/external/libopus/silk/ |
H A D | resampler.c | 181 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 ) );
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/external/aac/libPCMutils/src/ |
H A D | limiter.cpp | 222 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
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/external/libopus/silk/fixed/ |
H A D | noise_shape_analysis_FIX.c | 153 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;
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/external/libopus/silk/float/ |
H A D | noise_shape_analysis_FLP.c | 136 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;
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/external/webrtc/webrtc/modules/audio_device/ |
H A D | audio_device_buffer.cc | 379 // 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 D | func_test_manager.h | 50 size_t nSamples; member in struct:AudioPacket 89 const size_t nSamples, 99 int32_t NeedMorePlayData(const size_t nSamples,
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H A D | func_test_manager.cc | 195 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 D | block.cpp | 684 int fr, fl, tl, nSamples, nSpec; local 720 nSamples = imdct_block( 740 FDK_ASSERT(nSamples == frameLen);
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/external/aac/libAACenc/src/ |
H A D | psy_main.cpp | 420 INT nSamples, 425 for (k=0; k<nSamples; k++) { 418 FDKaacEnc_deinterleaveInputBuffer(INT_PCM *pOutputSamples, INT_PCM *pInputSamples, INT nSamples, INT nChannels) argument
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/external/webrtc/webrtc/voice_engine/ |
H A D | voe_base_impl.cc | 83 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
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H A D | transmit_mixer.cc | 321 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
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/external/pdfium/third_party/lcms2-2.6/include/ |
H A D | lcms2_plugin.h | 293 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
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/external/pdfium/third_party/lcms2-2.6/src/ |
H A D | cmsintrp.c | 104 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
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H A D | cmslut.c | 514 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 [all...] |
H A D | cmscgats.c | 123 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 [all...] |
/external/webrtc/webrtc/modules/audio_device/android/ |
H A D | audio_device_unittest.cc | 384 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
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/external/webrtc/webrtc/modules/audio_device/ios/ |
H A D | audio_device_unittest_ios.cc | 374 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
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/external/webrtc/webrtc/modules/audio_device/linux/ |
H A D | audio_device_pulse_linux.cc | 2774 uint32_t nSamples = local 2784 nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer); 2785 if (nSamples != numPlaySamples) 2789 nSamples);
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/external/webrtc/webrtc/modules/audio_device/mac/ |
H A D | audio_device_mac.cc | 2623 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;
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/external/webrtc/webrtc/modules/audio_device/win/ |
H A D | audio_device_wave_win.cc | 3461 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
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H A D | audio_device_core_win.cc | 3555 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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