timing_id_), "Codec timer error: %d", time_diff_ms); - assert(false); - } + assert(time_diff_ms >= 0); last_decode_ms_ = time_diff_ms; - if (master_) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(vcm_id_, - timing_id_), - "Frame decoded: time_stamp=%u dec_time=%d max_dec_time=%u, at %u", - time_stamp, time_diff_ms, max_dec_time, MaskWord64ToUWord32(now_ms)); - } return 0; } void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) { CriticalSectionScoped cs(crit_sect_); - ts_extrapolator_->Update(now_ms, time_stamp, master_); + ts_extrapolator_->Update(now_ms, time_stamp); } int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) const { CriticalSectionScoped cs(crit_sect_); const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms); - if (master_) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(vcm_id_, - timing_id_), "Render frame %u at %u. Render delay %u", - "jitter delay %u, max decode time %u, playout delay %u", - frame_timestamp, MaskWord64ToUWord32(render_time_ms), render_delay_ms_, - jitter_delay_ms_, MaxDecodeTimeMs(), min_playout_delay_ms_); - } return render_time_ms; } @@ -192,11 +162,6 @@ int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp, int64_t now_ms) const { int64_t estimated_complete_time_ms = ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp); - if (master_) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, timing_id_), "ExtrapolateLocalTime(%u)=%u ms", - frame_timestamp, MaskWord64ToUWord32(estimated_complete_time_ms)); - } if (estimated_complete_time_ms == -1) { estimated_complete_time_ms = now_ms; } @@ -210,11 +175,7 @@ int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp, int32_t VCMTiming::MaxDecodeTimeMs(FrameType frame_type /*= kVideoFrameDelta*/) const { const int32_t decode_time_ms = codec_timer_.RequiredDecodeTimeMs(frame_type); - if (decode_time_ms < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(vcm_id_, - timing_id_), "Negative maximum decode time: %d", decode_time_ms); - return -1; - } + assert(decode_time_ms >= 0); return decode_time_ms; } @@ -254,11 +215,6 @@ uint32_t VCMTiming::TargetVideoDelay() const { } uint32_t VCMTiming::TargetDelayInternal() const { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, - VCMId(vcm_id_, timing_id_), - "Delay: min_playout=%u jitter=%u max_decode=%u render=%u", - min_playout_delay_ms_, jitter_delay_ms_, MaxDecodeTimeMs(), - render_delay_ms_); return std::max(min_playout_delay_ms_, jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_); } diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/timing.h b/chromium/third_party/webrtc/modules/video_coding/main/source/timing.h index eb251b71153..1dca5e605af 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/timing.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/timing.h @@ -18,15 +18,13 @@ namespace webrtc { class Clock; -class VCMTimestampExtrapolator; +class TimestampExtrapolator; class VCMTiming { public: // The primary timing component should be passed // if this is the dual timing component. VCMTiming(Clock* clock, - int32_t vcm_id = 0, - int32_t timing_id = 0, VCMTiming* master_timing = NULL); ~VCMTiming(); @@ -101,11 +99,9 @@ class VCMTiming { private: CriticalSectionWrapper* crit_sect_; - int32_t vcm_id_; Clock* clock_; - int32_t timing_id_; bool master_; - VCMTimestampExtrapolator* ts_extrapolator_; + TimestampExtrapolator* ts_extrapolator_; VCMCodecTimer codec_timer_; uint32_t render_delay_ms_; uint32_t min_playout_delay_ms_; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding.gypi b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding.gypi index b4f6cb7b1c8..f19a585523d 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding.gypi +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding.gypi @@ -48,7 +48,6 @@ 'receiver.h', 'rtt_filter.h', 'session_info.h', - 'timestamp_extrapolator.h', 'timestamp_map.h', 'timing.h', 'video_coding_impl.h', @@ -72,7 +71,6 @@ 'receiver.cc', 'rtt_filter.cc', 'session_info.cc', - 'timestamp_extrapolator.cc', 'timestamp_map.cc', 'timing.cc', 'video_coding_impl.cc', diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.cc index 1decc2f112f..5b93a656717 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.cc @@ -16,7 +16,6 @@ #include "webrtc/modules/video_coding/main/source/packet.h" #include "webrtc/modules/video_coding/main/source/video_coding_impl.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -45,15 +44,44 @@ VCMProcessTimer::Processed() { } // namespace vcm namespace { +// This wrapper provides a way to modify the callback without the need to expose +// a register method all the way down to the function calling it. +class EncodedImageCallbackWrapper : public EncodedImageCallback { + public: + EncodedImageCallbackWrapper() + : cs_(CriticalSectionWrapper::CreateCriticalSection()), callback_(NULL) {} + + virtual ~EncodedImageCallbackWrapper() {} + + void Register(EncodedImageCallback* callback) { + CriticalSectionScoped cs(cs_.get()); + callback_ = callback; + } + + // TODO(andresp): Change to void as return value is ignored. + virtual int32_t Encoded(EncodedImage& encoded_image, + const CodecSpecificInfo* codec_specific_info, + const RTPFragmentationHeader* fragmentation) { + CriticalSectionScoped cs(cs_.get()); + if (callback_) + return callback_->Encoded( + encoded_image, codec_specific_info, fragmentation); + return 0; + } + + private: + scoped_ptr<CriticalSectionWrapper> cs_; + EncodedImageCallback* callback_ GUARDED_BY(cs_); +}; + class VideoCodingModuleImpl : public VideoCodingModule { public: - VideoCodingModuleImpl(const int32_t id, - Clock* clock, + VideoCodingModuleImpl(Clock* clock, EventFactory* event_factory, bool owns_event_factory) : VideoCodingModule(), - sender_(new vcm::VideoSender(id, clock)), - receiver_(new vcm::VideoReceiver(id, clock, event_factory)), + sender_(new vcm::VideoSender(clock, &post_encode_callback_)), + receiver_(new vcm::VideoReceiver(clock, event_factory)), own_event_factory_(owns_event_factory ? event_factory : NULL) {} virtual ~VideoCodingModuleImpl() { @@ -194,7 +222,8 @@ class VideoCodingModuleImpl : public VideoCodingModule { } virtual int StopDebugRecording() OVERRIDE { - return sender_->StopDebugRecording(); + sender_->StopDebugRecording(); + return VCM_OK; } virtual void SuspendBelowMinBitrate() { @@ -326,10 +355,11 @@ class VideoCodingModuleImpl : public VideoCodingModule { virtual void RegisterPostEncodeImageCallback( EncodedImageCallback* observer) OVERRIDE { - sender_->RegisterPostEncodeImageCallback(observer); + post_encode_callback_.Register(observer); } private: + EncodedImageCallbackWrapper post_encode_callback_; scoped_ptr<vcm::VideoSender> sender_; scoped_ptr<vcm::VideoReceiver> receiver_; scoped_ptr<EventFactory> own_event_factory_; @@ -354,17 +384,16 @@ int32_t VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec) { return VCMCodecDataBase::Codec(codecType, codec) ? 0 : -1; } -VideoCodingModule* VideoCodingModule::Create(const int32_t id) { +VideoCodingModule* VideoCodingModule::Create() { return new VideoCodingModuleImpl( - id, Clock::GetRealTimeClock(), new EventFactoryImpl, true); + Clock::GetRealTimeClock(), new EventFactoryImpl, true); } -VideoCodingModule* VideoCodingModule::Create(const int32_t id, - Clock* clock, +VideoCodingModule* VideoCodingModule::Create(Clock* clock, EventFactory* event_factory) { assert(clock); assert(event_factory); - return new VideoCodingModuleImpl(id, clock, event_factory, false); + return new VideoCodingModuleImpl(clock, event_factory, false); } void VideoCodingModule::Destroy(VideoCodingModule* module) { diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.h b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.h index d9564c04bd9..bf0bc7905ba 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.h +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_impl.h @@ -32,6 +32,8 @@ class EncodedFrameObserver; namespace vcm { +class DebugRecorder; + class VCMProcessTimer { public: VCMProcessTimer(uint32_t periodMs, Clock* clock) @@ -52,7 +54,8 @@ class VideoSender { public: typedef VideoCodingModule::SenderNackMode SenderNackMode; - VideoSender(const int32_t id, Clock* clock); + VideoSender(Clock* clock, EncodedImageCallback* post_encode_callback); + ~VideoSender(); int32_t