Roll WebRTC 11486:11495, Libjingle 11485:11495

content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
 
 #include <stddef.h>
 
 #include "base/logging.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/renderer/media/mock_peer_connection_impl.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
-#include "third_party/libjingle/source/talk/media/base/videocapturer.h"
 #include "third_party/webrtc/base/scoped_ref_ptr.h"
+#include "third_party/webrtc/media/base/videocapturer.h"
 
 using webrtc::AudioSourceInterface;
 using webrtc::AudioTrackInterface;
 using webrtc::AudioTrackVector;
 using webrtc::IceCandidateCollection;
 using webrtc::IceCandidateInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::ObserverInterface;
 using webrtc::SessionDescriptionInterface;
 using webrtc::VideoRendererInterface;
@@ skipping 502 matching lines @@
   return WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL,
                                              audio_source);
 }
 
 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack(
       WebRtcLocalAudioTrack* audio_track) {
   audio_track->Start();
 }
 
 }  // namespace content