Roll WebRTC 11523:11548, Libjingle 11522:11545

content/renderer/media/webrtc/media_stream_track_metrics_unittest.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 <stddef.h>
 
 #include "base/bind.h"
 #include "base/message_loop/message_loop.h"
 #include "base/run_loop.h"
 #include "base/threading/thread.h"
 #include "content/renderer/media/webrtc/media_stream_track_metrics.h"
 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
 
 using webrtc::AudioSourceInterface;
 using webrtc::AudioTrackInterface;
 using webrtc::AudioTrackSinkInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::ObserverInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::VideoRendererInterface;
 using webrtc::VideoSourceInterface;
 using webrtc::VideoTrackInterface;
@@ skipping 557 matching lines @@
                                   MediaStreamTrackMetrics::SENT_STREAM));
   EXPECT_CALL(*metrics_,
               SendLifetimeMessage("video3",
                                   MediaStreamTrackMetrics::VIDEO_TRACK,
                                   MediaStreamTrackMetrics::DISCONNECTED,
                                   MediaStreamTrackMetrics::SENT_STREAM));
   metrics_->RemoveStream(MediaStreamTrackMetrics::SENT_STREAM, stream_.get());
 }
 
 }  // namespace content