Searched defs:nSamples (Results 1 - 21 of 21) sorted by relevance

/external/libopus/silk/
H A Dresampler.c181 opus_int nSamples; local
188 nSamples = S->Fs_in_kHz - S->inputDelay;
191 silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) );
196 silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
200 silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
204 silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
208 silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16 ) );
/external/aac/libPCMutils/src/
H A Dlimiter.cpp222 const UINT nSamples)
230 FDK_ASSERT(gain_delay <= nSamples);
252 for (i = 0; i < nSamples; i++) {
216 applyLimiter(TDLimiterPtr limiter, INT_PCM* samples, FIXP_DBL* pGain, const INT* gain_scale, const UINT gain_size, const UINT gain_delay, const UINT nSamples) argument
/external/libopus/silk/fixed/
H A Dnoise_shape_analysis_FIX.c153 opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; local
210 nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 );
215 silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples );
216 nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/
223 pitch_res_ptr += nSamples;
/external/libopus/silk/float/
H A Dnoise_shape_analysis_FLP.c136 opus_int k, nSamples; local
182 nSamples = 2 * psEnc->sCmn.fs_kHz;
187 nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples );
193 pitch_res_ptr += nSamples;
/external/webrtc/webrtc/modules/audio_device/
H A Daudio_device_buffer.cc379 // 16-bit,48kHz mono, 10ms => nSamples=480 => _recSize=2*480=960 bytes
380 // 16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes
384 size_t nSamples)
394 _recSamples = nSamples;
395 _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
486 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) argument
509 _playSamples = nSamples;
510 _playSize = playBytesPerSample * nSamples; // {2,4}*nSamples
383 SetRecordedBuffer(const void* audioBuffer, size_t nSamples) argument
[all...]
/external/webrtc/webrtc/modules/audio_device/test/
H A Dfunc_test_manager.h50 size_t nSamples; member in struct:AudioPacket
89 const size_t nSamples,
99 int32_t NeedMorePlayData(const size_t nSamples,
H A Dfunc_test_manager.cc195 const size_t nSamples,
208 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
209 packet->nSamples = nSamples;
340 const size_t nSamples,
354 memset(audioSamples, 0, nBytesPerSample * nSamples);
367 const size_t nSamplesIn = packet->nSamples;
392 2 * nSamples, lenOut);
397 2 * nSamplesIn, tmpBuf_96kHz, 2 * nSamples,
404 for (size_t i = 0; i < nSamples;
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/aac/libAACdec/src/
H A Dblock.cpp684 int fr, fl, tl, nSamples, nSpec; local
720 nSamples = imdct_block(
740 FDK_ASSERT(nSamples == frameLen);
/external/aac/libAACenc/src/
H A Dpsy_main.cpp420 INT nSamples,
425 for (k=0; k<nSamples; k++) {
418 FDKaacEnc_deinterleaveInputBuffer(INT_PCM *pOutputSamples, INT_PCM *pInputSamples, INT nSamples, INT nChannels) argument
/external/webrtc/webrtc/voice_engine/
H A Dvoe_base_impl.cc83 const size_t nSamples,
93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples,
98 int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples, argument
106 GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true, local
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
H A Dtransmit_mixer.cc321 size_t nSamples,
330 "TransmitMixer::PrepareDemux(nSamples=%" PRIuS ", "
333 nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift,
338 nSamples, local
320 PrepareDemux(const void* audioSamples, size_t nSamples, size_t nChannels, uint32_t samplesPerSec, uint16_t totalDelayMS, int32_t clockDrift, uint16_t currentMicLevel, bool keyPressed) argument
/external/pdfium/third_party/lcms2-2.6/include/
H A Dlcms2_plugin.h293 cmsUInt32Number nSamples[MAX_INPUT_DIMENSIONS]; // Valid on all kinds of tables member in struct:_cms_interp_struc
294 cmsUInt32Number Domain[MAX_INPUT_DIMENSIONS]; // Domain = nSamples - 1
