| /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
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