| /external/libopus/silk/ |
| D | resampler.c | 181 opus_int nSamples; in silk_resampler() local
188 nSamples = S->Fs_in_kHz - S->inputDelay; in silk_resampler()
191 silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) ); in silk_resampler()
196 …silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in… in silk_resampler()
200 … silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); in silk_resampler()
204 …silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); in silk_resampler()
208 …silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16… in silk_resampler()
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| /external/webrtc/webrtc/modules/audio_device/ |
| D | audio_device_buffer.cc | 384 size_t nSamples) in SetRecordedBuffer() argument
394 _recSamples = nSamples; in SetRecordedBuffer()
395 _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples in SetRecordedBuffer()
486 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) in RequestPlayoutData() argument
509 _playSamples = nSamples; in RequestPlayoutData()
510 _playSize = playBytesPerSample * nSamples; // {2,4}*nSamples in RequestPlayoutData()
517 if (nSamples != _playSamples) in RequestPlayoutData()
519 …TraceWarning, kTraceAudioDevice, _id, "invalid number of samples to be played out (%d)", nSamples); in RequestPlayoutData()
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| D | mock_audio_device_buffer.h | 23 MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples));
26 int32_t(const void* audioBuffer, size_t nSamples));
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| D | audio_device_buffer.h | 53 size_t nSamples);
61 virtual int32_t RequestPlayoutData(size_t nSamples);
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| /external/webrtc/talk/app/webrtc/test/ |
| D | fakeaudiocapturemodule_unittest.cc | 59 const size_t nSamples, in RecordedDataIsAvailable() argument
69 rec_buffer_bytes_ = nSamples * nBytesPerSample; in RecordedDataIsAvailable()
83 int32_t NeedMorePlayData(const size_t nSamples, in NeedMorePlayData() argument
93 const size_t audio_buffer_size = nSamples * nBytesPerSample; in NeedMorePlayData()
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| /external/pdfium/third_party/lcms2-2.6/src/ |
| D | cmscgats.c | 123 int nSamples, nPatches; // Cols, Rows member
1442 t -> nSamples = (int) cmsIT8GetPropertyDbl(it8, "NUMBER_OF_FIELDS"); in AllocateDataFormat()
1444 if (t -> nSamples <= 0) { in AllocateDataFormat()
1447 t -> nSamples = 10; in AllocateDataFormat()
1450 … t -> DataFormat = (char**) AllocChunk (it8, ((cmsUInt32Number) t->nSamples + 1) * sizeof(char *)); in AllocateDataFormat()
1477 if (n > t -> nSamples) { in SetDataFormat()
1503 t-> nSamples = atoi(cmsIT8GetProperty(it8, "NUMBER_OF_FIELDS")); in AllocateDataSet()
1506 …t-> Data = (char**)AllocChunk (it8, ((cmsUInt32Number) t->nSamples + 1) * ((cmsUInt32Number) t->nP… in AllocateDataSet()
1518 int nSamples = t -> nSamples; in GetData() local
1521 if (nSet >= nPatches || nField >= nSamples) in GetData()
[all …]
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| D | cmslut.c | 514 Data ->Params ->nSamples, in CLUTElemDup()
754 cmsUInt32Number* nSamples; in cmsStageSampleCLut16bit() local
764 nSamples = clut->Params ->nSamples; in cmsStageSampleCLut16bit()
773 nTotalPoints = CubeSize(nSamples, nInputs); in cmsStageSampleCLut16bit()
782 cmsUInt32Number Colorant = rest % nSamples[t]; in cmsStageSampleCLut16bit()
784 rest /= nSamples[t]; in cmsStageSampleCLut16bit()
786 In[t] = _cmsQuantizeVal(Colorant, nSamples[t]); in cmsStageSampleCLut16bit()
816 cmsUInt32Number* nSamples; in cmsStageSampleCLutFloat() local
820 nSamples = clut->Params ->nSamples; in cmsStageSampleCLutFloat()
829 nTotalPoints = CubeSize(nSamples, nInputs); in cmsStageSampleCLutFloat()
[all …]
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| D | cmsintrp.c | 104 const cmsUInt32Number nSamples[], in _cmsComputeInterpParamsEx() argument
132 p -> nSamples[i] = nSamples[i]; in _cmsComputeInterpParamsEx()
133 p -> Domain[i] = nSamples[i] - 1; in _cmsComputeInterpParamsEx()
139 p ->opta[i] = p ->opta[i-1] * nSamples[InputChan-i]; in _cmsComputeInterpParamsEx()
154 cmsInterpParams* _cmsComputeInterpParams(cmsContext ContextID, int nSamples, int InputChan, int Out… in _cmsComputeInterpParams() argument
161 Samples[i] = nSamples; in _cmsComputeInterpParams()
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| D | lcms2_internal.h | 761 cmsInterpParams* _cmsComputeInterpParams(cmsContext ContextID, int nSamples, int InputChan, int…
762 cmsInterpParams* _cmsComputeInterpParamsEx(cmsContext ContextID, const cmsUInt32Number nSamples…
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| /external/webrtc/webrtc/modules/audio_device/test/ |
| D | func_test_manager.cc | 195 const size_t nSamples, in RecordedDataIsAvailable() argument
208 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample); in RecordedDataIsAvailable()
209 packet->nSamples = nSamples; in RecordedDataIsAvailable()
340 const size_t nSamples, in NeedMorePlayData() argument
354 memset(audioSamples, 0, nBytesPerSample * nSamples); in NeedMorePlayData()
367 const size_t nSamplesIn = packet->nSamples; in NeedMorePlayData()
392 2 * nSamples, lenOut); in NeedMorePlayData()
