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1 /* | |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #include "webrtc/api/test/fakeaudiocapturemodule.h" | |
12 | |
13 #include "webrtc/base/checks.h" | |
14 #include "webrtc/base/common.h" | |
15 #include "webrtc/base/refcount.h" | |
16 #include "webrtc/base/thread.h" | |
17 #include "webrtc/base/timeutils.h" | |
18 | |
19 // Audio sample value that is high enough that it doesn't occur naturally when | |
20 // frames are being faked. E.g. NetEq will not generate this large sample value | |
21 // unless it has received an audio frame containing a sample of this value. | |
22 // Even simpler buffers would likely just contain audio sample values of 0. | |
23 static const int kHighSampleValue = 10000; | |
24 | |
25 // Same value as src/modules/audio_device/main/source/audio_device_config.h in | |
26 // https://code.google.com/p/webrtc/ | |
27 static const int kAdmMaxIdleTimeProcess = 1000; | |
28 | |
29 // Constants here are derived by running VoE using a real ADM. | |
30 // The constants correspond to 10ms of mono audio at 44kHz. | |
31 static const int kTimePerFrameMs = 10; | |
32 static const uint8_t kNumberOfChannels = 1; | |
33 static const int kSamplesPerSecond = 44000; | |
34 static const int kTotalDelayMs = 0; | |
35 static const int kClockDriftMs = 0; | |
36 static const uint32_t kMaxVolume = 14392; | |
37 | |
38 enum { | |
39 MSG_START_PROCESS, | |
40 MSG_RUN_PROCESS, | |
41 }; | |
42 | |
43 FakeAudioCaptureModule::FakeAudioCaptureModule() | |
44 : last_process_time_ms_(0), | |
45 audio_callback_(nullptr), | |
46 recording_(false), | |
47 playing_(false), | |
48 play_is_initialized_(false), | |
49 rec_is_initialized_(false), | |
50 current_mic_level_(kMaxVolume), | |
51 started_(false), | |
52 next_frame_time_(0), | |
53 frames_received_(0) { | |
54 } | |
55 | |
56 FakeAudioCaptureModule::~FakeAudioCaptureModule() { | |
57 if (process_thread_) { | |
58 process_thread_->Stop(); | |
59 } | |
60 } | |
61 | |
62 rtc::scoped_refptr<FakeAudioCaptureModule> FakeAudioCaptureModule::Create() { | |
63 rtc::scoped_refptr<FakeAudioCaptureModule> capture_module( | |
64 new rtc::RefCountedObject<FakeAudioCaptureModule>()); | |
65 if (!capture_module->Initialize()) { | |
66 return nullptr; | |
67 } | |
68 return capture_module; | |
69 } | |
70 | |
71 int FakeAudioCaptureModule::frames_received() const { | |
72 rtc::CritScope cs(&crit_); | |
73 return frames_received_; | |
74 } | |
75 | |
76 int64_t FakeAudioCaptureModule::TimeUntilNextProcess() { | |
77 const int64_t current_time = rtc::TimeMillis(); | |
78 if (current_time < last_process_time_ms_) { | |
79 // TODO: wraparound could be handled more gracefully. | |
80 return 0; | |
81 } | |
82 const int64_t elapsed_time = current_time - last_process_time_ms_; | |
83 if (kAdmMaxIdleTimeProcess < elapsed_time) { | |
84 return 0; | |
85 } | |
86 return kAdmMaxIdleTimeProcess - elapsed_time; | |
87 } | |
88 | |
89 void FakeAudioCaptureModule::Process() { | |
90 last_process_time_ms_ = rtc::TimeMillis(); | |
91 } | |
92 | |
93 int32_t FakeAudioCaptureModule::ActiveAudioLayer( | |
94 AudioLayer* /*audio_layer*/) const { | |
95 RTC_NOTREACHED(); | |
