Support for remote audio into tracks

talk/app/webrtc/remoteaudiosource.h

Index: talk/app/webrtc/remoteaudiosource.h
diff --git a/talk/app/webrtc/remoteaudiosource.h b/talk/app/webrtc/remoteaudiosource.h
index e49aca56842091e3c29e17c6558c7c727ddc9e11..b0c65aa5221bcb7c13a099584ee06e733754a69f 100644
--- a/talk/app/webrtc/remoteaudiosource.h
+++ b/talk/app/webrtc/remoteaudiosource.h
@@ -29,23 +29,37 @@
 #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/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> {
+class RemoteAudioSource : public Notifier<AudioSourceInterface>,
+                          public cricket::AudioRenderer::Sink {
  public:
   // Creates an instance of RemoteAudioSource.
-  static rtc::scoped_refptr<RemoteAudioSource> Create();
+  static rtc::scoped_refptr<RemoteAudioSource> Create(
+      uint32_t ssrc,
+      AudioProviderInterface* provider);
+
+  void AddSink(AudioTrackSinkInterface* sink);
+  void RemoveSink(AudioTrackSinkInterface* sink);
 
  protected:
-  RemoteAudioSource();
-  virtual ~RemoteAudioSource();
+  RemoteAudioSource(uint32_t ssrc, AudioProviderInterface* provider);
+  ~RemoteAudioSource() override;
 
  private:
   typedef std::list<AudioObserver*> AudioObserverList;
@@ -58,7 +72,24 @@ class RemoteAudioSource : public Notifier<AudioSourceInterface> {
   void RegisterAudioObserver(AudioObserver* observer) override;
   void UnregisterAudioObserver(AudioObserver* observer) override;
 
+  // cricket::AudioRenderer::Sink implementation.
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              int number_of_channels,
+              size_t number_of_frames) override;
+  void OnClose() override;
+
+  class MessageHandler;
+  void OnMessage(rtc::Message* msg);
+
   AudioObserverList audio_observers_;
+  const uint32_t ssrc_;
+  AudioProviderInterface* provider_;
+  rtc::CriticalSection sink_lock_;
+  std::list<AudioTrackSinkInterface*> sinks_;
+  rtc::Thread* const main_thread_;
+  SourceState state_;
 };
 
 }  // namespace webrtc