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Issue 272043003: Renamed MediaStreamDependencyFactory to PeerConnectionDependencyFactory. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: rebased Created 6 years, 7 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include <vector> 5 #include <vector>
6 6
7 #include "content/renderer/media/audio_device_factory.h" 7 #include "content/renderer/media/audio_device_factory.h"
8 #include "content/renderer/media/audio_message_filter.h" 8 #include "content/renderer/media/audio_message_filter.h"
9 #include "content/renderer/media/media_stream_audio_renderer.h" 9 #include "content/renderer/media/media_stream_audio_renderer.h"
10 #include "content/renderer/media/mock_media_stream_dependency_factory.h" 10 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory. h"
11 #include "content/renderer/media/webrtc_audio_device_impl.h" 11 #include "content/renderer/media/webrtc_audio_device_impl.h"
12 #include "content/renderer/media/webrtc_audio_renderer.h" 12 #include "content/renderer/media/webrtc_audio_renderer.h"
13 #include "media/audio/audio_output_device.h" 13 #include "media/audio/audio_output_device.h"
14 #include "media/audio/audio_output_ipc.h" 14 #include "media/audio/audio_output_ipc.h"
15 #include "media/base/audio_bus.h" 15 #include "media/base/audio_bus.h"
16 #include "media/base/mock_audio_renderer_sink.h" 16 #include "media/base/mock_audio_renderer_sink.h"
17 #include "testing/gmock/include/gmock/gmock.h" 17 #include "testing/gmock/include/gmock/gmock.h"
18 #include "testing/gtest/include/gtest/gtest.h" 18 #include "testing/gtest/include/gtest/gtest.h"
19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
20 20
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144 } else { 144 } else {
145 // When the last proxy is stopped, the sink will stop. 145 // When the last proxy is stopped, the sink will stop.
146 EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get())); 146 EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
147 EXPECT_CALL(*mock_output_device_, Stop()); 147 EXPECT_CALL(*mock_output_device_, Stop());
148 } 148 }
149 renderer_proxies_[i]->Stop(); 149 renderer_proxies_[i]->Stop();
150 } 150 }
151 } 151 }
152 152
153 } // namespace content 153 } // namespace content
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