OLD | NEW |
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 #include "webrtc/modules/audio_device/audio_device_buffer.h" | 11 #include "webrtc/modules/audio_device/audio_device_buffer.h" |
12 | 12 |
13 #include "webrtc/base/bind.h" | |
14 #include "webrtc/base/checks.h" | 13 #include "webrtc/base/checks.h" |
15 #include "webrtc/base/logging.h" | 14 #include "webrtc/base/logging.h" |
16 #include "webrtc/base/format_macros.h" | 15 #include "webrtc/base/format_macros.h" |
17 #include "webrtc/base/timeutils.h" | |
18 #include "webrtc/modules/audio_device/audio_device_config.h" | 16 #include "webrtc/modules/audio_device/audio_device_config.h" |
| 17 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
19 | 18 |
20 namespace webrtc { | 19 namespace webrtc { |
21 | 20 |
22 static const int kHighDelayThresholdMs = 300; | 21 static const int kHighDelayThresholdMs = 300; |
23 static const int kLogHighDelayIntervalFrames = 500; // 5 seconds. | 22 static const int kLogHighDelayIntervalFrames = 500; // 5 seconds. |
24 | 23 |
25 static const char kTimerQueueName[] = "AudioDeviceBufferTimer"; | |
26 | |
27 // Time between two sucessive calls to LogStats(). | |
28 static const size_t kTimerIntervalInSeconds = 10; | |
29 static const size_t kTimerIntervalInMilliseconds = | |
30 kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec; | |
31 | |
32 AudioDeviceBuffer::AudioDeviceBuffer() | 24 AudioDeviceBuffer::AudioDeviceBuffer() |
33 : _ptrCbAudioTransport(nullptr), | 25 : _critSect(*CriticalSectionWrapper::CreateCriticalSection()), |
34 task_queue_(kTimerQueueName), | 26 _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()), |
35 timer_has_started_(false), | 27 _ptrCbAudioTransport(nullptr), |
36 _recSampleRate(0), | 28 _recSampleRate(0), |
37 _playSampleRate(0), | 29 _playSampleRate(0), |
38 _recChannels(0), | 30 _recChannels(0), |
39 _playChannels(0), | 31 _playChannels(0), |
40 _recChannel(AudioDeviceModule::kChannelBoth), | 32 _recChannel(AudioDeviceModule::kChannelBoth), |
41 _recBytesPerSample(0), | 33 _recBytesPerSample(0), |
42 _playBytesPerSample(0), | 34 _playBytesPerSample(0), |
43 _recSamples(0), | 35 _recSamples(0), |
44 _recSize(0), | 36 _recSize(0), |
45 _playSamples(0), | 37 _playSamples(0), |
46 _playSize(0), | 38 _playSize(0), |
47 _recFile(*FileWrapper::Create()), | 39 _recFile(*FileWrapper::Create()), |
48 _playFile(*FileWrapper::Create()), | 40 _playFile(*FileWrapper::Create()), |
49 _currentMicLevel(0), | 41 _currentMicLevel(0), |
50 _newMicLevel(0), | 42 _newMicLevel(0), |
51 _typingStatus(false), | 43 _typingStatus(false), |
52 _playDelayMS(0), | 44 _playDelayMS(0), |
53 _recDelayMS(0), | 45 _recDelayMS(0), |
54 _clockDrift(0), | 46 _clockDrift(0), |
55 // Set to the interval in order to log on the first occurrence. | 47 // Set to the interval in order to log on the first occurrence. |
56 high_delay_counter_(kLogHighDelayIntervalFrames), | 48 high_delay_counter_(kLogHighDelayIntervalFrames) { |
57 num_stat_reports_(0), | |
58 rec_callbacks_(0), | |
59 last_rec_callbacks_(0), | |
60 play_callbacks_(0), | |
61 last_play_callbacks_(0), | |
62 rec_samples_(0), | |
63 last_rec_samples_(0), | |
64 play_samples_(0), | |
65 last_play_samples_(0), | |
66 last_log_stat_time_(0) { | |
67 LOG(INFO) << "AudioDeviceBuffer::ctor"; | 49 LOG(INFO) << "AudioDeviceBuffer::ctor"; |
68 memset(_recBuffer, 0, kMaxBufferSizeBytes); | 50 memset(_recBuffer, 0, kMaxBufferSizeBytes); |
69 memset(_playBuffer, 0, kMaxBufferSizeBytes); | 51 memset(_playBuffer, 0, kMaxBufferSizeBytes); |
