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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 |
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391 enabled = true; | 391 enabled = true; |
392 return 0; | 392 return 0; |
393 } | 393 } |
394 | 394 |
395 int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const { | 395 int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const { |
396 return 0; | 396 return 0; |
397 } | 397 } |
398 | 398 |
399 int32_t FileAudioDevice::RecordingDelay(uint16_t& delayMS) const { return -1; } | 399 int32_t FileAudioDevice::RecordingDelay(uint16_t& delayMS) const { return -1; } |
400 | 400 |
401 bool FileAudioDevice::PlayoutWarning() const { return false; } | |
402 | |
403 bool FileAudioDevice::PlayoutError() const { return false; } | |
404 | |
405 bool FileAudioDevice::RecordingWarning() const { return false; } | |
406 | |
407 bool FileAudioDevice::RecordingError() const { return false; } | |
408 | |
409 void FileAudioDevice::ClearPlayoutWarning() {} | |
410 | |
411 void FileAudioDevice::ClearPlayoutError() {} | |
412 | |
413 void FileAudioDevice::ClearRecordingWarning() {} | |
414 | |
415 void FileAudioDevice::ClearRecordingError() {} | |
416 | |
417 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { | 401 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { |
418 rtc::CritScope lock(&_critSect); | 402 rtc::CritScope lock(&_critSect); |
419 | 403 |
420 _ptrAudioBuffer = audioBuffer; | 404 _ptrAudioBuffer = audioBuffer; |
421 | 405 |
422 // Inform the AudioBuffer about default settings for this implementation. | 406 // Inform the AudioBuffer about default settings for this implementation. |
423 // Set all values to zero here since the actual settings will be done by | 407 // Set all values to zero here since the actual settings will be done by |
424 // InitPlayout and InitRecording later. | 408 // InitPlayout and InitRecording later. |
425 _ptrAudioBuffer->SetRecordingSampleRate(0); | 409 _ptrAudioBuffer->SetRecordingSampleRate(0); |
426 _ptrAudioBuffer->SetPlayoutSampleRate(0); | 410 _ptrAudioBuffer->SetPlayoutSampleRate(0); |
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499 | 483 |
500 int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime; | 484 int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime; |
501 if (deltaTimeMillis < 10) { | 485 if (deltaTimeMillis < 10) { |
502 SleepMs(10 - deltaTimeMillis); | 486 SleepMs(10 - deltaTimeMillis); |
503 } | 487 } |
504 | 488 |
505 return true; | 489 return true; |
506 } | 490 } |
507 | 491 |
508 } // namespace webrtc | 492 } // namespace webrtc |
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