Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  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 "webrtc/pc/peerconnection.h"
 
 #include <algorithm>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/jsepicecandidate.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/mediastreamproxy.h"
 #include "webrtc/api/mediastreamtrackproxy.h"
-#include "webrtc/base/bind.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/stringencode.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/trace_event.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/sctp/sctptransport.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/pc/dtmfsender.h"
 #include "webrtc/pc/mediastream.h"
 #include "webrtc/pc/mediastreamobserver.h"
 #include "webrtc/pc/remoteaudiosource.h"
 #include "webrtc/pc/rtpreceiver.h"
 #include "webrtc/pc/rtpsender.h"
 #include "webrtc/pc/streamcollection.h"
 #include "webrtc/pc/videocapturertracksource.h"
 #include "webrtc/pc/videotrack.h"
+#include "webrtc/rtc_base/bind.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/stringencode.h"
+#include "webrtc/rtc_base/stringutils.h"
+#include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 
 using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::RTCError;
@@ skipping 2316 matching lines @@
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   if (event_log_) {
     event_log_->StopLogging();
   }
 }
 
 }  // namespace webrtc