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-rw-r--r--chromium/third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc319
1 files changed, 0 insertions, 319 deletions
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
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