1 files changed, 36 insertions, 6 deletions

diff --git a/talk/app/webrtc/remoteaudiosource.h b/talk/app/webrtc/remoteaudiosource.h
index e49aca5684..d648ba4604 100644
--- a/talk/app/webrtc/remoteaudiosource.h
+++ b/talk/app/webrtc/remoteaudiosource.h
@@ -29,36 +29,66 @@
 #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;
+  bool remote() const override;
+
+  void AddSink(AudioTrackSinkInterface* sink) override;
+  void RemoveSink(AudioTrackSinkInterface* sink) override;
 
  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