Searched refs:fir (Results 1 - 4 of 4) sorted by relevance
| /frameworks/av/tools/resampler_tools/ |
| H A D | Android.mk | 11 fir.cpp
13 LOCAL_MODULE := fir
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| /frameworks/av/media/libstagefright/codecs/amrwb/src/ |
| H A D | oversamp_12k8_to_16k.cpp | 124 const int16 * fir, /* filter coefficient */
277 const int16 *fir, /* filter coefficient */
282 const int16 *pt_fir = fir;
275 AmrWbInterpol( int16 * x, const int16 *fir, int16 nb_coef ) argument
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| /frameworks/av/media/libaudioprocessing/ |
| H A D | AudioResamplerFirProcess.h | 168 * Use fir() to compute the proper coefficient pointers for a polyphase
288 * below coefShift. See fir() for details.
376 void fir(TO* const out, function in namespace:android
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| H A D | AudioResamplerDyn.cpp | 575 // caution: fir() is inlined and may be large.
585 fir<CHANNELS, LOCKED, STRIDE>(
584 ALOG_ASSERT(phaseFraction < phaseWrapLimit); fir<CHANNELS, LOCKED, STRIDE>( &out[outputIndex], phaseFraction, phaseWrapLimit, coefShift, halfNumCoefs, coefs, impulse, volumeSimd); outputIndex += OUTPUT_CHANNELS; phaseFraction += phaseIncrement; while (phaseFraction >= phaseWrapLimit) argument
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