Roll WebRTC 11523:11548, Libjingle 11522:11545

content/renderer/media/webrtc_audio_renderer_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 "content/renderer/media/webrtc_audio_renderer.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/run_loop.h"
 #include "base/single_thread_task_runner.h"
 #include "build/build_config.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/audio/audio_output_device.h"
 #include "media/audio/audio_output_ipc.h"
 #include "media/base/audio_bus.h"
 #include "media/base/mock_audio_renderer_sink.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
-#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
 
 using testing::Return;
 using testing::_;
 
 namespace content {
 
 namespace {
 
 const int kHardwareSampleRate = 44100;
 const int kHardwareBufferSize = 512;
@@ skipping 273 matching lines @@
                  base::Unretained(this), &loop));
   loop.Run();
   EXPECT_EQ(kDefaultOutputDeviceId, mock_output_device_->GetDeviceId());
 
   EXPECT_CALL(*mock_output_device_.get(), Stop());
   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
   renderer_proxy_->Stop();
 }
 
 }  // namespace content