| /* |
| * Copyright (c) 2015 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/audio_send_stream.h" |
| |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "api/audio_codecs/audio_encoder.h" |
| #include "api/audio_codecs/audio_encoder_factory.h" |
| #include "api/audio_codecs/audio_format.h" |
| #include "api/call/transport.h" |
| #include "api/crypto/frameencryptorinterface.h" |
| #include "audio/audio_state.h" |
| #include "audio/channel_send.h" |
| #include "audio/conversion.h" |
| #include "call/rtp_config.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "common_audio/vad/include/vad.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "logging/rtc_event_log/rtc_stream_config.h" |
| #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/function_view.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/strings/audio_format_to_string.h" |
| #include "rtc_base/task_queue.h" |
| #include "rtc_base/timeutils.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| namespace internal { |
| namespace { |
| // TODO(eladalon): Subsequent CL will make these values experiment-dependent. |
| constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000; |
| constexpr size_t kPacketLossRateMinNumAckedPackets = 50; |
| constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40; |
| |
| void CallEncoder(const std::unique_ptr<voe::ChannelSendInterface>& channel_send, |
| rtc::FunctionView<void(AudioEncoder*)> lambda) { |
| channel_send->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) { |
| RTC_DCHECK(encoder_ptr); |
| lambda(encoder_ptr->get()); |
| }); |
| } |
| |
| void UpdateEventLogStreamConfig(RtcEventLog* event_log, |
| const AudioSendStream::Config& config, |
| const AudioSendStream::Config* old_config) { |
| using SendCodecSpec = AudioSendStream::Config::SendCodecSpec; |
| // Only update if any of the things we log have changed. |
| auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a, |
| const absl::optional<SendCodecSpec>& b) { |
| if (a.has_value() && b.has_value()) { |
| return a->format.name == b->format.name && |
| a->payload_type == b->payload_type; |
| } |
| return !a.has_value() && !b.has_value(); |
| }; |
| |
| if (old_config && config.rtp.ssrc == old_config->rtp.ssrc && |
| config.rtp.extensions == old_config->rtp.extensions && |
| payload_types_equal(config.send_codec_spec, |
| old_config->send_codec_spec)) { |
| return; |
| } |
| |
| auto rtclog_config = absl::make_unique<rtclog::StreamConfig>(); |
| rtclog_config->local_ssrc = config.rtp.ssrc; |
| rtclog_config->rtp_extensions = config.rtp.extensions; |
| if (config.send_codec_spec) { |
| rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name, |
| config.send_codec_spec->payload_type, 0); |
| } |
| event_log->Log(absl::make_unique<RtcEventAudioSendStreamConfig>( |
| std::move(rtclog_config))); |
| } |
| |
| } // namespace |
| |
| AudioSendStream::AudioSendStream( |
| const webrtc::AudioSendStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| rtc::TaskQueue* worker_queue, |
| ProcessThread* module_process_thread, |
| RtpTransportControllerSendInterface* rtp_transport, |
| BitrateAllocatorInterface* bitrate_allocator, |
| RtcEventLog* event_log, |
| RtcpRttStats* rtcp_rtt_stats, |
| const absl::optional<RtpState>& suspended_rtp_state) |
| : AudioSendStream(config, |
| audio_state, |
| worker_queue, |
| rtp_transport, |
| bitrate_allocator, |
| event_log, |
| rtcp_rtt_stats, |
| suspended_rtp_state, |
| voe::CreateChannelSend(worker_queue, |
| module_process_thread, |
| config.media_transport, |
| config.send_transport, |
| rtcp_rtt_stats, |
| event_log, |
| config.frame_encryptor, |
| config.crypto_options, |
| config.rtp.extmap_allow_mixed, |
| config.rtcp_report_interval_ms)) {} |
| |
| AudioSendStream::AudioSendStream( |
| const webrtc::AudioSendStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| rtc::TaskQueue* worker_queue, |
| RtpTransportControllerSendInterface* rtp_transport, |
| BitrateAllocatorInterface* bitrate_allocator, |
| RtcEventLog* event_log, |
| RtcpRttStats* rtcp_rtt_stats, |
| const absl::optional<RtpState>& suspended_rtp_state, |