96 return 0; | |
97 } | |
98 | |
99 webrtc::AudioDeviceModule::ErrorCode FakeAudioCaptureModule::LastError() const { | |
100 RTC_NOTREACHED(); | |
101 return webrtc::AudioDeviceModule::kAdmErrNone; | |
102 } | |
103 | |
104 int32_t FakeAudioCaptureModule::RegisterEventObserver( | |
105 webrtc::AudioDeviceObserver* /*event_callback*/) { | |
106 // Only used to report warnings and errors. This fake implementation won't | |
107 // generate any so discard this callback. | |
108 return 0; | |
109 } | |
110 | |
111 int32_t FakeAudioCaptureModule::RegisterAudioCallback( | |
112 webrtc::AudioTransport* audio_callback) { | |
113 rtc::CritScope cs(&crit_callback_); | |
114 audio_callback_ = audio_callback; | |
115 return 0; | |
116 } | |
117 | |
118 int32_t FakeAudioCaptureModule::Init() { | |
119 // Initialize is called by the factory method. Safe to ignore this Init call. | |
120 return 0; | |
121 } | |
122 | |
123 int32_t FakeAudioCaptureModule::Terminate() { | |
124 // Clean up in the destructor. No action here, just success. | |
125 return 0; | |
126 } | |
127 | |
128 bool FakeAudioCaptureModule::Initialized() const { | |
129 RTC_NOTREACHED(); | |
130 return 0; | |
131 } | |
132 | |
133 int16_t FakeAudioCaptureModule::PlayoutDevices() { | |
134 RTC_NOTREACHED(); | |
135 return 0; | |
136 } | |
137 | |
138 int16_t FakeAudioCaptureModule::RecordingDevices() { | |
139 RTC_NOTREACHED(); | |
140 return 0; | |
141 } | |
142 | |
143 int32_t FakeAudioCaptureModule::PlayoutDeviceName( | |
144 uint16_t /*index*/, | |
145 char /*name*/[webrtc::kAdmMaxDeviceNameSize], | |
146 char /*guid*/[webrtc::kAdmMaxGuidSize]) { | |
147 RTC_NOTREACHED(); | |
148 return 0; | |
149 } | |
150 | |
151 int32_t FakeAudioCaptureModule::RecordingDeviceName( | |
152 uint16_t /*index*/, | |
153 char /*name*/[webrtc::kAdmMaxDeviceNameSize], | |
154 char /*guid*/[webrtc::kAdmMaxGuidSize]) { | |
155 RTC_NOTREACHED(); | |
156 return 0; | |
157 } | |
158 | |
159 int32_t FakeAudioCaptureModule::SetPlayoutDevice(uint16_t /*index*/) { | |
160 // No playout device, just playing from file. Return success. | |
161 return 0; | |
162 } | |
163 | |
164 int32_t FakeAudioCaptureModule::SetPlayoutDevice(WindowsDeviceType /*device*/) { | |
165 if (play_is_initialized_) { | |
166 return -1; | |
167 } | |
168 return 0; | |
169 } | |
170 | |
171 int32_t FakeAudioCaptureModule::SetRecordingDevice(uint16_t /*index*/) { | |
172 // No recording device, just dropping audio. Return success. | |
173 return 0; | |
174 } | |
175 | |
176 int32_t FakeAudioCaptureModule::SetRecordingDevice( | |
177 WindowsDeviceType /*device*/) { | |
178 if (rec_is_initialized_) { | |
179 return -1; | |
180 } | |
181 return 0; | |
182 } | |
183 | |
184 int32_t FakeAudioCaptureModule::PlayoutIsAvailable(bool* /*available*/) { | |
185 RTC_NOTREACHED(); | |
186 return 0; | |
187 } | |
188 | |
189 int32_t FakeAudioCaptureModule::InitPlayout() { | |
190 play_is_initialized_ = true; | |
191 return 0; | |
192 } | |
193 | |
194 bool FakeAudioCaptureModule::PlayoutIsInitialized() const { | |
195 return play_is_initialized_; | |
196 } | |
197 | |
198 int32_t FakeAudioCaptureModule::RecordingIsAvailable(bool* /*available*/) { | |
