| /* |
| * 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 "audio/channel_receive.h" |
| |
| #include <algorithm> |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "audio/audio_level.h" |
| #include "audio/channel_send.h" |
| #include "audio/utility/audio_frame_operations.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_playout.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" |
| #include "modules/audio_coding/include/audio_coding_module.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/rtp_rtcp/include/receive_statistics.h" |
| #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "modules/rtp_rtcp/source/contributing_sources.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" |
| #include "modules/utility/include/process_thread.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/format_macros.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_minmax.h" |
| #include "rtc_base/race_checker.h" |
| #include "rtc_base/thread_checker.h" |
| #include "rtc_base/time_utils.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace webrtc { |
| namespace voe { |
| |
| namespace { |
| |
| constexpr double kAudioSampleDurationSeconds = 0.01; |
| |
| // Video Sync. |
| constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0; |
| constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000; |
| |
| RTPHeader CreateRTPHeaderForMediaTransportFrame( |
| const MediaTransportEncodedAudioFrame& frame, |
| uint64_t channel_id) { |
| webrtc::RTPHeader rtp_header; |
| rtp_header.payloadType = frame.payload_type(); |
| rtp_header.payload_type_frequency = frame.sampling_rate_hz(); |
| rtp_header.timestamp = frame.starting_sample_index(); |
| rtp_header.sequenceNumber = frame.sequence_number(); |
| |
| rtp_header.ssrc = static_cast<uint32_t>(channel_id); |
| |
| // The rest are initialized by the RTPHeader constructor. |
| return rtp_header; |
| } |
| |
| class ChannelReceive : public ChannelReceiveInterface, |
| public MediaTransportAudioSinkInterface { |
| public: |
| // Used for receive streams. |
| ChannelReceive(Clock* clock, |
| ProcessThread* module_process_thread, |
| AudioDeviceModule* audio_device_module, |
| const MediaTransportConfig& media_transport_config, |
| Transport* rtcp_send_transport, |
| RtcEventLog* rtc_event_log, |
| uint32_t remote_ssrc, |
| size_t jitter_buffer_max_packets, |
| bool jitter_buffer_fast_playout, |
| int jitter_buffer_min_delay_ms, |
| bool jitter_buffer_enable_rtx_handling, |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, |
| absl::optional<AudioCodecPairId> codec_pair_id, |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, |
| const webrtc::CryptoOptions& crypto_options); |
| ~ChannelReceive() override; |
| |
| void SetSink(AudioSinkInterface* sink) override; |
| |
| void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override; |
| |
| // API methods |
| |
| void StartPlayout() override; |
| void StopPlayout() override; |
| |
| // Codecs |
| absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec() |
| const override; |
| |
| void ReceivedRTCPPacket(const uint8_t* data, size_t length) override; |
| |
| // RtpPacketSinkInterface. |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| |
| // Muting, Volume and Level. |
| void SetChannelOutputVolumeScaling(float scaling) override; |
| int GetSpeechOutputLevelFullRange() const override; |
| // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy |
| double GetTotalOutputEnergy() const override; |
| double GetTotalOutputDuration() const override; |
| |
| // Stats. |
| NetworkStatistics GetNetworkStatistics() const override; |
| AudioDecodingCallStats GetDecodingCallStatistics() const override; |
| |
| // Audio+Video Sync. |
| uint32_t GetDelayEstimate() const override; |
| void SetMinimumPlayoutDelay(int delayMs) override; |
| uint32_t GetPlayoutTimestamp() const override; |
| |
| // Audio quality. |
| bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override; |
| int GetBaseMinimumPlayoutDelayMs() const override; |
| |
| // Produces the transport-related timestamps; current_delay_ms is left unset. |
| absl::optional<Syncable::Info> GetSyncInfo() const override; |
| |
| // RTP+RTCP |
| void SetLocalSSRC(unsigned int ssrc) override; |
| |
| void RegisterReceiverCongestionControlObjects( |
| PacketRouter* packet_router) override; |
| void ResetReceiverCongestionControlObjects() override; |