| std::unique_ptr<voe::ChannelSendInterface> channel_send) |
| : worker_queue_(worker_queue), |
| config_(Config(/*send_transport=*/nullptr, |
| /*media_transport=*/nullptr)), |
| audio_state_(audio_state), |
| channel_send_(std::move(channel_send)), |
| event_log_(event_log), |
| bitrate_allocator_(bitrate_allocator), |
| rtp_transport_(rtp_transport), |
| packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs, |
| kPacketLossRateMinNumAckedPackets, |
| kRecoverablePacketLossRateMinNumAckedPairs), |
| rtp_rtcp_module_(nullptr), |
| suspended_rtp_state_(suspended_rtp_state) { |
| RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc; |
| RTC_DCHECK(worker_queue_); |
| RTC_DCHECK(audio_state_); |
| RTC_DCHECK(channel_send_); |
| RTC_DCHECK(bitrate_allocator_); |
| // TODO(nisse): Eventually, we should have only media_transport. But for the |
| // time being, we can have either. When media transport is injected, there |
| // should be no rtp_transport, and below check should be strengthened to XOR |
| // (either rtp_transport or media_transport but not both). |
| RTC_DCHECK(rtp_transport || config.media_transport); |
| |
| rtp_rtcp_module_ = channel_send_->GetRtpRtcp(); |
| RTC_DCHECK(rtp_rtcp_module_); |
| |
| ConfigureStream(this, config, true); |
| |
| pacer_thread_checker_.DetachFromThread(); |
| if (rtp_transport_) { |
| // Signal congestion controller this object is ready for OnPacket* |
| // callbacks. |
| rtp_transport_->RegisterPacketFeedbackObserver(this); |
| } |
| } |
| |
| AudioSendStream::~AudioSendStream() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc; |
| RTC_DCHECK(!sending_); |
| if (rtp_transport_) { |
| rtp_transport_->DeRegisterPacketFeedbackObserver(this); |
| channel_send_->ResetSenderCongestionControlObjects(); |
| } |
| } |
| |
| const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return config_; |
| } |
| |
| void AudioSendStream::Reconfigure( |
| const webrtc::AudioSendStream::Config& new_config) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| ConfigureStream(this, new_config, false); |
| } |
| |
| AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds( |
| const std::vector<RtpExtension>& extensions) { |
| ExtensionIds ids; |
| for (const auto& extension : extensions) { |
| if (extension.uri == RtpExtension::kAudioLevelUri) { |
| ids.audio_level = extension.id; |
| } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
| ids.transport_sequence_number = extension.id; |
| } else if (extension.uri == RtpExtension::kMidUri) { |
| ids.mid = extension.id; |
| } |
| } |
| return ids; |
| } |
| |
| void AudioSendStream::ConfigureStream( |
| webrtc::internal::AudioSendStream* stream, |
| const webrtc::AudioSendStream::Config& new_config, |
| bool first_time) { |
| RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: " |
| << new_config.ToString(); |
| UpdateEventLogStreamConfig(stream->event_log_, new_config, |
| first_time ? nullptr : &stream->config_); |
| |
| const auto& channel_send = stream->channel_send_; |
| const auto& old_config = stream->config_; |
| |
| // Configuration parameters which cannot be changed. |
| RTC_DCHECK(first_time || |
| old_config.send_transport == new_config.send_transport); |
| |
| if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) { |
| channel_send->SetLocalSSRC(new_config.rtp.ssrc); |
| if (stream->suspended_rtp_state_) { |
| stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_); |
| } |
| } |
| if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) { |
| channel_send->SetRTCP_CNAME(new_config.rtp.c_name); |
| } |
| |
| // Enable the frame encryptor if a new frame encryptor has been provided. |
| if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) { |
| channel_send->SetFrameEncryptor(new_config.frame_encryptor); |
| } |
| |
| if (first_time || |
| new_config.rtp.extmap_allow_mixed != old_config.rtp.extmap_allow_mixed) { |
| channel_send->SetExtmapAllowMixed(new_config.rtp.extmap_allow_mixed); |
| } |
| |
| const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions); |
| const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions); |