199 RTC_NOTREACHED(); | |
200 return 0; | |
201 } | |
202 | |
203 int32_t FakeAudioCaptureModule::InitRecording() { | |
204 rec_is_initialized_ = true; | |
205 return 0; | |
206 } | |
207 | |
208 bool FakeAudioCaptureModule::RecordingIsInitialized() const { | |
209 return rec_is_initialized_; | |
210 } | |
211 | |
212 int32_t FakeAudioCaptureModule::StartPlayout() { | |
213 if (!play_is_initialized_) { | |
214 return -1; | |
215 } | |
216 { | |
217 rtc::CritScope cs(&crit_); | |
218 playing_ = true; | |
219 } | |
220 bool start = true; | |
221 UpdateProcessing(start); | |
222 return 0; | |
223 } | |
224 | |
225 int32_t FakeAudioCaptureModule::StopPlayout() { | |
226 bool start = false; | |
227 { | |
228 rtc::CritScope cs(&crit_); | |
229 playing_ = false; | |
230 start = ShouldStartProcessing(); | |
231 } | |
232 UpdateProcessing(start); | |
233 return 0; | |
234 } | |
235 | |
236 bool FakeAudioCaptureModule::Playing() const { | |
237 rtc::CritScope cs(&crit_); | |
238 return playing_; | |
239 } | |
240 | |
241 int32_t FakeAudioCaptureModule::StartRecording() { | |
242 if (!rec_is_initialized_) { | |
243 return -1; | |
244 } | |
245 { | |
246 rtc::CritScope cs(&crit_); | |
247 recording_ = true; | |
248 } | |
249 bool start = true; | |
250 UpdateProcessing(start); | |
251 return 0; | |
252 } | |
253 | |
254 int32_t FakeAudioCaptureModule::StopRecording() { | |
255 bool start = false; | |
256 { | |
257 rtc::CritScope cs(&crit_); | |
258 recording_ = false; | |
259 start = ShouldStartProcessing(); | |
260 } | |
261 UpdateProcessing(start); | |
262 return 0; | |
263 } | |
264 | |
265 bool FakeAudioCaptureModule::Recording() const { | |
266 rtc::CritScope cs(&crit_); | |
267 return recording_; | |
268 } | |
269 | |
270 int32_t FakeAudioCaptureModule::SetAGC(bool /*enable*/) { | |
271 // No AGC but not needed since audio is pregenerated. Return success. | |
272 return 0; | |
273 } | |
274 | |
275 bool FakeAudioCaptureModule::AGC() const { | |
276 RTC_NOTREACHED(); | |
277 return 0; | |
278 } | |
279 | |
280 int32_t FakeAudioCaptureModule::SetWaveOutVolume(uint16_t /*volume_left*/, | |
281 uint16_t /*volume_right*/) { | |
282 RTC_NOTREACHED(); | |
283 return 0; | |
284 } | |
285 | |
286 int32_t FakeAudioCaptureModule::WaveOutVolume( | |
287 uint16_t* /*volume_left*/, | |
288 uint16_t* /*volume_right*/) const { | |
289 RTC_NOTREACHED(); | |
290 return 0; | |
291 } | |
292 | |
293 int32_t FakeAudioCaptureModule::InitSpeaker() { | |
294 // No speaker, just playing from file. Return success. | |
295 return 0; | |
296 } | |
297 | |
298 bool FakeAudioCaptureModule::SpeakerIsInitialized() const { | |
299 RTC_NOTREACHED(); | |
300 return 0; | |
301 } | |
302 | |
303 int32_t FakeAudioCaptureModule::InitMicrophone() { | |
304 // No microphone, just playing from file. Return success. | |
305 return 0; | |
306 } | |
307 | |
308 bool FakeAudioCaptureModule::MicrophoneIsInitialized() const { | |
309 RTC_NOTREACHED(); | |
310 return 0; | |
311 } | |
312 | |
313 int32_t FakeAudioCaptureModule::SpeakerVolumeIsAvailable(bool* /*available*/) { | |
314 RTC_NOTREACHED(); | |
315 return 0; | |
316 } | |
317 | |
318 int32_t FakeAudioCaptureModule::SetSpeakerVolume(uint32_t /*volume*/) { | |
