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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 #include <algorithm> | 11 #include <algorithm> |
12 | 12 |
13 #include "webrtc/modules/audio_device/audio_device_buffer.h" | 13 #include "webrtc/modules/audio_device/audio_device_buffer.h" |
14 | 14 |
15 #include "webrtc/base/arraysize.h" | 15 #include "webrtc/base/arraysize.h" |
16 #include "webrtc/base/bind.h" | 16 #include "webrtc/base/bind.h" |
17 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
18 #include "webrtc/base/logging.h" | 18 #include "webrtc/base/logging.h" |
19 #include "webrtc/base/format_macros.h" | 19 #include "webrtc/base/format_macros.h" |
20 #include "webrtc/base/timeutils.h" | 20 #include "webrtc/base/timeutils.h" |
21 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 21 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
22 #include "webrtc/modules/audio_device/audio_device_config.h" | 22 #include "webrtc/modules/audio_device/audio_device_config.h" |
23 #include "webrtc/system_wrappers/include/metrics.h" | 23 #include "webrtc/system_wrappers/include/metrics.h" |
24 | 24 |
25 #include "webrtc/base/platform_thread.h" | |
26 | |
27 namespace webrtc { | 25 namespace webrtc { |
28 | 26 |
29 static const char kTimerQueueName[] = "AudioDeviceBufferTimer"; | 27 static const char kTimerQueueName[] = "AudioDeviceBufferTimer"; |
30 | 28 |
31 // Time between two sucessive calls to LogStats(). | 29 // Time between two sucessive calls to LogStats(). |
32 static const size_t kTimerIntervalInSeconds = 10; | 30 static const size_t kTimerIntervalInSeconds = 10; |
33 static const size_t kTimerIntervalInMilliseconds = | 31 static const size_t kTimerIntervalInMilliseconds = |
34 kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec; | 32 kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec; |
35 // Min time required to qualify an audio session as a "call". If playout or | 33 // Min time required to qualify an audio session as a "call". If playout or |
36 // recording has been active for less than this time we will not store any | 34 // recording has been active for less than this time we will not store any |
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294 LOG(LS_WARNING) << "Not implemented"; | 292 LOG(LS_WARNING) << "Not implemented"; |
295 return 0; | 293 return 0; |
296 } | 294 } |
297 | 295 |
298 int32_t AudioDeviceBuffer::StopOutputFileRecording() { | 296 int32_t AudioDeviceBuffer::StopOutputFileRecording() { |
299 LOG(LS_WARNING) << "Not implemented"; | 297 LOG(LS_WARNING) << "Not implemented"; |
300 return 0; | 298 return 0; |
301 } | 299 } |
302 | 300 |
303 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer, | 301 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer, |
304 size_t num_samples) { | 302 size_t samples_per_channel) { |
305 RTC_DCHECK_RUN_ON(&recording_thread_checker_); | 303 RTC_DCHECK_RUN_ON(&recording_thread_checker_); |
306 // Copy the complete input buffer to the local buffer. | 304 // Copy the complete input buffer to the local buffer. |
307 const size_t size_in_bytes = num_samples * rec_channels_ * sizeof(int16_t); | |
308 const size_t old_size = rec_buffer_.size(); | 305 const size_t old_size = rec_buffer_.size(); |
309 rec_buffer_.SetData(static_cast<const uint8_t*>(audio_buffer), size_in_bytes); | 306 rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer), |
| 307 rec_channels_ * samples_per_channel); |
310 // Keep track of the size of the recording buffer. Only updated when the | 308 // Keep track of the size of the recording buffer. Only updated when the |
311 // size changes, which is a rare event. | 309 // size changes, which is a rare event. |
312 if (old_size != rec_buffer_.size()) { | 310 if (old_size != rec_buffer_.size()) { |
313 LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size(); | 311 LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size(); |
314 } | 312 } |
| 313 |
