Searched defs:fcr (Results 1 - 2 of 2) sorted by path
/frameworks/av/media/libaudioprocessing/ |
H A D | AudioResamplerDyn.cpp | 228 double fcr; local 233 fcr = max(0.5*tbwCheat - tbw/2, tbw/2); 235 fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2); 238 firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten); 246 printf("L:%d hnc:%d stopBandAtten:%lf fcr:%lf atten:%lf tbw:%lf\n", 247 c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw); 249 double fp = (fcr - tbw/2)/c.mL; 250 double fs = (fcr + tbw/2)/c.mL;
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H A D | AudioResamplerFirGen.h | 634 * @param fcr is cutoff frequency/sampling rate (<0.5). At this point, the energy 635 * should be 6dB less. (fcr is where the amplitude drops by half). Use the 636 * firKaiserTbw() to calculate the transition bandwidth. fcr is the midpoint 644 double stopBandAtten, double fcr, double atten) { 670 const double xstep = (2. * M_PI) * fcr / L; 695 y = 2. * atten * fcr; // center of filter, sinc(0) = 1. 643 firKaiserGen(T* coef, int L, int halfNumCoef, double stopBandAtten, double fcr, double atten) argument
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