Move talk/media to webrtc/media

webrtc/media/webrtc/fakewebrtccall.cc

 /*
- * libjingle
- * Copyright 2015 Google Inc.
+ *  Copyright 2015 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 "talk/media/webrtc/fakewebrtccall.h"
+#include "webrtc/media/webrtc/fakewebrtccall.h"
 
 #include <algorithm>
 #include <utility>
 
-#include "talk/media/base/rtputils.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/media/base/rtputils.h"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
     const webrtc::AudioSendStream::Config& config) : config_(config) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
 const webrtc::AudioSendStream::Config&
     FakeAudioSendStream::GetConfig() const {
   return config_;
@@ skipping 387 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket