blob: 89af4db7144334cb55aae19438751900d8979148 [file] [log] [blame]
/*
* 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 <string>
#include "absl/types/optional.h"
#include "api/call/audio_sink.h"
#include "api/media_stream_interface.h"
#include "api/notifier.h"
#include "media/base/media_channel.h"
#include "rtc_base/message_handler.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_message.h"
namespace rtc {
struct Message;
class Thread;
} // namespace rtc
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>,
rtc::MessageHandler {
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(
rtc::Thread* worker_thread,
OnAudioChannelGoneAction on_audio_channel_gone_action);
// Register and unregister remote audio source with the underlying media
// engine.
void Start(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc);
void Stop(cricket::VoiceMediaChannel* media_channel,
absl::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();
void OnMessage(rtc::Message* msg) override;
rtc::Thread* const main_thread_;
rtc::Thread* 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_