InitializeSender(); @@ -68,7 +71,8 @@ class VideoSender { uint8_t payloadType, bool internalSource); - int32_t CodecConfigParameters(uint8_t* buffer, int32_t size); + int32_t CodecConfigParameters(uint8_t* buffer, int32_t size) const; + int32_t SentFrameCount(VCMFrameCount* frameCount); int Bitrate(unsigned int* bitrate) const; int FrameRate(unsigned int* framerate) const; @@ -88,7 +92,6 @@ class VideoSender { int32_t IntraFrameRequest(int stream_index); int32_t EnableFrameDropper(bool enable); - int32_t SentFrameCount(VCMFrameCount* frameCount) const; int SetSenderNackMode(SenderNackMode mode); int SetSenderReferenceSelection(bool enable); @@ -96,21 +99,19 @@ class VideoSender { int SetSenderKeyFramePeriod(int periodMs); int StartDebugRecording(const char* file_name_utf8); - int StopDebugRecording(); + void StopDebugRecording(); void SuspendBelowMinBitrate(); bool VideoSuspended() const; - void RegisterPostEncodeImageCallback( - EncodedImageCallback* post_encode_callback); - int32_t TimeUntilNextProcess(); int32_t Process(); private: - int32_t _id; Clock* clock_; + scoped_ptr<DebugRecorder> recorder_; + scoped_ptr<CriticalSectionWrapper> process_crit_sect_; CriticalSectionWrapper* _sendCritSect; VCMGenericEncoder* _encoder; @@ -118,17 +119,19 @@ class VideoSender { std::vector<FrameType> _nextFrameTypes; media_optimization::MediaOptimization _mediaOpt; VCMSendStatisticsCallback* _sendStatsCallback; - FILE* _encoderInputFile; VCMCodecDataBase _codecDataBase; bool frame_dropper_enabled_; VCMProcessTimer _sendStatsTimer; + + VCMQMSettingsCallback* qm_settings_callback_; + VCMProtectionCallback* protection_callback_; }; class VideoReceiver { public: typedef VideoCodingModule::ReceiverRobustness ReceiverRobustness; - VideoReceiver(const int32_t id, Clock* clock, EventFactory* event_factory); + VideoReceiver(Clock* clock, EventFactory* event_factory); ~VideoReceiver(); int32_t InitializeReceiver(); @@ -196,7 +199,6 @@ class VideoReceiver { // in any frame }; - int32_t _id; Clock* clock_; scoped_ptr<CriticalSectionWrapper> process_crit_sect_; CriticalSectionWrapper* _receiveCritSect; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc index fbb511448c4..435e18202f7 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc @@ -35,7 +35,7 @@ class VCMRobustnessTest : public ::testing::Test { virtual void SetUp() { clock_.reset(new SimulatedClock(0)); ASSERT_TRUE(clock_.get() != NULL); - vcm_ = VideoCodingModule::Create(0, clock_.get(), &event_factory_); + vcm_ = VideoCodingModule::Create(clock_.get(), &event_factory_); ASSERT_TRUE(vcm_ != NULL); ASSERT_EQ(0, vcm_->InitializeReceiver()); const size_t kMaxNackListSize = 250; diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_test.gypi b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_test.gypi index a64e02d02c8..b0fe510cf9c 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_test.gypi +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_coding_test.gypi @@ -20,6 +20,7 @@ '<(webrtc_root)/test/test.gyp:test_support', '<(webrtc_root)/test/metrics.gyp:metrics', '<(webrtc_root)/common_video/common_video.gyp:common_video', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', ], 'sources': [ # headers diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver.cc index 68668eae7de..5bc1c90f52c 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver.cc @@ -16,7 +16,7 @@ #include "webrtc/modules/video_coding/main/source/packet.h" #include "webrtc/modules/video_coding/main/source/video_coding_impl.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/trace_event.h" // #define DEBUG_DECODER_BIT_STREAM @@ -24,18 +24,15 @@ namespace webrtc { namespace vcm { -VideoReceiver::VideoReceiver(const int32_t id, - Clock* clock, - EventFactory* event_factory) - : _id(id), - clock_(clock), +VideoReceiver::VideoReceiver(Clock* clock, EventFactory* event_factory) + : clock_(clock), process_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), _receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()), _receiverInited(false), - _timing(clock_, id, 1), - _dualTiming(clock_, id, 2, &_timing), - _receiver(&_timing, clock_, event_factory, id, 1, true), - _dualReceiver(&_dualTiming, clock_, event_factory, id, 2, false), + _timing(clock_), + _dualTiming(clock_, &_timing), + _receiver(&_timing, clock_, event_factory, true), + _dualReceiver(&_dualTiming, clock_, event_factory, false), _decodedFrameCallback(_timing, clock_), _dualDecodedFrameCallback(_dualTiming, clock_), _frameTypeCallback(NULL), @@ -53,7 +50,7 @@ VideoReceiver::VideoReceiver(const int32_t id, _scheduleKeyRequest(false), max_nack_list_size_(0), pre_decode_image_callback_(NULL), - _codecDataBase(id), + _codecDataBase(), _receiveStatsTimer(1000, clock_), _retransmissionTimer(10, clock_), _keyRequestTimer(500, clock_) { @@ -121,8 +118,12 @@ int32_t VideoReceiver::Process() { // Key frame requests if (_keyRequestTimer.TimeUntilProcess() == 0) { _keyRequestTimer.Processed(); - CriticalSectionScoped cs(process_crit_sect_.get()); - if (_scheduleKeyRequest && _frameTypeCallback != NULL) { + bool request_key_frame = false; + { + CriticalSectionScoped cs(process_crit_sect_.get()); + request_key_frame = _scheduleKeyRequest && _frameTypeCallback != NULL; + } + if (request_key_frame) { const int32_t ret = RequestKeyFrame(); if (ret != VCM_OK && returnValue == VCM_OK) { returnValue = ret; @@ -135,16 +136,24 @@ int32_t VideoReceiver::Process() { // disabled when NACK is off. if (_retransmissionTimer.TimeUntilProcess() == 0) { _retransmissionTimer.Processed(); - CriticalSectionScoped cs(process_crit_sect_.get()); - if (_packetRequestCallback != NULL) { - uint16_t length = max_nack_list_size_; + bool callback_registered = false; + uint16_t length; + { + CriticalSectionScoped cs(process_crit_sect_.get()); + length = max_nack_list_size_; + callback_registered = _packetRequestCallback != NULL; + } + if (callback_registered && length > 0) { std::vector<uint16_t> nackList(length); const int32_t ret = NackList(&nackList[0], &length); if (ret != VCM_OK && returnValue == VCM_OK) { returnValue = ret; } - if (length > 0) { - _packetRequestCallback->ResendPackets(&nackList[0], length); + if (ret == VCM_OK && length > 0) { + CriticalSectionScoped cs(process_crit_sect_.get()); + if (_packetRequestCallback != NULL) { + _packetRequestCallback->ResendPackets(&nackList[0], length); + } } } } @@ -434,17 +443,9 @@ int32_t VideoReceiver::RequestSliceLossIndication( const int32_t ret = _frameTypeCallback->SliceLossIndicationRequest(pictureID); if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Failed to request key frame"); return ret; } } else { - WEBRTC_TRACE(webrtc::kTraceWarning, - webrtc::kTraceVideoCoding, - VCMId(_id), - "No frame type request callback registered"); return VCM_MISSING_CALLBACK; } return VCM_OK; @@ -452,22 +453,14 @@ int32_t VideoReceiver::RequestSliceLossIndication( int32_t VideoReceiver::RequestKeyFrame() { TRACE_EVENT0("webrtc", "RequestKeyFrame"); - CriticalSectionScoped cs(process_crit_sect_.get()); + CriticalSectionScoped process_cs(process_crit_sect_.get()); if (_frameTypeCallback != NULL) { const int32_t ret = _frameTypeCallback->RequestKeyFrame(); if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Failed to request key frame"); return ret; } _scheduleKeyRequest = false; } else { - WEBRTC_TRACE(webrtc::kTraceWarning, - webrtc::kTraceVideoCoding, - VCMId(_id), - "No frame type request callback registered"); return VCM_MISSING_CALLBACK; } return VCM_OK; @@ -490,29 +483,18 @@ int32_t VideoReceiver::DecodeDualFrame(uint16_t maxWaitTimeMs) { VCMEncodedFrame* dualFrame = _dualReceiver.FrameForDecoding(maxWaitTimeMs, dummyRenderTime); if (dualFrame != NULL && _dualDecoder != NULL) { - WEBRTC_TRACE(webrtc::kTraceStream, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Decoding frame %u with dual decoder", - dualFrame->TimeStamp()); // Decode dualFrame and try to catch up int32_t ret = _dualDecoder->Decode(*dualFrame, clock_->TimeInMilliseconds()); if (ret != WEBRTC_VIDEO_CODEC_OK) { - WEBRTC_TRACE(webrtc::kTraceWarning, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Failed to decode frame with dual decoder"); + LOG(LS_ERROR) << "Failed to decode frame with dual decoder. Error code: " + << ret; _dualReceiver.ReleaseFrame(dualFrame); return VCM_CODEC_ERROR; } if (_receiver.DualDecoderCaughtUp(dualFrame, _dualReceiver)) { // Copy the complete decoder state of the dual decoder // to the primary decoder. - WEBRTC_TRACE(webrtc::kTraceStream, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Dual decoder caught up"); _codecDataBase.CopyDecoder(*_dualDecoder); _codecDataBase.ReleaseDecoder(_dualDecoder); _dualDecoder = NULL; @@ -547,62 +529,65 @@ int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) { int32_t ret = _decoder->Decode(frame, clock_->TimeInMilliseconds()); // Check for failed decoding, run frame type request callback if needed. + bool request_key_frame = false; if (ret < 0) { if (ret == VCM_ERROR_REQUEST_SLI) { return RequestSliceLossIndication( _decodedFrameCallback.LastReceivedPictureID() + 1); } else { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Failed to decode frame %u, requesting key frame", - frame.TimeStamp()); - ret = RequestKeyFrame(); + request_key_frame = true; } } else if (ret == VCM_REQUEST_SLI) { ret = RequestSliceLossIndication( _decodedFrameCallback.LastReceivedPictureID() + 1); } if (!frame.Complete() || frame.MissingFrame()) { - CriticalSectionScoped cs(process_crit_sect_.get()); switch (_keyRequestMode) { case kKeyOnKeyLoss: { if (frame.FrameType() == kVideoFrameKey) { - _scheduleKeyRequest = true; - return VCM_OK; + request_key_frame = true; + ret = VCM_OK; } break; } case kKeyOnLoss: { - _scheduleKeyRequest = true; - return VCM_OK; + request_key_frame = true; + ret = VCM_OK; } default: break; } } + if (request_key_frame) { + CriticalSectionScoped cs(process_crit_sect_.get()); + _scheduleKeyRequest = true; + } TRACE_EVENT_ASYNC_END0("webrtc", "Video", frame.TimeStamp()); return ret; } // Reset the decoder state int32_t VideoReceiver::ResetDecoder() { - CriticalSectionScoped cs(_receiveCritSect); - if (_decoder != NULL) { - _receiver.Initialize(); - _timing.Reset(); - { - CriticalSectionScoped cs(process_crit_sect_.get()); - _scheduleKeyRequest = false; + bool reset_key_request = false; + { + CriticalSectionScoped cs(_receiveCritSect); + if (_decoder != NULL) { + _receiver.Initialize(); + _timing.Reset(); + reset_key_request = true; + _decoder->Reset(); + } + if (_dualReceiver.State() != kPassive) { + _dualReceiver.Initialize(); + } + if (_dualDecoder != NULL) { + _codecDataBase.ReleaseDecoder(_dualDecoder); + _dualDecoder = NULL; } - _decoder->Reset(); - } - if (_dualReceiver.State() != kPassive) { - _dualReceiver.Initialize(); } - if (_dualDecoder != NULL) { - _codecDataBase.ReleaseDecoder(_dualDecoder); - _dualDecoder = NULL; + if (reset_key_request) { + CriticalSectionScoped cs(process_crit_sect_.get()); + _scheduleKeyRequest = false; } return VCM_OK; } @@ -710,25 +695,8 @@ int32_t VideoReceiver::NackList(uint16_t* nackList, uint16_t* size) { nackStatus = _dualReceiver.NackList(nackList, *size, &nack_list_length); } *size = nack_list_length; - - switch (nackStatus) { - case kNackNeedMoreMemory: { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Out of memory"); - return VCM_MEMORY; - } - case kNackKeyFrameRequest: { - CriticalSectionScoped cs(_receiveCritSect); - WEBRTC_TRACE(webrtc::kTraceWarning, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Failed to get NACK list, requesting key frame"); + if (nackStatus == kNackKeyFrameRequest) { return RequestKeyFrame(); - } - default: - break; } return VCM_OK; } @@ -763,14 +731,17 @@ int VideoReceiver::SetReceiverRobustnessMode( _keyRequestMode = kKeyOnError; // TODO(hlundin): On long NACK list? break; case VideoCodingModule::kSoftNack: +#if 1 assert(false); // TODO(hlundin): Not completed. return VCM_NOT_IMPLEMENTED; +#else // Enable hybrid NACK/FEC. Always wait for retransmissions and don't add // extra delay when RTT is above kLowRttNackMs. _receiver.SetNackMode(kNack, media_optimization::kLowRttNackMs, -1); _dualReceiver.SetNackMode(kNoNack, -1, -1); _keyRequestMode = kKeyOnError; break; +#endif case VideoCodingModule::kDualDecoder: if (decode_error_mode == kNoErrors) { return VCM_PARAMETER_ERROR; @@ -783,14 +754,17 @@ int VideoReceiver::SetReceiverRobustnessMode( _keyRequestMode = kKeyOnError; break; case VideoCodingModule::kReferenceSelection: +#if 1 assert(false); // TODO(hlundin): Not completed. return VCM_NOT_IMPLEMENTED; +#else if (decode_error_mode == kNoErrors) { return VCM_PARAMETER_ERROR; } _receiver.SetNackMode(kNoNack, -1, -1); _dualReceiver.SetNackMode(kNoNack, -1, -1); break; +#endif } _receiver.SetDecodeErrorMode(decode_error_mode); // The dual decoder should never decode with errors. diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc index 4fd524d4bce..502dfa9dd2f 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc @@ -33,7 +33,7 @@ class TestVideoReceiver : public ::testing::Test { TestVideoReceiver() : clock_(0) {} virtual void SetUp() { - receiver_.reset(new VideoReceiver(0, &clock_, &event_factory_)); + receiver_.reset(new VideoReceiver(&clock_, &event_factory_)); EXPECT_EQ(0, receiver_->InitializeReceiver()); EXPECT_EQ(0, receiver_->RegisterExternalDecoder( diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender.cc index 948218b83a6..38ecc5479a4 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender.cc @@ -17,30 +17,66 @@ #include "webrtc/modules/video_coding/main/source/encoded_frame.h" #include "webrtc/modules/video_coding/main/source/video_coding_impl.h" #include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/logging.h" namespace webrtc { namespace vcm { -VideoSender::VideoSender(const int32_t id, Clock* clock) - : _id(id), - clock_(clock), +class DebugRecorder { + public: + DebugRecorder() + : cs_(CriticalSectionWrapper::CreateCriticalSection()), file_(NULL) {} + + ~DebugRecorder() { Stop(); } + + int Start(const char* file_name_utf8) { + CriticalSectionScoped cs(cs_.get()); + if (file_) + fclose(file_); + file_ = fopen(file_name_utf8, "wb"); + if (!file_) + return VCM_GENERAL_ERROR; + return VCM_OK; + } + + void Stop() { + CriticalSectionScoped cs(cs_.get()); + if (file_) { + fclose(file_); + file_ = NULL; + } + } + + void Add(const I420VideoFrame& frame) { + CriticalSectionScoped cs(cs_.get()); + if (file_) + PrintI420VideoFrame(frame, file_); + } + + private: + scoped_ptr<CriticalSectionWrapper> cs_; + FILE* file_ GUARDED_BY(cs_); +}; + +VideoSender::VideoSender(Clock* clock, + EncodedImageCallback* post_encode_callback) + : clock_(clock), + recorder_(new DebugRecorder()), process_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), _sendCritSect(CriticalSectionWrapper::CreateCriticalSection()), _encoder(), - _encodedFrameCallback(), + _encodedFrameCallback(post_encode_callback), _nextFrameTypes(1, kVideoFrameDelta), - _mediaOpt(id, clock_), + _mediaOpt(clock_), _sendStatsCallback(NULL), - _encoderInputFile(NULL), - _codecDataBase(id), + _codecDataBase(), frame_dropper_enabled_(true), - _sendStatsTimer(1000, clock_) {} + _sendStatsTimer(1000, clock_), + qm_settings_callback_(NULL), + protection_callback_(NULL) {} VideoSender::~VideoSender() { delete _sendCritSect; - if (_encoderInputFile != NULL) { - fclose(_encoderInputFile); - } } int32_t VideoSender::Process() { @@ -70,8 +106,6 @@ int32_t VideoSender::InitializeSender() { _codecDataBase.ResetSender(); _encoder = NULL; _encodedFrameCallback.SetTransportCallback(NULL); - // setting default bitRate and frameRate to 0 - _mediaOpt.SetEncodingData(kVideoCodecUnknown, 0, 0, 0, 0, 0, 0); _mediaOpt.Reset(); // Resetting frame dropper return VCM_OK; } @@ -97,10 +131,8 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec, _encoder = _codecDataBase.GetEncoder(); if (!ret) { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Failed to initialize encoder"); + LOG(LS_ERROR) << "Failed to initialize the encoder with payload name " + << sendCodec->plName << ". Error code: " << ret; return VCM_CODEC_ERROR; } @@ -125,9 +157,8 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec, sendCodec->startBitrate * 1000, sendCodec->width, sendCodec->height, - numLayers); - _mediaOpt.set_max_payload_size(maxPayloadSize); - + numLayers, + maxPayloadSize); return VCM_OK; } @@ -171,7 +202,8 @@ int32_t VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder, } // Get codec config parameters -int32_t