70 } | 52 } |
71 | 53 |
72 AudioDeviceBuffer::~AudioDeviceBuffer() { | 54 AudioDeviceBuffer::~AudioDeviceBuffer() { |
73 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
74 LOG(INFO) << "AudioDeviceBuffer::~dtor"; | 55 LOG(INFO) << "AudioDeviceBuffer::~dtor"; |
75 _recFile.Flush(); | 56 { |
76 _recFile.CloseFile(); | 57 CriticalSectionScoped lock(&_critSect); |
77 delete &_recFile; | |
78 | 58 |
79 _playFile.Flush(); | 59 _recFile.Flush(); |
80 _playFile.CloseFile(); | 60 _recFile.CloseFile(); |
81 delete &_playFile; | 61 delete &_recFile; |
| 62 |
| 63 _playFile.Flush(); |
| 64 _playFile.CloseFile(); |
| 65 delete &_playFile; |
| 66 } |
| 67 |
| 68 delete &_critSect; |
| 69 delete &_critSectCb; |
82 } | 70 } |
83 | 71 |
84 int32_t AudioDeviceBuffer::RegisterAudioCallback( | 72 int32_t AudioDeviceBuffer::RegisterAudioCallback( |
85 AudioTransport* audioCallback) { | 73 AudioTransport* audioCallback) { |
86 LOG(INFO) << __FUNCTION__; | 74 LOG(INFO) << __FUNCTION__; |
87 rtc::CritScope lock(&_critSectCb); | 75 CriticalSectionScoped lock(&_critSectCb); |
88 _ptrCbAudioTransport = audioCallback; | 76 _ptrCbAudioTransport = audioCallback; |
89 return 0; | 77 return 0; |
90 } | 78 } |
91 | 79 |
92 int32_t AudioDeviceBuffer::InitPlayout() { | 80 int32_t AudioDeviceBuffer::InitPlayout() { |
93 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
94 LOG(INFO) << __FUNCTION__; | 81 LOG(INFO) << __FUNCTION__; |
95 if (!timer_has_started_) { | |
96 StartTimer(); | |
97 timer_has_started_ = true; | |
98 } | |
99 return 0; | 82 return 0; |
100 } | 83 } |
101 | 84 |
102 int32_t AudioDeviceBuffer::InitRecording() { | 85 int32_t AudioDeviceBuffer::InitRecording() { |
103 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
104 LOG(INFO) << __FUNCTION__; | 86 LOG(INFO) << __FUNCTION__; |
105 if (!timer_has_started_) { | |
106 StartTimer(); | |
107 timer_has_started_ = true; | |
108 } | |
109 return 0; | 87 return 0; |
110 } | 88 } |
111 | 89 |
112 int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) { | 90 int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) { |
113 LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")"; | 91 LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")"; |
114 rtc::CritScope lock(&_critSect); | 92 CriticalSectionScoped lock(&_critSect); |
115 _recSampleRate = fsHz; | 93 _recSampleRate = fsHz; |
116 return 0; | 94 return 0; |
117 } | 95 } |
118 | 96 |
119 int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) { | 97 int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) { |
120 LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")"; | 98 LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")"; |
121 rtc::CritScope lock(&_critSect); | 99 CriticalSectionScoped lock(&_critSect); |
122 _playSampleRate = fsHz; | 100 _playSampleRate = fsHz; |
123 return 0; | 101 return 0; |
124 } | 102 } |
125 | 103 |
126 int32_t AudioDeviceBuffer::RecordingSampleRate() const { | 104 int32_t AudioDeviceBuffer::RecordingSampleRate() const { |
127 return _recSampleRate; | 105 return _recSampleRate; |
128 } | 106 } |
129 | 107 |
130 int32_t AudioDeviceBuffer::PlayoutSampleRate() const { | 108 int32_t AudioDeviceBuffer::PlayoutSampleRate() const { |
131 return _playSampleRate; | 109 return _playSampleRate; |
132 } | 110 } |
133 | 111 |
134 int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) { | 112 int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) { |
