| Index: webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
|
| diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
|
| deleted file mode 100644
|
| index 6c2893336a8290388800d8e4b24247c920f358e8..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
|
| +++ /dev/null
|
| @@ -1,540 +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.
|
| - */
|
| -
|
| -#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
|
| -
|
| -#include <stdlib.h> // malloc
|
| -
|
| -#include <algorithm> // sort
|
| -#include <vector>
|
| -
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/base/format_macros.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
| -#include "webrtc/common_types.h"
|
| -#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
|
| -#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/neteq/include/neteq.h"
|
| -#include "webrtc/system_wrappers/include/clock.h"
|
| -#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
|
| -#include "webrtc/system_wrappers/include/tick_util.h"
|
| -#include "webrtc/system_wrappers/include/trace.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -namespace acm2 {
|
| -
|
| -namespace {
|
| -
|
| -// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
|
| -// before the call to this function.
|
| -void SetAudioFrameActivityAndType(bool vad_enabled,
|
| - NetEqOutputType type,
|
| - AudioFrame* audio_frame) {
|
| - if (vad_enabled) {
|
| - switch (type) {
|
| - case kOutputNormal: {
|
| - audio_frame->vad_activity_ = AudioFrame::kVadActive;
|
| - audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
|
| - break;
|
| - }
|
| - case kOutputVADPassive: {
|
| - audio_frame->vad_activity_ = AudioFrame::kVadPassive;
|
| - audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
|
| - break;
|
| - }
|
| - case kOutputCNG: {
|
| - audio_frame->vad_activity_ = AudioFrame::kVadPassive;
|
| - audio_frame->speech_type_ = AudioFrame::kCNG;
|
| - break;
|
| - }
|
| - case kOutputPLC: {
|
| - // Don't change |audio_frame->vad_activity_|, it should be the same as
|
| - // |previous_audio_activity_|.
|
| - audio_frame->speech_type_ = AudioFrame::kPLC;
|
| - break;
|
| - }
|
| - case kOutputPLCtoCNG: {
|
| - audio_frame->vad_activity_ = AudioFrame::kVadPassive;
|
| - audio_frame->speech_type_ = AudioFrame::kPLCCNG;
|
| - break;
|
| - }
|
| - default:
|
| - assert(false);
|
| - }
|
| - } else {
|
| - // Always return kVadUnknown when receive VAD is inactive
|
| - audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
|
| - switch (type) {
|
| - case kOutputNormal: {
|
| - audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
|
| - break;
|
| - }
|
| - case kOutputCNG: {
|
| - audio_frame->speech_type_ = AudioFrame::kCNG;
|
| - break;
|
| - }
|
| - case kOutputPLC: {
|
| - audio_frame->speech_type_ = AudioFrame::kPLC;
|
| - break;
|
| - }
|
| - case kOutputPLCtoCNG: {
|
| - audio_frame->speech_type_ = AudioFrame::kPLCCNG;
|
| - break;
|
| - }
|
| - case kOutputVADPassive: {
|
| - // Normally, we should no get any VAD decision if post-decoding VAD is
|
| - // not active. However, if post-decoding VAD has been active then
|
| - // disabled, we might be here for couple of frames.
|
| - audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
|
| - LOG(WARNING) << "Post-decoding VAD is disabled but output is "
|
| - << "labeled VAD-passive";
|
| - break;
|
| - }
|
| - default:
|
| - assert(false);
|
| - }
|
| - }
|
| -}
|
| -
|
| -// Is the given codec a CNG codec?
|
| -// TODO(kwiberg): Move to RentACodec.
|
| -bool IsCng(int codec_id) {
|
| - auto i = RentACodec::CodecIdFromIndex(codec_id);
|
| - return (i && (*i == RentACodec::CodecId::kCNNB ||
|
| - *i == RentACodec::CodecId::kCNWB ||
|
| - *i == RentACodec::CodecId::kCNSWB ||
|
| - *i == RentACodec::CodecId::kCNFB));
|
| -}
|
| -
|
| -} // namespace
|
| -
|
| -AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
|
| - : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
|
| - id_(config.id),
|
| - last_audio_decoder_(nullptr),
|
| - previous_audio_activity_(AudioFrame::kVadPassive),
|
| - audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
|
| - last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
|
| - neteq_(NetEq::Create(config.neteq_config)),
|
| - vad_enabled_(config.neteq_config.enable_post_decode_vad),
|
| - clock_(config.clock),
|
| - resampled_last_output_frame_(true) {
|
| - assert(clock_);
|
| - memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
|
| - memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
|
| -}
|
| -
|
| -AcmReceiver::~AcmReceiver() {
|
| - delete neteq_;
|
| -}
|
| -
|
| -int AcmReceiver::SetMinimumDelay(int delay_ms) {
|
| - if (neteq_->SetMinimumDelay(delay_ms))
|
| - return 0;
|
| - LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
|
| - return -1;
|
| -}
|
| -
|
| -int AcmReceiver::SetMaximumDelay(int delay_ms) {
|
| - if (neteq_->SetMaximumDelay(delay_ms))
|
| - return 0;
|
| - LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
|
| - return -1;
|
| -}
|
| -
|
| -int AcmReceiver::LeastRequiredDelayMs() const {
|
| - return neteq_->LeastRequiredDelayMs();
|
| -}
|
| -
|
| -rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - return last_packet_sample_rate_hz_;
|
| -}
|
| -
|
| -int AcmReceiver::last_output_sample_rate_hz() const {
|
| - return neteq_->last_output_sample_rate_hz();
|
| -}
|
| -
|
| -int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
|
| - rtc::ArrayView<const uint8_t> incoming_payload) {
|
| - uint32_t receive_timestamp = 0;
|
| - const RTPHeader* header = &rtp_header.header; // Just a shorthand.
