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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_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |
12 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 12 #define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |
13 | 13 |
14 #include "webrtc/base/criticalsection.h" | |
15 #include "webrtc/base/task_queue.h" | |
14 #include "webrtc/modules/audio_device/include/audio_device.h" | 16 #include "webrtc/modules/audio_device/include/audio_device.h" |
15 #include "webrtc/system_wrappers/include/file_wrapper.h" | 17 #include "webrtc/system_wrappers/include/file_wrapper.h" |
16 #include "webrtc/typedefs.h" | 18 #include "webrtc/typedefs.h" |
17 | 19 |
18 namespace webrtc { | 20 namespace webrtc { |
19 class CriticalSectionWrapper; | 21 class CriticalSectionWrapper; |
20 | 22 |
21 const uint32_t kPulsePeriodMs = 1000; | 23 const uint32_t kPulsePeriodMs = 1000; |
22 const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz | 24 const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz |
23 | 25 |
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56 virtual int32_t GetPlayoutData(void* audioBuffer); | 58 virtual int32_t GetPlayoutData(void* audioBuffer); |
57 | 59 |
58 int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); | 60 int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
59 int32_t StopInputFileRecording(); | 61 int32_t StopInputFileRecording(); |
60 int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); | 62 int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
61 int32_t StopOutputFileRecording(); | 63 int32_t StopOutputFileRecording(); |
62 | 64 |
63 int32_t SetTypingStatus(bool typingStatus); | 65 int32_t SetTypingStatus(bool typingStatus); |
64 | 66 |
65 private: | 67 private: |
66 CriticalSectionWrapper& _critSect; | 68 // Posts the first delayed task in the task queue and starts the periodic |
67 CriticalSectionWrapper& _critSectCb; | 69 // timer. |
70 void StartTimer(); | |
71 | |
72 // Called periodically on the internal thread created by the TaskQueue. | |
73 // Members are only read and never modified by this method and all access is | |
74 // done without any locks. The stored data is only for logging purposes and | |
75 // minor deviations to to potential race issues are ignored. | |
76 // TODO(henrika): remove all usage of locks in this class, add thread checker | |
77 // and document the threading model. | |
78 void LogStats(); | |
79 | |
80 rtc::CriticalSection _critSect; | |
81 rtc::CriticalSection _critSectCb; | |
68 | 82 |
69 AudioTransport* _ptrCbAudioTransport; | 83 AudioTransport* _ptrCbAudioTransport; |
70 | 84 |
85 // Task queue posting delayed tasks periodically. Used as a timer and calls | |
86 // LogStats() in each task. | |
87 std::unique_ptr<rtc::TaskQueue> task_queue_; | |
stefan-webrtc
2016/07/07 15:22:14
Any point in allocating this dynamically?
henrika_webrtc
2016/07/08 12:46:48
Actually not. Will change.
| |
88 | |
89 // Ensures that the timer is only started once. | |
90 bool timer_has_started_; | |
91 | |
71 uint32_t _recSampleRate; | 92 uint32_t _recSampleRate; |
72 uint32_t _playSampleRate; | 93 uint32_t _playSampleRate; |
73 | 94 |
74 size_t _recChannels; | 95 size_t _recChannels; |
75 size_t _playChannels; | 96 size_t _playChannels; |
76 | 97 |
77 // selected recording channel (left/right/both) | 98 // selected recording channel (left/right/both) |
78 AudioDeviceModule::ChannelType _recChannel; | 99 AudioDeviceModule::ChannelType _recChannel; |
79 | 100 |
80 // 2 or 4 depending on mono or stereo | 101 // 2 or 4 depending on mono or stereo |
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100 | 121 |
101 uint32_t _currentMicLevel; | 122 uint32_t _currentMicLevel; |
102 uint32_t _newMicLevel; | 123 uint32_t _newMicLevel; |
103 | 124 |
104 bool _typingStatus; | 125 bool _typingStatus; |
105 | 126 |
106 int _playDelayMS; | 127 int _playDelayMS; |
107 int _recDelayMS; | 128 int _recDelayMS; |
108 int _clockDrift; | 129 int _clockDrift; |
109 int high_delay_counter_; | 130 int high_delay_counter_; |
131 | |
132 // Total number of recording callbacks where the source provides 10ms audio | |
133 // data each time. | |
134 uint64_t rec_callbacks_; | |
135 | |
136 // Total number of recording callbacks stored at the last timer task. | |
137 uint64_t last_rec_callbacks_; | |
138 | |
139 // Total number of playback callbacks where the sink asks for 10ms audio | |
140 // data each time. | |
141 uint64_t play_callbacks_; | |
142 | |
143 // Total number of playout callbacks stored at the last timer task. | |
144 uint64_t last_play_callbacks_; | |
145 | |
146 // Total number of recorded audio samples. | |
147 uint64_t rec_samples_; | |
148 | |
149 // Total number of recorded samples stored at the previous timer task. | |
150 uint64_t last_rec_samples_; | |
151 | |
152 // Total number of played audio samples. | |
153 uint64_t play_samples_; | |
154 | |
155 // Total number of played samples stored at the previous timer task. | |
156 uint64_t last_play_samples_; | |
110 }; | 157 }; |
111 | 158 |
112 } // namespace webrtc | 159 } // namespace webrtc |
113 | 160 |
114 #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 161 #endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |
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