Searched refs:fcr (Results 1 - 4 of 4) sorted by relevance
/frameworks/av/media/libaudioprocessing/ |
H A D | AudioResamplerDyn.cpp | 248 double fcr; // compute fcr, the 3 dB amplitude cut-off. local 250 fcr = max(0.5 * tbwCheat - halfbw, halfbw); 252 fcr = max(0.5 * tbwCheat * outSampleRate / inSampleRate - halfbw, halfbw); 254 createKaiserFir(c, stopBandAtten, fcr); 259 double stopBandAtten, double fcr) { 282 firKaiserGen(coefs, phases, halfLength, stopBandAtten, fcr, attenuation); 285 mNormalizedCutoffFrequency = fcr; 295 ALOGD("L:%d hnc:%d stopBandAtten:%lf fcr:%lf atten:%lf tbw:%lf\n", 296 c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, attenuatio 258 createKaiserFir(Constants &c, double stopBandAtten, double fcr) argument 365 double fcr = 0.; local [all...] |
H A D | AudioResamplerDyn.h | 148 void createKaiserFir(Constants &c, double stopBandAtten, double fcr);
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H A D | AudioResamplerFirGen.h | 706 * @param fcr is cutoff frequency/sampling rate (<0.5). At this point, the energy 707 * should be 6dB less. (fcr is where the amplitude drops by half). Use the 708 * firKaiserTbw() to calculate the transition bandwidth. fcr is the midpoint 716 double stopBandAtten, double fcr, double atten) { 719 const double xstep = (2. * M_PI) * fcr / L; 744 y = 2. * atten * fcr; // center of filter, sinc(0) = 1. 715 firKaiserGen(T* coef, int L, int halfNumCoef, double stopBandAtten, double fcr, double atten) argument
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/frameworks/av/media/libaudioprocessing/tests/ |
H A D | resampler_tests.cpp | 266 const double fcr = rdyn->getNormalizedCutoffFrequency(); local 271 const double fp = fcr - tbw / 2; 272 const double fs = fcr + tbw / 2; 276 " fcr:%lf fp:%lf fs:%lf tbw:%lf" 281 fcr, fp, fs, tbw,
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