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/*
* 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 WEBRTC_API_REMOTEAUDIOSOURCE_H_
#define WEBRTC_API_REMOTEAUDIOSOURCE_H_
#include <list>
#include <string>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/api/notifier.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/pc/channel.h"
namespace rtc {
struct Message;
class Thread;
} // namespace rtc
namespace webrtc {
// This class implements the audio source used by the remote audio track.
class RemoteAudioSource : public Notifier<AudioSourceInterface> {
public:
// Creates an instance of RemoteAudioSource.
static rtc::scoped_refptr<RemoteAudioSource> Create(
uint32_t ssrc,
cricket::VoiceChannel* channel);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
bool remote() const override;
void AddSink(AudioTrackSinkInterface* sink) override;
void RemoveSink(AudioTrackSinkInterface* sink) override;
protected:
RemoteAudioSource();
~RemoteAudioSource() override;
// Post construction initialize where we can do things like save a reference
// to ourselves (need to be fully constructed).
void Initialize(uint32_t ssrc, cricket::VoiceChannel* channel);
private:
typedef std::list<AudioObserver*> AudioObserverList;
// AudioSourceInterface implementation.
void SetVolume(double volume) override;
void RegisterAudioObserver(AudioObserver* observer) override;
void UnregisterAudioObserver(AudioObserver* observer) override;
class Sink;
void OnData(const AudioSinkInterface::Data& audio);
void OnAudioChannelGone();
class MessageHandler;
void OnMessage(rtc::Message* msg);
AudioObserverList audio_observers_;
rtc::CriticalSection sink_lock_;
std::list<AudioTrackSinkInterface*> sinks_;
rtc::Thread* const main_thread_;
SourceState state_;
};
} // namespace webrtc
#endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_
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