135 rtc::CritScope lock(&_critSect); | 113 CriticalSectionScoped lock(&_critSect); |
136 _recChannels = channels; | 114 _recChannels = channels; |
137 _recBytesPerSample = | 115 _recBytesPerSample = |
138 2 * channels; // 16 bits per sample in mono, 32 bits in stereo | 116 2 * channels; // 16 bits per sample in mono, 32 bits in stereo |
139 return 0; | 117 return 0; |
140 } | 118 } |
141 | 119 |
142 int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) { | 120 int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) { |
143 rtc::CritScope lock(&_critSect); | 121 CriticalSectionScoped lock(&_critSect); |
144 _playChannels = channels; | 122 _playChannels = channels; |
145 // 16 bits per sample in mono, 32 bits in stereo | 123 // 16 bits per sample in mono, 32 bits in stereo |
146 _playBytesPerSample = 2 * channels; | 124 _playBytesPerSample = 2 * channels; |
147 return 0; | 125 return 0; |
148 } | 126 } |
149 | 127 |
150 int32_t AudioDeviceBuffer::SetRecordingChannel( | 128 int32_t AudioDeviceBuffer::SetRecordingChannel( |
151 const AudioDeviceModule::ChannelType channel) { | 129 const AudioDeviceModule::ChannelType channel) { |
152 rtc::CritScope lock(&_critSect); | 130 CriticalSectionScoped lock(&_critSect); |
153 | 131 |
154 if (_recChannels == 1) { | 132 if (_recChannels == 1) { |
155 return -1; | 133 return -1; |
156 } | 134 } |
157 | 135 |
158 if (channel == AudioDeviceModule::kChannelBoth) { | 136 if (channel == AudioDeviceModule::kChannelBoth) { |
159 // two bytes per channel | 137 // two bytes per channel |
160 _recBytesPerSample = 4; | 138 _recBytesPerSample = 4; |
161 } else { | 139 } else { |
162 // only utilize one out of two possible channels (left or right) | 140 // only utilize one out of two possible channels (left or right) |
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208 } | 186 } |
209 } | 187 } |
210 | 188 |
211 _playDelayMS = playDelayMs; | 189 _playDelayMS = playDelayMs; |
212 _recDelayMS = recDelayMs; | 190 _recDelayMS = recDelayMs; |
213 _clockDrift = clockDrift; | 191 _clockDrift = clockDrift; |
214 } | 192 } |
215 | 193 |
216 int32_t AudioDeviceBuffer::StartInputFileRecording( | 194 int32_t AudioDeviceBuffer::StartInputFileRecording( |
217 const char fileName[kAdmMaxFileNameSize]) { | 195 const char fileName[kAdmMaxFileNameSize]) { |
218 rtc::CritScope lock(&_critSect); | 196 CriticalSectionScoped lock(&_critSect); |
219 | 197 |
220 _recFile.Flush(); | 198 _recFile.Flush(); |
221 _recFile.CloseFile(); | 199 _recFile.CloseFile(); |
222 | 200 |
223 return _recFile.OpenFile(fileName, false) ? 0 : -1; | 201 return _recFile.OpenFile(fileName, false) ? 0 : -1; |
224 } | 202 } |
225 | 203 |
226 int32_t AudioDeviceBuffer::StopInputFileRecording() { | 204 int32_t AudioDeviceBuffer::StopInputFileRecording() { |
227 rtc::CritScope lock(&_critSect); | 205 CriticalSectionScoped lock(&_critSect); |
228 | 206 |
229 _recFile.Flush(); | 207 _recFile.Flush(); |
230 _recFile.CloseFile(); | 208 _recFile.CloseFile(); |
231 | 209 |
232 return 0; | 210 return 0; |
233 } | 211 } |
234 | 212 |
235 int32_t AudioDeviceBuffer::StartOutputFileRecording( | 213 int32_t AudioDeviceBuffer::StartOutputFileRecording( |
236 const char fileName[kAdmMaxFileNameSize]) { | 214 const char fileName[kAdmMaxFileNameSize]) { |
237 rtc::CritScope lock(&_critSect); | 215 CriticalSectionScoped lock(&_critSect); |
238 | 216 |
239 _playFile.Flush(); | 217 _playFile.Flush(); |
240 _playFile.CloseFile(); | 218 _playFile.CloseFile(); |
241 | 219 |
