Adding the ability to create an RtpSender without a track.

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 82 matching lines @@
 class MediaStreamTrackInterface : public rtc::RefCountInterface,
                                   public NotifierInterface {
  public:
   enum TrackState {
     kInitializing,  // Track is beeing negotiated.
     kLive = 1,  // Track alive
     kEnded = 2,  // Track have ended
     kFailed = 3,  // Track negotiation failed.
   };
 
+  static const char kAudioTrackKind[];
+  static const char kVideoTrackKind[];
+
   virtual std::string kind() const = 0;
   virtual std::string id() const = 0;
   virtual bool enabled() const = 0;
   virtual TrackState state() const = 0;
   virtual bool set_enabled(bool enable) = 0;
   // These methods should be called by implementation only.
   virtual bool set_state(TrackState new_state) = 0;
 
  protected:
   virtual ~MediaStreamTrackInterface() {}
@@ skipping 151 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_