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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 #include "webrtc/base/platform_thread.h" |
10 #include "webrtc/modules/audio_device/dummy/file_audio_device.h" | 11 #include "webrtc/modules/audio_device/dummy/file_audio_device.h" |
11 #include "webrtc/system_wrappers/include/sleep.h" | 12 #include "webrtc/system_wrappers/include/sleep.h" |
12 #include "webrtc/system_wrappers/include/thread_wrapper.h" | |
13 | 13 |
14 namespace webrtc { | 14 namespace webrtc { |
15 | 15 |
16 const int kRecordingFixedSampleRate = 48000; | 16 const int kRecordingFixedSampleRate = 48000; |
17 const int kRecordingNumChannels = 2; | 17 const int kRecordingNumChannels = 2; |
18 const int kPlayoutFixedSampleRate = 48000; | 18 const int kPlayoutFixedSampleRate = 48000; |
19 const int kPlayoutNumChannels = 2; | 19 const int kPlayoutNumChannels = 2; |
20 const int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100 | 20 const int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100 |
21 * kPlayoutNumChannels * 2; | 21 * kPlayoutNumChannels * 2; |
22 const int kRecordingBufferSize = kRecordingFixedSampleRate / 100 | 22 const int kRecordingBufferSize = kRecordingFixedSampleRate / 100 |
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207 if (!_outputFilename.empty() && _outputFile.OpenFile( | 207 if (!_outputFilename.empty() && _outputFile.OpenFile( |
208 _outputFilename.c_str(), false, false, false) == -1) { | 208 _outputFilename.c_str(), false, false, false) == -1) { |
209 printf("Failed to open playout file %s!\n", _outputFilename.c_str()); | 209 printf("Failed to open playout file %s!\n", _outputFilename.c_str()); |
210 _playing = false; | 210 _playing = false; |
211 delete [] _playoutBuffer; | 211 delete [] _playoutBuffer; |
212 _playoutBuffer = NULL; | 212 _playoutBuffer = NULL; |
213 return -1; | 213 return -1; |
214 } | 214 } |
215 | 215 |
216 const char* threadName = "webrtc_audio_module_play_thread"; | 216 const char* threadName = "webrtc_audio_module_play_thread"; |
217 _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this, | 217 _ptrThreadPlay = |
218 threadName); | 218 PlatformThread::CreateThread(PlayThreadFunc, this, threadName); |
219 if (!_ptrThreadPlay->Start()) { | 219 if (!_ptrThreadPlay->Start()) { |
220 _ptrThreadPlay.reset(); | 220 _ptrThreadPlay.reset(); |
221 _playing = false; | 221 _playing = false; |
222 delete [] _playoutBuffer; | 222 delete [] _playoutBuffer; |
223 _playoutBuffer = NULL; | 223 _playoutBuffer = NULL; |
224 return -1; | 224 return -1; |
225 } | 225 } |
226 _ptrThreadPlay->SetPriority(kRealtimePriority); | 226 _ptrThreadPlay->SetPriority(kRealtimePriority); |
227 return 0; | 227 return 0; |
228 } | 228 } |
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270 _inputFilename.c_str(), true, true, false) == -1) { | 270 _inputFilename.c_str(), true, true, false) == -1) { |
271 printf("Failed to open audio input file %s!\n", | 271 printf("Failed to open audio input file %s!\n", |
272 _inputFilename.c_str()); | 272 _inputFilename.c_str()); |
273 _recording = false; | 273 _recording = false; |
274 delete[] _recordingBuffer; | 274 delete[] _recordingBuffer; |
275 _recordingBuffer = NULL; | 275 _recordingBuffer = NULL; |
276 return -1; | 276 return -1; |
277 } | 277 } |
278 | 278 |
279 const char* threadName = "webrtc_audio_module_capture_thread"; | 279 const char* threadName = "webrtc_audio_module_capture_thread"; |
280 _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this, threadName); | 280 _ptrThreadRec = PlatformThread::CreateThread(RecThreadFunc, this, threadName); |
281 | 281 |
282 if (!_ptrThreadRec->Start()) { | 282 if (!_ptrThreadRec->Start()) { |
283 _ptrThreadRec.reset(); | 283 _ptrThreadRec.reset(); |
284 _recording = false; | 284 _recording = false; |
285 delete [] _recordingBuffer; | 285 delete [] _recordingBuffer; |
286 _recordingBuffer = NULL; | 286 _recordingBuffer = NULL; |
287 return -1; | 287 return -1; |
288 } | 288 } |
289 _ptrThreadRec->SetPriority(kRealtimePriority); | 289 _ptrThreadRec->SetPriority(kRealtimePriority); |
290 | 290 |
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541 _critSect.Enter(); | 541 _critSect.Enter(); |
542 } | 542 } |
543 } | 543 } |
544 | 544 |
545 _critSect.Leave(); | 545 _critSect.Leave(); |
546 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); | 546 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); |
547 return true; | 547 return true; |
548 } | 548 } |
549 | 549 |
550 } // namespace webrtc | 550 } // namespace webrtc |
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