| // Audio level indication |
| if (first_time || new_ids.audio_level != old_ids.audio_level) { |
| channel_send->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0, |
| new_ids.audio_level); |
| } |
| bool transport_seq_num_id_changed = |
| new_ids.transport_sequence_number != old_ids.transport_sequence_number; |
| if (first_time || |
| (transport_seq_num_id_changed && |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC"))) { |
| if (!first_time) { |
| channel_send->ResetSenderCongestionControlObjects(); |
| } |
| |
| RtcpBandwidthObserver* bandwidth_observer = nullptr; |
| bool has_transport_sequence_number = |
| new_ids.transport_sequence_number != 0 && |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC"); |
| if (has_transport_sequence_number) { |
| channel_send->EnableSendTransportSequenceNumber( |
| new_ids.transport_sequence_number); |
| // Probing in application limited region is only used in combination with |
| // send side congestion control, wich depends on feedback packets which |
| // requires transport sequence numbers to be enabled. |
| if (stream->rtp_transport_) { |
| stream->rtp_transport_->EnablePeriodicAlrProbing(true); |
| bandwidth_observer = stream->rtp_transport_->GetBandwidthObserver(); |
| } |
| } |
| if (stream->rtp_transport_) { |
| channel_send->RegisterSenderCongestionControlObjects( |
| stream->rtp_transport_, bandwidth_observer); |
| } |
| } |
| // MID RTP header extension. |
| if ((first_time || new_ids.mid != old_ids.mid || |
| new_config.rtp.mid != old_config.rtp.mid) && |
| new_ids.mid != 0 && !new_config.rtp.mid.empty()) { |
| channel_send->SetMid(new_config.rtp.mid, new_ids.mid); |
| } |
| |
| if (!ReconfigureSendCodec(stream, new_config)) { |
| RTC_LOG(LS_ERROR) << "Failed to set up send codec state."; |
| } |
| |
| if (stream->sending_) { |
| ReconfigureBitrateObserver(stream, new_config); |
| } |
| stream->config_ = new_config; |
| } |
| |
| void AudioSendStream::Start() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| if (sending_) { |
| return; |
| } |
| |
| bool has_transport_sequence_number = |
| FindExtensionIds(config_.rtp.extensions).transport_sequence_number != 0 && |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC"); |
| if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 && |
| !config_.has_dscp && |
| (has_transport_sequence_number || |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") || |
| webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) { |
| // Audio BWE is enabled. |
| rtp_transport_->packet_sender()->SetAccountForAudioPackets(true); |
| rtp_rtcp_module_->SetAsPartOfAllocation(true); |
| ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps, |
| config_.bitrate_priority, |
| has_transport_sequence_number); |
| } else { |
| rtp_rtcp_module_->SetAsPartOfAllocation(false); |
| } |
| channel_send_->StartSend(); |
| sending_ = true; |
| audio_state()->AddSendingStream(this, encoder_sample_rate_hz_, |
| encoder_num_channels_); |
| } |
| |
| void AudioSendStream::Stop() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| if (!sending_) { |
| return; |
| } |
| |
| RemoveBitrateObserver(); |
| channel_send_->StopSend(); |
| sending_ = false; |
| audio_state()->RemoveSendingStream(this); |
| } |
| |
| void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) { |
| RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_); |
| channel_send_->ProcessAndEncodeAudio(std::move(audio_frame)); |
| } |
| |
| bool AudioSendStream::SendTelephoneEvent(int payload_type, |
| int payload_frequency, |
| int event, |
| int duration_ms) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_send_->SetSendTelephoneEventPayloadType(payload_type, |
| payload_frequency) && |
| channel_send_->SendTelephoneEventOutband(event, duration_ms); |
| } |
| |
| void AudioSendStream::SetMuted(bool muted) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_send_->SetInputMute(muted); |
| } |
| |
| webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
| return GetStats(true); |
| } |
| |
| webrtc::AudioSendStream::Stats AudioSendStream::GetStats( |
