Move talk/app/webrtc to webrtc/api

webrtc/api/rtpsenderreceiver_unittest.cc

 /*
- * 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.
  */
 
 #include <string>
 #include <utility>
 
-#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/mediastream.h"
-#include "talk/app/webrtc/remoteaudiosource.h"
-#include "talk/app/webrtc/rtpreceiver.h"
-#include "talk/app/webrtc/rtpsender.h"
-#include "talk/app/webrtc/streamcollection.h"
-#include "talk/app/webrtc/videosource.h"
-#include "talk/app/webrtc/videotrack.h"
+#include "webrtc/api/audiotrack.h"
+#include "webrtc/api/mediastream.h"
+#include "webrtc/api/remoteaudiosource.h"
+#include "webrtc/api/rtpreceiver.h"
+#include "webrtc/api/rtpsender.h"
+#include "webrtc/api/streamcollection.h"
+#include "webrtc/api/videosource.h"
+#include "webrtc/api/videotrack.h"
 #include "talk/media/base/fakevideocapturer.h"
 #include "talk/media/base/mediachannel.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/gunit.h"
 
 using ::testing::_;
 using ::testing::Exactly;
 
 static const char kStreamLabel1[] = "local_stream_1";
@@ skipping 456 matching lines @@
                                video_track_->GetSource()->GetVideoCapturer()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
   sender->SetSsrc(kVideoSsrc2);
 
   // Calls expected from destructor.
   EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
 }
 
 }  // namespace webrtc