Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sour…

talk/app/webrtc/mediastreaminterface.h

 /*
  * libjingle
  * Copyright 2012 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 53 matching lines @@
 class NotifierInterface {
  public:
   virtual void RegisterObserver(ObserverInterface* observer) = 0;
   virtual void UnregisterObserver(ObserverInterface* observer) = 0;
 
   virtual ~NotifierInterface() {}
 };
 
 // Base class for sources. A MediaStreamTrack have an underlying source that
 // provide media. A source can be shared with multiple tracks.
 // TODO(perkj): Implement sources for local and remote audio tracks and

[perkj_webrtc, 2015/12/15 10:15:31]

Remove this todo please.

[tommi, 2015/12/15 11:00:54]

Done.

 // remote video tracks.
 class MediaSourceInterface : public rtc::RefCountInterface,
                              public NotifierInterface {
  public:
   enum SourceState {
     kInitializing,
     kLive,
     kEnded,
     kMuted
   };
 
   virtual SourceState state() const = 0;
 
+  virtual bool remote() const = 0;
+
  protected:
   virtual ~MediaSourceInterface() {}
 };
 
 // Information about a track.
 class MediaStreamTrackInterface : public rtc::RefCountInterface,
                                   public NotifierInterface {
  public:
   enum TrackState {
     kInitializing,  // Track is beeing negotiated.
@@ skipping 47 matching lines @@
   virtual void AddRenderer(VideoRendererInterface* renderer) = 0;
   // Deregister a renderer.
   virtual void RemoveRenderer(VideoRendererInterface* renderer) = 0;
 
   virtual VideoSourceInterface* GetSource() const = 0;
 
  protected:
   virtual ~VideoTrackInterface() {}
 };
 
+// Interface for receiving audio data from a AudioTrack.
+class AudioTrackSinkInterface {
+ public:
+  virtual void OnData(const void* audio_data,
+                      int bits_per_sample,
+                      int sample_rate,
+                      int number_of_channels,
+                      size_t number_of_frames) = 0;
+
+ protected:
+  virtual ~AudioTrackSinkInterface() {}
+};
+
 // AudioSourceInterface is a reference counted source used for AudioTracks.
 // The same source can be used in multiple AudioTracks.
 class AudioSourceInterface : public MediaSourceInterface {
  public:
   class AudioObserver {
    public:
     virtual void OnSetVolume(double volume) = 0;
 
    protected:
     virtual ~AudioObserver() {}
   };
 
   // TODO(xians): Makes all the interface pure virtual after Chrome has their
   // implementations.
   // Sets the volume to the source. |volume| is in  the range of [0, 10].
   // TODO(tommi): This method should be on the track and ideally volume should
   // be applied in the track in a way that does not affect clones of the track.
   virtual void SetVolume(double volume) {}
 
   // Registers/unregisters observer to the audio source.
   virtual void RegisterAudioObserver(AudioObserver* observer) {}
   virtual void UnregisterAudioObserver(AudioObserver* observer) {}
-};
 
-// Interface for receiving audio data from a AudioTrack.
-class AudioTrackSinkInterface {
- public:
-  virtual void OnData(const void* audio_data,
-                      int bits_per_sample,
-                      int sample_rate,
-                      int number_of_channels,
-                      size_t number_of_frames) = 0;
- protected:
-  virtual ~AudioTrackSinkInterface() {}
+  // TODO(tommi): Make pure virtual.
+  virtual void AddSink(AudioTrackSinkInterface* sink) {}
+  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
 };
 
 // Interface of the audio processor used by the audio track to collect
 // statistics.
 class AudioProcessorInterface : public rtc::RefCountInterface {
  public:
   struct AudioProcessorStats {
     AudioProcessorStats() : typing_noise_detected(false),
                             echo_return_loss(0),
                             echo_return_loss_enhancement(0),
@@ skipping 70 matching lines @@
   virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
   virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
 
  protected:
   virtual ~MediaStreamInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_