webrtc/test/fake_audio_device.h - Issue 1613643004: Remove mutable from rtc::CriticalSection members.

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Side by Side Diff: webrtc/test/fake_audio_device.h

Issue 1613643004: Remove mutable from rtc::CriticalSection members. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 4 years, 11 months ago

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1 /*	1 /*

2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.	2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.

3 *	3 *

4 * Use of this source code is governed by a BSD-style license	4 * Use of this source code is governed by a BSD-style license

5 * that can be found in the LICENSE file in the root of the source	5 * that can be found in the LICENSE file in the root of the source

6 * tree. An additional intellectual property rights grant can be found	6 * tree. An additional intellectual property rights grant can be found

7 * in the file PATENTS. All contributing project authors may	7 * in the file PATENTS. All contributing project authors may

8 * be found in the AUTHORS file in the root of the source tree.	8 * be found in the AUTHORS file in the root of the source tree.

9 */	9 */

10 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_	10 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_

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51 static const size_t kBufferSizeBytes = 2 * kFrequencyHz;	51 static const size_t kBufferSizeBytes = 2 * kFrequencyHz;

52	52

53 AudioTransport* audio_callback_;	53 AudioTransport* audio_callback_;

54 bool capturing_;	54 bool capturing_;

55 int8_t captured_audio_[kBufferSizeBytes];	55 int8_t captured_audio_[kBufferSizeBytes];

56 int8_t playout_buffer_[kBufferSizeBytes];	56 int8_t playout_buffer_[kBufferSizeBytes];

57 int64_t last_playout_ms_;	57 int64_t last_playout_ms_;

58	58

59 Clock* clock_;	59 Clock* clock_;

60 rtc::scoped_ptr<EventTimerWrapper> tick_;	60 rtc::scoped_ptr<EventTimerWrapper> tick_;

61 mutable rtc::CriticalSection lock_;	61 rtc::CriticalSection lock_;

62 rtc::PlatformThread thread_;	62 rtc::PlatformThread thread_;

63 rtc::scoped_ptr<ModuleFileUtility> file_utility_;	63 rtc::scoped_ptr<ModuleFileUtility> file_utility_;

64 rtc::scoped_ptr<FileWrapper> input_stream_;	64 rtc::scoped_ptr<FileWrapper> input_stream_;

65 };	65 };

66 } // namespace test	66 } // namespace test

67 } // namespace webrtc	67 } // namespace webrtc

68	68

69 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_	69 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_

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