Mercurial > mplayer.hg
view libao2/firfilter.c @ 11290:82f9b6d890ea
1l
author | diego |
---|---|
date | Mon, 27 Oct 2003 10:25:56 +0000 |
parents | f99944f9f427 |
children |
line wrap: on
line source
#include <inttypes.h> #include <math.h> static double desired_7kHz_lowpass[] = {1.0, 0.0}; static double weights_7kHz_lowpass[] = {0.2, 2.0}; double *calc_coefficients_7kHz_lowpass(int rate) { double *result = (double *)malloc(32*sizeof(double)); double bands[4]; bands[0] = 0.0; bands[1] = 6800.0/rate; bands[2] = 8500.0/rate; bands[3] = 0.5; remez(result, 32, 2, bands, desired_7kHz_lowpass, weights_7kHz_lowpass, BANDPASS); return result; } #if 0 static double desired_125Hz_lowpass[] = {1.0, 0.0}; static double weights_125Hz_lowpass[] = {0.2, 2.0}; double *calc_coefficients_125Hz_lowpass(int rate) { double *result = (double *)malloc(256*sizeof(double)); double bands[4]; bands[0] = 0.0; bands[1] = 125.0/rate; bands[2] = 175.0/rate; bands[3] = 0.5; remez(result, 256, 2, bands, desired_125Hz_lowpass, weights_125Hz_lowpass, BANDPASS); return result; } #endif int16_t firfilter(int16_t *buf, int pos, int len, int count, double *coefficients) { double result = 0.0; int count1, count2; int16_t *ptr; if (pos >= count) { pos -= count; count1 = count; count2 = 0; } else { count2 = pos; count1 = count - pos; pos = len - count1; } //fprintf(stderr, "pos=%d, count1=%d, count2=%d\n", pos, count1, count2); // high part of window ptr = &buf[pos]; while (count1--) result += *ptr++ * *coefficients++; // wrapped part of window while (count2--) result += *buf++ * *coefficients++; return result; } void dump_filter_coefficients(double *coefficients) { int i; fprintf(stderr, "pl_surround: Filter coefficients are: \n"); for (i=0; (i<32); i++) { fprintf(stderr, " [%2d]: %23.20f\n", i, coefficients[i]); } } #ifdef TESTING #define PI 3.1415926536 // For testing purposes, fill a buffer with some sine-wave tone void sinewave(int16_t *output, int samples, int incr, int freq, double phase, int samplerate) { double radians_per_sample = 2*PI / ((0.0+samplerate) / freq), r; //fprintf(stderr, "samples=%d tone freq=%d, samplerate=%d, radians/sample=%f\n", // samples, freq, samplerate, radians_per_sample); r = phase; while (samples--) { *output = sin(r)*10000; output = &output[incr]; r += radians_per_sample; } } // Pass various frequencies through a FIR filter and report amplitudes void testfilter(double *coefficients, int count, int samplerate) { int16_t wavein[8192]; //, waveout[2048]; int sample, samples, maxsample, minsample, totsample; int nyquist=samplerate/2; int freq, i; for (freq=25; freq<nyquist; freq+=25) { // Make input tone sinewave(wavein, 8192, 1, freq, 0.0, samplerate); //for (i=0; i<32; i++) // fprintf(stderr, "%5d\n", wavein[i]); // Filter through the filter, measure results maxsample=0; minsample=1000000; totsample=0; samples=0; for (i=2048; i<8192; i++) { //waveout[i] = wavein[i]; sample = abs(firfilter(wavein, i, 8192, count, coefficients)); if (sample > maxsample) maxsample=sample; if (sample < minsample) minsample=sample; totsample += sample; samples++; } // Report results fprintf(stderr, "%5d %5d %5d %5d %f\n", freq, totsample/samples, maxsample, minsample, 10*log((totsample/samples)/6500.0)); } } #endif