diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc | 516 |
1 files changed, 516 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc new file mode 100644 index 00000000000..6c7269a35fe --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc @@ -0,0 +1,516 @@ +/* + * 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/audio_decoder_impl.h" + +#include <assert.h> +#include <string.h> // memmove + +#ifdef WEBRTC_CODEC_CELT +#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h" +#endif +#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" +#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" +#ifdef WEBRTC_CODEC_G722 +#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" +#endif +#ifdef WEBRTC_CODEC_ILBC +#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" +#endif +#ifdef WEBRTC_CODEC_ISACFX +#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" +#endif +#ifdef WEBRTC_CODEC_ISAC +#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" +#endif +#ifdef WEBRTC_CODEC_OPUS +#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" +#endif +#ifdef WEBRTC_CODEC_PCM16 +#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" +#endif + +namespace webrtc { + +// PCMu +int AudioDecoderPcmU::Decode(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 = WebRtcG711_DecodeU( + state_, reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)), + static_cast<int16_t>(encoded_len), decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded, + size_t encoded_len) { + // One encoded byte per sample per channel. + return static_cast<int>(encoded_len / channels_); +} + +// PCMa +int AudioDecoderPcmA::Decode(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 = WebRtcG711_DecodeA( + state_, reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)), + static_cast<int16_t>(encoded_len), decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded, + size_t encoded_len) { + // One encoded byte per sample per channel. + return static_cast<int>(encoded_len / channels_); +} + +// PCM16B +#ifdef WEBRTC_CODEC_PCM16 +AudioDecoderPcm16B::AudioDecoderPcm16B(enum NetEqDecoder type) + : AudioDecoder(type) { + assert(type == kDecoderPCM16B || + type == kDecoderPCM16Bwb || + type == kDecoderPCM16Bswb32kHz || + type == kDecoderPCM16Bswb48kHz); +} + +int AudioDecoderPcm16B::Decode(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 = WebRtcPcm16b_DecodeW16( + state_, reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)), + static_cast<int16_t>(encoded_len), decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded, + size_t encoded_len) { + // Two encoded byte per sample per channel. + return static_cast<int>(encoded_len / (2 * channels_)); +} + +AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh( + enum NetEqDecoder type) + : AudioDecoderPcm16B(kDecoderPCM16B) { // This will be changed below. + codec_type_ = type; // Changing to actual type here. + switch (codec_type_) { + case kDecoderPCM16B_2ch: + case kDecoderPCM16Bwb_2ch: + case kDecoderPCM16Bswb32kHz_2ch: + case kDecoderPCM16Bswb48kHz_2ch: + channels_ = 2; + break; + case kDecoderPCM16B_5ch: + channels_ = 5; + break; + default: + assert(false); + } +} +#endif + +// iLBC +#ifdef WEBRTC_CODEC_ILBC +AudioDecoderIlbc::AudioDecoderIlbc() : AudioDecoder(kDecoderILBC) { + WebRtcIlbcfix_DecoderCreate(reinterpret_cast<iLBC_decinst_t**>(&state_)); +} + +AudioDecoderIlbc::~AudioDecoderIlbc() { + WebRtcIlbcfix_DecoderFree(static_cast<iLBC_decinst_t*>(state_)); +} + +int AudioDecoderIlbc::Decode(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 = WebRtcIlbcfix_Decode(static_cast<iLBC_decinst_t*>(state_), + reinterpret_cast<const int16_t*>(encoded), + static_cast<int16_t>(encoded_len), decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) { + return WebRtcIlbcfix_NetEqPlc(static_cast<iLBC_decinst_t*>(state_), + decoded, num_frames); +} + +int AudioDecoderIlbc::Init() { + return WebRtcIlbcfix_Decoderinit30Ms(static_cast<iLBC_decinst_t*>(state_)); +} +#endif + +// iSAC float +#ifdef WEBRTC_CODEC_ISAC +AudioDecoderIsac::AudioDecoderIsac() : AudioDecoder(kDecoderISAC) { + WebRtcIsac_Create(reinterpret_cast<ISACStruct**>(&state_)); + WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_), 16000); +} + +AudioDecoderIsac::~AudioDecoderIsac() { + WebRtcIsac_Free(static_cast<ISACStruct*>(state_)); +} + +int AudioDecoderIsac::Decode(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 = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderIsac::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 = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) { + return WebRtcIsac_DecodePlc(static_cast<ISACStruct*>(state_), + decoded, num_frames); +} + +int AudioDecoderIsac::Init() { + return WebRtcIsac_DecoderInit(static_cast<ISACStruct*>(state_)); +} + +int AudioDecoderIsac::IncomingPacket(const uint8_t* payload, + size_t payload_len, + uint16_t rtp_sequence_number, + uint32_t rtp_timestamp, + uint32_t arrival_timestamp) { + return WebRtcIsac_UpdateBwEstimate(static_cast<ISACStruct*>(state_), + reinterpret_cast<const uint16_t*>(payload), + static_cast<int32_t>(payload_len), + rtp_sequence_number, + rtp_timestamp, + arrival_timestamp); +} + +int AudioDecoderIsac::ErrorCode() { + return WebRtcIsac_GetErrorCode(static_cast<ISACStruct*>(state_)); +} + +// iSAC SWB +AudioDecoderIsacSwb::AudioDecoderIsacSwb() : AudioDecoderIsac() { + codec_type_ = kDecoderISACswb; + WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_), 32000); +} + +// iSAC FB +AudioDecoderIsacFb::AudioDecoderIsacFb() : AudioDecoderIsacSwb() { + codec_type_ = kDecoderISACfb; +} +#endif + +// iSAC fix +#ifdef WEBRTC_CODEC_ISACFX +AudioDecoderIsacFix::AudioDecoderIsacFix() : AudioDecoder(kDecoderISAC) { + WebRtcIsacfix_Create(reinterpret_cast<ISACFIX_MainStruct**>(&state_)); +} + +AudioDecoderIsacFix::~AudioDecoderIsacFix() { + WebRtcIsacfix_Free(static_cast<ISACFIX_MainStruct*>(state_)); +} + +int AudioDecoderIsacFix::Decode(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 = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderIsacFix::Init() { + return WebRtcIsacfix_DecoderInit(static_cast<ISACFIX_MainStruct*>(state_)); +} + +int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload, + size_t payload_len, + uint16_t rtp_sequence_number, + uint32_t rtp_timestamp, + uint32_t arrival_timestamp) { + return WebRtcIsacfix_UpdateBwEstimate( + static_cast<ISACFIX_MainStruct*>(state_), + reinterpret_cast<const uint16_t*>(payload), + static_cast<int32_t>(payload_len), + rtp_sequence_number, rtp_timestamp, arrival_timestamp); +} + +int AudioDecoderIsacFix::ErrorCode() { + return WebRtcIsacfix_GetErrorCode(static_cast<ISACFIX_MainStruct*>(state_)); +} +#endif + +// G.722 +#ifdef WEBRTC_CODEC_G722 +AudioDecoderG722::AudioDecoderG722() : AudioDecoder(kDecoderG722) { + WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_)); +} + +AudioDecoderG722::~AudioDecoderG722() { + WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_)); +} + +int AudioDecoderG722::Decode(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 = WebRtcG722_Decode( + static_cast<G722DecInst*>(state_), + const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)), + static_cast<int16_t>(encoded_len), decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderG722::Init() { + return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_)); +} + +int AudioDecoderG722::PacketDuration(const uint8_t* encoded, + size_t encoded_len) { + // 1/2 encoded byte per sample per channel. + return static_cast<int>(2 * encoded_len / channels_); +} + +AudioDecoderG722Stereo::AudioDecoderG722Stereo() + : AudioDecoderG722(), + state_left_(state_), // Base member |state_| is used for left channel. + state_right_(NULL) { + channels_ = 2; + // |state_left_| already created by the base class AudioDecoderG722. + WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_right_)); +} + +AudioDecoderG722Stereo::~AudioDecoderG722Stereo() { + // |state_left_| will be freed by the base class AudioDecoderG722. + WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_right_)); +} + +int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len, + int16_t* decoded, SpeechType* speech_type) { + int16_t temp_type = 1; // Default is speech. + // De-interleave the bit-stream into two separate payloads. + uint8_t* encoded_deinterleaved = new uint8_t[encoded_len]; + SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved); + // Decode left and right. + int16_t ret = WebRtcG722_Decode( + static_cast<G722DecInst*>(state_left_), + reinterpret_cast<int16_t*>(encoded_deinterleaved), + static_cast<int16_t>(encoded_len / 2), decoded, &temp_type); + if (ret >= 0) { + int decoded_len = ret; + ret = WebRtcG722_Decode( + static_cast<G722DecInst*>(state_right_), + reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]), + static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type); + if (ret == decoded_len) { + decoded_len += ret; + // Interleave output. + for (int k = decoded_len / 2; k < decoded_len; k++) { + int16_t temp = decoded[k]; + memmove(&decoded[2 * k - decoded_len + 2], + &decoded[2 * k - decoded_len + 1], + (decoded_len - k - 1) * sizeof(int16_t)); + decoded[2 * k - decoded_len + 1] = temp; + } + ret = decoded_len; // Return total number of samples. + } + } + *speech_type = ConvertSpeechType(temp_type); + delete [] encoded_deinterleaved; + return ret; +} + +int AudioDecoderG722Stereo::Init() { + int ret = WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_right_)); + if (ret != 0) { + return ret; + } + return AudioDecoderG722::Init(); +} + +// Split the stereo packet and place left and right channel after each other +// in the output array. +void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded, + size_t encoded_len, + uint8_t* encoded_deinterleaved) { + assert(encoded); + // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ..., + // where "lx" is 4 bits representing left sample number x, and "rx" right + // sample. Two samples fit in one byte, represented with |...|. + for (size_t i = 0; i + 1 < encoded_len; i += 2) { + uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F); + encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4); + encoded_deinterleaved[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 (size_t i = 0; i < encoded_len / 2; i++) { + uint8_t right_byte = encoded_deinterleaved[i + 1]; + memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2], + encoded_len - i - 2); + encoded_deinterleaved[encoded_len - 1] = right_byte; + } +} +#endif + +// CELT +#ifdef WEBRTC_CODEC_CELT +AudioDecoderCelt::AudioDecoderCelt(enum NetEqDecoder type) + : AudioDecoder(type) { + assert(type == kDecoderCELT_32 || type == kDecoderCELT_32_2ch); + if (type == kDecoderCELT_32) { + channels_ = 1; + } else { + channels_ = 2; + } + WebRtcCelt_CreateDec(reinterpret_cast<CELT_decinst_t**>(&state_), + static_cast<int>(channels_)); +} + +AudioDecoderCelt::~AudioDecoderCelt() { + WebRtcCelt_FreeDec(static_cast<CELT_decinst_t*>(state_)); +} + +int AudioDecoderCelt::Decode(const uint8_t* encoded, size_t encoded_len, + int16_t* decoded, SpeechType* speech_type) { + int16_t temp_type = 1; // Default to speech. + int ret = WebRtcCelt_DecodeUniversal(static_cast<CELT_decinst_t*>(state_), + encoded, static_cast<int>(encoded_len), + decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + if (ret < 0) { + return -1; + } + // Return the total number of samples. + return ret * static_cast<int>(channels_); +} + +int AudioDecoderCelt::Init() { + return WebRtcCelt_DecoderInit(static_cast<CELT_decinst_t*>(state_)); +} + +bool AudioDecoderCelt::HasDecodePlc() const { return true; } + +int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) { + int ret = WebRtcCelt_DecodePlc(static_cast<CELT_decinst_t*>(state_), + decoded, num_frames); + if (ret < 0) { + return -1; + } + // Return the total number of samples. + return ret * static_cast<int>(channels_); +} +#endif + +// Opus +#ifdef WEBRTC_CODEC_OPUS +AudioDecoderOpus::AudioDecoderOpus(enum NetEqDecoder type) + : AudioDecoder(type) { + if (type == kDecoderOpus_2ch) { + channels_ = 2; + } else { + channels_ = 1; + } + WebRtcOpus_DecoderCreate(reinterpret_cast<OpusDecInst**>(&state_), + static_cast<int>(channels_)); +} + +AudioDecoderOpus::~AudioDecoderOpus() { + WebRtcOpus_DecoderFree(static_cast<OpusDecInst*>(state_)); +} + +int AudioDecoderOpus::Decode(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_DecodeNew(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::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_)); +} + +int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, + size_t encoded_len) { + 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) + : AudioDecoder(type) { + assert(type == kDecoderCNGnb || type == kDecoderCNGwb || + kDecoderCNGswb32kHz || type == kDecoderCNGswb48kHz); + WebRtcCng_CreateDec(reinterpret_cast<CNG_dec_inst**>(&state_)); + assert(state_); +} + +AudioDecoderCng::~AudioDecoderCng() { + if (state_) { + WebRtcCng_FreeDec(static_cast<CNG_dec_inst*>(state_)); + } +} + +int AudioDecoderCng::Init() { + assert(state_); + return WebRtcCng_InitDec(static_cast<CNG_dec_inst*>(state_)); +} + +} // namespace webrtc |