Mercurial > mplayer.hg
view libao2/firfilter.c @ 3639:64ee21b3bd09
Modified the sync code once again, commented out hardware pts sync (I'll likely burn in hell before understanding how to get this bastard to sync well)
Added automagic setup of aspect ratio, will remove the "aspect-bug" (I hope). As well as please you rich 16:9 doods ;)
author | mswitch |
---|---|
date | Thu, 20 Dec 2001 20:50:35 +0000 |
parents | cc1c879533ee |
children | f99944f9f427 |
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#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