Move talk/session/media -> webrtc/pc

webrtc/pc/channel.cc

 /*
  * libjingle
  * Copyright 2004 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 <utility>
 
-#include "talk/session/media/channel.h"
+#include "webrtc/pc/channel.h"
 
-#include "talk/session/media/channelmanager.h"
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/media/base/constants.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/pc/channelmanager.h"
 
 namespace cricket {
 using rtc::Bind;
 
 namespace {
 // See comment below for why we need to use a pointer to a scoped_ptr.
 bool SetRawAudioSink_w(VoiceMediaChannel* channel,
                        uint32_t ssrc,
                        rtc::scoped_ptr<webrtc::AudioSinkInterface>* sink) {
   channel->SetRawAudioSink(ssrc, std::move(*sink));
@@ skipping 2211 matching lines @@
   return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
 }
 
 }  // namespace cricket