319 RTC_NOTREACHED(); | |
320 return 0; | |
321 } | |
322 | |
323 int32_t FakeAudioCaptureModule::SpeakerVolume(uint32_t* /*volume*/) const { | |
324 RTC_NOTREACHED(); | |
325 return 0; | |
326 } | |
327 | |
328 int32_t FakeAudioCaptureModule::MaxSpeakerVolume( | |
329 uint32_t* /*max_volume*/) const { | |
330 RTC_NOTREACHED(); | |
331 return 0; | |
332 } | |
333 | |
334 int32_t FakeAudioCaptureModule::MinSpeakerVolume( | |
335 uint32_t* /*min_volume*/) const { | |
336 RTC_NOTREACHED(); | |
337 return 0; | |
338 } | |
339 | |
340 int32_t FakeAudioCaptureModule::SpeakerVolumeStepSize( | |
341 uint16_t* /*step_size*/) const { | |
342 RTC_NOTREACHED(); | |
343 return 0; | |
344 } | |
345 | |
346 int32_t FakeAudioCaptureModule::MicrophoneVolumeIsAvailable( | |
347 bool* /*available*/) { | |
348 RTC_NOTREACHED(); | |
349 return 0; | |
350 } | |
351 | |
352 int32_t FakeAudioCaptureModule::SetMicrophoneVolume(uint32_t volume) { | |
353 rtc::CritScope cs(&crit_); | |
354 current_mic_level_ = volume; | |
355 return 0; | |
356 } | |
357 | |
358 int32_t FakeAudioCaptureModule::MicrophoneVolume(uint32_t* volume) const { | |
359 rtc::CritScope cs(&crit_); | |
360 *volume = current_mic_level_; | |
361 return 0; | |
362 } | |
363 | |
364 int32_t FakeAudioCaptureModule::MaxMicrophoneVolume( | |
365 uint32_t* max_volume) const { | |
366 *max_volume = kMaxVolume; | |
367 return 0; | |
368 } | |
369 | |
370 int32_t FakeAudioCaptureModule::MinMicrophoneVolume( | |
371 uint32_t* /*min_volume*/) const { | |
372 RTC_NOTREACHED(); | |
373 return 0; | |
374 } | |
375 | |
376 int32_t FakeAudioCaptureModule::MicrophoneVolumeStepSize( | |
377 uint16_t* /*step_size*/) const { | |
378 RTC_NOTREACHED(); | |
379 return 0; | |
380 } | |
381 | |
382 int32_t FakeAudioCaptureModule::SpeakerMuteIsAvailable(bool* /*available*/) { | |
383 RTC_NOTREACHED(); | |
384 return 0; | |
385 } | |
386 | |
387 int32_t FakeAudioCaptureModule::SetSpeakerMute(bool /*enable*/) { | |
388 RTC_NOTREACHED(); | |
389 return 0; | |
390 } | |
391 | |
392 int32_t FakeAudioCaptureModule::SpeakerMute(bool* /*enabled*/) const { | |
393 RTC_NOTREACHED(); | |
394 return 0; | |
395 } | |
396 | |
397 int32_t FakeAudioCaptureModule::MicrophoneMuteIsAvailable(bool* /*available*/) { | |
398 RTC_NOTREACHED(); | |
399 return 0; | |
400 } | |
401 | |
402 int32_t FakeAudioCaptureModule::SetMicrophoneMute(bool /*enable*/) { | |
403 RTC_NOTREACHED(); | |
404 return 0; | |
405 } | |
406 | |
407 int32_t FakeAudioCaptureModule::MicrophoneMute(bool* /*enabled*/) const { | |
408 RTC_NOTREACHED(); | |
409 return 0; | |
410 } | |
411 | |
412 int32_t FakeAudioCaptureModule::MicrophoneBoostIsAvailable( | |
413 bool* /*available*/) { | |
414 RTC_NOTREACHED(); | |
415 return 0; | |
416 } | |
417 | |
418 int32_t FakeAudioCaptureModule::SetMicrophoneBoost(bool /*enable*/) { | |
419 RTC_NOTREACHED(); | |
420 return 0; | |
421 } | |
422 | |
423 int32_t FakeAudioCaptureModule::MicrophoneBoost(bool* /*enabled*/) const { | |
424 RTC_NOTREACHED(); | |
425 return 0; | |
426 } | |
427 | |
428 int32_t FakeAudioCaptureModule::StereoPlayoutIsAvailable( | |
429 bool* available) const { | |