|
| -
|
| - {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| -
|
| - const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload[0]);
|
| - if (!decoder) {
|
| - LOG_F(LS_ERROR) << "Payload-type "
|
| - << static_cast<int>(header->payloadType)
|
| - << " is not registered.";
|
| - return -1;
|
| - }
|
| - const int sample_rate_hz = [&decoder] {
|
| - const auto ci = RentACodec::CodecIdFromIndex(decoder->acm_codec_id);
|
| - return ci ? RentACodec::CodecInstById(*ci)->plfreq : -1;
|
| - }();
|
| - receive_timestamp = NowInTimestamp(sample_rate_hz);
|
| -
|
| - // If this is a CNG while the audio codec is not mono, skip pushing in
|
| - // packets into NetEq.
|
| - if (IsCng(decoder->acm_codec_id) && last_audio_decoder_ &&
|
| - last_audio_decoder_->channels > 1)
|
| - return 0;
|
| - if (!IsCng(decoder->acm_codec_id) &&
|
| - decoder->acm_codec_id !=
|
| - *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
|
| - last_audio_decoder_ = decoder;
|
| - last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
|
| - }
|
| -
|
| - } // |crit_sect_| is released.
|
| -
|
| - if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
|
| - 0) {
|
| - LOG(LERROR) << "AcmReceiver::InsertPacket "
|
| - << static_cast<int>(header->payloadType)
|
| - << " Failed to insert packet";
|
| - return -1;
|
| - }
|
| - return 0;
|
| -}
|
| -
|
| -int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
|
| - enum NetEqOutputType type;
|
| - size_t samples_per_channel;
|
| - int num_channels;
|
| -
|
| - // Accessing members, take the lock.
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| -
|
| - // Always write the output to |audio_buffer_| first.
|
| - if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
|
| - audio_buffer_.get(),
|
| - &samples_per_channel,
|
| - &num_channels,
|
| - &type) != NetEq::kOK) {
|
| - LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
|
| - return -1;
|
| - }
|
| -
|
| - const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
|
| -
|
| - // Update if resampling is required.
|
| - const bool need_resampling =
|
| - (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
|
| -
|
| - if (need_resampling && !resampled_last_output_frame_) {
|
| - // Prime the resampler with the last frame.
|
| - int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
|
| - int samples_per_channel_int = resampler_.Resample10Msec(
|
| - last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
|
| - num_channels, AudioFrame::kMaxDataSizeSamples, temp_output);
|
| - if (samples_per_channel_int < 0) {
|
| - LOG(LERROR) << "AcmReceiver::GetAudio - "
|
| - "Resampling last_audio_buffer_ failed.";
|
| - return -1;
|
| - }
|
| - samples_per_channel = static_cast<size_t>(samples_per_channel_int);
|
| - }
|
| -
|
| - // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
|
| - // through resampling, or through straight memcpy.
|
| - // TODO(henrik.lundin) Glitches in the output may appear if the output rate
|
| - // from NetEq changes. See WebRTC issue 3923.
|
| - if (need_resampling) {
|
| - int samples_per_channel_int = resampler_.Resample10Msec(
|
| - audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
|
| - num_channels, AudioFrame::kMaxDataSizeSamples, audio_frame->data_);
|
| - if (samples_per_channel_int < 0) {
|
| - LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
|
| - return -1;
|
| - }
|
| - samples_per_channel = static_cast<size_t>(samples_per_channel_int);
|
| - resampled_last_output_frame_ = true;
|
| - } else {
|
| - resampled_last_output_frame_ = false;
|
| - // We might end up here ONLY if codec is changed.
|
| - memcpy(audio_frame->data_,
|
| - audio_buffer_.get(),
|
| - samples_per_channel * num_channels * sizeof(int16_t));
|
| - }
|
| -
|
| - // Swap buffers, so that the current audio is stored in |last_audio_buffer_|
|
| - // for next time.