| bool has_remote_tracks) const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| webrtc::AudioSendStream::Stats stats; |
| stats.local_ssrc = config_.rtp.ssrc; |
| stats.target_bitrate_bps = channel_send_->GetBitrate(); |
| |
| webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics(); |
| stats.bytes_sent = call_stats.bytesSent; |
| stats.packets_sent = call_stats.packetsSent; |
| // RTT isn't known until a RTCP report is received. Until then, VoiceEngine |
| // returns 0 to indicate an error value. |
| if (call_stats.rttMs > 0) { |
| stats.rtt_ms = call_stats.rttMs; |
| } |
| if (config_.send_codec_spec) { |
| const auto& spec = *config_.send_codec_spec; |
| stats.codec_name = spec.format.name; |
| stats.codec_payload_type = spec.payload_type; |
| |
| // Get data from the last remote RTCP report. |
| for (const auto& block : channel_send_->GetRemoteRTCPReportBlocks()) { |
| // Lookup report for send ssrc only. |
| if (block.source_SSRC == stats.local_ssrc) { |
| stats.packets_lost = block.cumulative_num_packets_lost; |
| stats.fraction_lost = Q8ToFloat(block.fraction_lost); |
| stats.ext_seqnum = block.extended_highest_sequence_number; |
| // Convert timestamps to milliseconds. |
| if (spec.format.clockrate_hz / 1000 > 0) { |
| stats.jitter_ms = |
| block.interarrival_jitter / (spec.format.clockrate_hz / 1000); |
| } |
| break; |
| } |
| } |
| } |
| |
| AudioState::Stats input_stats = audio_state()->GetAudioInputStats(); |
| stats.audio_level = input_stats.audio_level; |
| stats.total_input_energy = input_stats.total_energy; |
| stats.total_input_duration = input_stats.total_duration; |
| |
| stats.typing_noise_detected = audio_state()->typing_noise_detected(); |
| stats.ana_statistics = channel_send_->GetANAStatistics(); |
| RTC_DCHECK(audio_state_->audio_processing()); |
| stats.apm_statistics = |
| audio_state_->audio_processing()->GetStatistics(has_remote_tracks); |
| |
| return stats; |
| } |
| |
| void AudioSendStream::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| } |
| |
| bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // TODO(solenberg): Tests call this function on a network thread, libjingle |
| // calls on the worker thread. We should move towards always using a network |
| // thread. Then this check can be enabled. |
| // RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread()); |
| return channel_send_->ReceivedRTCPPacket(packet, length); |
| } |
| |
| uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) { |
| // A send stream may be allocated a bitrate of zero if the allocator decides |
| // to disable it. For now we ignore this decision and keep sending on min |
| // bitrate. |
| if (update.target_bitrate.IsZero()) { |
| update.target_bitrate = DataRate::bps(config_.min_bitrate_bps); |
| } |
| RTC_DCHECK_GE(update.target_bitrate.bps<int>(), config_.min_bitrate_bps); |
| // The bitrate allocator might allocate an higher than max configured bitrate |
| // if there is room, to allow for, as example, extra FEC. Ignore that for now. |
| const DataRate max_bitrate = DataRate::bps(config_.max_bitrate_bps); |
| if (update.target_bitrate > max_bitrate) |
| update.target_bitrate = max_bitrate; |
| |
| channel_send_->OnBitrateAllocation(update); |
| |
| // The amount of audio protection is not exposed by the encoder, hence |
| // always returning 0. |
| return 0; |
| } |
| |
| void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) { |
| RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread()); |
| // Only packets that belong to this stream are of interest. |
| if (ssrc == config_.rtp.ssrc) { |
| rtc::CritScope lock(&packet_loss_tracker_cs_); |
| // TODO(eladalon): This function call could potentially reset the window, |
| // setting both PLR and RPLR to unknown. Consider (during upcoming |
| // refactoring) passing an indication of such an event. |
| packet_loss_tracker_.OnPacketAdded(seq_num, rtc::TimeMillis()); |
| } |
| } |
| |
| void AudioSendStream::OnPacketFeedbackVector( |
| const std::vector<PacketFeedback>& packet_feedback_vector) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| absl::optional<float> plr; |
