| /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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