242 return _playFile.OpenFile(fileName, false) ? 0 : -1; | 220 return _playFile.OpenFile(fileName, false) ? 0 : -1; |
243 } | 221 } |
244 | 222 |
245 int32_t AudioDeviceBuffer::StopOutputFileRecording() { | 223 int32_t AudioDeviceBuffer::StopOutputFileRecording() { |
246 rtc::CritScope lock(&_critSect); | 224 CriticalSectionScoped lock(&_critSect); |
247 | 225 |
248 _playFile.Flush(); | 226 _playFile.Flush(); |
249 _playFile.CloseFile(); | 227 _playFile.CloseFile(); |
250 | 228 |
251 return 0; | 229 return 0; |
252 } | 230 } |
253 | 231 |
254 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer, | 232 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer, |
255 size_t nSamples) { | 233 size_t nSamples) { |
256 rtc::CritScope lock(&_critSect); | 234 CriticalSectionScoped lock(&_critSect); |
257 | 235 |
258 if (_recBytesPerSample == 0) { | 236 if (_recBytesPerSample == 0) { |
259 assert(false); | 237 assert(false); |
260 return -1; | 238 return -1; |
261 } | 239 } |
262 | 240 |
263 _recSamples = nSamples; | 241 _recSamples = nSamples; |
264 _recSize = _recBytesPerSample * nSamples; // {2,4}*nSamples | 242 _recSize = _recBytesPerSample * nSamples; // {2,4}*nSamples |
265 if (_recSize > kMaxBufferSizeBytes) { | 243 if (_recSize > kMaxBufferSizeBytes) { |
266 assert(false); | 244 assert(false); |
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285 ptr16In++; | 263 ptr16In++; |
286 ptr16In++; | 264 ptr16In++; |
287 } | 265 } |
288 } | 266 } |
289 | 267 |
290 if (_recFile.is_open()) { | 268 if (_recFile.is_open()) { |
291 // write to binary file in mono or stereo (interleaved) | 269 // write to binary file in mono or stereo (interleaved) |
292 _recFile.Write(&_recBuffer[0], _recSize); | 270 _recFile.Write(&_recBuffer[0], _recSize); |
293 } | 271 } |
294 | 272 |
295 // Update some stats but do it on the task queue to ensure that the members | |
296 // are modified and read on the same thread. | |
297 task_queue_.PostTask( | |
298 rtc::Bind(&AudioDeviceBuffer::UpdateRecStats, this, nSamples)); | |
299 | |
300 return 0; | 273 return 0; |
301 } | 274 } |
302 | 275 |
303 int32_t AudioDeviceBuffer::DeliverRecordedData() { | 276 int32_t AudioDeviceBuffer::DeliverRecordedData() { |
304 rtc::CritScope lock(&_critSectCb); | 277 CriticalSectionScoped lock(&_critSectCb); |
305 // Ensure that user has initialized all essential members | 278 // Ensure that user has initialized all essential members |
306 if ((_recSampleRate == 0) || (_recSamples == 0) || | 279 if ((_recSampleRate == 0) || (_recSamples == 0) || |
307 (_recBytesPerSample == 0) || (_recChannels == 0)) { | 280 (_recBytesPerSample == 0) || (_recChannels == 0)) { |
308 RTC_NOTREACHED(); | 281 RTC_NOTREACHED(); |
309 return -1; | 282 return -1; |
310 } | 283 } |
311 | 284 |
312 if (!_ptrCbAudioTransport) { | 285 if (!_ptrCbAudioTransport) { |
313 LOG(LS_WARNING) << "Invalid audio transport"; | 286 LOG(LS_WARNING) << "Invalid audio transport"; |
314 return 0; | 287 return 0; |
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329 } | 302 } |
330 | 303 |
331 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) { | 304 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) { |
332 uint32_t playSampleRate = 0; | 305 uint32_t playSampleRate = 0; |
333 size_t playBytesPerSample = 0; | 306 size_t playBytesPerSample = 0; |
334 size_t playChannels = 0; | 307 size_t playChannels = 0; |
335 | 308 |
336 // TOOD(henrika): improve bad locking model and make it more clear that only | 309 // TOOD(henrika): improve bad locking model and make it more clear that only |
