| /* | |
| * 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. | |
| */ | |
| #ifndef PC_REMOTE_AUDIO_SOURCE_H_ | |
| #define PC_REMOTE_AUDIO_SOURCE_H_ | |
| #include <stdint.h> | |
| #include <list> | |
| #include <optional> | |
| #include "api/call/audio_sink.h" | |
| #include "api/media_stream_interface.h" | |
| #include "api/notifier.h" | |
| #include "api/task_queue/task_queue_base.h" | |
| #include "media/base/media_channel.h" | |
| #include "rtc_base/synchronization/mutex.h" | |
| namespace webrtc { | |
| // This class implements the audio source used by the remote audio track. | |
| // This class works by configuring itself as a sink with the underlying media | |
| // engine, then when receiving data will fan out to all added sinks. | |
| class RemoteAudioSource : public Notifier<AudioSourceInterface> { | |
| public: | |
| // In Unified Plan, receivers map to m= sections and their tracks and sources | |
| // survive SSRCs being reconfigured. The life cycle of the remote audio source | |
| // is associated with the life cycle of the m= section, and thus even if an | |
| // audio channel is destroyed the RemoteAudioSource should kSurvive. | |
| // | |
| // In Plan B however, remote audio sources map 1:1 with an SSRCs and if an | |
| // audio channel is destroyed, the RemoteAudioSource should kEnd. | |
| enum class OnAudioChannelGoneAction { | |
| kSurvive, | |
| kEnd, | |
| }; | |
| explicit RemoteAudioSource( | |
| TaskQueueBase* worker_thread, | |
| OnAudioChannelGoneAction on_audio_channel_gone_action); | |
| // Register and unregister remote audio source with the underlying media | |
| // engine. | |
| void Start(VoiceMediaReceiveChannelInterface* media_channel, | |
| std::optional<uint32_t> ssrc); | |
| void Stop(VoiceMediaReceiveChannelInterface* media_channel, | |
| std::optional<uint32_t> ssrc); | |
| void SetState(SourceState new_state); | |
| // MediaSourceInterface implementation. | |
| MediaSourceInterface::SourceState state() const override; | |
| bool remote() const override; | |
| // AudioSourceInterface implementation. | |
| void SetVolume(double volume) override; | |
| void RegisterAudioObserver(AudioObserver* observer) override; | |
| void UnregisterAudioObserver(AudioObserver* observer) override; | |
| void AddSink(AudioTrackSinkInterface* sink) override; | |
| void RemoveSink(AudioTrackSinkInterface* sink) override; | |
| protected: | |
| ~RemoteAudioSource() override; | |
| private: | |
| // These are callbacks from the media engine. | |
| class AudioDataProxy; | |
| void OnData(const AudioSinkInterface::Data& audio); | |
| void OnAudioChannelGone(); | |
| TaskQueueBase* const main_thread_; | |
| TaskQueueBase* const worker_thread_; | |
| const OnAudioChannelGoneAction on_audio_channel_gone_action_; | |
| std::list<AudioObserver*> audio_observers_; | |
| Mutex sink_lock_; | |
| std::list<AudioTrackSinkInterface*> sinks_; | |
| SourceState state_; | |
| }; | |
| } // namespace webrtc | |
| #endif // PC_REMOTE_AUDIO_SOURCE_H_ |