| absl::optional<float> rplr; |
| { |
| rtc::CritScope lock(&packet_loss_tracker_cs_); |
| packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector); |
| plr = packet_loss_tracker_.GetPacketLossRate(); |
| rplr = packet_loss_tracker_.GetRecoverablePacketLossRate(); |
| } |
| // TODO(eladalon): If R/PLR go back to unknown, no indication is given that |
| // the previously sent value is no longer relevant. This will be taken care |
| // of with some refactoring which is now being done. |
| if (plr) { |
| channel_send_->OnTwccBasedUplinkPacketLossRate(*plr); |
| } |
| if (rplr) { |
| channel_send_->OnRecoverableUplinkPacketLossRate(*rplr); |
| } |
| } |
| |
| void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_send_->SetTransportOverhead(transport_overhead_per_packet); |
| } |
| |
| RtpState AudioSendStream::GetRtpState() const { |
| return rtp_rtcp_module_->GetRtpState(); |
| } |
| |
| const voe::ChannelSendInterface* AudioSendStream::GetChannel() const { |
| return channel_send_.get(); |
| } |
| |
| internal::AudioState* AudioSendStream::audio_state() { |
| internal::AudioState* audio_state = |
| static_cast<internal::AudioState*>(audio_state_.get()); |
| RTC_DCHECK(audio_state); |
| return audio_state; |
| } |
| |
| const internal::AudioState* AudioSendStream::audio_state() const { |
| internal::AudioState* audio_state = |
| static_cast<internal::AudioState*>(audio_state_.get()); |
| RTC_DCHECK(audio_state); |
| return audio_state; |
| } |
| |
| void AudioSendStream::StoreEncoderProperties(int sample_rate_hz, |
| size_t num_channels) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| encoder_sample_rate_hz_ = sample_rate_hz; |
| encoder_num_channels_ = num_channels; |
| if (sending_) { |
| // Update AudioState's information about the stream. |
| audio_state()->AddSendingStream(this, sample_rate_hz, num_channels); |
| } |
| } |
| |
| // Apply current codec settings to a single voe::Channel used for sending. |
| bool AudioSendStream::SetupSendCodec(AudioSendStream* stream, |
| const Config& new_config) { |
| RTC_DCHECK(new_config.send_codec_spec); |
| const auto& spec = *new_config.send_codec_spec; |
| |
| RTC_DCHECK(new_config.encoder_factory); |
| std::unique_ptr<AudioEncoder> encoder = |
| new_config.encoder_factory->MakeAudioEncoder( |
| spec.payload_type, spec.format, new_config.codec_pair_id); |
| |
| if (!encoder) { |
| RTC_DLOG(LS_ERROR) << "Unable to create encoder for " |
| << rtc::ToString(spec.format); |
| return false; |
| } |
| |
| // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is |
| // not enabled, do not update target audio bitrate if we are in |
| // WebRTC-Audio-SendSideBwe-For-Video experiment |
| const bool do_not_update_target_bitrate = |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") && |
| webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") && |
| !FindExtensionIds(new_config.rtp.extensions).transport_sequence_number; |
| // If a bitrate has been specified for the codec, use it over the |
| // codec's default. |
| if (!do_not_update_target_bitrate && spec.target_bitrate_bps) { |
| encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps); |
| } |
| |
| // Enable ANA if configured (currently only used by Opus). |
| if (new_config.audio_network_adaptor_config) { |
| if (encoder->EnableAudioNetworkAdaptor( |
| *new_config.audio_network_adaptor_config, stream->event_log_)) { |
| RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC " |
| << new_config.rtp.ssrc; |
| } else { |
| RTC_NOTREACHED(); |
| } |
| } |
| |
| // Wrap the encoder in a an AudioEncoderCNG, if VAD is enabled. |
| if (spec.cng_payload_type) { |
| AudioEncoderCngConfig cng_config; |
| cng_config.num_channels = encoder->NumChannels(); |
| cng_config.payload_type = *spec.cng_payload_type; |
| cng_config.speech_encoder = std::move(encoder); |
| cng_config.vad_mode = Vad::kVadNormal; |
| encoder = CreateComfortNoiseEncoder(std::move(cng_config)); |
| |
| stream->RegisterCngPayloadType( |
| *spec.cng_payload_type, |
| new_config.send_codec_spec->format.clockrate_hz); |
| } |
| |
| stream->StoreEncoderProperties(encoder->SampleRateHz(), |
