/external/webrtc/webrtc/modules/audio_device/test/ |
H A D | func_test_manager.h | 51 size_t nBytesPerSample; member in struct:AudioPacket 90 const size_t nBytesPerSample, 100 const size_t nBytesPerSample,
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H A D | func_test_manager.cc | 196 const size_t nBytesPerSample, 208 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample); 210 packet->nBytesPerSample = nBytesPerSample; 314 } else if ((nChannels == 2) && (nBytesPerSample == 2) && addMarker) 329 if (nChannels == 2 && nBytesPerSample == 2) 341 const size_t nBytesPerSample, 354 memset(audioSamples, 0, nBytesPerSample * nSamples); 370 const size_t nBytesPerSampleIn = packet->nBytesPerSample; 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
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/external/webrtc/webrtc/voice_engine/ |
H A D | voe_base_impl.cc | 84 const size_t nBytesPerSample, 99 const size_t nBytesPerSample, 82 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 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
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/external/webrtc/webrtc/modules/audio_device/android/ |
H A D | audio_device_unittest.cc | 385 const size_t nBytesPerSample, 395 const size_t nBytesPerSample, 425 const size_t nBytesPerSample, 447 const size_t nBytesPerSample, 423 RealRecordedDataIsAvailable(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 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
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/external/webrtc/webrtc/modules/audio_device/ios/ |
H A D | audio_device_unittest_ios.cc | 375 const size_t nBytesPerSample, 385 const size_t nBytesPerSample, 415 const size_t nBytesPerSample, 439 const size_t nBytesPerSample, 413 RealRecordedDataIsAvailable(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 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
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/external/webrtc/webrtc/modules/audio_device/win/ |
H A D | audio_device_wave_win.cc | 2634 const uint8_t nBytesPerSample = 2*_recChannels; local 2638 _waveHeaderIn[n].dwBufferLength = nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES; 2643 memset(_recBuffer[n], 0, nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES);
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