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Side by Side Diff: webrtc/modules/audio_device/audio_device_buffer.h

Issue 2132613002: Adds data logging in native AudioDeviceBuffer class (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 months ago
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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
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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