315 // Derive a new level value twice per second and check if it is non-zero. | 314 // Derive a new level value twice per second and check if it is non-zero. |
316 int16_t max_abs = 0; | 315 int16_t max_abs = 0; |
317 RTC_DCHECK_LT(rec_stat_count_, 50); | 316 RTC_DCHECK_LT(rec_stat_count_, 50); |
318 if (++rec_stat_count_ >= 50) { | 317 if (++rec_stat_count_ >= 50) { |
319 const size_t size = num_samples * rec_channels_; | |
320 // Returns the largest absolute value in a signed 16-bit vector. | 318 // Returns the largest absolute value in a signed 16-bit vector. |
321 max_abs = WebRtcSpl_MaxAbsValueW16( | 319 max_abs = WebRtcSpl_MaxAbsValueW16(rec_buffer_.data(), rec_buffer_.size()); |
322 reinterpret_cast<const int16_t*>(rec_buffer_.data()), size); | |
323 rec_stat_count_ = 0; | 320 rec_stat_count_ = 0; |
324 // Set |only_silence_recorded_| to false as soon as at least one detection | 321 // Set |only_silence_recorded_| to false as soon as at least one detection |
325 // of a non-zero audio packet is found. It can only be restored to true | 322 // of a non-zero audio packet is found. It can only be restored to true |
326 // again by restarting the call. | 323 // again by restarting the call. |
327 if (max_abs > 0) { | 324 if (max_abs > 0) { |
328 only_silence_recorded_ = false; | 325 only_silence_recorded_ = false; |
329 } | 326 } |
330 } | 327 } |
331 // Update some stats but do it on the task queue to ensure that the members | 328 // Update some stats but do it on the task queue to ensure that the members |
332 // are modified and read on the same thread. Note that |max_abs| will be | 329 // are modified and read on the same thread. Note that |max_abs| will be |
333 // zero in most calls and then have no effect of the stats. It is only updated | 330 // zero in most calls and then have no effect of the stats. It is only updated |
334 // approximately two times per second and can then change the stats. | 331 // approximately two times per second and can then change the stats. |
335 task_queue_.PostTask( | 332 task_queue_.PostTask([this, max_abs, samples_per_channel] { |
336 [this, max_abs, num_samples] { UpdateRecStats(max_abs, num_samples); }); | 333 UpdateRecStats(max_abs, samples_per_channel); |
| 334 }); |
337 return 0; | 335 return 0; |
338 } | 336 } |
339 | 337 |
340 int32_t AudioDeviceBuffer::DeliverRecordedData() { | 338 int32_t AudioDeviceBuffer::DeliverRecordedData() { |
341 RTC_DCHECK_RUN_ON(&recording_thread_checker_); | 339 RTC_DCHECK_RUN_ON(&recording_thread_checker_); |
342 if (!audio_transport_cb_) { | 340 if (!audio_transport_cb_) { |
343 LOG(LS_WARNING) << "Invalid audio transport"; | 341 LOG(LS_WARNING) << "Invalid audio transport"; |
344 return 0; | 342 return 0; |
345 } | 343 } |
346 const size_t rec_bytes_per_sample = rec_channels_ * sizeof(int16_t); | 344 const size_t frames = rec_buffer_.size() / rec_channels_; |
| 345 const size_t bytes_per_frame = rec_channels_ * sizeof(int16_t); |
347 uint32_t new_mic_level(0); | 346 uint32_t new_mic_level(0); |
348 uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_; | 347 uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_; |
349 size_t num_samples = rec_buffer_.size() / rec_bytes_per_sample; | |
350 int32_t res = audio_transport_cb_->RecordedDataIsAvailable( | 348 int32_t res = audio_transport_cb_->RecordedDataIsAvailable( |
351 rec_buffer_.data(), num_samples, rec_bytes_per_sample, rec_channels_, | 349 rec_buffer_.data(), frames, bytes_per_frame, rec_channels_, |
352 rec_sample_rate_, total_delay_ms, clock_drift_, current_mic_level_, | 350 rec_sample_rate_, total_delay_ms, clock_drift_, current_mic_level_, |
353 typing_status_, new_mic_level); | 351 typing_status_, new_mic_level); |
354 if (res != -1) { | 352 if (res != -1) { |
355 new_mic_level_ = new_mic_level; | 353 new_mic_level_ = new_mic_level; |
356 } else { | 354 } else { |
357 LOG(LS_ERROR) << "RecordedDataIsAvailable() failed"; | 355 LOG(LS_ERROR) << "RecordedDataIsAvailable() failed"; |
