content/renderer/media/webrtc_local_audio_source_provider_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_source_provider_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, 11 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 <stddef.h>

	6

5 #include "base/logging.h"	7 #include "base/logging.h"

6 #include "base/strings/utf_string_conversions.h"	8 #include "base/strings/utf_string_conversions.h"

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

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

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

10 #include "content/renderer/media/webrtc_local_audio_source_provider.h"	12 #include "content/renderer/media/webrtc_local_audio_source_provider.h"

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

12 #include "media/audio/audio_parameters.h"	14 #include "media/audio/audio_parameters.h"

13 #include "media/base/audio_bus.h"	15 #include "media/base/audio_bus.h"

14 #include "testing/gtest/include/gtest/gtest.h"	16 #include "testing/gtest/include/gtest/gtest.h"

(...skipping 119 matching lines...) Expand 10 before \| Expand all \| Expand 10 after Loading...
134 // Stop the audio track.	136 // Stop the audio track.

135 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(	137 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(

136 MediaStreamTrack::GetTrack(blink_track_));	138 MediaStreamTrack::GetTrack(blink_track_));

137 native_track->Stop();	139 native_track->Stop();

138	140

139 // Delete the source provider.	141 // Delete the source provider.

140 source_provider_.reset();	142 source_provider_.reset();

141 }	143 }

142	144

143 } // namespace content	145 } // namespace content

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