Move talk/app/webrtc to webrtc/api

webrtc/api/peerconnection.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
  * 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_PEERCONNECTION_H_
-#define TALK_APP_WEBRTC_PEERCONNECTION_H_
+#ifndef WEBRTC_API_PEERCONNECTION_H_
+#define WEBRTC_API_PEERCONNECTION_H_
 
 #include <string>
 
-#include "talk/app/webrtc/dtlsidentitystore.h"
-#include "talk/app/webrtc/peerconnectionfactory.h"
-#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/app/webrtc/rtpreceiverinterface.h"
-#include "talk/app/webrtc/rtpsenderinterface.h"
-#include "talk/app/webrtc/statscollector.h"
-#include "talk/app/webrtc/streamcollection.h"
-#include "talk/app/webrtc/webrtcsession.h"
+#include "webrtc/api/dtlsidentitystore.h"
+#include "webrtc/api/peerconnectionfactory.h"
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/api/rtpreceiverinterface.h"
+#include "webrtc/api/rtpsenderinterface.h"
+#include "webrtc/api/statscollector.h"
+#include "webrtc/api/streamcollection.h"
+#include "webrtc/api/webrtcsession.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
 class MediaStreamObserver;
 class RemoteMediaStreamFactory;
 
 // Populates |session_options| from |rtc_options|, and returns true if options
 // are valid.
 bool ConvertRtcOptionsForOffer(
@@ skipping 338 matching lines @@
   // The session_ scoped_ptr is declared at the bottom of PeerConnection
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_
+#endif  // WEBRTC_API_PEERCONNECTION_H_