| |
| CallReceiveStatistics GetRTCPStatistics() const override; |
| void SetNACKStatus(bool enable, int maxNumberOfPackets) override; |
| |
| AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
| int sample_rate_hz, |
| AudioFrame* audio_frame) override; |
| |
| int PreferredSampleRate() const override; |
| |
| // Associate to a send channel. |
| // Used for obtaining RTT for a receive-only channel. |
| void SetAssociatedSendChannel(const ChannelSendInterface* channel) override; |
| |
| std::vector<RtpSource> GetSources() const override; |
| |
| // TODO(sukhanov): Return const pointer. It requires making media transport |
| // getters like GetLatestTargetTransferRate to be also const. |
| MediaTransportInterface* media_transport() const { |
| return media_transport_config_.media_transport; |
| } |
| |
| private: |
| bool ReceivePacket(const uint8_t* packet, |
| size_t packet_length, |
| const RTPHeader& header); |
| int ResendPackets(const uint16_t* sequence_numbers, int length); |
| void UpdatePlayoutTimestamp(bool rtcp); |
| |
| int GetRtpTimestampRateHz() const; |
| int64_t GetRTT() const; |
| |
| // MediaTransportAudioSinkInterface override; |
| void OnData(uint64_t channel_id, |
| MediaTransportEncodedAudioFrame frame) override; |
| |
| int32_t OnReceivedPayloadData(const uint8_t* payloadData, |
| size_t payloadSize, |
| const RTPHeader& rtpHeader); |
| |
| bool Playing() const { |
| rtc::CritScope lock(&playing_lock_); |
| return playing_; |
| } |
| |
| // Thread checkers document and lock usage of some methods to specific threads |
| // we know about. The goal is to eventually split up voe::ChannelReceive into |
| // parts with single-threaded semantics, and thereby reduce the need for |
| // locks. |
| rtc::ThreadChecker worker_thread_checker_; |
| rtc::ThreadChecker module_process_thread_checker_; |
| // Methods accessed from audio and video threads are checked for sequential- |
| // only access. We don't necessarily own and control these threads, so thread |
| // checkers cannot be used. E.g. Chromium may transfer "ownership" from one |
| // audio thread to another, but access is still sequential. |
| rtc::RaceChecker audio_thread_race_checker_; |
| rtc::RaceChecker video_capture_thread_race_checker_; |
| rtc::CriticalSection _callbackCritSect; |
| rtc::CriticalSection volume_settings_critsect_; |
| |
| rtc::CriticalSection playing_lock_; |
| bool playing_ RTC_GUARDED_BY(&playing_lock_) = false; |
| |
| RtcEventLog* const event_log_; |
| |
| // Indexed by payload type. |
| std::map<uint8_t, int> payload_type_frequencies_; |
| |
| std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; |
| std::unique_ptr<RtpRtcp> _rtpRtcpModule; |
| const uint32_t remote_ssrc_; |
| |
| // Info for GetSources and GetSyncInfo is updated on network or worker thread, |
| // queried on the worker thread. |
| rtc::CriticalSection rtp_sources_lock_; |
| ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_); |
| absl::optional<uint32_t> last_received_rtp_timestamp_ |
| RTC_GUARDED_BY(&rtp_sources_lock_); |
| absl::optional<int64_t> last_received_rtp_system_time_ms_ |
| RTC_GUARDED_BY(&rtp_sources_lock_); |
| absl::optional<uint8_t> last_received_rtp_audio_level_ |
| RTC_GUARDED_BY(&rtp_sources_lock_); |
| |
| std::unique_ptr<AudioCodingModule> audio_coding_; |
| AudioSinkInterface* audio_sink_ = nullptr; |
| AudioLevel _outputAudioLevel; |
| |
| RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_); |
| |
| // Timestamp of the audio pulled from NetEq. |
| absl::optional<uint32_t> jitter_buffer_playout_timestamp_; |
| |
| rtc::CriticalSection video_sync_lock_; |
| uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_); |
| uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_); |
| |
| rtc::CriticalSection ts_stats_lock_; |
| |
| std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_; |
| // The rtp timestamp of the first played out audio frame. |
| int64_t capture_start_rtp_time_stamp_; |
| // The capture ntp time (in local timebase) of the first played out audio |
| // frame. |
| int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_); |
| |
| // uses |
| ProcessThread* _moduleProcessThreadPtr; |
| AudioDeviceModule* _audioDeviceModulePtr; |
| float _outputGain RTC_GUARDED_BY(volume_settings_critsect_); |
| |
| // An associated send channel. |
| rtc::CriticalSection assoc_send_channel_lock_; |
