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/*
* libjingle
* Copyright 2014 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
#define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
#include <list>
#include <string>
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/notifier.h"
#include "talk/media/base/audiorenderer.h"
#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/criticalsection.h"
namespace rtc {
struct Message;
class Thread;
} // namespace rtc
namespace webrtc {
class AudioProviderInterface;
// 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,
AudioProviderInterface* provider);
// 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, AudioProviderInterface* provider);
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 OnAudioProviderGone();
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 // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
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