| /* |
| * Copyright 2019 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 "pc/audio_rtp_receiver.h" |
| |
| #include <stddef.h> |
| |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/sequence_checker.h" |
| #include "pc/audio_track.h" |
| #include "pc/media_stream_track_proxy.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| AudioRtpReceiver::AudioRtpReceiver( |
| rtc::Thread* worker_thread, |
| std::string receiver_id, |
| std::vector<std::string> stream_ids, |
| bool is_unified_plan, |
| cricket::VoiceMediaReceiveChannelInterface* voice_channel /*= nullptr*/) |
| : AudioRtpReceiver(worker_thread, |
| receiver_id, |
| CreateStreamsFromIds(std::move(stream_ids)), |
| is_unified_plan, |
| voice_channel) {} |
| |
| AudioRtpReceiver::AudioRtpReceiver( |
| rtc::Thread* worker_thread, |
| const std::string& receiver_id, |
| const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams, |
| bool is_unified_plan, |
| cricket::VoiceMediaReceiveChannelInterface* voice_channel /*= nullptr*/) |
| : worker_thread_(worker_thread), |
| id_(receiver_id), |
| source_(rtc::make_ref_counted<RemoteAudioSource>( |
| worker_thread, |
| is_unified_plan |
| ? RemoteAudioSource::OnAudioChannelGoneAction::kSurvive |
| : RemoteAudioSource::OnAudioChannelGoneAction::kEnd)), |
| track_(AudioTrackProxyWithInternal<AudioTrack>::Create( |
| rtc::Thread::Current(), |
| AudioTrack::Create(receiver_id, source_))), |
| media_channel_(voice_channel), |
| cached_track_enabled_(track_->internal()->enabled()), |
| attachment_id_(GenerateUniqueId()), |
| worker_thread_safety_(PendingTaskSafetyFlag::CreateDetachedInactive()) { |
| RTC_DCHECK(worker_thread_); |
| RTC_DCHECK(track_->GetSource()->remote()); |
| track_->RegisterObserver(this); |
| track_->GetSource()->RegisterAudioObserver(this); |
| SetStreams(streams); |
| } |
| |
| AudioRtpReceiver::~AudioRtpReceiver() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RTC_DCHECK(!media_channel_); |
| |
| track_->GetSource()->UnregisterAudioObserver(this); |
| track_->UnregisterObserver(this); |
| } |
| |
| void AudioRtpReceiver::OnChanged() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| const bool enabled = track_->internal()->enabled(); |
| if (cached_track_enabled_ == enabled) |
| return; |
| cached_track_enabled_ = enabled; |
| worker_thread_->PostTask(SafeTask(worker_thread_safety_, [this, enabled]() { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| Reconfigure(enabled); |
| })); |
| } |
| |
| void AudioRtpReceiver::SetOutputVolume_w(double volume) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| RTC_DCHECK_GE(volume, 0.0); |
| RTC_DCHECK_LE(volume, 10.0); |
| |
| if (!media_channel_) |
| return; |
| |
| signaled_ssrc_ ? media_channel_->SetOutputVolume(*signaled_ssrc_, volume) |
| : media_channel_->SetDefaultOutputVolume(volume); |
| } |
| |
| void AudioRtpReceiver::OnSetVolume(double volume) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RTC_DCHECK_GE(volume, 0); |
| RTC_DCHECK_LE(volume, 10); |
| |
| bool track_enabled = track_->internal()->enabled(); |
| worker_thread_->BlockingCall([&]() { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| // Update the cached_volume_ even when stopped, to allow clients to set |
| // the volume before starting/restarting, eg see crbug.com/1272566. |
| cached_volume_ = volume; |
| // When the track is disabled, the volume of the source, which is the |
| // corresponding WebRtc Voice Engine channel will be 0. So we do not |
| // allow setting the volume to the source when the track is disabled. |
| if (track_enabled) |
| SetOutputVolume_w(volume); |
| }); |
| } |
| |
| rtc::scoped_refptr<DtlsTransportInterface> AudioRtpReceiver::dtls_transport() |
| const { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| return dtls_transport_; |
| } |
| |
| std::vector<std::string> AudioRtpReceiver::stream_ids() const { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| std::vector<std::string> stream_ids(streams_.size()); |
| for (size_t i = 0; i < streams_.size(); ++i) |
| stream_ids[i] = streams_[i]->id(); |
| return stream_ids; |
| } |
| |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>> |
| AudioRtpReceiver::streams() const { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| return streams_; |
| } |
| |
| RtpParameters AudioRtpReceiver::GetParameters() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (!media_channel_) |
| return RtpParameters(); |
| auto current_ssrc = ssrc(); |
| return current_ssrc.has_value() |
| ? media_channel_->GetRtpReceiverParameters(current_ssrc.value()) |
| : media_channel_->GetDefaultRtpReceiveParameters(); |
| } |
| |
| void AudioRtpReceiver::SetFrameDecryptor( |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| frame_decryptor_ = std::move(frame_decryptor); |
| // Special Case: Set the frame decryptor to any value on any existing channel. |
| if (media_channel_ && signaled_ssrc_) { |
| media_channel_->SetFrameDecryptor(*signaled_ssrc_, frame_decryptor_); |
| } |
| } |
| |
| rtc::scoped_refptr<FrameDecryptorInterface> |
| AudioRtpReceiver::GetFrameDecryptor() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| return frame_decryptor_; |
| } |
| |
| void AudioRtpReceiver::Stop() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| source_->SetState(MediaSourceInterface::kEnded); |
| track_->internal()->set_ended(); |
| } |
| |
| void AudioRtpReceiver::RestartMediaChannel(std::optional<uint32_t> ssrc) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| bool enabled = track_->internal()->enabled(); |