| const ChannelSendInterface* associated_send_channel_ |
| RTC_GUARDED_BY(assoc_send_channel_lock_); |
| |
| PacketRouter* packet_router_ = nullptr; |
| |
| rtc::ThreadChecker construction_thread_; |
| |
| MediaTransportConfig media_transport_config_; |
| |
| // E2EE Audio Frame Decryption |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_; |
| webrtc::CryptoOptions crypto_options_; |
| }; |
| |
| int32_t ChannelReceive::OnReceivedPayloadData(const uint8_t* payloadData, |
| size_t payloadSize, |
| const RTPHeader& rtp_header) { |
| // We should not be receiving any RTP packets if media_transport is set. |
| RTC_CHECK(!media_transport()); |
| |
| if (!Playing()) { |
| // Avoid inserting into NetEQ when we are not playing. Count the |
| // packet as discarded. |
| return 0; |
| } |
| |
| // Push the incoming payload (parsed and ready for decoding) into the ACM |
| if (audio_coding_->IncomingPacket(payloadData, payloadSize, rtp_header) != |
| 0) { |
| RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to " |
| "push data to the ACM"; |
| return -1; |
| } |
| |
| int64_t round_trip_time = 0; |
| _rtpRtcpModule->RTT(remote_ssrc_, &round_trip_time, NULL, NULL, NULL); |
| |
| std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time); |
| if (!nack_list.empty()) { |
| // Can't use nack_list.data() since it's not supported by all |
| // compilers. |
| ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size())); |
| } |
| return 0; |
| } |
| |
| // MediaTransportAudioSinkInterface override. |
| void ChannelReceive::OnData(uint64_t channel_id, |
| MediaTransportEncodedAudioFrame frame) { |
| RTC_CHECK(media_transport()); |
| |
| if (!Playing()) { |
| // Avoid inserting into NetEQ when we are not playing. Count the |
| // packet as discarded. |
| return; |
| } |
| |
| // Send encoded audio frame to Decoder / NetEq. |
| if (audio_coding_->IncomingPacket( |
| frame.encoded_data().data(), frame.encoded_data().size(), |
| CreateRTPHeaderForMediaTransportFrame(frame, channel_id)) != 0) { |
| RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to " |
| "push data to the ACM"; |
| } |
| } |
| |
| AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo( |
| int sample_rate_hz, |
| AudioFrame* audio_frame) { |
| RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); |
| audio_frame->sample_rate_hz_ = sample_rate_hz; |
| |
| event_log_->Log(absl::make_unique<RtcEventAudioPlayout>(remote_ssrc_)); |
| |
| // Get 10ms raw PCM data from the ACM (mixer limits output frequency) |
| bool muted; |
| if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame, |
| &muted) == -1) { |
| RTC_DLOG(LS_ERROR) |
| << "ChannelReceive::GetAudioFrame() PlayoutData10Ms() failed!"; |
| // In all likelihood, the audio in this frame is garbage. We return an |
| // error so that the audio mixer module doesn't add it to the mix. As |
| // a result, it won't be played out and the actions skipped here are |
| // irrelevant. |
| return AudioMixer::Source::AudioFrameInfo::kError; |
| } |
| |
| if (muted) { |
| // TODO(henrik.lundin): We should be able to do better than this. But we |
| // will have to go through all the cases below where the audio samples may |
| // be used, and handle the muted case in some way. |
| AudioFrameOperations::Mute(audio_frame); |
| } |
| |
| { |
| // Pass the audio buffers to an optional sink callback, before applying |
| // scaling/panning, as that applies to the mix operation. |
| // External recipients of the audio (e.g. via AudioTrack), will do their |
| // own mixing/dynamic processing. |
| rtc::CritScope cs(&_callbackCritSect); |
| if (audio_sink_) { |
| AudioSinkInterface::Data data( |
| audio_frame->data(), audio_frame->samples_per_channel_, |
| audio_frame->sample_rate_hz_, audio_frame->num_channels_, |
| audio_frame->timestamp_); |
| audio_sink_->OnData(data); |
| } |
| } |
| |
| float output_gain = 1.0f; |
| { |
| rtc::CritScope cs(&volume_settings_critsect_); |
| output_gain = _outputGain; |
| } |
| |
| // Output volume scaling |
| if (output_gain < 0.99f || output_gain > 1.01f) { |
| // TODO(solenberg): Combine with mute state - this can cause clicks! |
| AudioFrameOperations::ScaleWithSat(output_gain, audio_frame); |
| } |
| |
| // Measure audio level (0-9) |
| // TODO(henrik.lundin) Use the |muted| information here too. |
| // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see |
