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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 |
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200 return 0; | 200 return 0; |
201 } | 201 } |
202 | 202 |
203 CriticalSectionScoped lock(&_critSect); | 203 CriticalSectionScoped lock(&_critSect); |
204 | 204 |
205 _mixerManager.Close(); | 205 _mixerManager.Close(); |
206 | 206 |
207 // RECORDING | 207 // RECORDING |
208 if (_ptrThreadRec) | 208 if (_ptrThreadRec) |
209 { | 209 { |
210 ThreadWrapper* tmpThread = _ptrThreadRec.release(); | 210 PlatformThread* tmpThread = _ptrThreadRec.release(); |
211 _critSect.Leave(); | 211 _critSect.Leave(); |
212 | 212 |
213 tmpThread->Stop(); | 213 tmpThread->Stop(); |
214 delete tmpThread; | 214 delete tmpThread; |
215 | 215 |
216 _critSect.Enter(); | 216 _critSect.Enter(); |
217 } | 217 } |
218 | 218 |
219 // PLAYOUT | 219 // PLAYOUT |
220 if (_ptrThreadPlay) | 220 if (_ptrThreadPlay) |
221 { | 221 { |
222 ThreadWrapper* tmpThread = _ptrThreadPlay.release(); | 222 PlatformThread* tmpThread = _ptrThreadPlay.release(); |
223 _critSect.Leave(); | 223 _critSect.Leave(); |
224 | 224 |
225 tmpThread->Stop(); | 225 tmpThread->Stop(); |
226 delete tmpThread; | 226 delete tmpThread; |
227 | 227 |
228 _critSect.Enter(); | 228 _critSect.Enter(); |
229 } | 229 } |
230 #if defined(USE_X11) | 230 #if defined(USE_X11) |
231 if (_XDisplay) | 231 if (_XDisplay) |
232 { | 232 { |
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1357 _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS]; | 1357 _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS]; |
1358 if (!_recordingBuffer) | 1358 if (!_recordingBuffer) |
1359 { | 1359 { |
1360 WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, | 1360 WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
1361 " failed to alloc recording buffer"); | 1361 " failed to alloc recording buffer"); |
1362 _recording = false; | 1362 _recording = false; |
1363 return -1; | 1363 return -1; |
1364 } | 1364 } |
1365 // RECORDING | 1365 // RECORDING |
1366 const char* threadName = "webrtc_audio_module_capture_thread"; | 1366 const char* threadName = "webrtc_audio_module_capture_thread"; |
1367 _ptrThreadRec = ThreadWrapper::CreateThread( | 1367 _ptrThreadRec = |
1368 RecThreadFunc, this, threadName); | 1368 PlatformThread::CreateThread(RecThreadFunc, this, threadName); |
1369 | 1369 |
1370 if (!_ptrThreadRec->Start()) | 1370 if (!_ptrThreadRec->Start()) |
1371 { | 1371 { |
1372 WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, | 1372 WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
1373 " failed to start the rec audio thread"); | 1373 " failed to start the rec audio thread"); |
1374 _recording = false; | 1374 _recording = false; |
1375 _ptrThreadRec.reset(); | 1375 _ptrThreadRec.reset(); |
1376 delete [] _recordingBuffer; | 1376 delete [] _recordingBuffer; |
1377 _recordingBuffer = NULL; | 1377 _recordingBuffer = NULL; |
1378 return -1; | 1378 return -1; |
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1511 if (!_playoutBuffer) | 1511 if (!_playoutBuffer) |
1512 { | 1512 { |
1513 WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, | 1513 WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
1514 " failed to alloc playout buf"); | 1514 " failed to alloc playout buf"); |
1515 _playing = false; | 1515 _playing = false; |
1516 return -1; | 1516 return -1; |
1517 } | 1517 } |
1518 | 1518 |
1519 // PLAYOUT | 1519 // PLAYOUT |
1520 const char* threadName = "webrtc_audio_module_play_thread"; | 1520 const char* threadName = "webrtc_audio_module_play_thread"; |
1521 _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this, | 1521 _ptrThreadPlay = |
1522 threadName); | 1522 PlatformThread::CreateThread(PlayThreadFunc, this, threadName); |
1523 if (!_ptrThreadPlay->Start()) | 1523 if (!_ptrThreadPlay->Start()) |
1524 { | 1524 { |
1525 WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, | 1525 WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
1526 " failed to start the play audio thread"); | 1526 " failed to start the play audio thread"); |
1527 _playing = false; | 1527 _playing = false; |
1528 _ptrThreadPlay.reset(); | 1528 _ptrThreadPlay.reset(); |
1529 delete [] _playoutBuffer; | 1529 delete [] _playoutBuffer; |
1530 _playoutBuffer = NULL; | 1530 _playoutBuffer = NULL; |
1531 return -1; | 1531 return -1; |
1532 } | 1532 } |
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2238 state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i]; | 2238 state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i]; |
2239 | 2239 |
2240 // Save old state | 2240 // Save old state |
2241 memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState)); | 2241 memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState)); |
2242 return (state != 0); | 2242 return (state != 0); |
2243 #else | 2243 #else |
2244 return false; | 2244 return false; |
2245 #endif | 2245 #endif |
2246 } | 2246 } |
2247 } // namespace webrtc | 2247 } // namespace webrtc |
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