/* * Copyright 2014 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/remote_audio_source.h" #include #include #include #include #include "absl/algorithm/container.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/strings/string_format.h" #include "rtc_base/trace_event.h" namespace webrtc { // This proxy is passed to the underlying media engine to receive audio data as // they come in. The data will then be passed back up to the RemoteAudioSource // which will fan it out to all the sinks that have been added to it. class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface { public: explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) { RTC_DCHECK(source); } AudioDataProxy() = delete; AudioDataProxy(const AudioDataProxy&) = delete; AudioDataProxy& operator=(const AudioDataProxy&) = delete; ~AudioDataProxy() override { source_->OnAudioChannelGone(); } // AudioSinkInterface implementation. void OnData(const AudioSinkInterface::Data& audio) override { source_->OnData(audio); } private: const rtc::scoped_refptr source_; }; RemoteAudioSource::RemoteAudioSource( TaskQueueBase* worker_thread, OnAudioChannelGoneAction on_audio_channel_gone_action) : main_thread_(TaskQueueBase::Current()), worker_thread_(worker_thread), on_audio_channel_gone_action_(on_audio_channel_gone_action), state_(MediaSourceInterface::kInitializing) { RTC_DCHECK(main_thread_); RTC_DCHECK(worker_thread_); } RemoteAudioSource::~RemoteAudioSource() { RTC_DCHECK(audio_observers_.empty()); if (!sinks_.empty()) { RTC_LOG(LS_WARNING) << "RemoteAudioSource destroyed while sinks_ is non-empty."; } } void RemoteAudioSource::Start( cricket::VoiceMediaReceiveChannelInterface* media_channel, absl::optional ssrc) { RTC_DCHECK_RUN_ON(worker_thread_); // Register for callbacks immediately before AddSink so that we always get // notified when a channel goes out of scope (signaled when "AudioDataProxy" // is destroyed). RTC_DCHECK(media_channel); ssrc ? media_channel->SetRawAudioSink(*ssrc, std::make_unique(this)) : media_channel->SetDefaultRawAudioSink( std::make_unique(this)); } void RemoteAudioSource::Stop( cricket::VoiceMediaReceiveChannelInterface* media_channel, absl::optional ssrc) { RTC_DCHECK_RUN_ON(worker_thread_); RTC_DCHECK(media_channel); ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr) : media_channel->SetDefaultRawAudioSink(nullptr); } void RemoteAudioSource::SetState(SourceState new_state) { RTC_DCHECK_RUN_ON(main_thread_); if (state_ != new_state) { state_ = new_state; FireOnChanged(); } } MediaSourceInterface::SourceState RemoteAudioSource::state() const { RTC_DCHECK_RUN_ON(main_thread_); return state_; } bool RemoteAudioSource::remote() const { RTC_DCHECK_RUN_ON(main_thread_); return true; } void RemoteAudioSource::SetVolume(double volume) { RTC_DCHECK_GE(volume, 0); RTC_DCHECK_LE(volume, 10); RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__, volume); for (auto* observer : audio_observers_) { observer->OnSetVolume(volume); } } void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { RTC_DCHECK(observer != NULL); RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer)); audio_observers_.push_back(observer); } void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { RTC_DCHECK(observer != NULL); audio_observers_.remove(observer); } void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { RTC_DCHECK_RUN_ON(main_thread_); RTC_DCHECK(sink); MutexLock lock(&sink_lock_); RTC_DCHECK(!absl::c_linear_search(sinks_, sink)); sinks_.push_back(sink); } void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { RTC_DCHECK_RUN_ON(main_thread_); RTC_DCHECK(sink); MutexLock lock(&sink_lock_); sinks_.remove(sink); } void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { // Called on the externally-owned audio callback thread, via/from webrtc. TRACE_EVENT0("webrtc", "RemoteAudioSource::OnData"); MutexLock lock(&sink_lock_); for (auto* sink : sinks_) { // When peerconnection acts as an audio source, it should not provide // absolute capture timestamp. sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, audio.samples_per_channel, /*absolute_capture_timestamp_ms=*/absl::nullopt); } } void RemoteAudioSource::OnAudioChannelGone() { if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) { return; } // Called when the audio channel is deleted. It may be the worker thread or // may be a different task queue. // This object needs to live long enough for the cleanup logic in the posted // task to run, so take a reference to it. Sometimes the task may not be // processed (because the task queue was destroyed shortly after this call), // but that is fine because the task queue destructor will take care of // destroying task which will release the reference on RemoteAudioSource. rtc::scoped_refptr thiz(this); main_thread_->PostTask([thiz = std::move(thiz)] { thiz->sinks_.clear(); thiz->SetState(MediaSourceInterface::kEnded); }); } } // namespace webrtc