VideoSender::CodecConfigParameters(uint8_t* buffer, int32_t size) { +int32_t VideoSender::CodecConfigParameters(uint8_t* buffer, + int32_t size) const { CriticalSectionScoped cs(_sendCritSect); if (_encoder != NULL) { return _encoder->CodecConfigParameters(buffer, size); @@ -179,6 +211,14 @@ int32_t VideoSender::CodecConfigParameters(uint8_t* buffer, int32_t size) { return VCM_UNINITIALIZED; } +// TODO(andresp): Make const once media_opt is thread-safe and this has a +// pointer to it. +int32_t VideoSender::SentFrameCount(VCMFrameCount* frameCount) { + CriticalSectionScoped cs(_sendCritSect); + *frameCount = _mediaOpt.SentFrameCount(); + return VCM_OK; +} + // Get encode bitrate int VideoSender::Bitrate(unsigned int* bitrate) const { CriticalSectionScoped cs(_sendCritSect); @@ -208,8 +248,11 @@ int32_t VideoSender::SetChannelParameters(uint32_t target_bitrate, int32_t ret = 0; { CriticalSectionScoped sendCs(_sendCritSect); - uint32_t targetRate = - _mediaOpt.SetTargetRates(target_bitrate, lossRate, rtt); + uint32_t targetRate = _mediaOpt.SetTargetRates(target_bitrate, + lossRate, + rtt, + protection_callback_, + qm_settings_callback_); if (_encoder != NULL) { ret = _encoder->SetChannelParameters(lossRate, rtt); if (ret < 0) { @@ -247,17 +290,19 @@ int32_t VideoSender::RegisterSendStatisticsCallback( // Register a video quality settings callback which will be called when frame // rate/dimensions need to be updated for video quality optimization int32_t VideoSender::RegisterVideoQMCallback( - VCMQMSettingsCallback* videoQMSettings) { + VCMQMSettingsCallback* qm_settings_callback) { CriticalSectionScoped cs(_sendCritSect); - return _mediaOpt.RegisterVideoQMCallback(videoQMSettings); + qm_settings_callback_ = qm_settings_callback; + _mediaOpt.EnableQM(qm_settings_callback_ != NULL); + return VCM_OK; } // Register a video protection callback which will be called to deliver the // requested FEC rate and NACK status (on/off). int32_t VideoSender::RegisterProtectionCallback( - VCMProtectionCallback* protection) { + VCMProtectionCallback* protection_callback) { CriticalSectionScoped cs(_sendCritSect); - _mediaOpt.RegisterProtectionCallback(protection); + protection_callback_ = protection_callback; return VCM_OK; } @@ -314,33 +359,19 @@ int32_t VideoSender::AddVideoFrame(const I420VideoFrame& videoFrame, if (_nextFrameTypes[0] == kFrameEmpty) { return VCM_OK; } - _mediaOpt.UpdateIncomingFrameRate(); - if (_mediaOpt.DropFrame()) { - WEBRTC_TRACE(webrtc::kTraceStream, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Drop frame due to bitrate"); - } else { - _mediaOpt.UpdateContentData(contentMetrics); - int32_t ret = - _encoder->Encode(videoFrame, codecSpecificInfo, _nextFrameTypes); - if (_encoderInputFile != NULL) { - if (PrintI420VideoFrame(videoFrame, _encoderInputFile) < 0) { - return -1; - } - } - if (ret < 0) { - WEBRTC_TRACE(webrtc::kTraceError, - webrtc::kTraceVideoCoding, - VCMId(_id), - "Encode error: %d", - ret); - return ret; - } - for (size_t i = 0; i < _nextFrameTypes.size(); ++i) { - _nextFrameTypes[i] = kVideoFrameDelta; // Default frame type. - } + return VCM_OK; + } + _mediaOpt.UpdateContentData(contentMetrics); + int32_t ret = + _encoder->Encode(videoFrame, codecSpecificInfo, _nextFrameTypes); + recorder_->Add(videoFrame); + if (ret < 0) { + LOG(LS_ERROR) << "Failed to encode frame. Error code: " << ret; + return ret; + } + for (size_t i = 0; i < _nextFrameTypes.size(); ++i) { + _nextFrameTypes[i] = kVideoFrameDelta; // Default frame type. } return VCM_OK; } @@ -369,11 +400,6 @@ int32_t VideoSender::EnableFrameDropper(bool enable) { return VCM_OK; } -int32_t VideoSender::SentFrameCount(VCMFrameCount* frameCount) const { - CriticalSectionScoped cs(_sendCritSect); - return _mediaOpt.SentFrameCount(frameCount); -} - int VideoSender::SetSenderNackMode(SenderNackMode mode) { CriticalSectionScoped cs(_sendCritSect); @@ -406,20 +432,11 @@ int VideoSender::SetSenderKeyFramePeriod(int periodMs) { } int VideoSender::StartDebugRecording(const char* file_name_utf8) { - CriticalSectionScoped cs(_sendCritSect); - _encoderInputFile = fopen(file_name_utf8, "wb"); - if (_encoderInputFile == NULL) - return VCM_GENERAL_ERROR; - return VCM_OK; + return recorder_->Start(file_name_utf8); } -int VideoSender::StopDebugRecording() { - CriticalSectionScoped cs(_sendCritSect); - if (_encoderInputFile != NULL) { - fclose(_encoderInputFile); - _encoderInputFile = NULL; - } - return VCM_OK; +void VideoSender::StopDebugRecording() { + recorder_->Stop(); } void VideoSender::SuspendBelowMinBitrate() { @@ -443,14 +460,7 @@ void VideoSender::SuspendBelowMinBitrate() { bool VideoSender::VideoSuspended() const { CriticalSectionScoped cs(_sendCritSect); - return _mediaOpt.video_suspended(); + return _mediaOpt.IsVideoSuspended(); } - -void VideoSender::RegisterPostEncodeImageCallback( - EncodedImageCallback* observer) { - CriticalSectionScoped cs(_sendCritSect); - _encodedFrameCallback.RegisterPostEncodeImageCallback(observer); -} - } // namespace vcm } // namespace webrtc diff --git a/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender_unittest.cc index 513a99ee74b..67b3e7aeccb 100644 --- a/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender_unittest.cc +++ b/chromium/third_party/webrtc/modules/video_coding/main/source/video_sender_unittest.cc @@ -52,7 +52,7 @@ struct Vp8StreamInfo { MATCHER_P(MatchesVp8StreamInfo, expected, "") { bool res = true; for (int tl = 0; tl < kMaxNumberOfTemporalLayers; ++tl) { - if (abs(expected.framerate_fps[tl] - arg.framerate_fps[tl]) > 0.5) { + if (fabs(expected.framerate_fps[tl] - arg.framerate_fps[tl]) > 0.5) { *result_listener << " framerate_fps[" << tl << "] = " << arg.framerate_fps[tl] << " (expected " << expected.framerate_fps[tl] << ") "; @@ -173,7 +173,7 @@ class TestVideoSender : public ::testing::Test { TestVideoSender() : clock_(1000), packetization_callback_(&clock_) {} virtual void SetUp() { - sender_.reset(new VideoSender(0, &clock_)); + sender_.reset(new VideoSender(&clock_, &post_encode_callback_)); EXPECT_EQ(0, sender_->InitializeSender()); EXPECT_EQ(0, sender_->RegisterTransportCallback(&packetization_callback_)); } @@ -185,6 +185,7 @@ class TestVideoSender : public ::testing::Test { SimulatedClock clock_; PacketizationCallback packetization_callback_; + MockEncodedImageCallback post_encode_callback_; scoped_ptr<VideoSender> sender_; scoped_ptr<FrameGenerator> generator_; }; @@ -344,6 +345,8 @@ class TestVideoSenderWithVp8 : public TestVideoSender { void InsertFrames(float framerate, float seconds) { for (int i = 0; i < seconds * framerate; ++i) { clock_.AdvanceTimeMilliseconds(1000.0f / framerate); + EXPECT_CALL(post_encode_callback_, Encoded(_, NULL, NULL)) + .WillOnce(Return(0)); AddFrame(); // SetChannelParameters needs to be called frequently to propagate diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/OWNERS b/chromium/third_party/webrtc/modules/video_coding/utility/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_coding/utility/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_processing/main/interface/video_processing.h b/chromium/third_party/webrtc/modules/video_processing/main/interface/video_processing.h index b3e0483d06a..817d43d9bd6 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/interface/video_processing.h +++ b/chromium/third_party/webrtc/modules/video_processing/main/interface/video_processing.h @@ -236,14 +236,6 @@ class VideoProcessingModule : public Module { uint32_t frame_rate) = 0; /** - Set max frame rate - \param[in] max_frame_rate: maximum frame rate (limited to native frame rate) - - \return VPM_OK on success, a negative value on error (see error codes) - */ - virtual int32_t SetMaxFramerate(uint32_t max_frame_rate) = 0; - - /** Get decimated(target) frame rate */ virtual uint32_t Decimatedframe_rate() = 0; diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/OWNERS b/chromium/third_party/webrtc/modules/video_processing/main/source/OWNERS new file mode 100644 index 00000000000..3ee6b4bf5f9 --- /dev/null +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/brighten.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/brighten.cc index ffabbf7fff7..907a549064a 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/brighten.