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

webrtc/pc/peerconnection_integrationtest.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.
  */
 
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 #if !defined(THREAD_SANITIZER)
 
 #include <stdio.h>
 
 #include <algorithm>
 #include <functional>
 #include <list>
 #include <map>
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/fakemetricsobserver.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/base/asyncinvoker.h"
-#include "webrtc/base/fakenetwork.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/ssladapter.h"
-#include "webrtc/base/sslstreamadapter.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/engine/fakewebrtcvideoengine.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/portinterface.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "webrtc/pc/dtmfsender.h"
 #include "webrtc/pc/localaudiosource.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/peerconnection.h"
 #include "webrtc/pc/peerconnectionfactory.h"
 #include "webrtc/pc/test/fakeaudiocapturemodule.h"
 #include "webrtc/pc/test/fakeperiodicvideocapturer.h"
 #include "webrtc/pc/test/fakertccertificategenerator.h"
 #include "webrtc/pc/test/fakevideotrackrenderer.h"
 #include "webrtc/pc/test/mockpeerconnectionobservers.h"
+#include "webrtc/rtc_base/asyncinvoker.h"
+#include "webrtc/rtc_base/fakenetwork.h"
+#include "webrtc/rtc_base/gunit.h"
+#include "webrtc/rtc_base/helpers.h"
+#include "webrtc/rtc_base/ssladapter.h"
+#include "webrtc/rtc_base/sslstreamadapter.h"
+#include "webrtc/rtc_base/thread.h"
+#include "webrtc/rtc_base/virtualsocketserver.h"
 
 using cricket::ContentInfo;
 using cricket::FakeWebRtcVideoDecoder;
 using cricket::FakeWebRtcVideoDecoderFactory;
 using cricket::FakeWebRtcVideoEncoder;
 using cricket::FakeWebRtcVideoEncoderFactory;
 using cricket::MediaContentDescription;
 using webrtc::DataBuffer;
 using webrtc::DataChannelInterface;
 using webrtc::DtmfSender;
@@ skipping 2907 matching lines @@
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for additional audio frames to be received by the callee.
   ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
                                   kMaxWaitForFramesMs);
 }
 
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)