| encoder->NumChannels()); |
| stream->channel_send_->SetEncoder(new_config.send_codec_spec->payload_type, |
| std::move(encoder)); |
| return true; |
| } |
| |
| bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream, |
| const Config& new_config) { |
| const auto& old_config = stream->config_; |
| |
| if (!new_config.send_codec_spec) { |
| // We cannot de-configure a send codec. So we will do nothing. |
| // By design, the send codec should have not been configured. |
| RTC_DCHECK(!old_config.send_codec_spec); |
| return true; |
| } |
| |
| if (new_config.send_codec_spec == old_config.send_codec_spec && |
| new_config.audio_network_adaptor_config == |
| old_config.audio_network_adaptor_config) { |
| return true; |
| } |
| |
| // If we have no encoder, or the format or payload type's changed, create a |
| // new encoder. |
| if (!old_config.send_codec_spec || |
| new_config.send_codec_spec->format != |
| old_config.send_codec_spec->format || |
| new_config.send_codec_spec->payload_type != |
| old_config.send_codec_spec->payload_type) { |
| return SetupSendCodec(stream, new_config); |
| } |
| |
| // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is |
| // not enabled, do not update target audio bitrate if we are in |
| // WebRTC-Audio-SendSideBwe-For-Video experiment |
| const bool do_not_update_target_bitrate = |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") && |
| webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") && |
| !FindExtensionIds(new_config.rtp.extensions).transport_sequence_number; |
| |
| const absl::optional<int>& new_target_bitrate_bps = |
| new_config.send_codec_spec->target_bitrate_bps; |
| // If a bitrate has been specified for the codec, use it over the |
| // codec's default. |
| if (!do_not_update_target_bitrate && new_target_bitrate_bps && |
| new_target_bitrate_bps != |
| old_config.send_codec_spec->target_bitrate_bps) { |
| CallEncoder(stream->channel_send_, [&](AudioEncoder* encoder) { |
| encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps); |
| }); |
| } |
| |
| ReconfigureANA(stream, new_config); |
| ReconfigureCNG(stream, new_config); |
| |
| return true; |
| } |
| |
| void AudioSendStream::ReconfigureANA(AudioSendStream* stream, |
| const Config& new_config) { |
| if (new_config.audio_network_adaptor_config == |
| stream->config_.audio_network_adaptor_config) { |
| return; |
| } |
| if (new_config.audio_network_adaptor_config) { |
| CallEncoder(stream->channel_send_, [&](AudioEncoder* encoder) { |
| if (encoder->EnableAudioNetworkAdaptor( |
| *new_config.audio_network_adaptor_config, stream->event_log_)) { |
| RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC " |
| << new_config.rtp.ssrc; |
| } else { |
| RTC_NOTREACHED(); |
| } |
| }); |
| } else { |
| CallEncoder(stream->channel_send_, [&](AudioEncoder* encoder) { |
| encoder->DisableAudioNetworkAdaptor(); |
| }); |
| RTC_DLOG(LS_INFO) << "Audio network adaptor disabled on SSRC " |
| << new_config.rtp.ssrc; |
| } |
| } |
| |
| void AudioSendStream::ReconfigureCNG(AudioSendStream* stream, |
| const Config& new_config) { |
| if (new_config.send_codec_spec->cng_payload_type == |
| stream->config_.send_codec_spec->cng_payload_type) { |
| return; |
| } |
| |
| // Register the CNG payload type if it's been added, don't do anything if CNG |
| // is removed. Payload types must not be redefined. |
| if (new_config.send_codec_spec->cng_payload_type) { |
| stream->RegisterCngPayloadType( |
| *new_config.send_codec_spec->cng_payload_type, |
| new_config.send_codec_spec->format.clockrate_hz); |
| } |
| |
| // Wrap or unwrap the encoder in an AudioEncoderCNG. |
| stream->channel_send_->ModifyEncoder( |
| [&](std::unique_ptr<AudioEncoder>* encoder_ptr) { |
| std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr)); |
| auto sub_encoders = old_encoder->ReclaimContainedEncoders(); |
| if (!sub_encoders.empty()) { |
| // Replace enc with its sub encoder. We need to put the sub |
| // encoder in a temporary first, since otherwise the old value |
| // of enc would be destroyed before the new value got assigned, |
