Move talk/session/media -> webrtc/pc

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/constants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
 // TODO(juberti): re-evaluate this include
-#include "talk/session/media/audiomonitor.h"
+#include "webrtc/pc/audiomonitor.h"
 
 namespace rtc {
 class Buffer;
 class RateLimiter;
 class Timing;
 }
 
 namespace webrtc {
 class AudioSinkInterface;
 }
@@ skipping 1070 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_