| MediaSourceInterface::SourceState state = source_->state(); |
| worker_thread_->BlockingCall([&]() { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| RestartMediaChannel_w(std::move(ssrc), enabled, state); |
| }); |
| source_->SetState(MediaSourceInterface::kLive); |
| } |
| |
| void AudioRtpReceiver::RestartMediaChannel_w( |
| std::optional<uint32_t> ssrc, |
| bool track_enabled, |
| MediaSourceInterface::SourceState state) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (!media_channel_) |
| return; // Can't restart. |
| |
| // Make sure the safety flag is marked as `alive` for cases where the media |
| // channel was provided via the ctor and not an explicit call to |
| // SetMediaChannel. |
| worker_thread_safety_->SetAlive(); |
| |
| if (state != MediaSourceInterface::kInitializing) { |
| if (signaled_ssrc_ == ssrc) |
| return; |
| source_->Stop(media_channel_, signaled_ssrc_); |
| } |
| |
| signaled_ssrc_ = std::move(ssrc); |
| source_->Start(media_channel_, signaled_ssrc_); |
| if (signaled_ssrc_) { |
| media_channel_->SetBaseMinimumPlayoutDelayMs(*signaled_ssrc_, |
| delay_.GetMs()); |
| } |
| |
| Reconfigure(track_enabled); |
| } |
| |
| void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RestartMediaChannel(ssrc); |
| } |
| |
| void AudioRtpReceiver::SetupUnsignaledMediaChannel() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RestartMediaChannel(std::nullopt); |
| } |
| |
| std::optional<uint32_t> AudioRtpReceiver::ssrc() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (!signaled_ssrc_.has_value() && media_channel_) { |
| return media_channel_->GetUnsignaledSsrc(); |
| } |
| return signaled_ssrc_; |
| } |
| |
| void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| SetStreams(CreateStreamsFromIds(std::move(stream_ids))); |
| } |
| |
| void AudioRtpReceiver::set_transport( |
| rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| dtls_transport_ = std::move(dtls_transport); |
| } |
| |
| void AudioRtpReceiver::SetStreams( |
| const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| // Remove remote track from any streams that are going away. |
| for (const auto& existing_stream : streams_) { |
| bool removed = true; |
| for (const auto& stream : streams) { |
| if (existing_stream->id() == stream->id()) { |
| RTC_DCHECK_EQ(existing_stream.get(), stream.get()); |
| removed = false; |
| break; |
| } |
| } |
| if (removed) { |
| existing_stream->RemoveTrack(audio_track()); |
| } |
| } |
| // Add remote track to any streams that are new. |
| for (const auto& stream : streams) { |
| bool added = true; |
| for (const auto& existing_stream : streams_) { |
| if (stream->id() == existing_stream->id()) { |
| RTC_DCHECK_EQ(stream.get(), existing_stream.get()); |
| added = false; |
| break; |
| } |
| } |
| if (added) { |
| stream->AddTrack(audio_track()); |
| } |
| } |
| streams_ = streams; |
| } |
| |
| std::vector<RtpSource> AudioRtpReceiver::GetSources() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| auto current_ssrc = ssrc(); |
| if (!media_channel_ || !current_ssrc.has_value()) { |
| return {}; |
| } |
| return media_channel_->GetSources(current_ssrc.value()); |
| } |
| |
| void AudioRtpReceiver::SetFrameTransformer( |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (media_channel_) { |
| media_channel_->SetDepacketizerToDecoderFrameTransformer( |
| signaled_ssrc_.value_or(0), frame_transformer); |
| } |
| frame_transformer_ = std::move(frame_transformer); |
| } |
| |
| void AudioRtpReceiver::Reconfigure(bool track_enabled) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| RTC_DCHECK(media_channel_); |
| |
| SetOutputVolume_w(track_enabled ? cached_volume_ : 0); |
| |
| if (signaled_ssrc_ && frame_decryptor_) { |
| // Reattach the frame decryptor if we were reconfigured. |
| media_channel_->SetFrameDecryptor(*signaled_ssrc_, frame_decryptor_); |
| } |
| |
| if (frame_transformer_) { |
| media_channel_->SetDepacketizerToDecoderFrameTransformer( |
| signaled_ssrc_.value_or(0), frame_transformer_); |
| } |
| } |
| |
| void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| observer_ = observer; |
| // Deliver any notifications the observer may have missed by being set late. |
| if (received_first_packet_ && observer_) { |
| observer_->OnFirstPacketReceived(media_type()); |
| } |
| } |
| |
| void AudioRtpReceiver::SetJitterBufferMinimumDelay( |
| std::optional<double> delay_seconds) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| delay_.Set(delay_seconds); |
| if (media_channel_ && signaled_ssrc_) |
| media_channel_->SetBaseMinimumPlayoutDelayMs(*signaled_ssrc_, |
| delay_.GetMs()); |
| } |
| |
| void AudioRtpReceiver::SetMediaChannel( |
| cricket::MediaReceiveChannelInterface* media_channel) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| RTC_DCHECK(media_channel == nullptr || |
| media_channel->media_type() == media_type()); |
| if (!media_channel && media_channel_) |
| SetOutputVolume_w(0.0); |
| |
| media_channel ? worker_thread_safety_->SetAlive() |
| : worker_thread_safety_->SetNotAlive(); |
| media_channel_ = |
| static_cast<cricket::VoiceMediaReceiveChannelInterface*>(media_channel); |
| } |
| |
| void AudioRtpReceiver::NotifyFirstPacketReceived() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| if (observer_) { |
| observer_->OnFirstPacketReceived(media_type()); |
| } |
| received_first_packet_ = true; |
| } |
| |
| } // namespace webrtc |