Move talk/app/webrtc to webrtc/api

webrtc/api/webrtcsession_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 <utility>
 #include <vector>
 
-#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/fakemediacontroller.h"
-#include "talk/app/webrtc/fakemetricsobserver.h"
-#include "talk/app/webrtc/jsepicecandidate.h"
-#include "talk/app/webrtc/jsepsessiondescription.h"
-#include "talk/app/webrtc/peerconnection.h"
-#include "talk/app/webrtc/sctputils.h"
-#include "talk/app/webrtc/streamcollection.h"
-#include "talk/app/webrtc/streamcollection.h"
-#include "talk/app/webrtc/test/fakeconstraints.h"
-#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
-#include "talk/app/webrtc/videotrack.h"
-#include "talk/app/webrtc/webrtcsession.h"
-#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
+#include "webrtc/api/audiotrack.h"
+#include "webrtc/api/fakemediacontroller.h"
+#include "webrtc/api/fakemetricsobserver.h"
+#include "webrtc/api/jsepicecandidate.h"
+#include "webrtc/api/jsepsessiondescription.h"
+#include "webrtc/api/peerconnection.h"
+#include "webrtc/api/sctputils.h"
+#include "webrtc/api/streamcollection.h"
+#include "webrtc/api/streamcollection.h"
+#include "webrtc/api/test/fakeconstraints.h"
+#include "webrtc/api/test/fakedtlsidentitystore.h"
+#include "webrtc/api/videotrack.h"
+#include "webrtc/api/webrtcsession.h"
+#include "webrtc/api/webrtcsessiondescriptionfactory.h"
 #include "talk/media/base/fakemediaengine.h"
 #include "talk/media/base/fakevideorenderer.h"
 #include "talk/media/base/mediachannel.h"
 #include "talk/media/webrtc/fakewebrtccall.h"
-#include "webrtc/p2p/base/stunserver.h"
-#include "webrtc/p2p/base/teststunserver.h"
-#include "webrtc/p2p/base/testturnserver.h"
-#include "webrtc/p2p/base/transportchannel.h"
-#include "webrtc/p2p/client/basicportallocator.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/fakenetwork.h"
 #include "webrtc/base/firewallsocketserver.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/p2p/client/basicportallocator.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \
   if (!(feature())) {                               \
     LOG(LS_INFO) << "Feature disabled... skipping"; \
     return;                                         \
   }
 
 using cricket::FakeVoiceMediaChannel;
 using cricket::TransportInfo;
 using rtc::SocketAddress;
@@ skipping 4249 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));