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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ | 11 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H |
12 #define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ | 12 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H |
13 | 13 |
14 #include "webrtc/base/criticalsection.h" | |
15 #include "webrtc/base/task_queue.h" | |
16 #include "webrtc/base/thread_checker.h" | |
17 #include "webrtc/modules/audio_device/include/audio_device.h" | 14 #include "webrtc/modules/audio_device/include/audio_device.h" |
18 #include "webrtc/system_wrappers/include/file_wrapper.h" | 15 #include "webrtc/system_wrappers/include/file_wrapper.h" |
19 #include "webrtc/typedefs.h" | 16 #include "webrtc/typedefs.h" |
20 | 17 |
21 namespace webrtc { | 18 namespace webrtc { |
22 class CriticalSectionWrapper; | 19 class CriticalSectionWrapper; |
23 | 20 |
24 const uint32_t kPulsePeriodMs = 1000; | 21 const uint32_t kPulsePeriodMs = 1000; |
25 const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz | 22 const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz |
26 | 23 |
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59 virtual int32_t GetPlayoutData(void* audioBuffer); | 56 virtual int32_t GetPlayoutData(void* audioBuffer); |
60 | 57 |
61 int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); | 58 int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
62 int32_t StopInputFileRecording(); | 59 int32_t StopInputFileRecording(); |
63 int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); | 60 int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
64 int32_t StopOutputFileRecording(); | 61 int32_t StopOutputFileRecording(); |
65 | 62 |
66 int32_t SetTypingStatus(bool typingStatus); | 63 int32_t SetTypingStatus(bool typingStatus); |
67 | 64 |
68 private: | 65 private: |
69 // Posts the first delayed task in the task queue and starts the periodic | 66 CriticalSectionWrapper& _critSect; |
70 // timer. | 67 CriticalSectionWrapper& _critSectCb; |
71 void StartTimer(); | |
72 | |
73 // Called periodically on the internal thread created by the TaskQueue. | |
74 void LogStats(); | |
75 | |
76 // Updates counters in each play/record callback but does it on the task | |
77 // queue to ensure that they can be read by LogStats() without any locks since | |
78 // each task is serialized by the task queue. | |
79 void UpdateRecStats(size_t num_samples); | |
80 void UpdatePlayStats(size_t num_samples); | |
81 | |
82 // Ensures that methods are called on the same thread as the thread that | |
83 // creates this object. | |
84 rtc::ThreadChecker thread_checker_; | |
85 | |
86 rtc::CriticalSection _critSect; | |
87 rtc::CriticalSection _critSectCb; | |
88 | 68 |
89 AudioTransport* _ptrCbAudioTransport; | 69 AudioTransport* _ptrCbAudioTransport; |
90 | 70 |
91 // Task queue used to invoke LogStats() periodically. Tasks are executed on a | |
92 // worker thread but it does not necessarily have to be the same thread for | |
93 // each task. | |
94 rtc::TaskQueue task_queue_; | |
95 | |
96 // Ensures that the timer is only started once. | |
97 bool timer_has_started_; | |
98 | |
99 uint32_t _recSampleRate; | 71 uint32_t _recSampleRate; |
100 uint32_t _playSampleRate; | 72 uint32_t _playSampleRate; |
101 | 73 |
102 size_t _recChannels; | 74 size_t _recChannels; |
103 size_t _playChannels; | 75 size_t _playChannels; |
104 | 76 |
105 // selected recording channel (left/right/both) | 77 // selected recording channel (left/right/both) |
106 AudioDeviceModule::ChannelType _recChannel; | 78 AudioDeviceModule::ChannelType _recChannel; |
107 | 79 |
108 // 2 or 4 depending on mono or stereo | 80 // 2 or 4 depending on mono or stereo |
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128 | 100 |
129 uint32_t _currentMicLevel; | 101 uint32_t _currentMicLevel; |
130 uint32_t _newMicLevel; | 102 uint32_t _newMicLevel; |
131 | 103 |
132 bool _typingStatus; | 104 bool _typingStatus; |
133 | 105 |
134 int _playDelayMS; | 106 int _playDelayMS; |
135 int _recDelayMS; | 107 int _recDelayMS; |
136 int _clockDrift; | 108 int _clockDrift; |
137 int high_delay_counter_; | 109 int high_delay_counter_; |
138 | |
139 // Counts number of times LogStats() has been called. | |
140 size_t num_stat_reports_; | |
141 | |
142 // Total number of recording callbacks where the source provides 10ms audio | |
143 // data each time. | |
144 uint64_t rec_callbacks_; | |
145 | |
146 // Total number of recording callbacks stored at the last timer task. | |
147 uint64_t last_rec_callbacks_; | |
148 | |
149 // Total number of playback callbacks where the sink asks for 10ms audio | |
150 // data each time. | |
151 uint64_t play_callbacks_; | |
152 | |
153 // Total number of playout callbacks stored at the last timer task. | |
154 uint64_t last_play_callbacks_; | |
155 | |
156 // Total number of recorded audio samples. | |
157 uint64_t rec_samples_; | |
158 | |
159 // Total number of recorded samples stored at the previous timer task. | |
160 uint64_t last_rec_samples_; | |
161 | |
162 // Total number of played audio samples. | |
163 uint64_t play_samples_; | |
164 | |
165 // Total number of played samples stored at the previous timer task. | |
166 uint64_t last_play_samples_; | |
167 | |
168 // Time stamp of last stat report. | |
169 uint64_t last_log_stat_time_; | |
170 }; | 110 }; |
171 | 111 |
172 } // namespace webrtc | 112 } // namespace webrtc |
173 | 113 |
174 #endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ | 114 #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H |
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