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/brighten.cc @@ -12,22 +12,15 @@ #include <stdlib.h> -#include "webrtc/system_wrappers/interface/trace.h" - namespace webrtc { namespace VideoProcessing { int32_t Brighten(I420VideoFrame* frame, int delta) { assert(frame); if (frame->IsZeroSize()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, - "zero size frame"); return VPM_PARAMETER_ERROR; } - if (frame->width() <= 0 || frame->height() <= 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, - "Invalid frame size"); return VPM_PARAMETER_ERROR; } diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/brightness_detection.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/brightness_detection.cc index 8817bac434b..f33117d1376 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/brightness_detection.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/brightness_detection.cc @@ -10,7 +10,6 @@ #include "webrtc/modules/video_processing/main/interface/video_processing.h" #include "webrtc/modules/video_processing/main/source/brightness_detection.h" -#include "webrtc/system_wrappers/interface/trace.h" #include <math.h> @@ -37,16 +36,12 @@ int32_t VPMBrightnessDetection::ProcessFrame( const I420VideoFrame& frame, const VideoProcessingModule::FrameStats& stats) { if (frame.IsZeroSize()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "Null frame pointer"); return VPM_PARAMETER_ERROR; } int width = frame.width(); int height = frame.height(); if (!VideoProcessingModule::ValidFrameStats(stats)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "Invalid frame stats"); return VPM_PARAMETER_ERROR; } @@ -58,7 +53,7 @@ int32_t VPMBrightnessDetection::ProcessFrame( for (uint32_t i = 0; i < low_th; i++) { prop_low += stats.hist[i]; } -prop_low /= stats.num_pixels; + prop_low /= stats.num_pixels; // Get proportion in highest bins. unsigned char high_th = 230; diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/color_enhancement.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/color_enhancement.cc index eeec01659c0..aaa3a462256 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/color_enhancement.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/color_enhancement.cc @@ -12,44 +12,38 @@ #include "webrtc/modules/video_processing/main/source/color_enhancement.h" #include "webrtc/modules/video_processing/main/source/color_enhancement_private.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { namespace VideoProcessing { int32_t ColorEnhancement(I420VideoFrame* frame) { -assert(frame); -// Pointers to U and V color pixels. -uint8_t* ptr_u; -uint8_t* ptr_v; -uint8_t temp_chroma; -if (frame->IsZeroSize()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, - -1, "Null frame pointer"); - return VPM_GENERAL_ERROR; -} - -if (frame->width() == 0 || frame->height() == 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, - -1, "Invalid frame size"); - return VPM_GENERAL_ERROR; -} - -// Set pointers to first U and V pixels (skip luminance). -ptr_u = frame->buffer(kUPlane); -ptr_v = frame->buffer(kVPlane); -int size_uv = ((frame->width() + 1) / 2) * ((frame->height() + 1) / 2); - -// Loop through all chrominance pixels and modify color. -for (int ix = 0; ix < size_uv; ix++) { - temp_chroma = colorTable[*ptr_u][*ptr_v]; - *ptr_v = colorTable[*ptr_v][*ptr_u]; - *ptr_u = temp_chroma; - - ptr_u++; - ptr_v++; -} -return VPM_OK; + assert(frame); + // Pointers to U and V color pixels. + uint8_t* ptr_u; + uint8_t* ptr_v; + uint8_t temp_chroma; + if (frame->IsZeroSize()) { + return VPM_GENERAL_ERROR; + } + if (frame->width() == 0 || frame->height() == 0) { + return VPM_GENERAL_ERROR; + } + + // Set pointers to first U and V pixels (skip luminance). + ptr_u = frame->buffer(kUPlane); + ptr_v = frame->buffer(kVPlane); + int size_uv = ((frame->width() + 1) / 2) * ((frame->height() + 1) / 2); + + // Loop through all chrominance pixels and modify color. + for (int ix = 0; ix < size_uv; ix++) { + temp_chroma = colorTable[*ptr_u][*ptr_v]; + *ptr_v = colorTable[*ptr_v][*ptr_u]; + *ptr_u = temp_chroma; + + ptr_u++; + ptr_v++; + } + return VPM_OK; } } // namespace VideoProcessing diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/deflickering.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/deflickering.cc index 898fd80f473..cdc6174883f 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/deflickering.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/deflickering.cc @@ -14,8 +14,8 @@ #include <stdlib.h> #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" +#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/sort.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { @@ -102,21 +102,16 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame, int height = frame->height(); if (frame->IsZeroSize()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "Null frame pointer"); return VPM_GENERAL_ERROR; } // Stricter height check due to subsampling size calculation below. if (height < 2) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "Invalid frame size"); + LOG(LS_ERROR) << "Invalid frame size."; return VPM_GENERAL_ERROR; } if (!VideoProcessingModule::ValidFrameStats(*stats)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "Invalid frame stats"); return VPM_GENERAL_ERROR; } @@ -152,8 +147,7 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame, // Ensure we won't get an overflow below. // In practice, the number of subsampled pixels will not become this large. if (y_sub_size > (1 << 21) - 1) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "Subsampled number of pixels too large"); + LOG(LS_ERROR) << "Subsampled number of pixels too large."; return -1; } diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/denoising.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/denoising.cc index 79c4bcc3d1b..4c8dcb439f6 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/denoising.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/denoising.cc @@ -9,7 +9,6 @@ */ #include "webrtc/modules/video_processing/main/source/denoising.h" -#include "webrtc/system_wrappers/interface/trace.h" #include <string.h> @@ -78,8 +77,6 @@ int32_t VPMDenoising::ProcessFrame(I420VideoFrame* frame) { int32_t num_pixels_changed = 0; if (frame->IsZeroSize()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, id_, - "zero size frame"); return VPM_GENERAL_ERROR; } diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.cc index de4907029bc..e1cd04ff711 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.cc @@ -9,14 +9,12 @@ */ #include "webrtc/modules/video_processing/main/source/frame_preprocessor.h" -#include "webrtc/system_wrappers/interface/trace.h" namespace webrtc { VPMFramePreprocessor::VPMFramePreprocessor() : id_(0), content_metrics_(NULL), - max_frame_rate_(0), resampled_frame_(), enable_ca_(false), frame_cnt_(0) { @@ -60,14 +58,6 @@ void VPMFramePreprocessor::SetInputFrameResampleMode( spatial_resampler_->SetInputFrameResampleMode(resampling_mode); } -int32_t VPMFramePreprocessor::SetMaxFramerate(uint32_t max_frame_rate) { - if (max_frame_rate == 0) return VPM_PARAMETER_ERROR; - - // Max allowed frame_rate. - max_frame_rate_ = max_frame_rate; - return vd_->SetMaxFramerate(max_frame_rate); -} - int32_t VPMFramePreprocessor::SetTargetResolution( uint32_t width, uint32_t height, uint32_t frame_rate) { if ( (width == 0) || (height == 0) || (frame_rate == 0)) { @@ -78,7 +68,7 @@ int32_t VPMFramePreprocessor::SetTargetResolution( if (ret_val < 0) return ret_val; - ret_val = vd_->SetTargetframe_rate(frame_rate); + ret_val = vd_->SetTargetFramerate(frame_rate); if (ret_val < 0) return ret_val; return VPM_OK; @@ -112,8 +102,6 @@ int32_t VPMFramePreprocessor::PreprocessFrame(const I420VideoFrame& frame, vd_->UpdateIncomingframe_rate(); if (vd_->DropFrame()) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, id_, - "Drop frame due to frame rate"); return 1; // drop 1 frame } diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.h b/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.h index ca62d38fc6d..64a5797b9bc 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.h +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.h @@ -39,9 +39,6 @@ class VPMFramePreprocessor { // Enable content analysis. void