Move talk/media to webrtc/media

webrtc/media/sctp/sctpdataengine_unittest.cc

 /*
- * libjingle
- * Copyright 2013 Google Inc.
+ *  Copyright 2013 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 <errno.h>
 #include <stdarg.h>
 #include <stdio.h>
 #include <string>
 #include <vector>
 
-#include "talk/media/base/constants.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/sctp/sctpdataengine.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 
 enum {
   MSG_PACKET = 1,
 };
 
 // Fake NetworkInterface that sends/receives sctp packets.  The one in
-// talk/media/base/fakenetworkinterface.h only works with rtp/rtcp.
+// webrtc/media/base/fakenetworkinterface.h only works with rtp/rtcp.
 class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
                                  public rtc::MessageHandler {
  public:
   explicit SctpFakeNetworkInterface(rtc::Thread* thread)
     : thread_(thread),
       dest_(NULL) {
   }
 
   void SetDestination(cricket::DataMediaChannel* dest) { dest_ = dest; }
 
@@ skipping 470 matching lines @@
   // Channel 1 is gone now.
 
   // Create a new channel 1.
   AddStream(1);
   ASSERT_TRUE(SendData(channel1(), 1, "hi?", &result));
   EXPECT_EQ(cricket::SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hi?"), 1000);
   channel1()->RemoveSendStream(1);
   EXPECT_TRUE_WAIT(chan_2_sig_receiver.StreamCloseCount(1) == 2, 1000);
 }