Move talk/app/webrtc to webrtc/api

webrtc/api/peerconnection.h

 /*
- * libjingle
- * Copyright 2012 Google Inc.
+ *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
- * 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.
+ *  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 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 339 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_