358 } | 356 } |
359 return 0; | 357 return 0; |
360 } | 358 } |
361 | 359 |
362 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t num_samples) { | 360 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) { |
363 RTC_DCHECK_RUN_ON(&playout_thread_checker_); | 361 RTC_DCHECK_RUN_ON(&playout_thread_checker_); |
364 // The consumer can change the request size on the fly and we therefore | 362 // The consumer can change the requested size on the fly and we therefore |
365 // resize the buffer accordingly. Also takes place at the first call to this | 363 // resize the buffer accordingly. Also takes place at the first call to this |
366 // method. | 364 // method. |
367 const size_t play_bytes_per_sample = play_channels_ * sizeof(int16_t); | 365 const size_t total_samples = play_channels_ * samples_per_channel; |
368 const size_t size_in_bytes = num_samples * play_bytes_per_sample; | 366 if (play_buffer_.size() != total_samples) { |
369 if (play_buffer_.size() != size_in_bytes) { | 367 play_buffer_.SetSize(total_samples); |
370 play_buffer_.SetSize(size_in_bytes); | |
371 LOG(LS_INFO) << "Size of playout buffer: " << play_buffer_.size(); | 368 LOG(LS_INFO) << "Size of playout buffer: " << play_buffer_.size(); |
372 } | 369 } |
373 | 370 |
374 size_t num_samples_out(0); | 371 size_t num_samples_out(0); |
375 // It is currently supported to start playout without a valid audio | 372 // It is currently supported to start playout without a valid audio |
376 // transport object. Leads to warning and silence. | 373 // transport object. Leads to warning and silence. |
377 if (!audio_transport_cb_) { | 374 if (!audio_transport_cb_) { |
378 LOG(LS_WARNING) << "Invalid audio transport"; | 375 LOG(LS_WARNING) << "Invalid audio transport"; |
379 return 0; | 376 return 0; |
380 } | 377 } |
381 | 378 |
382 // Retrieve new 16-bit PCM audio data using the audio transport instance. | 379 // Retrieve new 16-bit PCM audio data using the audio transport instance. |
383 int64_t elapsed_time_ms = -1; | 380 int64_t elapsed_time_ms = -1; |
384 int64_t ntp_time_ms = -1; | 381 int64_t ntp_time_ms = -1; |
| 382 const size_t bytes_per_frame = play_channels_ * sizeof(int16_t); |
385 uint32_t res = audio_transport_cb_->NeedMorePlayData( | 383 uint32_t res = audio_transport_cb_->NeedMorePlayData( |
386 num_samples, play_bytes_per_sample, play_channels_, play_sample_rate_, | 384 samples_per_channel, bytes_per_frame, play_channels_, play_sample_rate_, |
387 play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms); | 385 play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms); |
388 if (res != 0) { | 386 if (res != 0) { |
389 LOG(LS_ERROR) << "NeedMorePlayData() failed"; | 387 LOG(LS_ERROR) << "NeedMorePlayData() failed"; |
390 } | 388 } |
391 | 389 |
392 // Derive a new level value twice per second. | 390 // Derive a new level value twice per second. |
393 int16_t max_abs = 0; | 391 int16_t max_abs = 0; |
394 RTC_DCHECK_LT(play_stat_count_, 50); | 392 RTC_DCHECK_LT(play_stat_count_, 50); |
395 if (++play_stat_count_ >= 50) { | 393 if (++play_stat_count_ >= 50) { |
396 const size_t size = num_samples * play_channels_; | |
397 // Returns the largest absolute value in a signed 16-bit vector. | 394 // Returns the largest absolute value in a signed 16-bit vector. |
398 max_abs = WebRtcSpl_MaxAbsValueW16( | 395 max_abs = |
399 reinterpret_cast<const int16_t*>(play_buffer_.data()), size); | 396 WebRtcSpl_MaxAbsValueW16(play_buffer_.data(), play_buffer_.size()); |
400 play_stat_count_ = 0; | 397 play_stat_count_ = 0; |
401 } | 398 } |
402 // Update some stats but do it on the task queue to ensure that the members | 399 // Update some stats but do it on the task queue to ensure that the members |
403 // are modified and read on the same thread. Note that |max_abs| will be | 400 // are modified and read on the same thread. Note that |max_abs| will be |
