Move talk/session/media -> webrtc/pc

webrtc/api/rtpsenderinterface.h

 /*
  *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 // This file contains interfaces for RtpSenders
 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
 #ifndef WEBRTC_API_RTPSENDERINTERFACE_H_
 #define WEBRTC_API_RTPSENDERINTERFACE_H_
 
 #include <string>
 
-#include "talk/session/media/mediasession.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/proxy.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/pc/mediasession.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.
   virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
   virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
 
@@ skipping 30 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  // WEBRTC_API_RTPSENDERINTERFACE_H_