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

Issue 3012853002: Update thread annotiation macros to use RTC_ prefix (Closed)
Patch Set: Rebase Created 3 years, 3 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_
(...skipping 103 matching lines...) Expand 10 before | Expand all | Expand 10 after
114 // Returns false if |timeout_ms| passes before that happens. 114 // Returns false if |timeout_ms| passes before that happens.
115 bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever); 115 bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever);
116 // Blocks until the Recorder stops producing data. 116 // Blocks until the Recorder stops producing data.
117 // Returns false if |timeout_ms| passes before that happens. 117 // Returns false if |timeout_ms| passes before that happens.
118 bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever); 118 bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever);
119 119
120 private: 120 private:
121 static bool Run(void* obj); 121 static bool Run(void* obj);
122 void ProcessAudio(); 122 void ProcessAudio();
123 123
124 const std::unique_ptr<Capturer> capturer_ GUARDED_BY(lock_); 124 const std::unique_ptr<Capturer> capturer_ RTC_GUARDED_BY(lock_);
125 const std::unique_ptr<Renderer> renderer_ GUARDED_BY(lock_); 125 const std::unique_ptr<Renderer> renderer_ RTC_GUARDED_BY(lock_);
126 const float speed_; 126 const float speed_;
127 127
128 rtc::CriticalSection lock_; 128 rtc::CriticalSection lock_;
129 AudioTransport* audio_callback_ GUARDED_BY(lock_); 129 AudioTransport* audio_callback_ RTC_GUARDED_BY(lock_);
130 bool rendering_ GUARDED_BY(lock_); 130 bool rendering_ RTC_GUARDED_BY(lock_);
131 bool capturing_ GUARDED_BY(lock_); 131 bool capturing_ RTC_GUARDED_BY(lock_);
132 rtc::Event done_rendering_; 132 rtc::Event done_rendering_;
133 rtc::Event done_capturing_; 133 rtc::Event done_capturing_;
134 134
135 std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_); 135 std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
136 rtc::BufferT<int16_t> recording_buffer_ GUARDED_BY(lock_); 136 rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
137 137
138 std::unique_ptr<EventTimerWrapper> tick_; 138 std::unique_ptr<EventTimerWrapper> tick_;
139 rtc::PlatformThread thread_; 139 rtc::PlatformThread thread_;
140 }; 140 };
141 } // namespace test 141 } // namespace test
142 } // namespace webrtc 142 } // namespace webrtc
143 143
144 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ 144 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
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