content/renderer/media/webrtc_local_audio_track_unittest.cc - Issue 1547073003: Switch to standard integer types in content/renderer/.

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Side by Side Diff: content/renderer/media/webrtc_local_audio_track_unittest.cc

Issue 1547073003: Switch to standard integer types in content/renderer/. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master

Patch Set: Created 4 years, 12 months ago

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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/synchronization/waitable_event.h"	6 #include "base/synchronization/waitable_event.h"

6 #include "base/test/test_timeouts.h"	7 #include "base/test/test_timeouts.h"

	8 #include "build/build_config.h"

7 #include "content/public/renderer/media_stream_audio_sink.h"	9 #include "content/public/renderer/media_stream_audio_sink.h"

8 #include "content/renderer/media/media_stream_audio_source.h"	10 #include "content/renderer/media/media_stream_audio_source.h"

9 #include "content/renderer/media/mock_media_constraint_factory.h"	11 #include "content/renderer/media/mock_media_constraint_factory.h"

10 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"	12 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"

11 #include "content/renderer/media/webrtc_audio_capturer.h"	13 #include "content/renderer/media/webrtc_audio_capturer.h"

12 #include "content/renderer/media/webrtc_local_audio_track.h"	14 #include "content/renderer/media/webrtc_local_audio_track.h"

13 #include "media/audio/audio_parameters.h"	15 #include "media/audio/audio_parameters.h"

14 #include "media/base/audio_bus.h"	16 #include "media/base/audio_bus.h"

15 #include "media/base/audio_capturer_source.h"	17 #include "media/base/audio_capturer_source.h"

16 #include "testing/gmock/include/gmock/gmock.h"	18 #include "testing/gmock/include/gmock/gmock.h"

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500 // Stopping the new source will stop the second track.	502 // Stopping the new source will stop the second track.

501 EXPECT_CALL(*source.get(), OnStop()).Times(1);	503 EXPECT_CALL(*source.get(), OnStop()).Times(1);

502 capturer->Stop();	504 capturer->Stop();

503	505

504 // Even though this test don't use \|capturer_source_\| it will be stopped	506 // Even though this test don't use \|capturer_source_\| it will be stopped

505 // during teardown of the test harness.	507 // during teardown of the test harness.

506 EXPECT_CALL(*capturer_source_.get(), OnStop());	508 EXPECT_CALL(*capturer_source_.get(), OnStop());

507 }	509 }

508	510

509 } // namespace content	511 } // namespace content

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