EnableContentAnalysis(bool enable); - // Set max frame rate. - int32_t SetMaxFramerate(uint32_t max_frame_rate); - // Set target resolution: frame rate and dimension. int32_t SetTargetResolution(uint32_t width, uint32_t height, uint32_t frame_rate); @@ -68,7 +65,6 @@ class VPMFramePreprocessor { int32_t id_; VideoContentMetrics* content_metrics_; - uint32_t max_frame_rate_; I420VideoFrame resampled_frame_; VPMSpatialResampler* spatial_resampler_; VPMContentAnalysis* ca_; diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.cc index 8fd3d036919..bf05bd71545 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.cc @@ -16,15 +16,7 @@ namespace webrtc { -VPMVideoDecimator::VPMVideoDecimator() - : overshoot_modifier_(0), - drop_count_(0), - keep_count_(0), - target_frame_rate_(30), - incoming_frame_rate_(0.0f), - max_frame_rate_(30), - incoming_frame_times_(), - enable_temporal_decimation_(true) { +VPMVideoDecimator::VPMVideoDecimator() { Reset(); } @@ -36,7 +28,6 @@ void VPMVideoDecimator::Reset() { keep_count_ = 0; target_frame_rate_ = 30; incoming_frame_rate_ = 0.0f; - max_frame_rate_ = 30; memset(incoming_frame_times_, 0, sizeof(incoming_frame_times_)); enable_temporal_decimation_ = true; } @@ -45,26 +36,10 @@ void VPMVideoDecimator::EnableTemporalDecimation(bool enable) { enable_temporal_decimation_ = enable; } -int32_t VPMVideoDecimator::SetMaxFramerate(uint32_t max_frame_rate) { - if (max_frame_rate == 0) return VPM_PARAMETER_ERROR; - - max_frame_rate_ = max_frame_rate; - - if (target_frame_rate_ > max_frame_rate_) - target_frame_rate_ = max_frame_rate_; - - return VPM_OK; -} - -int32_t VPMVideoDecimator::SetTargetframe_rate(uint32_t frame_rate) { +int32_t VPMVideoDecimator::SetTargetFramerate(uint32_t frame_rate) { if (frame_rate == 0) return VPM_PARAMETER_ERROR; - if (frame_rate > max_frame_rate_) { - // Override. - target_frame_rate_ = max_frame_rate_; - } else { - target_frame_rate_ = frame_rate; - } + target_frame_rate_ = frame_rate; return VPM_OK; } diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.h b/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.h index d17da618802..fca74aeae15 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.h +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/video_decimator.h @@ -25,8 +25,7 @@ class VPMVideoDecimator { void EnableTemporalDecimation(bool enable); - int32_t SetMaxFramerate(uint32_t max_frame_rate); - int32_t SetTargetframe_rate(uint32_t frame_rate); + int32_t SetTargetFramerate(uint32_t frame_rate); bool DropFrame(); @@ -50,7 +49,6 @@ class VPMVideoDecimator { uint32_t keep_count_; uint32_t target_frame_rate_; float incoming_frame_rate_; - uint32_t max_frame_rate_; int64_t incoming_frame_times_[kFrameCountHistory_size]; bool enable_temporal_decimation_; }; diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.cc b/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.cc index af1bfe1a412..3560030c86c 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.cc +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.cc @@ -11,7 +11,7 @@ #include "webrtc/modules/video_processing/main/source/video_processing_impl.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/system_wrappers/interface/logging.h" #include <assert.h> @@ -68,13 +68,9 @@ VideoProcessingModuleImpl::VideoProcessingModuleImpl(const int32_t id) deflickering_.ChangeUniqueId(id); denoising_.ChangeUniqueId(id); frame_pre_processor_.ChangeUniqueId(id); - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideoPreocessing, id_, - "Created"); } VideoProcessingModuleImpl::~VideoProcessingModuleImpl() { - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideoPreocessing, id_, - "Destroyed"); delete &mutex_; } @@ -89,8 +85,7 @@ void VideoProcessingModuleImpl::Reset() { int32_t VideoProcessingModule::GetFrameStats(FrameStats* stats, const I420VideoFrame& frame) { if (frame.IsZeroSize()) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, - "zero size frame"); + LOG(LS_ERROR) << "Zero size frame."; return VPM_PARAMETER_ERROR; } @@ -121,7 +116,10 @@ int32_t VideoProcessingModule::GetFrameStats(FrameStats* stats, } bool VideoProcessingModule::ValidFrameStats(const FrameStats& stats) { - if (stats.num_pixels == 0) return false; + if (stats.num_pixels == 0) { + LOG(LS_WARNING) << "Invalid frame stats."; + return false; + } return true; } @@ -173,11 +171,6 @@ void VideoProcessingModuleImpl::SetInputFrameResampleMode(VideoFrameResampling frame_pre_processor_.SetInputFrameResampleMode(resampling_mode); } -int32_t VideoProcessingModuleImpl::SetMaxFramerate(uint32_t max_frame_rate) { - CriticalSectionScoped cs(&mutex_); - return frame_pre_processor_.SetMaxFramerate(max_frame_rate); -} - int32_t VideoProcessingModuleImpl::SetTargetResolution(uint32_t width, uint32_t height, uint32_t frame_rate) { diff --git a/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.h b/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.h index 913bb648364..deae6ff657e 100644 --- a/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.h +++ b/chromium/third_party/webrtc/modules/video_processing/main/source/video_processing_impl.h @@ -51,9 +51,6 @@ class VideoProcessingModuleImpl : public VideoProcessingModule { // Enable content analysis virtual void EnableContentAnalysis(bool enable); - // Set max frame rate - virtual int32_t SetMaxFramerate(uint32_t max_frame_rate); - // Set Target Resolution: frame rate and dimension virtual int32_t SetTargetResolution(uint32_t width, uint32_t height, diff --git a/chromium/third_party/webrtc/modules/video_render/OWNERS b/chromium/third_party/webrtc/modules/video_render/OWNERS index 5e8ed090910..7dd4a3624b0 100644 --- a/chromium/third_party/webrtc/modules/video_render/OWNERS +++ b/chromium/third_party/webrtc/modules/video_render/OWNERS @@ -3,3 +3,10 @@ mflodman@webrtc.org perkj@webrtc.org wu@webrtc.org mallinath@webrtc.org + +per-file *.isolate=kjellander@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc index 2057e282f48..c2afbbd6396 100644 --- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc +++ b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc @@ -245,7 +245,6 @@ AndroidNativeOpenGl2Channel::AndroidNativeOpenGl2Channel( AndroidNativeOpenGl2Channel::~AndroidNativeOpenGl2Channel() { WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "AndroidNativeOpenGl2Channel dtor"); - delete &_renderCritSect; if (_jvm) { // get the JNI env for this thread bool isAttached = false; @@ -277,6 +276,8 @@ AndroidNativeOpenGl2Channel::~AndroidNativeOpenGl2Channel() { } } } + + delete &_renderCritSect; } int32_t AndroidNativeOpenGl2Channel::Init(int32_t zOrder, diff --git a/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm b/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm index 0707a714b9f..9dfa69d1b79 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm +++ b/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm @@ -8,6 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + // This files is mostly copied from // webrtc/modules/video_render/android/video_render_opengles20.h diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm index 2a3ca180f72..02814b22234 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm +++ b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm @@ -8,6 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #include "webrtc/modules/video_render/ios/video_render_ios_channel.h" using namespace webrtc; diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm index 7bc46bfdad5..bbada099edf 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm +++ b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm @@ -8,6 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #include "webrtc/modules/video_render/ios/video_render_ios_gles20.