OLD | NEW |
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 Loading... |
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_ |
OLD | NEW |