430 // No recording device, just dropping audio. Stereo can be dropped just | |
431 // as easily as mono. | |
432 *available = true; | |
433 return 0; | |
434 } | |
435 | |
436 int32_t FakeAudioCaptureModule::SetStereoPlayout(bool /*enable*/) { | |
437 // No recording device, just dropping audio. Stereo can be dropped just | |
438 // as easily as mono. | |
439 return 0; | |
440 } | |
441 | |
442 int32_t FakeAudioCaptureModule::StereoPlayout(bool* /*enabled*/) const { | |
443 RTC_NOTREACHED(); | |
444 return 0; | |
445 } | |
446 | |
447 int32_t FakeAudioCaptureModule::StereoRecordingIsAvailable( | |
448 bool* available) const { | |
449 // Keep thing simple. No stereo recording. | |
450 *available = false; | |
451 return 0; | |
452 } | |
453 | |
454 int32_t FakeAudioCaptureModule::SetStereoRecording(bool enable) { | |
455 if (!enable) { | |
456 return 0; | |
457 } | |
458 return -1; | |
459 } | |
460 | |
461 int32_t FakeAudioCaptureModule::StereoRecording(bool* /*enabled*/) const { | |
462 RTC_NOTREACHED(); | |
463 return 0; | |
464 } | |
465 | |
466 int32_t FakeAudioCaptureModule::SetRecordingChannel( | |
467 const ChannelType channel) { | |
468 if (channel != AudioDeviceModule::kChannelBoth) { | |
469 // There is no right or left in mono. I.e. kChannelBoth should be used for | |
470 // mono. | |
471 RTC_NOTREACHED(); | |
472 return -1; | |
473 } | |
474 return 0; | |
475 } | |
476 | |
477 int32_t FakeAudioCaptureModule::RecordingChannel(ChannelType* channel) const { | |
478 // Stereo recording not supported. However, WebRTC ADM returns kChannelBoth | |
479 // in that case. Do the same here. | |
480 *channel = AudioDeviceModule::kChannelBoth; | |
481 return 0; | |
482 } | |
483 | |
484 int32_t FakeAudioCaptureModule::SetPlayoutBuffer(const BufferType /*type*/, | |
485 uint16_t /*size_ms*/) { | |
486 RTC_NOTREACHED(); | |
487 return 0; | |
488 } | |
489 | |
490 int32_t FakeAudioCaptureModule::PlayoutBuffer(BufferType* /*type*/, | |
491 uint16_t* /*size_ms*/) const { | |
492 RTC_NOTREACHED(); | |
493 return 0; | |
494 } | |
495 | |
496 int32_t FakeAudioCaptureModule::PlayoutDelay(uint16_t* delay_ms) const { | |
497 // No delay since audio frames are dropped. | |
498 *delay_ms = 0; | |
499 return 0; | |
500 } | |
501 | |
502 int32_t FakeAudioCaptureModule::RecordingDelay(uint16_t* /*delay_ms*/) const { | |
503 RTC_NOTREACHED(); | |
504 return 0; | |
505 } | |
506 | |
507 int32_t FakeAudioCaptureModule::CPULoad(uint16_t* /*load*/) const { | |
508 RTC_NOTREACHED(); | |
509 return 0; | |
510 } | |
511 | |
512 int32_t FakeAudioCaptureModule::StartRawOutputFileRecording( | |
513 const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) { | |
514 RTC_NOTREACHED(); | |
515 return 0; | |
516 } | |
517 | |
518 int32_t FakeAudioCaptureModule::StopRawOutputFileRecording() { | |
519 RTC_NOTREACHED(); | |
520 return 0; | |
521 } | |
522 | |
523 int32_t FakeAudioCaptureModule::StartRawInputFileRecording( | |
524 const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) { | |
525 RTC_NOTREACHED(); | |
526 return 0; | |
527 } | |
528 | |
529 int32_t FakeAudioCaptureModule::StopRawInputFileRecording() { | |
530 RTC_NOTREACHED(); | |
531 return 0; | |
532 } | |
