Move talk/app/webrtc to webrtc/api

talk/app/webrtc/remoteaudiosource.h

-/*
- * 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 "webrtc/audio/audio_sink.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/media/base/audiorenderer.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_