| // https://crbug.com/webrtc/7517). |
| _outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds); |
| |
| if (capture_start_rtp_time_stamp_ < 0 && audio_frame->timestamp_ != 0) { |
| // The first frame with a valid rtp timestamp. |
| capture_start_rtp_time_stamp_ = audio_frame->timestamp_; |
| } |
| |
| if (capture_start_rtp_time_stamp_ >= 0) { |
| // audio_frame.timestamp_ should be valid from now on. |
| |
| // Compute elapsed time. |
| int64_t unwrap_timestamp = |
| rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_); |
| audio_frame->elapsed_time_ms_ = |
| (unwrap_timestamp - capture_start_rtp_time_stamp_) / |
| (GetRtpTimestampRateHz() / 1000); |
| |
| { |
| rtc::CritScope lock(&ts_stats_lock_); |
| // Compute ntp time. |
| audio_frame->ntp_time_ms_ = |
| ntp_estimator_.Estimate(audio_frame->timestamp_); |
| // |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received. |
| if (audio_frame->ntp_time_ms_ > 0) { |
| // Compute |capture_start_ntp_time_ms_| so that |
| // |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_| |
| capture_start_ntp_time_ms_ = |
| audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_; |
| } |
| } |
| } |
| |
| { |
| RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs", |
| audio_coding_->TargetDelayMs()); |
| const int jitter_buffer_delay = audio_coding_->FilteredCurrentDelayMs(); |
| rtc::CritScope lock(&video_sync_lock_); |
| RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs", |
| jitter_buffer_delay + playout_delay_ms_); |
| RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs", |
| jitter_buffer_delay); |
| RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs", |
| playout_delay_ms_); |
| } |
| |
| return muted ? AudioMixer::Source::AudioFrameInfo::kMuted |
| : AudioMixer::Source::AudioFrameInfo::kNormal; |
| } |
| |
| int ChannelReceive::PreferredSampleRate() const { |
| RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); |
| // Return the bigger of playout and receive frequency in the ACM. |
| return std::max(audio_coding_->ReceiveFrequency(), |
| audio_coding_->PlayoutFrequency()); |
| } |
| |
| ChannelReceive::ChannelReceive( |
| Clock* clock, |
| ProcessThread* module_process_thread, |
| AudioDeviceModule* audio_device_module, |
| const MediaTransportConfig& media_transport_config, |
| Transport* rtcp_send_transport, |
| RtcEventLog* rtc_event_log, |
| uint32_t remote_ssrc, |
| size_t jitter_buffer_max_packets, |
| bool jitter_buffer_fast_playout, |
| int jitter_buffer_min_delay_ms, |
| bool jitter_buffer_enable_rtx_handling, |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, |
| absl::optional<AudioCodecPairId> codec_pair_id, |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, |
| const webrtc::CryptoOptions& crypto_options) |
| : event_log_(rtc_event_log), |
| rtp_receive_statistics_(ReceiveStatistics::Create(clock)), |
| remote_ssrc_(remote_ssrc), |
| _outputAudioLevel(), |
| ntp_estimator_(clock), |
| playout_timestamp_rtp_(0), |
| playout_delay_ms_(0), |
| rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()), |
| capture_start_rtp_time_stamp_(-1), |
| capture_start_ntp_time_ms_(-1), |
| _moduleProcessThreadPtr(module_process_thread), |
| _audioDeviceModulePtr(audio_device_module), |
| _outputGain(1.0f), |
| associated_send_channel_(nullptr), |
| media_transport_config_(media_transport_config), |
| frame_decryptor_(frame_decryptor), |
| crypto_options_(crypto_options) { |
| // TODO(nisse): Use _moduleProcessThreadPtr instead? |
| module_process_thread_checker_.Detach(); |
| |
| RTC_DCHECK(module_process_thread); |
| RTC_DCHECK(audio_device_module); |
| AudioCodingModule::Config acm_config; |
| acm_config.decoder_factory = decoder_factory; |
| acm_config.neteq_config.codec_pair_id = codec_pair_id; |
| acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets; |
| acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout; |
| acm_config.neteq_config.min_delay_ms = jitter_buffer_min_delay_ms; |
| acm_config.neteq_config.enable_muted_state = true; |
| acm_config.neteq_config.enable_rtx_handling = |
| jitter_buffer_enable_rtx_handling; |
| audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
| |
| _outputAudioLevel.ResetLevelFullRange(); |
| |
| rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true); |
| RtpRtcp::Configuration configuration; |
| configuration.clock = clock; |
| configuration.audio = true; |