|
| - audio_buffer_.swap(last_audio_buffer_);
|
| -
|
| - audio_frame->num_channels_ = num_channels;
|
| - audio_frame->samples_per_channel_ = samples_per_channel;
|
| - audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
|
| -
|
| - // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
|
| - audio_frame->vad_activity_ = previous_audio_activity_;
|
| - 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 -
|
| - static_cast<uint32_t>(audio_frame->samples_per_channel_);
|
| - } else {
|
| - // Remain 0 until we have a valid |playout_timestamp|.
|
| - audio_frame->timestamp_ = 0;
|
| - }
|
| -
|
| - return 0;
|
| -}
|
| -
|
| -int32_t AcmReceiver::AddCodec(int acm_codec_id,
|
| - uint8_t payload_type,
|
| - int channels,
|
| - int sample_rate_hz,
|
| - AudioDecoder* audio_decoder) {
|
| - const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
|
| - if (acm_codec_id == -1)
|
| - return NetEqDecoder::kDecoderArbitrary; // External decoder.
|
| - const rtc::Optional<RentACodec::CodecId> cid =
|
| - RentACodec::CodecIdFromIndex(acm_codec_id);
|
| - RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
|
| - const rtc::Optional<NetEqDecoder> ned =
|
| - RentACodec::NetEqDecoderFromCodecId(*cid, channels);
|
| - RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
|
| - return *ned;
|
| - }();
|
| -
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| -
|
| - // The corresponding NetEq decoder ID.
|
| - // If this codec has been registered before.
|
| - auto it = decoders_.find(payload_type);
|
| - if (it != decoders_.end()) {
|
| - const Decoder& decoder = it->second;
|
| - if (acm_codec_id != -1 && decoder.acm_codec_id == acm_codec_id &&
|
| - decoder.channels == channels &&
|
| - decoder.sample_rate_hz == sample_rate_hz) {
|
| - // Re-registering the same codec. Do nothing and return.
|
| - return 0;
|
| - }
|
| -
|
| - // Changing codec. First unregister the old codec, then register the new
|
| - // one.
|
| - if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
|
| - LOG(LERROR) << "Cannot remove payload " << static_cast<int>(payload_type);
|
| - return -1;
|
| - }
|
| -
|
| - decoders_.erase(it);
|
| - }
|
| -
|
| - int ret_val;
|
| - if (!audio_decoder) {
|
| - ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
|
| - } else {
|
| - ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
|
| - payload_type, sample_rate_hz);
|
| - }
|
| - if (ret_val != NetEq::kOK) {
|
| - LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
|
| - << static_cast<int>(payload_type)
|
| - << " channels: " << channels;
|
| - return -1;
|
| - }
|
| -
|
| - Decoder decoder;
|
| - decoder.acm_codec_id = acm_codec_id;
|
| - decoder.payload_type = payload_type;
|
| - decoder.channels = channels;
|
| - decoder.sample_rate_hz = sample_rate_hz;
|
| - decoders_[payload_type] = decoder;
|
| - return 0;
|
| -}
|
| -
|
| -void AcmReceiver::EnableVad() {
|
| - neteq_->EnableVad();
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - vad_enabled_ = true;
|
| -}
|
| -
|
| -void AcmReceiver::DisableVad() {
|
| - neteq_->DisableVad();
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - vad_enabled_ = false;
|
| -}
|
| -
|
| -void AcmReceiver::FlushBuffers() {
|
| - neteq_->FlushBuffers();
|
| -}
|
| -
|
| -// If failed in removing one of the codecs, this method continues to remove as
|
| -// many as it can.
|
| -int AcmReceiver::RemoveAllCodecs() {
|
| - int ret_val = 0;
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - for (auto it = decoders_.begin(); it != decoders_.end(); ) {
|
| - auto cur = it;
|
| - ++it; // it will be valid even if we erase cur
|
| - if (neteq_->RemovePayloadType(cur->second.payload_type) == 0) {
|
| - decoders_.erase(cur);
|
| - } else {
|
| - LOG_F(LS_ERROR) << "Cannot remove payload "
|
| - << static_cast<int>(cur->second.payload_type);
|
| - ret_val = -1;
|
| - }
|
| - }
|
| -
|
| - // No codec is registered, invalidate last audio decoder.
|
| - last_audio_decoder_ = nullptr;
|
| - last_packet_sample_rate_hz_ = rtc::Optional<int>();
|
| - return ret_val;
|
| -}
|
| -
|
| -int AcmReceiver::RemoveCodec(uint8_t payload_type) {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - auto it = decoders_.find(payload_type);
|
| - if (it == decoders_.end()) { // Such a payload-type is not registered.
