Move talk/media to webrtc/media

talk/app/webrtc/rtpsenderinterface.h

 /*
  * libjingle
  * Copyright 2015 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 15 matching lines @@
  */
 
 // This file contains interfaces for RtpSenders
 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
 #ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
 #define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
 
 #include <string>
 
+#include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/proxy.h"
-#include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 class RtpSenderInterface : public rtc::RefCountInterface {
  public:
   // Returns true if successful in setting the track.
   // Fails if an audio track is set on a video RtpSender, or vice-versa.
@@ skipping 33 matching lines @@
 PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_METHOD1(void, set_stream_id, const std::string&)
 PROXY_CONSTMETHOD0(std::string, stream_id)
 PROXY_METHOD0(void, Stop)
 END_PROXY()
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_