| configuration.receiver_only = true; |
| configuration.outgoing_transport = rtcp_send_transport; |
| configuration.receive_statistics = rtp_receive_statistics_.get(); |
| |
| configuration.event_log = event_log_; |
| |
| _rtpRtcpModule = RtpRtcp::Create(configuration); |
| _rtpRtcpModule->SetSendingMediaStatus(false); |
| _rtpRtcpModule->SetRemoteSSRC(remote_ssrc_); |
| |
| _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE); |
| |
| // Ensure that RTCP is enabled by default for the created channel. |
| // Note that, the module will keep generating RTCP until it is explicitly |
| // disabled by the user. |
| // After StopListen (when no sockets exists), RTCP packets will no longer |
| // be transmitted since the Transport object will then be invalid. |
| // RTCP is enabled by default. |
| _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); |
| |
| if (media_transport()) { |
| media_transport()->SetReceiveAudioSink(this); |
| } |
| } |
| |
| ChannelReceive::~ChannelReceive() { |
| RTC_DCHECK(construction_thread_.IsCurrent()); |
| |
| if (media_transport()) { |
| media_transport()->SetReceiveAudioSink(nullptr); |
| } |
| |
| StopPlayout(); |
| |
| int error = audio_coding_->RegisterTransportCallback(NULL); |
| RTC_DCHECK_EQ(0, error); |
| |
| if (_moduleProcessThreadPtr) |
| _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()); |
| } |
| |
| void ChannelReceive::SetSink(AudioSinkInterface* sink) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| rtc::CritScope cs(&_callbackCritSect); |
| audio_sink_ = sink; |
| } |
| |
| void ChannelReceive::StartPlayout() { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| rtc::CritScope lock(&playing_lock_); |
| playing_ = true; |
| } |
| |
| void ChannelReceive::StopPlayout() { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| rtc::CritScope lock(&playing_lock_); |
| playing_ = false; |
| _outputAudioLevel.ResetLevelFullRange(); |
| } |
| |
| absl::optional<std::pair<int, SdpAudioFormat>> |
| ChannelReceive::GetReceiveCodec() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| return audio_coding_->ReceiveCodec(); |
| } |
| |
| std::vector<webrtc::RtpSource> ChannelReceive::GetSources() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| int64_t now_ms = rtc::TimeMillis(); |
| std::vector<RtpSource> sources; |
| { |
| rtc::CritScope cs(&rtp_sources_lock_); |
| sources = contributing_sources_.GetSources(now_ms); |
| if (last_received_rtp_system_time_ms_ >= |
| now_ms - ContributingSources::kHistoryMs) { |
| RTC_DCHECK(last_received_rtp_timestamp_.has_value()); |
| sources.emplace_back(*last_received_rtp_system_time_ms_, remote_ssrc_, |
| RtpSourceType::SSRC, last_received_rtp_audio_level_, |
| *last_received_rtp_timestamp_); |
| } |
| } |
| return sources; |
| } |
| |
| void ChannelReceive::SetReceiveCodecs( |
| const std::map<int, SdpAudioFormat>& codecs) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| for (const auto& kv : codecs) { |
| RTC_DCHECK_GE(kv.second.clockrate_hz, 1000); |
| payload_type_frequencies_[kv.first] = kv.second.clockrate_hz; |
| } |
| audio_coding_->SetReceiveCodecs(codecs); |
| } |
| |
| // May be called on either worker thread or network thread. |
| void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) { |
| int64_t now_ms = rtc::TimeMillis(); |
| uint8_t audio_level; |
| bool voice_activity; |
| bool has_audio_level = |
| packet.GetExtension<::webrtc::AudioLevel>(&voice_activity, &audio_level); |
| |
| { |
| rtc::CritScope cs(&rtp_sources_lock_); |
| last_received_rtp_timestamp_ = packet.Timestamp(); |
| last_received_rtp_system_time_ms_ = now_ms; |
| if (has_audio_level) |
| last_received_rtp_audio_level_ = audio_level; |
| std::vector<uint32_t> csrcs = packet.Csrcs(); |
| contributing_sources_.Update( |
| now_ms, csrcs, |
| has_audio_level ? absl::optional<uint8_t>(audio_level) : absl::nullopt, |
| packet.Timestamp()); |
| } |
| |
| // Store playout timestamp for the received RTP packet |
| UpdatePlayoutTimestamp(false); |
| |
| const auto& it = payload_type_frequencies_.find(packet.PayloadType()); |
| if (it == payload_type_frequencies_.end()) |
| return; |
| // TODO(nisse): Set payload_type_frequency earlier, when packet is parsed. |
| RtpPacketReceived packet_copy(packet); |
| packet_copy.set_payload_type_frequency(it->second); |
| |
| rtp_receive_statistics_->OnRtpPacket(packet_copy); |
