| /external/webrtc/talk/app/webrtc/test/ |
| D | fakeaudiocapturemodule_unittest.cc | 58 int32_t RecordedDataIsAvailable(const void* audioSamples, in RecordedDataIsAvailable() argument
76 memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_); in RecordedDataIsAvailable()
87 void* audioSamples, in NeedMorePlayData() argument
95 CopyFromRecBuffer(audioSamples, audio_buffer_size): in NeedMorePlayData()
96 GenerateZeroBuffer(audioSamples, audio_buffer_size); in NeedMorePlayData()
|
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| D | audio_device_unittest_ios.cc | 373 int32_t(const void* audioSamples,
388 void* audioSamples,
413 int32_t RealRecordedDataIsAvailable(const void* audioSamples, in RealRecordedDataIsAvailable() argument
428 audio_stream_->Write(audioSamples, nSamples); in RealRecordedDataIsAvailable()
442 void* audioSamples, in RealNeedMorePlayData() argument
452 audio_stream_->Read(audioSamples, nSamples); in RealNeedMorePlayData()
|
| /external/webrtc/webrtc/modules/audio_device/test/ |
| D | func_test_manager.h | 88 int32_t RecordedDataIsAvailable(const void* audioSamples,
103 void* audioSamples,
|
| D | func_test_manager.cc | 194 const void* audioSamples, in RecordedDataIsAvailable() argument
208 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample); in RecordedDataIsAvailable()
344 void* audioSamples, in NeedMorePlayData() argument
354 memset(audioSamples, 0, nBytesPerSample * nSamples); in NeedMorePlayData()
391 2 * nSamplesIn, (int16_t*) audioSamples, in NeedMorePlayData()
401 ptr16Out = (int16_t*) audioSamples; in NeedMorePlayData()
431 (int16_t*) audioSamples, nSamples, lenOut); in NeedMorePlayData()
439 ptr16Out = (int16_t*) audioSamples; in NeedMorePlayData()
480 memcpy(audioSamples, fileBuf, 2 * nSamples); in NeedMorePlayData()
485 int16_t* audio16 = (int16_t*) audioSamples; in NeedMorePlayData()
|
| D | audio_device_test_api.cc | 85 int32_t RecordedDataIsAvailable(const void* audioSamples, in RecordedDataIsAvailable() argument
115 void* audioSamples, in NeedMorePlayData() argument
|
| /external/webrtc/webrtc/voice_engine/ |
| D | voe_base_impl.h | 57 int32_t RecordedDataIsAvailable(const void* audioSamples,
71 void* audioSamples,
|
| D | transmit_mixer.h | 53 int32_t PrepareDemux(const void* audioSamples,
175 void GenerateAudioFrame(const int16_t audioSamples[],
|
| D | voe_base_impl.cc | 82 int32_t VoEBaseImpl::RecordedDataIsAvailable(const void* audioSamples, in RecordedDataIsAvailable() argument
93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples, in RecordedDataIsAvailable()
102 void* audioSamples, in NeedMorePlayData() argument
107 audioSamples, elapsed_time_ms, ntp_time_ms); in NeedMorePlayData()
|
| D | transmit_mixer.cc | 320 TransmitMixer::PrepareDemux(const void* audioSamples, in PrepareDemux() argument
337 GenerateAudioFrame(static_cast<const int16_t*>(audioSamples), in PrepareDemux()
|
| /external/webrtc/webrtc/modules/audio_device/include/ |
| D | audio_device_defines.h | 49 virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
64 void* audioSamples,
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_device_unittest.cc | 383 int32_t(const void* audioSamples,
398 void* audioSamples,
423 int32_t RealRecordedDataIsAvailable(const void* audioSamples, in RealRecordedDataIsAvailable() argument
438 audio_stream_->Write(audioSamples, nSamples); in RealRecordedDataIsAvailable()
450 void* audioSamples, in RealNeedMorePlayData() argument
460 audio_stream_->Read(audioSamples, nSamples); in RealNeedMorePlayData()
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| D | RTPencode.cc | 125 void stereoDeInterleave(int16_t* audioSamples, size_t numSamples);
1825 void stereoDeInterleave(int16_t* audioSamples, size_t numSamples) { in stereoDeInterleave() argument
1838 memcpy(tempVec, audioSamples, numSamples * sizeof(int16_t)); in stereoDeInterleave()
1840 writeL = audioSamples; in stereoDeInterleave()
1841 writeR = &audioSamples[numSamples / 2]; in stereoDeInterleave()
|