337 // 10ms buffer sizes is supported in WebRTC. | 310 // 10ms buffer sizes is supported in WebRTC. |
338 { | 311 { |
339 rtc::CritScope lock(&_critSect); | 312 CriticalSectionScoped lock(&_critSect); |
340 | 313 |
341 // Store copies under lock and use copies hereafter to avoid race with | 314 // Store copies under lock and use copies hereafter to avoid race with |
342 // setter methods. | 315 // setter methods. |
343 playSampleRate = _playSampleRate; | 316 playSampleRate = _playSampleRate; |
344 playBytesPerSample = _playBytesPerSample; | 317 playBytesPerSample = _playBytesPerSample; |
345 playChannels = _playChannels; | 318 playChannels = _playChannels; |
346 | 319 |
347 // Ensure that user has initialized all essential members | 320 // Ensure that user has initialized all essential members |
348 if ((playBytesPerSample == 0) || (playChannels == 0) || | 321 if ((playBytesPerSample == 0) || (playChannels == 0) || |
349 (playSampleRate == 0)) { | 322 (playSampleRate == 0)) { |
350 RTC_NOTREACHED(); | 323 RTC_NOTREACHED(); |
351 return -1; | 324 return -1; |
352 } | 325 } |
353 | 326 |
354 _playSamples = nSamples; | 327 _playSamples = nSamples; |
355 _playSize = playBytesPerSample * nSamples; // {2,4}*nSamples | 328 _playSize = playBytesPerSample * nSamples; // {2,4}*nSamples |
356 RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes); | 329 RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes); |
357 RTC_CHECK_EQ(nSamples, _playSamples); | 330 RTC_CHECK_EQ(nSamples, _playSamples); |
358 } | 331 } |
359 | 332 |
360 size_t nSamplesOut(0); | 333 size_t nSamplesOut(0); |
361 | 334 |
362 rtc::CritScope lock(&_critSectCb); | 335 CriticalSectionScoped lock(&_critSectCb); |
363 | 336 |
364 // It is currently supported to start playout without a valid audio | 337 // It is currently supported to start playout without a valid audio |
365 // transport object. Leads to warning and silence. | 338 // transport object. Leads to warning and silence. |
366 if (!_ptrCbAudioTransport) { | 339 if (!_ptrCbAudioTransport) { |
367 LOG(LS_WARNING) << "Invalid audio transport"; | 340 LOG(LS_WARNING) << "Invalid audio transport"; |
368 return 0; | 341 return 0; |
369 } | 342 } |
370 | 343 |
371 uint32_t res(0); | 344 uint32_t res(0); |
372 int64_t elapsed_time_ms = -1; | 345 int64_t elapsed_time_ms = -1; |
373 int64_t ntp_time_ms = -1; | 346 int64_t ntp_time_ms = -1; |
374 res = _ptrCbAudioTransport->NeedMorePlayData( | 347 res = _ptrCbAudioTransport->NeedMorePlayData( |
375 _playSamples, playBytesPerSample, playChannels, playSampleRate, | 348 _playSamples, playBytesPerSample, playChannels, playSampleRate, |
376 &_playBuffer[0], nSamplesOut, &elapsed_time_ms, &ntp_time_ms); | 349 &_playBuffer[0], nSamplesOut, &elapsed_time_ms, &ntp_time_ms); |
377 if (res != 0) { | 350 if (res != 0) { |
378 LOG(LS_ERROR) << "NeedMorePlayData() failed"; | 351 LOG(LS_ERROR) << "NeedMorePlayData() failed"; |
379 } | 352 } |
380 | 353 |
381 // Update some stats but do it on the task queue to ensure that access of | |
382 // members is serialized hence avoiding usage of locks. | |
383 task_queue_.PostTask( | |
384 rtc::Bind(&AudioDeviceBuffer::UpdatePlayStats, this, nSamplesOut)); | |
385 | |
386 return static_cast<int32_t>(nSamplesOut); | 354 return static_cast<int32_t>(nSamplesOut); |
387 } | 355 } |
388 | 356 |
389 int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) { | 357 int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) { |