| |
| RTPHeader header; |
| packet_copy.GetHeader(&header); |
| |
| ReceivePacket(packet_copy.data(), packet_copy.size(), header); |
| } |
| |
| bool ChannelReceive::ReceivePacket(const uint8_t* packet, |
| size_t packet_length, |
| const RTPHeader& header) { |
| const uint8_t* payload = packet + header.headerLength; |
| assert(packet_length >= header.headerLength); |
| size_t payload_length = packet_length - header.headerLength; |
| |
| size_t payload_data_length = payload_length - header.paddingLength; |
| |
| // E2EE Custom Audio Frame Decryption (This is optional). |
| // Keep this buffer around for the lifetime of the OnReceivedPayloadData call. |
| rtc::Buffer decrypted_audio_payload; |
| if (frame_decryptor_ != nullptr) { |
| const size_t max_plaintext_size = frame_decryptor_->GetMaxPlaintextByteSize( |
| cricket::MEDIA_TYPE_AUDIO, payload_length); |
| decrypted_audio_payload.SetSize(max_plaintext_size); |
| |
| const std::vector<uint32_t> csrcs(header.arrOfCSRCs, |
| header.arrOfCSRCs + header.numCSRCs); |
| const FrameDecryptorInterface::Result decrypt_result = |
| frame_decryptor_->Decrypt( |
| cricket::MEDIA_TYPE_AUDIO, csrcs, |
| /*additional_data=*/nullptr, |
| rtc::ArrayView<const uint8_t>(payload, payload_data_length), |
| decrypted_audio_payload); |
| |
| if (decrypt_result.IsOk()) { |
| decrypted_audio_payload.SetSize(decrypt_result.bytes_written); |
| } else { |
| // Interpret failures as a silent frame. |
| decrypted_audio_payload.SetSize(0); |
| } |
| |
| payload = decrypted_audio_payload.data(); |
| payload_data_length = decrypted_audio_payload.size(); |
| } else if (crypto_options_.sframe.require_frame_encryption) { |
| RTC_DLOG(LS_ERROR) |
| << "FrameDecryptor required but not set, dropping packet"; |
| payload_data_length = 0; |
| } |
| |
| if (payload_data_length == 0) { |
| return OnReceivedPayloadData(nullptr, 0, header); |
| } |
| return OnReceivedPayloadData(payload, payload_data_length, header); |
| } |
| |
| // May be called on either worker thread or network thread. |
| void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) { |
| // Store playout timestamp for the received RTCP packet |
| UpdatePlayoutTimestamp(true); |
| |
| // Deliver RTCP packet to RTP/RTCP module for parsing |
| _rtpRtcpModule->IncomingRtcpPacket(data, length); |
| |
| int64_t rtt = GetRTT(); |
| if (rtt == 0) { |
| // Waiting for valid RTT. |
| return; |
| } |
| |
| uint32_t ntp_secs = 0; |
| uint32_t ntp_frac = 0; |
| uint32_t rtp_timestamp = 0; |
| if (0 != _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, |
| &rtp_timestamp)) { |
| // Waiting for RTCP. |
| return; |
| } |
| |
| { |
| rtc::CritScope lock(&ts_stats_lock_); |
| ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); |
| } |
| } |
| |
| int ChannelReceive::GetSpeechOutputLevelFullRange() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| return _outputAudioLevel.LevelFullRange(); |
| } |
| |
| double ChannelReceive::GetTotalOutputEnergy() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| return _outputAudioLevel.TotalEnergy(); |
| } |
| |
| double ChannelReceive::GetTotalOutputDuration() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| return _outputAudioLevel.TotalDuration(); |
| } |
| |
| void ChannelReceive::SetChannelOutputVolumeScaling(float scaling) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| rtc::CritScope cs(&volume_settings_critsect_); |
| _outputGain = scaling; |
| } |
| |
| void ChannelReceive::SetLocalSSRC(uint32_t ssrc) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| _rtpRtcpModule->SetSSRC(ssrc); |
| } |
| |
| void ChannelReceive::RegisterReceiverCongestionControlObjects( |
| PacketRouter* packet_router) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| RTC_DCHECK(packet_router); |
| RTC_DCHECK(!packet_router_); |
| constexpr bool remb_candidate = false; |
| packet_router->AddReceiveRtpModule(_rtpRtcpModule.get(), remb_candidate); |
| packet_router_ = packet_router; |
| } |
| |
| void ChannelReceive::ResetReceiverCongestionControlObjects() { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| RTC_DCHECK(packet_router_); |
| packet_router_->RemoveReceiveRtpModule(_rtpRtcpModule.get()); |
| packet_router_ = nullptr; |
| } |
| |
| CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| // --- RtcpStatistics |
| CallReceiveStatistics stats; |
| |