/external/pdfium/third_party/lcms2-2.6/src/
H A Dcmsintrp.c104 const cmsUInt32Number nSamples[],
132 p -> nSamples[i] = nSamples[i];
133 p -> Domain[i] = nSamples[i] - 1;
139 p ->opta[i] = p ->opta[i-1] * nSamples[InputChan-i];
154 cmsInterpParams* _cmsComputeInterpParams(cmsContext ContextID, int nSamples, int InputChan, int OutputChan, const void* Table, cmsUInt32Number dwFlags) argument
161 Samples[i] = nSamples;
103 _cmsComputeInterpParamsEx(cmsContext ContextID, const cmsUInt32Number nSamples[], int InputChan, int OutputChan, const void *Table, cmsUInt32Number dwFlags) argument
H A Dcmslut.c514 Data ->Params ->nSamples,
754 cmsUInt32Number* nSamples; local
764 nSamples = clut->Params ->nSamples;
773 nTotalPoints = CubeSize(nSamples, nInputs);
782 cmsUInt32Number Colorant = rest % nSamples[t];
784 rest /= nSamples[t];
786 In[t] = _cmsQuantizeVal(Colorant, nSamples[t]);
816 cmsUInt32Number* nSamples; local
820 nSamples
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H A Dcmscgats.c123 int nSamples, nPatches; // Cols, Rows member in struct:_Table
1442 t -> nSamples = (int) cmsIT8GetPropertyDbl(it8, "NUMBER_OF_FIELDS");
1444 if (t -> nSamples <= 0) {
1447 t -> nSamples = 10;
1450 t -> DataFormat = (char**) AllocChunk (it8, ((cmsUInt32Number) t->nSamples + 1) * sizeof(char *));
1477 if (n > t -> nSamples) {
1503 t-> nSamples = atoi(cmsIT8GetProperty(it8, "NUMBER_OF_FIELDS"));
1506 t-> Data = (char**)AllocChunk (it8, ((cmsUInt32Number) t->nSamples + 1) * ((cmsUInt32Number) t->nPatches + 1) *sizeof (char*));
1518 int nSamples = t -> nSamples; local
1696 int i, nSamples; local
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/external/webrtc/webrtc/modules/audio_device/android/
H A Daudio_device_unittest.cc384 const size_t nSamples,
394 int32_t(const size_t nSamples,
424 const size_t nSamples,
438 audio_stream_->Write(audioSamples, nSamples);
446 int32_t RealNeedMorePlayData(const size_t nSamples, argument
456 nSamplesOut = nSamples;
460 audio_stream_->Read(audioSamples, nSamples);
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
/external/webrtc/webrtc/modules/audio_device/ios/
H A Daudio_device_unittest_ios.cc374 const size_t nSamples,
384 int32_t(const size_t nSamples,
414 const size_t nSamples,
428 audio_stream_->Write(audioSamples, nSamples);
438 int32_t RealNeedMorePlayData(const size_t nSamples, argument
448 nSamplesOut = nSamples;
452 audio_stream_->Read(audioSamples, nSamples);
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
/external/webrtc/webrtc/modules/audio_device/linux/
H A Daudio_device_pulse_linux.cc2774 uint32_t nSamples = local
2784 nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
2785 if (nSamples != numPlaySamples)
2789 nSamples);
/external/webrtc/webrtc/modules/audio_device/mac/
H A Daudio_device_mac.cc2623 uint32_t nSamples = local
2626 nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
2627 if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES) {
2629 " invalid number of output samples(%d)", nSamples);
2632 uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
/external/webrtc/webrtc/modules/audio_device/win/
H A Daudio_device_wave_win.cc3461 uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES); local
3470 nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
3471 if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES)
3473 WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples);
3528 int32_t AudioDeviceWindowsWave::Write(int8_t* data, uint16_t nSamples) argument
3543 const int16_t nBytes = (2*_playChannels)*nSamples;
3575 _writtenSamples += nSamples; // each sample is 2 or 4 bytes
H A Daudio_device_core_win.cc3555 int32_t nSamples = local
3559 if (nSamples == -1)
3574 if (nSamples != static_cast<int32_t>(_playBlockSize))
3576 WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
3580 nSamples = _ptrAudioBuffer->GetPlayoutData((int8_t*)pData);

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