Mercurial > mplayer.hg
view libao2/firfilter.c @ 13249:a6642a4330fa
ensure that avi files have a valid header as soon as possible.
without this, the header says 0x0 video size, which works with mplayer
when the video size is stored in the codec data, but it does NOT work
with other players or with codecs that don't store size (e.g. snow).
actually i don't like having seeks in the muxer module, but i don't
know any other way to implement this fix without major changes to
mencoder. if you have a better fix, please reverse this and commit
yours.
author | rfelker |
---|---|
date | Sun, 05 Sep 2004 16:51:15 +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