Mercurial > mplayer.hg
view libao2/firfilter.c @ 7946:f483ab704252
postprocessing cleanup:
remove opendivx #ifdefs
remove rk1 filter
remove unused / obsolete stuff
add -1,4,2,4,-1 deinterlacing filter (ffmpeg uses that)
threadsafe / no more non-const globals
some optimizations
different strides for Y,U,V possible
remove ebx usage (someone really should fix gcc, this is really lame)
change the dering filter slightly (tell me if its worse for any files)
author | michael |
---|---|
date | Mon, 28 Oct 2002 19:31:04 +0000 |
parents | f99944f9f427 |
children |
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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