Searched defs:fcr (Results 1 - 2 of 2) sorted by path

/frameworks/av/media/libaudioprocessing/
H A DAudioResamplerDyn.cpp228 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;
H A DAudioResamplerFirGen.h634 * @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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