Move talk/app/webrtc to webrtc/api

webrtc/api/webrtcsdp_unittest.cc

 /*
- * libjingle
- * Copyright 2011 Google Inc.
+ *  Copyright 2011 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 <set>
 #include <string>
 #include <vector>
 
-#include "talk/app/webrtc/jsepsessiondescription.h"
+#include "webrtc/api/jsepsessiondescription.h"
 #ifdef WEBRTC_ANDROID
-#include "talk/app/webrtc/test/androidtestinitializer.h"
+#include "webrtc/api/test/androidtestinitializer.h"
 #endif
-#include "talk/app/webrtc/webrtcsdp.h"
+#include "webrtc/api/webrtcsdp.h"
 #include "talk/media/base/constants.h"
-#include "webrtc/p2p/base/constants.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslfingerprint.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/p2p/base/constants.h"
 
 using cricket::AudioCodec;
 using cricket::AudioContentDescription;
 using cricket::Candidate;
 using cricket::ContentInfo;
 using cricket::CryptoParams;
 using cricket::ContentGroup;
 using cricket::DataCodec;
 using cricket::DataContentDescription;
 using cricket::ICE_CANDIDATE_COMPONENT_RTCP;
@@ skipping 2695 matching lines @@
       const cricket::MediaContentDescription* mdesc =
           static_cast<const cricket::MediaContentDescription*>(
               desc->contents()[i].description);
       EXPECT_EQ(media_types[media_content_in_sdp[i]], mdesc->type());
     }
 
     std::string serialized_sdp = webrtc::SdpSerialize(jdesc);
     EXPECT_EQ(sdp_string, serialized_sdp);
   }
 }