| // which would be bad since the new value is a part of the old |
| // value. |
| auto tmp = std::move(sub_encoders[0]); |
| old_encoder = std::move(tmp); |
| } |
| if (new_config.send_codec_spec->cng_payload_type) { |
| AudioEncoderCngConfig config; |
| config.speech_encoder = std::move(old_encoder); |
| config.num_channels = config.speech_encoder->NumChannels(); |
| config.payload_type = *new_config.send_codec_spec->cng_payload_type; |
| config.vad_mode = Vad::kVadNormal; |
| *encoder_ptr = CreateComfortNoiseEncoder(std::move(config)); |
| } else { |
| *encoder_ptr = std::move(old_encoder); |
| } |
| }); |
| } |
| |
| void AudioSendStream::ReconfigureBitrateObserver( |
| AudioSendStream* stream, |
| const webrtc::AudioSendStream::Config& new_config) { |
| // Since the Config's default is for both of these to be -1, this test will |
| // allow us to configure the bitrate observer if the new config has bitrate |
| // limits set, but would only have us call RemoveBitrateObserver if we were |
| // previously configured with bitrate limits. |
| int new_transport_seq_num_id = |
| FindExtensionIds(new_config.rtp.extensions).transport_sequence_number; |
| if (stream->config_.min_bitrate_bps == new_config.min_bitrate_bps && |
| stream->config_.max_bitrate_bps == new_config.max_bitrate_bps && |
| stream->config_.bitrate_priority == new_config.bitrate_priority && |
| (FindExtensionIds(stream->config_.rtp.extensions) |
| .transport_sequence_number == new_transport_seq_num_id || |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) { |
| return; |
| } |
| |
| bool has_transport_sequence_number = new_transport_seq_num_id != 0; |
| if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1 && |
| !new_config.has_dscp && |
| (has_transport_sequence_number || |
| !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) { |
| stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true); |
| stream->ConfigureBitrateObserver( |
| new_config.min_bitrate_bps, new_config.max_bitrate_bps, |
| new_config.bitrate_priority, has_transport_sequence_number); |
| stream->rtp_rtcp_module_->SetAsPartOfAllocation(true); |
| } else { |
| stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false); |
| stream->RemoveBitrateObserver(); |
| stream->rtp_rtcp_module_->SetAsPartOfAllocation(false); |
| } |
| } |
| |
| void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps, |
| int max_bitrate_bps, |
| double bitrate_priority, |
| bool has_packet_feedback) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps); |
| rtc::Event thread_sync_event; |
| worker_queue_->PostTask([&] { |
| // We may get a callback immediately as the observer is registered, so make |
| // sure the bitrate limits in config_ are up-to-date. |
| config_.min_bitrate_bps = min_bitrate_bps; |
| config_.max_bitrate_bps = max_bitrate_bps; |
| config_.bitrate_priority = bitrate_priority; |
| // This either updates the current observer or adds a new observer. |
| bitrate_allocator_->AddObserver( |
| this, MediaStreamAllocationConfig{ |
| static_cast<uint32_t>(min_bitrate_bps), |
| static_cast<uint32_t>(max_bitrate_bps), 0, true, |
| config_.track_id, bitrate_priority, has_packet_feedback}); |
| thread_sync_event.Set(); |
| }); |
| thread_sync_event.Wait(rtc::Event::kForever); |
| } |
| |
| void AudioSendStream::RemoveBitrateObserver() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| rtc::Event thread_sync_event; |
| worker_queue_->PostTask([this, &thread_sync_event] { |
| bitrate_allocator_->RemoveObserver(this); |
| thread_sync_event.Set(); |
| }); |
| thread_sync_event.Wait(rtc::Event::kForever); |
| } |
| |
| void AudioSendStream::RegisterCngPayloadType(int payload_type, |
| int clockrate_hz) { |
| const CodecInst codec = {payload_type, "CN", clockrate_hz, 0, 1, 0}; |
| if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) { |
| rtp_rtcp_module_->DeRegisterSendPayload(codec.pltype); |
| if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) { |
| RTC_DLOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to " |
| "RTP/RTCP module"; |
| } |
| } |
| } |
| } // namespace internal |
| } // namespace webrtc |
