view libao2/firfilter.c @ 4754:ad892563d9c4

Added VOCTRL_PAUSE and VOCTRL_RESUME Fixed playback bug in vo_dxr3 which would cause syncloss when playing ntsc movies on pal and vice versa (this does not affect mpeg playback as we still don't know how to set the internal fps registers)

#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