| // The jitter statistics is updated for each received RTP packet and is |
| // based on received packets. |
| RtcpStatistics statistics; |
| StreamStatistician* statistician = |
| rtp_receive_statistics_->GetStatistician(remote_ssrc_); |
| if (statistician) { |
| statistician->GetStatistics(&statistics, |
| _rtpRtcpModule->RTCP() == RtcpMode::kOff); |
| } |
| |
| stats.cumulativeLost = statistics.packets_lost; |
| stats.extendedMax = statistics.extended_highest_sequence_number; |
| stats.jitterSamples = statistics.jitter; |
| |
| // --- RTT |
| stats.rttMs = GetRTT(); |
| |
| // --- Data counters |
| if (statistician) { |
| StreamDataCounters data_counters; |
| statistician->GetReceiveStreamDataCounters(&data_counters); |
| // TODO(http://crbug.com/webrtc/10525): Bytes received should only include |
| // payload bytes, not header and padding bytes. |
| stats.bytesReceived = data_counters.transmitted.payload_bytes + |
| data_counters.transmitted.header_bytes + |
| data_counters.transmitted.padding_bytes; |
| stats.packetsReceived = data_counters.transmitted.packets; |
| stats.last_packet_received_timestamp_ms = |
| data_counters.last_packet_received_timestamp_ms; |
| } else { |
| stats.bytesReceived = 0; |
| stats.packetsReceived = 0; |
| stats.last_packet_received_timestamp_ms = absl::nullopt; |
| } |
| |
| // --- Timestamps |
| { |
| rtc::CritScope lock(&ts_stats_lock_); |
| stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_; |
| } |
| return stats; |
| } |
| |
| void ChannelReceive::SetNACKStatus(bool enable, int max_packets) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| // None of these functions can fail. |
| if (enable) { |
| rtp_receive_statistics_->SetMaxReorderingThreshold(remote_ssrc_, |
| max_packets); |
| audio_coding_->EnableNack(max_packets); |
| } else { |
| rtp_receive_statistics_->SetMaxReorderingThreshold( |
| remote_ssrc_, kDefaultMaxReorderingThreshold); |
| audio_coding_->DisableNack(); |
| } |
| } |
| |
| // Called when we are missing one or more packets. |
| int ChannelReceive::ResendPackets(const uint16_t* sequence_numbers, |
| int length) { |
| return _rtpRtcpModule->SendNACK(sequence_numbers, length); |
| } |
| |
| void ChannelReceive::SetAssociatedSendChannel( |
| const ChannelSendInterface* channel) { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| rtc::CritScope lock(&assoc_send_channel_lock_); |
| associated_send_channel_ = channel; |
| } |
| |
| NetworkStatistics ChannelReceive::GetNetworkStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| NetworkStatistics stats; |
| int error = audio_coding_->GetNetworkStatistics(&stats); |
| RTC_DCHECK_EQ(0, error); |
| return stats; |
| } |
| |
| AudioDecodingCallStats ChannelReceive::GetDecodingCallStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent()); |
| AudioDecodingCallStats stats; |
| audio_coding_->GetDecodingCallStatistics(&stats); |
| return stats; |
| } |
| |
| uint32_t ChannelReceive::GetDelayEstimate() const { |
| RTC_DCHECK(worker_thread_checker_.IsCurrent() || |
| module_process_thread_checker_.IsCurrent()); |
| rtc::CritScope lock(&video_sync_lock_); |
| return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_; |
| } |
| |
| void ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) { |
| RTC_DCHECK(module_process_thread_checker_.IsCurrent()); |
| // Limit to range accepted by both VoE and ACM, so we're at least getting as |
| // close as possible, instead of failing. |
| delay_ms = rtc::SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs, |
| kVoiceEngineMaxMinPlayoutDelayMs); |
| if (audio_coding_->SetMinimumPlayoutDelay(delay_ms) != 0) { |
| RTC_DLOG(LS_ERROR) |
| << "SetMinimumPlayoutDelay() failed to set min playout delay"; |
| } |
| } |
| |
| uint32_t ChannelReceive::GetPlayoutTimestamp() const { |
| RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_); |
| { |
| rtc::CritScope lock(&video_sync_lock_); |
| return playout_timestamp_rtp_; |
| } |
| } |
| |
| bool ChannelReceive::SetBaseMinimumPlayoutDelayMs(int delay_ms) { |
| return audio_coding_->SetBaseMinimumPlayoutDelayMs(delay_ms); |
| } |
| |
| int ChannelReceive::GetBaseMinimumPlayoutDelayMs() const { |
| return audio_coding_->GetBaseMinimumPlayoutDelayMs(); |
| } |
| |
| absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const { |
| RTC_DCHECK(module_process_thread_checker_.IsCurrent()); |
| Syncable::Info info; |
