Move talk/app/webrtc to webrtc/api

webrtc/api/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,
  *     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 "talk/app/webrtc/peerconnection.h"
+#include "webrtc/api/peerconnection.h"
 
 #include <algorithm>
 #include <cctype>  // for isdigit
 #include <utility>
 #include <vector>
 
-#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/dtmfsender.h"
-#include "talk/app/webrtc/jsepicecandidate.h"
-#include "talk/app/webrtc/jsepsessiondescription.h"
-#include "talk/app/webrtc/mediaconstraintsinterface.h"
-#include "talk/app/webrtc/mediastream.h"
-#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/session/media/channelmanager.h"
+#include "webrtc/api/audiotrack.h"
+#include "webrtc/api/dtmfsender.h"
+#include "webrtc/api/jsepicecandidate.h"
+#include "webrtc/api/jsepsessiondescription.h"
+#include "webrtc/api/mediaconstraintsinterface.h"
+#include "webrtc/api/mediastream.h"
+#include "webrtc/api/mediastreamobserver.h"
+#include "webrtc/api/mediastreamproxy.h"
+#include "webrtc/api/mediastreamtrackproxy.h"
+#include "webrtc/api/remoteaudiosource.h"
+#include "webrtc/api/remotevideocapturer.h"
+#include "webrtc/api/rtpreceiver.h"
+#include "webrtc/api/rtpsender.h"
+#include "webrtc/api/streamcollection.h"
+#include "webrtc/api/videosource.h"
+#include "webrtc/api/videotrack.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 {
@@ 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