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h" @@ -246,7 +250,7 @@ int VideoRenderIosGles20::GetWindowRect(Rect& rect) { int VideoRenderIosGles20::ChangeWindow(void* new_window) { CriticalSectionScoped cs(gles_crit_sec_.get()); - view_ = (VideoRenderIosView*)new_window; + view_ = (__bridge VideoRenderIosView*)new_window; return 0; } diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h index 138a524a0db..e38ed7ae4b1 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h +++ b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h @@ -99,7 +99,7 @@ class VideoRenderIosImpl : IVideoRender { bool full_screen_; CriticalSectionWrapper* crit_sec_; - VideoRenderIosGles20* ptr_ios_render_; + webrtc::scoped_ptr<VideoRenderIosGles20> ptr_ios_render_; }; } // namespace webrtc #endif // WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_IMPL_H_ diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm index 089b1e76988..9b8e1d62438 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm +++ b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm @@ -8,6 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #include "webrtc/modules/video_render/ios/video_render_ios_impl.h" #include "webrtc/modules/video_render/ios/video_render_ios_gles20.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" @@ -33,18 +37,13 @@ VideoRenderIosImpl::VideoRenderIosImpl(const int32_t id, VideoRenderIosImpl::~VideoRenderIosImpl() { delete crit_sec_; - - if (ptr_ios_render_) { - delete ptr_ios_render_; - ptr_ios_render_ = NULL; - } } int32_t VideoRenderIosImpl::Init() { CriticalSectionScoped cs(crit_sec_); - ptr_ios_render_ = new VideoRenderIosGles20( - (VideoRenderIosView*)ptr_window_, full_screen_, id_); + ptr_ios_render_.reset(new VideoRenderIosGles20( + (__bridge VideoRenderIosView*)ptr_window_, full_screen_, id_)); return ptr_ios_render_->Init(); ; diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h index db053560607..915c0f71e20 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h +++ b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h @@ -16,15 +16,7 @@ #include "webrtc/modules/video_render/ios/open_gles20.h" -@interface VideoRenderIosView : UIView { - @private // NOLINT - EAGLContext* context_; - webrtc::OpenGles20* gles_renderer20_; - int _frameBufferWidth; - int _frameBufferHeight; - unsigned int _defaultFrameBuffer; - unsigned int _colorRenderBuffer; -} +@interface VideoRenderIosView : UIView - (BOOL)createContext; - (BOOL)presentFramebuffer; diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm index 662c9676a61..2e00e097eec 100644 --- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm +++ b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm @@ -8,12 +8,23 @@ * be found in the AUTHORS file in the root of the source tree. */ +#if !defined(__has_feature) || !__has_feature(objc_arc) +#error "This file requires ARC support." +#endif + #include "webrtc/modules/video_render/ios/video_render_ios_view.h" #include "webrtc/system_wrappers/interface/trace.h" using namespace webrtc; -@implementation VideoRenderIosView +@implementation VideoRenderIosView { + EAGLContext* _context; + webrtc::scoped_ptr<webrtc::OpenGles20> _gles_renderer20; + int _frameBufferWidth; + int _frameBufferHeight; + unsigned int _defaultFrameBuffer; + unsigned int _colorRenderBuffer; +} @synthesize context = context_; @@ -25,7 +36,7 @@ using namespace webrtc; // init super class self = [super initWithCoder:coder]; if (self) { - gles_renderer20_ = new OpenGles20(); + _gles_renderer20.reset(new OpenGles20()); } return self; } @@ -34,7 +45,7 @@ using namespace webrtc; // init super class self = [super init]; if (self) { - gles_renderer20_ = new OpenGles20(); + _gles_renderer20.reset(new OpenGles20()); } return self; } @@ -43,7 +54,7 @@ using namespace webrtc; // init super class self = [super initWithFrame:frame]; if (self) { - gles_renderer20_ = new OpenGles20(); + _gles_renderer20.reset(new OpenGles20()); } return self; } @@ -59,13 +70,7 @@ using namespace webrtc; _colorRenderBuffer = 0; } - context_ = nil; - - if (gles_renderer20_) { - delete gles_renderer20_; - } - - [super dealloc]; + [EAGLContext setCurrentContext:nil]; } - (NSString*)description { @@ -84,14 +89,13 @@ using namespace webrtc; kEAGLColorFormatRGBA8, kEAGLDrawablePropertyColorFormat, nil]; - context_ = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2]; + _context = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2]; - if (!context_) { + if (!_context) { return NO; } - // set current EAGLContext to self context_ - if (![EAGLContext setCurrentContext:context_]) { + if (![EAGLContext setCurrentContext:_context]) { return NO; } @@ -102,7 +106,7 @@ using namespace webrtc; // Create color render buffer and allocate backing store. glGenRenderbuffers(1, &_colorRenderBuffer); glBindRenderbuffer(GL_RENDERBUFFER, _colorRenderBuffer); - [context_ renderbufferStorage:GL_RENDERBUFFER + [_context renderbufferStorage:GL_RENDERBUFFER fromDrawable:(CAEAGLLayer*)self.layer]; glGetRenderbufferParameteriv( GL_RENDERBUFFER, GL_RENDERBUFFER_WIDTH, &_frameBufferWidth); @@ -121,12 +125,12 @@ using namespace webrtc; glBindFramebuffer(GL_FRAMEBUFFER, _defaultFrameBuffer); glViewport(0, 0, self.frame.size.width, self.frame.size.height); - return gles_renderer20_->Setup([self bounds].size.width, + return _gles_renderer20->Setup([self bounds].size.width, [self bounds].size.height); } - (BOOL)presentFramebuffer { - if (![context_ presentRenderbuffer:GL_RENDERBUFFER]) { + if (![_context presentRenderbuffer:GL_RENDERBUFFER]) { WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, 0, @@ -135,21 +139,15 @@ using namespace webrtc; __FUNCTION__, __LINE__); } - - // update UI stuff on the main thread - [self performSelectorOnMainThread:@selector(setNeedsDisplay) - withObject:nil - waitUntilDone:NO]; - return YES; } - (BOOL)renderFrame:(I420VideoFrame*)frameToRender { - if (![EAGLContext setCurrentContext:context_]) { + if (![EAGLContext setCurrentContext:_context]) { return NO; } - return gles_renderer20_->Render(*frameToRender); + return _gles_renderer20->Render(*frameToRender); } - (BOOL)setCoordinatesForZOrder:(const float)zOrder @@ -157,7 +155,7 @@ using namespace webrtc; Top:(const float)top Right:(const float)right Bottom:(const float)bottom { - return gles_renderer20_->SetCoordinates(zOrder, left, top, right, bottom); + return _gles_renderer20->SetCoordinates(zOrder, left, top, right, bottom); } @end diff --git a/chromium/third_party/webrtc/modules/video_render/video_render.gypi b/chromium/third_party/webrtc/modules/video_render/video_render.gypi index 4f3844406e0..71d969baba2 100644 --- a/chromium/third_party/webrtc/modules/video_render/video_render.gypi +++ b/chromium/third_party/webrtc/modules/video_render/video_render.gypi @@ -87,6 +87,12 @@ 'android/video_render_android_surface_view.cc', 'android/video_render_opengles20.cc', ], + }, { + 'link_settings': { + 'libraries': [ + '-lGLESv2', + ], + }, }], ['OS!="ios" or include_internal_video_render==0', { 'sources!': [ @@ -112,6 +118,12 @@ 'linux/video_x11_channel.cc', 'linux/video_x11_render.cc', ], + }, { + 'link_settings': { + 'libraries': [ + '-lXext', + ], + }, }], ['OS!="mac" or include_internal_video_render==0', { 'sources!': [ @@ -130,6 +142,9 @@ ], }], ['OS=="ios"', { + 'xcode_settings': { + 'CLANG_ENABLE_OBJC_ARC': 'YES', + }, 'all_dependent_settings': { 'xcode_settings': { 'OTHER_LDFLAGS': [ @@ -205,11 +220,11 @@ ], }], ['OS=="linux"', { - 'libraries': [ - '-lrt', - '-lXext', - '-lX11', - ], + 'link_settings': { + 'libraries': [ + '-lX11', + ], + }, }], ['OS=="mac"', { 'xcode_settings': { diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_frames.cc b/chromium/third_party/webrtc/modules/video_render/video_render_frames.cc index be5cac9aaa6..d790877e31f 100644 --- a/chromium/third_party/webrtc/modules/video_render/video_render_frames.cc +++ b/chromium/third_party/webrtc/modules/video_render/video_render_frames.