| if (_rtpRtcpModule->RemoteNTP(&info.capture_time_ntp_secs, |
| &info.capture_time_ntp_frac, nullptr, nullptr, |
| &info.capture_time_source_clock) != 0) { |
| return absl::nullopt; |
| } |
| { |
| rtc::CritScope cs(&rtp_sources_lock_); |
| if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) { |
| return absl::nullopt; |
| } |
| info.latest_received_capture_timestamp = *last_received_rtp_timestamp_; |
| info.latest_receive_time_ms = *last_received_rtp_system_time_ms_; |
| } |
| return info; |
| } |
| |
| void ChannelReceive::UpdatePlayoutTimestamp(bool rtcp) { |
| jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp(); |
| |
| if (!jitter_buffer_playout_timestamp_) { |
| // This can happen if this channel has not received any RTP packets. In |
| // this case, NetEq is not capable of computing a playout timestamp. |
| return; |
| } |
| |
| uint16_t delay_ms = 0; |
| if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) { |
| RTC_DLOG(LS_WARNING) |
| << "ChannelReceive::UpdatePlayoutTimestamp() failed to read" |
| << " playout delay from the ADM"; |
| return; |
| } |
| |
| RTC_DCHECK(jitter_buffer_playout_timestamp_); |
| uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_; |
| |
| // Remove the playout delay. |
| playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000)); |
| |
| { |
| rtc::CritScope lock(&video_sync_lock_); |
| if (!rtcp) { |
| playout_timestamp_rtp_ = playout_timestamp; |
| } |
| playout_delay_ms_ = delay_ms; |
| } |
| } |
| |
| int ChannelReceive::GetRtpTimestampRateHz() const { |
| const auto decoder = audio_coding_->ReceiveCodec(); |
| // Default to the playout frequency if we've not gotten any packets yet. |
| // TODO(ossu): Zero clockrate can only happen if we've added an external |
| // decoder for a format we don't support internally. Remove once that way of |
| // adding decoders is gone! |
| return (decoder && decoder->second.clockrate_hz != 0) |
| ? decoder->second.clockrate_hz |
| : audio_coding_->PlayoutFrequency(); |
| } |
| |
| int64_t ChannelReceive::GetRTT() const { |
| if (media_transport()) { |
| auto target_rate = media_transport()->GetLatestTargetTransferRate(); |
| if (target_rate.has_value()) { |
| return target_rate->network_estimate.round_trip_time.ms(); |
| } |
| |
| return 0; |
| } |
| RtcpMode method = _rtpRtcpModule->RTCP(); |
| if (method == RtcpMode::kOff) { |
| return 0; |
| } |
| std::vector<RTCPReportBlock> report_blocks; |
| _rtpRtcpModule->RemoteRTCPStat(&report_blocks); |
| |
| // TODO(nisse): Could we check the return value from the ->RTT() call below, |
| // instead of checking if we have any report blocks? |
| if (report_blocks.empty()) { |
| rtc::CritScope lock(&assoc_send_channel_lock_); |
| // Tries to get RTT from an associated channel. |
| if (!associated_send_channel_) { |
| return 0; |
| } |
| return associated_send_channel_->GetRTT(); |
| } |
| |
| int64_t rtt = 0; |
| int64_t avg_rtt = 0; |
| int64_t max_rtt = 0; |
| int64_t min_rtt = 0; |
| // TODO(nisse): This method computes RTT based on sender reports, even though |
| // a receive stream is not supposed to do that. |
| if (_rtpRtcpModule->RTT(remote_ssrc_, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
| 0) { |
| return 0; |
| } |
| return rtt; |
| } |
| |
| } // namespace |
| |
| std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive( |
| Clock* clock, |
| ProcessThread* module_process_thread, |
| AudioDeviceModule* audio_device_module, |
| const MediaTransportConfig& media_transport_config, |
| Transport* rtcp_send_transport, |
| RtcEventLog* rtc_event_log, |
| uint32_t remote_ssrc, |
| size_t jitter_buffer_max_packets, |
| bool jitter_buffer_fast_playout, |
| int jitter_buffer_min_delay_ms, |
| bool jitter_buffer_enable_rtx_handling, |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, |
| absl::optional<AudioCodecPairId> codec_pair_id, |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, |
| const webrtc::CryptoOptions& crypto_options) { |
| return absl::make_unique<ChannelReceive>( |
| clock, module_process_thread, audio_device_module, media_transport_config, |
| rtcp_send_transport, rtc_event_log, remote_ssrc, |
| jitter_buffer_max_packets, jitter_buffer_fast_playout, |
| jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling, |
| decoder_factory, codec_pair_id, frame_decryptor, crypto_options); |
| } |
| |
| } // namespace voe |
| } // namespace webrtc |