|
| - return 0;
|
| - }
|
| - if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
|
| - LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
|
| - return -1;
|
| - }
|
| - if (last_audio_decoder_ == &it->second) {
|
| - last_audio_decoder_ = nullptr;
|
| - last_packet_sample_rate_hz_ = rtc::Optional<int>();
|
| - }
|
| - decoders_.erase(it);
|
| - return 0;
|
| -}
|
| -
|
| -void AcmReceiver::set_id(int id) {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - id_ = id;
|
| -}
|
| -
|
| -bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
|
| - return neteq_->GetPlayoutTimestamp(timestamp);
|
| -}
|
| -
|
| -int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - if (!last_audio_decoder_) {
|
| - return -1;
|
| - }
|
| - *codec = *RentACodec::CodecInstById(
|
| - *RentACodec::CodecIdFromIndex(last_audio_decoder_->acm_codec_id));
|
| - codec->pltype = last_audio_decoder_->payload_type;
|
| - codec->channels = last_audio_decoder_->channels;
|
| - codec->plfreq = last_audio_decoder_->sample_rate_hz;
|
| - return 0;
|
| -}
|
| -
|
| -void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
|
| - NetEqNetworkStatistics neteq_stat;
|
| - // NetEq function always returns zero, so we don't check the return value.
|
| - neteq_->NetworkStatistics(&neteq_stat);
|
| -
|
| - acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
|
| - acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
|
| - acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
|
| - acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
|
| - acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
|
| - acm_stat->currentExpandRate = neteq_stat.expand_rate;
|
| - acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
|
| - acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
|
| - acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
|
| - acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
|
| - acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
|
| - acm_stat->addedSamples = neteq_stat.added_zero_samples;
|
| - acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
|
| - acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
|
| - acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
|
| - acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
|
| -}
|
| -
|
| -int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
|
| - CodecInst* codec) const {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - auto it = decoders_.find(payload_type);
|
| - if (it == decoders_.end()) {
|
| - LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
|
| - << static_cast<int>(payload_type);
|
| - return -1;
|
| - }
|
| - const Decoder& decoder = it->second;
|
| - *codec = *RentACodec::CodecInstById(
|
| - *RentACodec::CodecIdFromIndex(decoder.acm_codec_id));
|
| - codec->pltype = decoder.payload_type;
|
| - codec->channels = decoder.channels;
|
| - codec->plfreq = decoder.sample_rate_hz;
|
| - return 0;
|
| -}
|
| -
|
| -int AcmReceiver::EnableNack(size_t max_nack_list_size) {
|
| - neteq_->EnableNack(max_nack_list_size);
|
| - return 0;
|
| -}
|
| -
|
| -void AcmReceiver::DisableNack() {
|
| - neteq_->DisableNack();
|
| -}
|
| -
|
| -std::vector<uint16_t> AcmReceiver::GetNackList(
|
| - int64_t round_trip_time_ms) const {
|
| - return neteq_->GetNackList(round_trip_time_ms);
|
| -}
|
| -
|
| -void AcmReceiver::ResetInitialDelay() {
|
| - neteq_->SetMinimumDelay(0);
|
| - // TODO(turajs): Should NetEq Buffer be flushed?
|
| -}
|
| -
|
| -const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder(
|
| - const RTPHeader& rtp_header,
|
| - uint8_t payload_type) const {
|
| - auto it = decoders_.find(rtp_header.payloadType);
|
| - const auto red_index =
|
| - RentACodec::CodecIndexFromId(RentACodec::CodecId::kRED);
|
| - if (red_index && // This ensures that RED is defined in WebRTC.
|
| - it != decoders_.end() && it->second.acm_codec_id == *red_index) {
|
| - // This is a RED packet, get the payload of the audio codec.
|
| - it = decoders_.find(payload_type & 0x7F);
|
| - }
|
| -
|
| - // Check if the payload is registered.
|
| - return it != decoders_.end() ? &it->second : nullptr;
|
| -}
|
| -
|
| -uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
|
| - // 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 there is no overflow in
|
| - // the conversion from milliseconds to timestamp.
|
| - const uint32_t now_in_ms = static_cast<uint32_t>(
|
| - clock_->TimeInMilliseconds() & 0x03ffffff);
|
| - return static_cast<uint32_t>(
|
| - (decoder_sampling_rate / 1000) * now_in_ms);
|
| -}
|
| -
|
| -void AcmReceiver::GetDecodingCallStatistics(
|
| - AudioDecodingCallStats* stats) const {
|
| - CriticalSectionScoped lock(crit_sect_.get());
|
| - *stats = call_stats_.GetDecodingStatistics();
|
| -}
|
| -
|
| -} // namespace acm2
|
| -
|
| -} // namespace webrtc
|
|
|