Move talk/media to webrtc/media

talk/app/webrtc/peerconnection.cc

 /*
  * libjingle
  * Copyright 2012 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,
@@ skipping 30 matching lines @@
 #include "talk/app/webrtc/mediastreamobserver.h"
 #include "talk/app/webrtc/mediastreamproxy.h"
 #include "talk/app/webrtc/mediastreamtrackproxy.h"
 #include "talk/app/webrtc/remoteaudiosource.h"
 #include "talk/app/webrtc/remotevideocapturer.h"
 #include "talk/app/webrtc/rtpreceiver.h"
 #include "talk/app/webrtc/rtpsender.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 
 using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::RtpSenderInterface;
@@ skipping 2020 matching lines @@
 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
   for (const auto& channel : sctp_data_channels_) {
     if (channel->id() == sid) {
       return channel;
     }
   }
   return nullptr;
 }
 
 }  // namespace webrtc