Support for remote audio into tracks

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,
@@ skipping 11 matching lines @@
  * 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 {
 
-using webrtc::AudioSourceInterface;
+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();
+  static rtc::scoped_refptr<RemoteAudioSource> Create(
+      uint32_t ssrc,
+      AudioProviderInterface* provider);
+
+  // MediaSourceInterface implementation.
+  MediaSourceInterface::SourceState state() const override;
+
+  void AddSink(AudioTrackSinkInterface* sink);
+  void RemoveSink(AudioTrackSinkInterface* sink);
 
  protected:
   RemoteAudioSource();
-  virtual ~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;
 
-  // MediaSourceInterface implementation.
-  MediaSourceInterface::SourceState state() const override;
-
   // 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_