Move talk/app/webrtc to webrtc/api

talk/media/webrtc/webrtcvoiceengine_unittest.cc

 /*
  * libjingle
  * Copyright 2008 Google Inc.
  *
  * 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.
  */
 
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/byteorder.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/call.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/fakemediaengine.h"
 #include "talk/media/base/fakenetworkinterface.h"
 #include "talk/media/base/fakertp.h"
 #include "talk/media/webrtc/fakewebrtccall.h"
 #include "talk/media/webrtc/fakewebrtcvoiceengine.h"
 #include "talk/media/webrtc/webrtcvoiceengine.h"
+#include "talk/session/media/channel.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/byteorder.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/call.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
-#include "talk/session/media/channel.h"
 
 using cricket::kRtpAudioLevelHeaderExtension;
 using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
 
 namespace {
 
 const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
 const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
 const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0);
 const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0);
@@ skipping 3244 matching lines @@
   cricket::WebRtcVoiceEngine engine;
   EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
   rtc::scoped_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config()));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::AudioOptions(),
                                            call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }