Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/pc/rtpsender.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 classes that implement RtpSenderInterface.
 // An RtpSender associates a MediaStreamTrackInterface with an underlying
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
-#ifndef WEBRTC_API_RTPSENDER_H_
-#define WEBRTC_API_RTPSENDER_H_
+#ifndef WEBRTC_PC_RTPSENDER_H_
+#define WEBRTC_PC_RTPSENDER_H_
 
 #include <memory>
 #include <string>
 
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
-#include "webrtc/api/statscollector.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/pc/channel.h"
+#include "webrtc/pc/statscollector.h"
 
 namespace webrtc {
 
 // Internal interface used by PeerConnection.
 class RtpSenderInternal : public RtpSenderInterface {
  public:
   // Used to set the SSRC of the sender, once a local description has been set.
   // If |ssrc| is 0, this indiates that the sender should disconnect from the
   // underlying transport (this occurs if the sender isn't seen in a local
   // description).
@@ skipping 187 matching lines @@
   rtc::scoped_refptr<VideoTrackInterface> track_;
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   VideoTrackInterface::ContentHint cached_track_content_hint_ =
       VideoTrackInterface::ContentHint::kNone;
   bool stopped_ = false;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_API_RTPSENDER_H_
+#endif  // WEBRTC_PC_RTPSENDER_H_