Move talk/app/webrtc to webrtc/api

webrtc/api/dtmfsenderinterface.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,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#ifndef TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
-#define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
+#ifndef WEBRTC_API_DTMFSENDERINTERFACE_H_
+#define WEBRTC_API_DTMFSENDERINTERFACE_H_
 
 #include <string>
 
-#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/refcount.h"
 
 // This file contains interfaces for DtmfSender.
 
 namespace webrtc {
 
 // DtmfSender callback interface. Application should implement this interface
 // to get notifications from the DtmfSender.
 class DtmfSenderObserverInterface {
@@ skipping 51 matching lines @@
   // This value will be the value last set via the InsertDtmf() method, or the
   // default value of 50 ms if InsertDtmf() was never called.
   virtual int inter_tone_gap() const = 0;
 
  protected:
   virtual ~DtmfSenderInterface() {}
 };
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
+#endif  // WEBRTC_API_DTMFSENDERINTERFACE_H_