390 rtc::CritScope lock(&_critSect); | 358 CriticalSectionScoped lock(&_critSect); |
391 RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes); | 359 RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes); |
392 | 360 |
393 memcpy(audioBuffer, &_playBuffer[0], _playSize); | 361 memcpy(audioBuffer, &_playBuffer[0], _playSize); |
394 | 362 |
395 if (_playFile.is_open()) { | 363 if (_playFile.is_open()) { |
396 // write to binary file in mono or stereo (interleaved) | 364 // write to binary file in mono or stereo (interleaved) |
397 _playFile.Write(&_playBuffer[0], _playSize); | 365 _playFile.Write(&_playBuffer[0], _playSize); |
398 } | 366 } |
399 | 367 |
400 return static_cast<int32_t>(_playSamples); | 368 return static_cast<int32_t>(_playSamples); |
401 } | 369 } |
402 | 370 |
403 void AudioDeviceBuffer::StartTimer() { | |
404 last_log_stat_time_ = rtc::TimeMillis(); | |
405 task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this), | |
406 kTimerIntervalInMilliseconds); | |
407 } | |
408 | |
409 void AudioDeviceBuffer::LogStats() { | |
410 RTC_DCHECK(task_queue_.IsCurrent()); | |
411 | |
412 int64_t now_time = rtc::TimeMillis(); | |
413 int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds; | |
414 int64_t time_since_last = rtc::TimeDiff(now_time, last_log_stat_time_); | |
415 last_log_stat_time_ = now_time; | |
416 | |
417 // Log the latest statistics but skip the first 10 seconds since we are not | |
418 // sure of the exact starting point. I.e., the first log printout will be | |
419 // after ~20 seconds. | |
420 if (++num_stat_reports_ > 1) { | |
421 uint32_t diff_samples = rec_samples_ - last_rec_samples_; | |
422 uint32_t rate = diff_samples / kTimerIntervalInSeconds; | |
423 LOG(INFO) << "[REC : " << time_since_last << "msec, " | |
424 << _recSampleRate / 1000 | |
425 << "kHz] callbacks: " << rec_callbacks_ - last_rec_callbacks_ | |
426 << ", " | |
427 << "samples: " << diff_samples << ", " | |
428 << "rate: " << rate; | |
429 | |
430 diff_samples = play_samples_ - last_play_samples_; | |
431 rate = diff_samples / kTimerIntervalInSeconds; | |
432 LOG(INFO) << "[PLAY: " << time_since_last << "msec, " | |
433 << _playSampleRate / 1000 | |
434 << "kHz] callbacks: " << play_callbacks_ - last_play_callbacks_ | |
435 << ", " | |
436 << "samples: " << diff_samples << ", " | |
437 << "rate: " << rate; | |
438 } | |
439 | |
440 last_rec_callbacks_ = rec_callbacks_; | |
441 last_play_callbacks_ = play_callbacks_; | |
442 last_rec_samples_ = rec_samples_; | |
443 last_play_samples_ = play_samples_; | |
444 | |
445 int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis(); | |
446 RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval"; | |
447 | |
448 // Update some stats but do it on the task queue to ensure that access of | |
449 // members is serialized hence avoiding usage of locks. | |
450 task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this), | |
451 time_to_wait_ms); | |
452 } | |
453 | |
454 void AudioDeviceBuffer::UpdateRecStats(size_t num_samples) { | |
455 RTC_DCHECK(task_queue_.IsCurrent()); | |
456 ++rec_callbacks_; | |
457 rec_samples_ += num_samples; | |
458 } | |
459 | |
460 void AudioDeviceBuffer::UpdatePlayStats(size_t num_samples) { | |
461 RTC_DCHECK(task_queue_.IsCurrent()); | |
462 ++play_callbacks_; | |
463 play_samples_ += num_samples; | |
464 } | |
465 | |
466 } // namespace webrtc | 371 } // namespace webrtc |
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