blob: 4d87331f91dfb28a2fefe1d0190916aa267a73fc [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.
#include <list>
#include <string>
#include "api/call/audio_sink.h"
#include "api/notifier.h"
#include "pc/channel.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/messagehandler.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 {
explicit RemoteAudioSource(rtc::Thread* worker_thread);
// Register and unregister remote audio source with the underlying media
// engine.
void Start(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
void Stop(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
// 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;
~RemoteAudioSource() override;
// 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_;
std::list<AudioObserver*> audio_observers_;
rtc::CriticalSection sink_lock_;
std::list<AudioTrackSinkInterface*> sinks_;
SourceState state_;
} // namespace webrtc