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/video_render//video_render_frames.h" +#include "webrtc/modules/video_render/video_render_frames.h" #include <assert.h> @@ -19,13 +19,12 @@ namespace webrtc { -const int32_t KEventMaxWaitTimeMs = 200; +const uint32_t KEventMaxWaitTimeMs = 200; const uint32_t kMinRenderDelayMs = 10; const uint32_t kMaxRenderDelayMs= 500; VideoRenderFrames::VideoRenderFrames() - : incoming_frames_(), - render_delay_ms_(10) { + : render_delay_ms_(10) { } VideoRenderFrames::~VideoRenderFrames() { @@ -35,12 +34,19 @@ VideoRenderFrames::~VideoRenderFrames() { int32_t VideoRenderFrames::AddFrame(I420VideoFrame* new_frame) { const int64_t time_now = TickTime::MillisecondTimestamp(); - if (new_frame->render_time_ms() + KOldRenderTimestampMS < time_now) { - WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1, + // Drop old frames only when there are other frames in the queue, otherwise, a + // really slow system never renders any frames. + if (!incoming_frames_.empty() && + new_frame->render_time_ms() + KOldRenderTimestampMS < time_now) { + WEBRTC_TRACE(kTraceWarning, + kTraceVideoRenderer, + -1, "%s: too old frame, timestamp=%u.", - __FUNCTION__, new_frame->timestamp()); + __FUNCTION__, + new_frame->timestamp()); return -1; } + if (new_frame->render_time_ms() > time_now + KFutureRenderTimestampMS) { WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1, "%s: frame too long into the future, timestamp=%u.", @@ -49,26 +55,18 @@ int32_t VideoRenderFrames::AddFrame(I420VideoFrame* new_frame) { } if (new_frame->native_handle() != NULL) { - incoming_frames_.PushBack(new TextureVideoFrame( - static_cast<NativeHandle*>(new_frame->native_handle()), - new_frame->width(), - new_frame->height(), - new_frame->timestamp(), - new_frame->render_time_ms())); - return incoming_frames_.GetSize(); + incoming_frames_.push_back(new_frame->CloneFrame()); + return static_cast<int32_t>(incoming_frames_.size()); } // Get an empty frame I420VideoFrame* frame_to_add = NULL; - if (!empty_frames_.Empty()) { - ListItem* item = empty_frames_.First(); - if (item) { - frame_to_add = static_cast<I420VideoFrame*>(item->GetItem()); - empty_frames_.Erase(item); - } + if (!empty_frames_.empty()) { + frame_to_add = empty_frames_.front(); + empty_frames_.pop_front(); } if (!frame_to_add) { - if (empty_frames_.GetSize() + incoming_frames_.GetSize() > + if (empty_frames_.size() + incoming_frames_.size() > KMaxNumberOfFrames) { // Already allocated too many frames. WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, @@ -80,7 +78,7 @@ int32_t VideoRenderFrames::AddFrame(I420VideoFrame* new_frame) { // Allocate new memory. WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, -1, "%s: allocating buffer %d", __FUNCTION__, - empty_frames_.GetSize() + incoming_frames_.GetSize()); + empty_frames_.size() + incoming_frames_.size()); frame_to_add = new I420VideoFrame(); if (!frame_to_add) { @@ -97,33 +95,28 @@ int32_t VideoRenderFrames::AddFrame(I420VideoFrame* new_frame) { // TODO(mflodman) Change this! // Remove const ness. Copying will be costly. frame_to_add->SwapFrame(new_frame); - incoming_frames_.PushBack(frame_to_add); + incoming_frames_.push_back(frame_to_add); - return incoming_frames_.GetSize(); + return static_cast<int32_t>(incoming_frames_.size()); } I420VideoFrame* VideoRenderFrames::FrameToRender() { I420VideoFrame* render_frame = NULL; - while (!incoming_frames_.Empty()) { - ListItem* item = incoming_frames_.First(); - if (item) { - I420VideoFrame* oldest_frame_in_list = - static_cast<I420VideoFrame*>(item->GetItem()); - if (oldest_frame_in_list->render_time_ms() <= - TickTime::MillisecondTimestamp() + render_delay_ms_) { - // This is the oldest one so far and it's OK to render. - if (render_frame) { - // This one is older than the newly found frame, remove this one. - ReturnFrame(render_frame); - } - render_frame = oldest_frame_in_list; - incoming_frames_.Erase(item); - } else { - // We can't release this one yet, we're done here. - break; + FrameList::iterator iter = incoming_frames_.begin(); + while(iter != incoming_frames_.end()) { + I420VideoFrame* oldest_frame_in_list = *iter; + if (oldest_frame_in_list->render_time_ms() <= + TickTime::MillisecondTimestamp() + render_delay_ms_) { + // This is the oldest one so far and it's OK to render. + if (render_frame) { + // This one is older than the newly found frame, remove this one. + ReturnFrame(render_frame); } + render_frame = oldest_frame_in_list; + iter = incoming_frames_.erase(iter); } else { - assert(false); + // We can't release this one yet, we're done here. + break; } } return render_frame; @@ -135,7 +128,7 @@ int32_t VideoRenderFrames::ReturnFrame(I420VideoFrame* old_frame) { old_frame->ResetSize(); old_frame->set_timestamp(0); old_frame->set_render_time_ms(0); - empty_frames_.PushBack(old_frame); + empty_frames_.push_back(old_frame); } else { delete old_frame; } @@ -143,40 +136,29 @@ int32_t VideoRenderFrames::ReturnFrame(I420VideoFrame* old_frame) { } int32_t VideoRenderFrames::ReleaseAllFrames() { - while (!incoming_frames_.Empty()) { - ListItem* item = incoming_frames_.First(); - if (item) { - I420VideoFrame* frame = static_cast<I420VideoFrame*>(item->GetItem()); - assert(frame != NULL); - delete frame; - } - incoming_frames_.Erase(item); + for (FrameList::iterator iter = incoming_frames_.begin(); + iter != incoming_frames_.end(); ++iter) { + delete *iter; } - while (!empty_frames_.Empty()) { - ListItem* item = empty_frames_.First(); - if (item) { - I420VideoFrame* frame = static_cast<I420VideoFrame*>(item->GetItem()); - assert(frame != NULL); - delete frame; - } - empty_frames_.Erase(item); + incoming_frames_.clear(); + + for (FrameList::iterator iter = empty_frames_.begin(); + iter != empty_frames_.end(); ++iter) { + delete *iter; } + empty_frames_.clear(); return 0; } uint32_t VideoRenderFrames::TimeToNextFrameRelease() { - int64_t time_to_release = 0; - ListItem* item = incoming_frames_.First(); - if (item) { - I420VideoFrame* oldest_frame = - static_cast<I420VideoFrame*>(item->GetItem()); - time_to_release = oldest_frame->render_time_ms() - render_delay_ms_ - - TickTime::MillisecondTimestamp(); - if (time_to_release < 0) { - time_to_release = 0; - } - } else { - time_to_release = KEventMaxWaitTimeMs; + if (incoming_frames_.empty()) { + return KEventMaxWaitTimeMs; + } + I420VideoFrame* oldest_frame = incoming_frames_.front(); + int64_t time_to_release = oldest_frame->render_time_ms() - render_delay_ms_ + - TickTime::MillisecondTimestamp(); + if (time_to_release < 0) { + time_to_release = 0; } return static_cast<uint32_t>(time_to_release); } diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_frames.h b/chromium/third_party/webrtc/modules/video_render/video_render_frames.h index 1a5d64d8072..d2e887f8272 100644 --- a/chromium/third_party/webrtc/modules/video_render/video_render_frames.h +++ b/chromium/third_party/webrtc/modules/video_render/video_render_frames.h @@ -11,8 +11,9 @@ #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_ // NOLINT #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_ // NOLINT +#include <list> + #include "webrtc/modules/video_render/include/video_render.h" -#include "webrtc/system_wrappers/interface/list_wrapper.h" namespace webrtc { @@ -41,6 +42,8 @@ class VideoRenderFrames { int32_t SetRenderDelay(const uint32_t render_delay); private: + typedef std::list<I420VideoFrame*> FrameList; + // 10 seconds for 30 fps. enum { KMaxNumberOfFrames = 300 }; // Don't render frames with timestamp older than 500ms from now. @@ -49,9 +52,9 @@ class VideoRenderFrames { enum { KFutureRenderTimestampMS = 10000 }; // Sorted list with framed to be rendered, oldest first. - ListWrapper incoming_frames_; + FrameList incoming_frames_; // Empty frames. - ListWrapper empty_frames_; + FrameList empty_frames_; // Estimated delay from a frame is released until it's rendered. uint32_t render_delay_ms_; diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_tests.isolate b/chromium/third_party/webrtc/modules/video_render/video_render_tests.isolate index 397ec04e8c7..15c80141dcb 100644 --- a/chromium/third_party/webrtc/modules/video_render/video_render_tests.isolate +++ b/chromium/third_party/webrtc/modules/video_render/video_render_tests.isolate @@ -8,27 +8,25 @@ { 'conditions': [ ['OS=="android"', { - # When doing Android builds, the WebRTC code is put in third_party/webrtc - # of a Chromium checkout, this is one level above the standalone build. 'variables': { 'isolate_dependency_untracked': [ - '../../../../data/', - '../../../../resources/', + '<(DEPTH)/data/', + '<(DEPTH)/resources/', ], }, }], ['OS=="linux" or OS=="mac" or OS=="win"', { 'variables': { 'command': [ - '../../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/video_render_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_tracked': [ - '../../../testing/test_env.py', + '<(DEPTH)/testing/test_env.py', '<(PRODUCT_DIR)/video_render_tests<(EXECUTABLE_SUFFIX)', ], 'isolate_dependency_untracked': [ - '../../../tools/swarming_client/', + '<(DEPTH)/tools/swarming_client/', ], }, }], |