diff options
Diffstat (limited to 'webrtc/modules/audio_coding/main/acm2/acm_receiver.cc')
| -rw-r--r-- | webrtc/modules/audio_coding/main/acm2/acm_receiver.cc | 739 |
1 files changed, 0 insertions, 739 deletions
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 cf486ce06a..0000000000 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc +++ /dev/null @@ -1,739 +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_common_defs.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), - current_sample_rate_hz_(config.neteq_config.sample_rate_hz), - audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), - last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), - neteq_(NetEq::Create(config.neteq_config)), - vad_enabled_(true), - clock_(config.clock), - resampled_last_output_frame_(true), - av_sync_(false), - initial_delay_manager_(), - missing_packets_sync_stream_(), - late_packets_sync_stream_() { - assert(clock_); - - // Make sure we are on the same page as NetEq. Post-decode VAD is disabled by - // default in NetEq4, however, Audio Conference Mixer relies on VAD decision - // and fails if VAD decision is not provided. - if (vad_enabled_) - neteq_->EnableVad(); - else - neteq_->DisableVad(); - - 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::SetInitialDelay(int delay_ms) { - if (delay_ms < 0 || delay_ms > 10000) { - return -1; - } - CriticalSectionScoped lock(crit_sect_.get()); - - if (delay_ms == 0) { - av_sync_ = false; - initial_delay_manager_.reset(); - missing_packets_sync_stream_.reset(); - late_packets_sync_stream_.reset(); - neteq_->SetMinimumDelay(0); - return 0; - } - - if (av_sync_ && initial_delay_manager_->PacketBuffered()) { - // Too late for this API. Only works before a call is started. - return -1; - } - - // Most of places NetEq calls are not within AcmReceiver's critical section to - // improve performance. Here, this call has to be placed before the following - // block, therefore, we keep it inside critical section. Otherwise, we have to - // release |neteq_crit_sect_| and acquire it again, which seems an overkill. - if (!neteq_->SetMinimumDelay(delay_ms)) - return -1; - - const int kLatePacketThreshold = 5; - av_sync_ = true; - initial_delay_manager_.reset(new InitialDelayManager(delay_ms, - kLatePacketThreshold)); - missing_packets_sync_stream_.reset(new InitialDelayManager::SyncStream); - late_packets_sync_stream_.reset(new InitialDelayManager::SyncStream); - return 0; -} - -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(); -} - -int AcmReceiver::current_sample_rate_hz() const { - CriticalSectionScoped lock(crit_sect_.get()); - return current_sample_rate_hz_; -} - -int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, - const uint8_t* incoming_payload, - size_t length_payload) { - uint32_t receive_timestamp = 0; - InitialDelayManager::PacketType packet_type = - InitialDelayManager::kUndefinedPacket; - bool new_codec = false; - const RTPHeader* header = &rtp_header.header; // Just a shorthand. - - { - CriticalSectionScoped lock(crit_sect_.get()); - - const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload); - if (!decoder) { - LOG_F(LS_ERROR) << "Payload-type " - << static_cast<int>(header->payloadType) - << " is not registered."; - return -1; - } - const int sample_rate_hz = ACMCodecDB::CodecFreq(decoder->acm_codec_id); - receive_timestamp = NowInTimestamp(sample_rate_hz); - - if (IsCng(decoder->acm_codec_id)) { - // If this is a CNG while the audio codec is not mono skip pushing in - // packets into NetEq. - if (last_audio_decoder_ && last_audio_decoder_->channels > 1) - return 0; - packet_type = InitialDelayManager::kCngPacket; - } else if (decoder->acm_codec_id == - *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) { - packet_type = InitialDelayManager::kAvtPacket; - } else { - if (decoder != last_audio_decoder_) { - // This is either the first audio packet or send codec is changed. - // Therefore, either NetEq buffer is empty or will be flushed when this - // packet is inserted. - new_codec = true; - last_audio_decoder_ = decoder; - } - packet_type = InitialDelayManager::kAudioPacket; - } - - if (av_sync_) { - assert(initial_delay_manager_.get()); - assert(missing_packets_sync_stream_.get()); - // This updates |initial_delay_manager_| and specifies an stream of - // sync-packets, if required to be inserted. We insert the sync-packets - // when AcmReceiver lock is released and |decoder_lock_| is acquired. - initial_delay_manager_->UpdateLastReceivedPacket( - rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz, - missing_packets_sync_stream_.get()); - } - } // |crit_sect_| is released. - - // If |missing_packets_sync_stream_| is allocated then we are in AV-sync and - // we may need to insert sync-packets. We don't check |av_sync_| as we are - // outside AcmReceiver's critical section. - if (missing_packets_sync_stream_.get()) { - InsertStreamOfSyncPackets(missing_packets_sync_stream_.get()); - } - - if (neteq_->InsertPacket(rtp_header, incoming_payload, length_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; - bool return_silence = false; - - { - // Accessing members, take the lock. - CriticalSectionScoped lock(crit_sect_.get()); - - if (av_sync_) { - assert(initial_delay_manager_.get()); - assert(late_packets_sync_stream_.get()); - return_silence = GetSilence(desired_freq_hz, audio_frame); - uint32_t timestamp_now = NowInTimestamp(current_sample_rate_hz_); - initial_delay_manager_->LatePackets(timestamp_now, - late_packets_sync_stream_.get()); - } - } - - // If |late_packets_sync_stream_| is allocated then we have been in AV-sync - // mode and we might have to insert sync-packets. - if (late_packets_sync_stream_.get()) { - InsertStreamOfSyncPackets(late_packets_sync_stream_.get()); - if (return_silence) // Silence generated, don't pull from NetEq. - return 0; - } - - // 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; - } - - // NetEq always returns 10 ms of audio. - current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100); - - // Update if resampling is required. - 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::Maybe<RentACodec::CodecId> cid = - RentACodec::CodecIdFromIndex(acm_codec_id); - RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id; - const rtc::Maybe<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; - 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; - decoders_.erase(it); - return 0; -} - -void AcmReceiver::set_id(int id) { - CriticalSectionScoped lock(crit_sect_.get()); - id_ = id; -} - -bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) { - if (av_sync_) { - assert(initial_delay_manager_.get()); - if (initial_delay_manager_->buffering()) { - return initial_delay_manager_->GetPlayoutTimestamp(timestamp); - } - } - return neteq_->GetPlayoutTimestamp(timestamp); -} - -int AcmReceiver::last_audio_codec_id() const { - CriticalSectionScoped lock(crit_sect_.get()); - return last_audio_decoder_ ? last_audio_decoder_->acm_codec_id : -1; -} - -int AcmReceiver::RedPayloadType() const { - const auto red_index = - RentACodec::CodecIndexFromId(RentACodec::CodecId::kRED); - if (red_index) { - CriticalSectionScoped lock(crit_sect_.get()); - for (const auto& decoder_pair : decoders_) { - const Decoder& decoder = decoder_pair.second; - if (decoder.acm_codec_id == *red_index) - return decoder.payload_type; - } - } - LOG(WARNING) << "RED is not registered."; - return -1; -} - -int AcmReceiver::LastAudioCodec(CodecInst* codec) const { - CriticalSectionScoped lock(crit_sect_.get()); - if (!last_audio_decoder_) { - return -1; - } - memcpy(codec, &ACMCodecDB::database_[last_audio_decoder_->acm_codec_id], - sizeof(CodecInst)); - 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; - memcpy(codec, &ACMCodecDB::database_[decoder.acm_codec_id], - sizeof(CodecInst)); - 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() { - { - CriticalSectionScoped lock(crit_sect_.get()); - av_sync_ = false; - initial_delay_manager_.reset(NULL); - missing_packets_sync_stream_.reset(NULL); - late_packets_sync_stream_.reset(NULL); - } - neteq_->SetMinimumDelay(0); - // TODO(turajs): Should NetEq Buffer be flushed? -} - -// This function is called within critical section, no need to acquire a lock. -bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) { - assert(av_sync_); - assert(initial_delay_manager_.get()); - if (!initial_delay_manager_->buffering()) { - return false; - } - - // We stop accumulating packets, if the number of packets or the total size - // exceeds a threshold. - int num_packets; - int max_num_packets; - const float kBufferingThresholdScale = 0.9f; - neteq_->PacketBufferStatistics(&num_packets, &max_num_packets); - if (num_packets > max_num_packets * kBufferingThresholdScale) { - initial_delay_manager_->DisableBuffering(); - return false; - } - - // Update statistics. - call_stats_.DecodedBySilenceGenerator(); - - // Set the values if already got a packet, otherwise set to default values. - if (last_audio_decoder_) { - current_sample_rate_hz_ = - ACMCodecDB::database_[last_audio_decoder_->acm_codec_id].plfreq; - frame->num_channels_ = last_audio_decoder_->channels; - } else { - frame->num_channels_ = 1; - } - - // Set the audio frame's sampling frequency. - if (desired_sample_rate_hz > 0) { - frame->sample_rate_hz_ = desired_sample_rate_hz; - } else { - frame->sample_rate_hz_ = current_sample_rate_hz_; - } - - frame->samples_per_channel_ = - static_cast<size_t>(frame->sample_rate_hz_ / 100); // Always 10 ms. - frame->speech_type_ = AudioFrame::kCNG; - frame->vad_activity_ = AudioFrame::kVadPassive; - size_t samples = frame->samples_per_channel_ * frame->num_channels_; - memset(frame->data_, 0, samples * sizeof(int16_t)); - return true; -} - -const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder( - const RTPHeader& rtp_header, - const uint8_t* payload) 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[0] & 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); -} - -// This function only interacts with |neteq_|, therefore, it does not have to -// be within critical section of AcmReceiver. It is inserting packets -// into NetEq, so we call it when |decode_lock_| is acquired. However, this is -// not essential as sync-packets do not interact with codecs (especially BWE). -void AcmReceiver::InsertStreamOfSyncPackets( - InitialDelayManager::SyncStream* sync_stream) { - assert(sync_stream); - assert(av_sync_); - for (int n = 0; n < sync_stream->num_sync_packets; ++n) { - neteq_->InsertSyncPacket(sync_stream->rtp_info, - sync_stream->receive_timestamp); - ++sync_stream->rtp_info.header.sequenceNumber; - sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step; - sync_stream->receive_timestamp += sync_stream->timestamp_step; - } -} - -void AcmReceiver::GetDecodingCallStatistics( - AudioDecodingCallStats* stats) const { - CriticalSectionScoped lock(crit_sect_.get()); - *stats = call_stats_.GetDecodingStatistics(); -} - -} // namespace acm2 - -} // namespace webrtc |