533 | |
534 int32_t FakeAudioCaptureModule::SetRecordingSampleRate( | |
535 const uint32_t /*samples_per_sec*/) { | |
536 RTC_NOTREACHED(); | |
537 return 0; | |
538 } | |
539 | |
540 int32_t FakeAudioCaptureModule::RecordingSampleRate( | |
541 uint32_t* /*samples_per_sec*/) const { | |
542 RTC_NOTREACHED(); | |
543 return 0; | |
544 } | |
545 | |
546 int32_t FakeAudioCaptureModule::SetPlayoutSampleRate( | |
547 const uint32_t /*samples_per_sec*/) { | |
548 RTC_NOTREACHED(); | |
549 return 0; | |
550 } | |
551 | |
552 int32_t FakeAudioCaptureModule::PlayoutSampleRate( | |
553 uint32_t* /*samples_per_sec*/) const { | |
554 RTC_NOTREACHED(); | |
555 return 0; | |
556 } | |
557 | |
558 int32_t FakeAudioCaptureModule::ResetAudioDevice() { | |
559 RTC_NOTREACHED(); | |
560 return 0; | |
561 } | |
562 | |
563 int32_t FakeAudioCaptureModule::SetLoudspeakerStatus(bool /*enable*/) { | |
564 RTC_NOTREACHED(); | |
565 return 0; | |
566 } | |
567 | |
568 int32_t FakeAudioCaptureModule::GetLoudspeakerStatus(bool* /*enabled*/) const { | |
569 RTC_NOTREACHED(); | |
570 return 0; | |
571 } | |
572 | |
573 void FakeAudioCaptureModule::OnMessage(rtc::Message* msg) { | |
574 switch (msg->message_id) { | |
575 case MSG_START_PROCESS: | |
576 StartProcessP(); | |
577 break; | |
578 case MSG_RUN_PROCESS: | |
579 ProcessFrameP(); | |
580 break; | |
581 default: | |
582 // All existing messages should be caught. Getting here should never | |
583 // happen. | |
584 RTC_NOTREACHED(); | |
585 } | |
586 } | |
587 | |
588 bool FakeAudioCaptureModule::Initialize() { | |
589 // Set the send buffer samples high enough that it would not occur on the | |
590 // remote side unless a packet containing a sample of that magnitude has been | |
591 // sent to it. Note that the audio processing pipeline will likely distort the | |
592 // original signal. | |
593 SetSendBuffer(kHighSampleValue); | |
594 last_process_time_ms_ = rtc::TimeMillis(); | |
595 return true; | |
596 } | |
597 | |
598 void FakeAudioCaptureModule::SetSendBuffer(int value) { | |
599 Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_); | |
600 const size_t buffer_size_in_samples = | |
601 sizeof(send_buffer_) / kNumberBytesPerSample; | |
602 for (size_t i = 0; i < buffer_size_in_samples; ++i) { | |
603 buffer_ptr[i] = value; | |
604 } | |
605 } | |
606 | |
607 void FakeAudioCaptureModule::ResetRecBuffer() { | |
608 memset(rec_buffer_, 0, sizeof(rec_buffer_)); | |
609 } | |
610 | |
611 bool FakeAudioCaptureModule::CheckRecBuffer(int value) { | |
612 const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_); | |
613 const size_t buffer_size_in_samples = | |
614 sizeof(rec_buffer_) / kNumberBytesPerSample; | |
615 for (size_t i = 0; i < buffer_size_in_samples; ++i) { | |
616 if (buffer_ptr[i] >= value) return true; | |
617 } | |
618 return false; | |
619 } | |
620 | |
621 bool FakeAudioCaptureModule::ShouldStartProcessing() { | |
622 return recording_ || playing_; | |
623 } | |
624 | |
625 void FakeAudioCaptureModule::UpdateProcessing(bool start) { | |
626 if (start) { | |
627 if (!process_thread_) { | |
628 process_thread_.reset(new rtc::Thread()); | |
629 process_thread_->Start(); | |
630 } | |