397 2 * nSamplesIn, tmpBuf_96kHz, 2 * nSamples, in NeedMorePlayData()
404 for (size_t i = 0; i < nSamples; i++) in NeedMorePlayData()
412 assert(2*nSamples == lenOut); in NeedMorePlayData()
[all …]
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| D | func_test_manager.h | 50 size_t nSamples; member
89 const size_t nSamples,
99 int32_t NeedMorePlayData(const size_t nSamples,
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| /external/libopus/silk/float/ |
| D | noise_shape_analysis_FLP.c | 136 opus_int k, nSamples; in silk_noise_shape_analysis_FLP() local
182 nSamples = 2 * psEnc->sCmn.fs_kHz; in silk_noise_shape_analysis_FLP()
187 nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples ); in silk_noise_shape_analysis_FLP()
193 pitch_res_ptr += nSamples; in silk_noise_shape_analysis_FLP()
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| /external/webrtc/webrtc/modules/audio_device/ios/ |
| D | audio_device_unittest_ios.cc | 374 const size_t nSamples,
384 int32_t(const size_t nSamples,
414 const size_t nSamples, in RealRecordedDataIsAvailable() argument
428 audio_stream_->Write(audioSamples, nSamples); in RealRecordedDataIsAvailable()
438 int32_t RealNeedMorePlayData(const size_t nSamples, in RealNeedMorePlayData() argument
448 nSamplesOut = nSamples; in RealNeedMorePlayData()
452 audio_stream_->Read(audioSamples, nSamples); in RealNeedMorePlayData()
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_device_unittest.cc | 384 const size_t nSamples,
394 int32_t(const size_t nSamples,
424 const size_t nSamples, in RealRecordedDataIsAvailable() argument
438 audio_stream_->Write(audioSamples, nSamples); in RealRecordedDataIsAvailable()
446 int32_t RealNeedMorePlayData(const size_t nSamples, in RealNeedMorePlayData() argument
456 nSamplesOut = nSamples; in RealNeedMorePlayData()
460 audio_stream_->Read(audioSamples, nSamples); in RealNeedMorePlayData()
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| /external/webrtc/webrtc/voice_engine/ |
| D | voe_base_impl.h | 58 const size_t nSamples,
67 int32_t NeedMorePlayData(const size_t nSamples,
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| D | transmit_mixer.h | 54 size_t nSamples,
176 size_t nSamples,
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| D | voe_base_impl.cc | 83 const size_t nSamples, in RecordedDataIsAvailable() argument
93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples, in RecordedDataIsAvailable()
98 int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples, in NeedMorePlayData() argument
106 GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true, in NeedMorePlayData()
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| D | transmit_mixer.cc | 321 size_t nSamples, in PrepareDemux() argument
333 nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift, in PrepareDemux()
338 nSamples, in PrepareDemux()
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| /external/webrtc/webrtc/modules/audio_device/include/ |
| D | audio_device_defines.h | 50 const size_t nSamples,
60 virtual int32_t NeedMorePlayData(const size_t nSamples,
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| /external/libopus/silk/fixed/ |
| D | noise_shape_analysis_FIX.c | 153 opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; in silk_noise_shape_analysis_FIX() local
210 nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 ); in silk_noise_shape_analysis_FIX()
215 silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples ); in silk_noise_shape_analysis_FIX()
216 nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/ in silk_noise_shape_analysis_FIX()
223 pitch_res_ptr += nSamples; in silk_noise_shape_analysis_FIX()
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| /external/aac/libPCMutils/src/ |
| D | limiter.cpp | 222 const UINT nSamples) in applyLimiter() argument
230 FDK_ASSERT(gain_delay <= nSamples); in applyLimiter()
252 for (i = 0; i < nSamples; i++) { in applyLimiter()
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| /external/aac/libPCMutils/include/ |
| D | limiter.h | 179 const UINT nSamples);
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| /external/aac/libAACdec/src/ |
| D | block.cpp | 684 int fr, fl, tl, nSamples, nSpec; in CBlock_FrequencyToTime() local
720 nSamples = imdct_block( in CBlock_FrequencyToTime()
740 FDK_ASSERT(nSamples == frameLen); in CBlock_FrequencyToTime()
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| /external/webrtc/webrtc/modules/audio_device/win/ |
| D | audio_device_wave_win.cc | 3461 uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES); in PlayProc() local
3470 nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer); in PlayProc()
3471 if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES) in PlayProc()
3473 … WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples); in PlayProc()
3528 int32_t AudioDeviceWindowsWave::Write(int8_t* data, uint16_t nSamples) in Write() argument
3543 const int16_t nBytes = (2*_playChannels)*nSamples; in Write()
3575 _writtenSamples += nSamples; // each sample is 2 or 4 bytes in Write()
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| /external/aac/libAACenc/src/ |
| D | metadata_main.cpp | 235 const INT nSamples
486 const INT nSamples in ProcessCompressor() argument
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