Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/test/fakemediastreamsignaling.h

 /*
  * libjingle
  * Copyright 2013 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 75 matching lines @@
 
   // Implements MediaStreamSignalingObserver.
   virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {
   }
   virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {
   }
   virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {
   }
   virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
                                     webrtc::AudioTrackInterface* audio_track,
-                                    uint32 ssrc) {
-  }
+                                    uint32 ssrc,
+                                    const std::string& mid) {}
   virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
                                     webrtc::VideoTrackInterface* video_track,
-                                    uint32 ssrc) {
-  }
+                                    uint32 ssrc,
+                                    const std::string& mid) {}
   virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
                                      webrtc::AudioTrackInterface* audio_track,
-                                     uint32 ssrc) {
-  }
+                                     uint32 ssrc,
+                                     const std::string& mid) {}
 
   virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
                                      webrtc::VideoTrackInterface* video_track,
-                                     uint32 ssrc) {
-  }
+                                     uint32 ssrc,
+                                     const std::string& mid) {}
 
   virtual void OnRemoveRemoteAudioTrack(
       webrtc::MediaStreamInterface* stream,
       webrtc::AudioTrackInterface* audio_track) {
   }
 
   virtual void OnRemoveRemoteVideoTrack(
       webrtc::MediaStreamInterface* stream,
       webrtc::VideoTrackInterface* video_track) {
   }
@@ skipping 27 matching lines @@
     if (!video_track_id.empty()) {
       rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
           webrtc::VideoTrack::Create(video_track_id, NULL));
       stream->AddTrack(video_track);
     }
     return stream;
   }
 };
 
 #endif  // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_