631 process_thread_->Post(RTC_FROM_HERE, this, MSG_START_PROCESS); | |
632 } else { | |
633 if (process_thread_) { | |
634 process_thread_->Stop(); | |
635 process_thread_.reset(nullptr); | |
636 } | |
637 started_ = false; | |
638 } | |
639 } | |
640 | |
641 void FakeAudioCaptureModule::StartProcessP() { | |
642 RTC_CHECK(process_thread_->IsCurrent()); | |
643 if (started_) { | |
644 // Already started. | |
645 return; | |
646 } | |
647 ProcessFrameP(); | |
648 } | |
649 | |
650 void FakeAudioCaptureModule::ProcessFrameP() { | |
651 RTC_CHECK(process_thread_->IsCurrent()); | |
652 if (!started_) { | |
653 next_frame_time_ = rtc::TimeMillis(); | |
654 started_ = true; | |
655 } | |
656 | |
657 { | |
658 rtc::CritScope cs(&crit_); | |
659 // Receive and send frames every kTimePerFrameMs. | |
660 if (playing_) { | |
661 ReceiveFrameP(); | |
662 } | |
663 if (recording_) { | |
664 SendFrameP(); | |
665 } | |
666 } | |
667 | |
668 next_frame_time_ += kTimePerFrameMs; | |
669 const int64_t current_time = rtc::TimeMillis(); | |
670 const int64_t wait_time = | |
671 (next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0; | |
672 process_thread_->PostDelayed(RTC_FROM_HERE, wait_time, this, MSG_RUN_PROCESS); | |
673 } | |
674 | |
675 void FakeAudioCaptureModule::ReceiveFrameP() { | |
676 RTC_CHECK(process_thread_->IsCurrent()); | |
677 { | |
678 rtc::CritScope cs(&crit_callback_); | |
679 if (!audio_callback_) { | |
680 return; | |
681 } | |
682 ResetRecBuffer(); | |
683 size_t nSamplesOut = 0; | |
684 int64_t elapsed_time_ms = 0; | |
685 int64_t ntp_time_ms = 0; | |
686 if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample, | |
687 kNumberOfChannels, kSamplesPerSecond, | |
688 rec_buffer_, nSamplesOut, | |
689 &elapsed_time_ms, &ntp_time_ms) != 0) { | |
690 RTC_NOTREACHED(); | |
691 } | |
692 RTC_CHECK(nSamplesOut == kNumberSamples); | |
693 } | |
694 // The SetBuffer() function ensures that after decoding, the audio buffer | |
695 // should contain samples of similar magnitude (there is likely to be some | |
696 // distortion due to the audio pipeline). If one sample is detected to | |
697 // have the same or greater magnitude somewhere in the frame, an actual frame | |
698 // has been received from the remote side (i.e. faked frames are not being | |
699 // pulled). | |
700 if (CheckRecBuffer(kHighSampleValue)) { | |
701 rtc::CritScope cs(&crit_); | |
702 ++frames_received_; | |
703 } | |
704 } | |
705 | |
706 void FakeAudioCaptureModule::SendFrameP() { | |
707 RTC_CHECK(process_thread_->IsCurrent()); | |
708 rtc::CritScope cs(&crit_callback_); | |
709 if (!audio_callback_) { | |
710 return; | |
711 } | |
712 bool key_pressed = false; | |
713 uint32_t current_mic_level = 0; | |
714 MicrophoneVolume(¤t_mic_level); | |
715 if (audio_callback_->RecordedDataIsAvailable(send_buffer_, kNumberSamples, | |
716 kNumberBytesPerSample, | |
717 kNumberOfChannels, | |
718 kSamplesPerSecond, kTotalDelayMs, | |
719 kClockDriftMs, current_mic_level, | |
720 key_pressed, | |
721 current_mic_level) != 0) { | |
722 RTC_NOTREACHED(); | |
723 } | |
724 SetMicrophoneVolume(current_mic_level); | |
725 } | |
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