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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 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 "base/macros.h" | 5 #include "base/macros.h" |
6 #include "base/synchronization/waitable_event.h" | 6 #include "base/synchronization/waitable_event.h" |
7 #include "base/test/test_timeouts.h" | 7 #include "base/test/test_timeouts.h" |
8 #include "build/build_config.h" | 8 #include "build/build_config.h" |
9 #include "content/public/renderer/media_stream_audio_sink.h" | 9 #include "content/public/renderer/media_stream_audio_sink.h" |
10 #include "content/renderer/media/media_stream_audio_source.h" | 10 #include "content/renderer/media/media_stream_audio_source.h" |
11 #include "content/renderer/media/mock_media_constraint_factory.h" | 11 #include "content/renderer/media/mock_media_constraint_factory.h" |
12 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 12 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
13 #include "content/renderer/media/webrtc_audio_capturer.h" | 13 #include "content/renderer/media/webrtc_audio_capturer.h" |
14 #include "content/renderer/media/webrtc_local_audio_track.h" | 14 #include "content/renderer/media/webrtc_local_audio_track.h" |
15 #include "media/audio/audio_parameters.h" | 15 #include "media/audio/audio_parameters.h" |
16 #include "media/base/audio_bus.h" | 16 #include "media/base/audio_bus.h" |
17 #include "media/base/audio_capturer_source.h" | 17 #include "media/base/audio_capturer_source.h" |
18 #include "testing/gmock/include/gmock/gmock.h" | 18 #include "testing/gmock/include/gmock/gmock.h" |
19 #include "testing/gtest/include/gtest/gtest.h" | 19 #include "testing/gtest/include/gtest/gtest.h" |
20 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 20 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
21 #include "third_party/WebKit/public/web/WebHeap.h" | 21 #include "third_party/WebKit/public/web/WebHeap.h" |
22 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 22 #include "third_party/webrtc/api/mediastreaminterface.h" |
23 | 23 |
24 using ::testing::_; | 24 using ::testing::_; |
25 using ::testing::AnyNumber; | 25 using ::testing::AnyNumber; |
26 using ::testing::AtLeast; | 26 using ::testing::AtLeast; |
27 using ::testing::Return; | 27 using ::testing::Return; |
28 | 28 |
29 namespace content { | 29 namespace content { |
30 | 30 |
31 namespace { | 31 namespace { |
32 | 32 |
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510 // Stopping the new source will stop the second track. | 510 // Stopping the new source will stop the second track. |
511 EXPECT_CALL(*source.get(), OnStop()).Times(1); | 511 EXPECT_CALL(*source.get(), OnStop()).Times(1); |
512 capturer->Stop(); | 512 capturer->Stop(); |
513 | 513 |
514 // Even though this test don't use |capturer_source_| it will be stopped | 514 // Even though this test don't use |capturer_source_| it will be stopped |
515 // during teardown of the test harness. | 515 // during teardown of the test harness. |
516 EXPECT_CALL(*capturer_source_.get(), OnStop()); | 516 EXPECT_CALL(*capturer_source_.get(), OnStop()); |
517 } | 517 } |
518 | 518 |
519 } // namespace content | 519 } // namespace content |
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