404 // zero in most calls and then have no effect of the stats. It is only updated | 401 // zero in most calls and then have no effect of the stats. It is only updated |
405 // approximately two times per second and can then change the stats. | 402 // approximately two times per second and can then change the stats. |
406 task_queue_.PostTask([this, max_abs, num_samples_out] { | 403 task_queue_.PostTask([this, max_abs, num_samples_out] { |
407 UpdatePlayStats(max_abs, num_samples_out); | 404 UpdatePlayStats(max_abs, num_samples_out); |
408 }); | 405 }); |
409 return static_cast<int32_t>(num_samples_out); | 406 return static_cast<int32_t>(num_samples_out); |
410 } | 407 } |
411 | 408 |
412 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) { | 409 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) { |
413 RTC_DCHECK_RUN_ON(&playout_thread_checker_); | 410 RTC_DCHECK_RUN_ON(&playout_thread_checker_); |
414 RTC_DCHECK_GT(play_buffer_.size(), 0u); | 411 RTC_DCHECK_GT(play_buffer_.size(), 0u); |
415 const size_t play_bytes_per_sample = play_channels_ * sizeof(int16_t); | 412 const size_t bytes_per_sample = sizeof(int16_t); |
416 memcpy(audio_buffer, play_buffer_.data(), play_buffer_.size()); | 413 memcpy(audio_buffer, play_buffer_.data(), |
417 return static_cast<int32_t>(play_buffer_.size() / play_bytes_per_sample); | 414 play_buffer_.size() * bytes_per_sample); |
| 415 // Return samples per channel or number of frames. |
| 416 return static_cast<int32_t>(play_buffer_.size() / play_channels_); |
418 } | 417 } |
419 | 418 |
420 void AudioDeviceBuffer::StartPeriodicLogging() { | 419 void AudioDeviceBuffer::StartPeriodicLogging() { |
421 task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this, | 420 task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this, |
422 AudioDeviceBuffer::LOG_START)); | 421 AudioDeviceBuffer::LOG_START)); |
423 } | 422 } |
424 | 423 |
425 void AudioDeviceBuffer::StopPeriodicLogging() { | 424 void AudioDeviceBuffer::StopPeriodicLogging() { |
426 task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this, | 425 task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this, |
427 AudioDeviceBuffer::LOG_STOP)); | 426 AudioDeviceBuffer::LOG_STOP)); |
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497 | 496 |
498 void AudioDeviceBuffer::ResetPlayStats() { | 497 void AudioDeviceBuffer::ResetPlayStats() { |
499 RTC_DCHECK_RUN_ON(&task_queue_); | 498 RTC_DCHECK_RUN_ON(&task_queue_); |
500 play_callbacks_ = 0; | 499 play_callbacks_ = 0; |
501 last_play_callbacks_ = 0; | 500 last_play_callbacks_ = 0; |
502 play_samples_ = 0; | 501 play_samples_ = 0; |
503 last_play_samples_ = 0; | 502 last_play_samples_ = 0; |
504 max_play_level_ = 0; | 503 max_play_level_ = 0; |
505 } | 504 } |
506 | 505 |
507 void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs, size_t num_samples) { | 506 void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs, |
| 507 size_t samples_per_channel) { |
508 RTC_DCHECK_RUN_ON(&task_queue_); | 508 RTC_DCHECK_RUN_ON(&task_queue_); |
509 ++rec_callbacks_; | 509 ++rec_callbacks_; |
510 rec_samples_ += num_samples; | 510 rec_samples_ += samples_per_channel; |
511 if (max_abs > max_rec_level_) { | 511 if (max_abs > max_rec_level_) { |
512 max_rec_level_ = max_abs; | 512 max_rec_level_ = max_abs; |
513 } | 513 } |
514 } | 514 } |
515 | 515 |
516 void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs, size_t num_samples) { | 516 void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs, |
| 517 size_t samples_per_channel) { |
517 RTC_DCHECK_RUN_ON(&task_queue_); | 518 RTC_DCHECK_RUN_ON(&task_queue_); |
518 ++play_callbacks_; | 519 ++play_callbacks_; |
519 play_samples_ += num_samples; | 520 play_samples_ += samples_per_channel; |
520 if (max_abs > max_play_level_) { | 521 if (max_abs > max_play_level_) { |
521 max_play_level_ = max_abs; | 522 max_play_level_ = max_abs; |
522 } | 523 } |
523 } | 524 } |
524 | 525 |
525 } // namespace webrtc | 526 } // namespace webrtc |
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