/external/webrtc/webrtc/modules/audio_device/test/ |
H A D | func_test_manager.cc | 194 const void* audioSamples, 208 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample); 344 void* audioSamples, 354 memset(audioSamples, 0, nBytesPerSample * nSamples); 391 2 * nSamplesIn, (int16_t*) audioSamples, 401 ptr16Out = (int16_t*) audioSamples; 431 (int16_t*) audioSamples, nSamples, lenOut); 439 ptr16Out = (int16_t*) audioSamples; 480 memcpy(audioSamples, fileBuf, 2 * nSamples); 485 int16_t* audio16 = (int16_t*) audioSamples; 193 RecordedDataIsAvailable( const void* audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument 339 NeedMorePlayData( const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument [all...] |
/external/webrtc/webrtc/voice_engine/ |
H A D | voe_base_impl.cc | 82 int32_t VoEBaseImpl::RecordedDataIsAvailable(const void* audioSamples, argument 93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples, 102 void* audioSamples, 107 audioSamples, elapsed_time_ms, ntp_time_ms); local 98 NeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
|
H A D | transmit_mixer.cc | 320 TransmitMixer::PrepareDemux(const void* audioSamples, argument 337 GenerateAudioFrame(static_cast<const int16_t*>(audioSamples),
|
/external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
H A D | RTPencode.cc | 125 void stereoDeInterleave(int16_t* audioSamples, size_t numSamples); 1825 void stereoDeInterleave(int16_t* audioSamples, size_t numSamples) { argument 1838 memcpy(tempVec, audioSamples, numSamples * sizeof(int16_t)); 1840 writeL = audioSamples; 1841 writeR = &audioSamples[numSamples / 2];
|
/external/webrtc/webrtc/modules/audio_device/android/ |
H A D | audio_device_unittest.cc | 383 int32_t(const void* audioSamples, 398 void* audioSamples, 423 int32_t RealRecordedDataIsAvailable(const void* audioSamples, argument 438 audio_stream_->Write(audioSamples, nSamples); 450 void* audioSamples, 460 audio_stream_->Read(audioSamples, nSamples); 446 RealNeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
|
/external/webrtc/webrtc/modules/audio_device/ios/ |
H A D | audio_device_unittest_ios.cc | 373 int32_t(const void* audioSamples, 388 void* audioSamples, 413 int32_t RealRecordedDataIsAvailable(const void* audioSamples, argument 428 audio_stream_->Write(audioSamples, nSamples); 442 void* audioSamples, 452 audio_stream_->Read(audioSamples, nSamples); 438 RealNeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
|