Move talk/app/webrtc to webrtc/api

webrtc/api/test/peerconnectiontestwrapper.h

 /*
  * libjingle
  * Copyright 2013 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
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
-#define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
 
-#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
-#include "talk/app/webrtc/test/fakeconstraints.h"
-#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/api/test/fakeaudiocapturemodule.h"
+#include "webrtc/api/test/fakeconstraints.h"
+#include "webrtc/api/test/fakevideotrackrenderer.h"
 #include "webrtc/base/sigslot.h"
 
 class PeerConnectionTestWrapper
     : public webrtc::PeerConnectionObserver,
       public webrtc::CreateSessionDescriptionObserver,
       public sigslot::has_slots<> {
  public:
   static void Connect(PeerConnectionTestWrapper* caller,
                       PeerConnectionTestWrapper* callee);
 
@@ skipping 60 matching lines @@
       bool video, const webrtc::FakeConstraints& video_constraints);
 
   std::string name_;
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
       peer_connection_factory_;
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
 };
 
-#endif  // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#endif  // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_