ref: 9443f3a29bca510330c5153916f06494141f9662
dir: /src/stat.c/
/*
* Sound Tools statistics "effect" file.
*
* Compute various statistics on file and print them.
*
* Output is unmodified from input.
*
*/
/*
* July 5, 1991
* Copyright 1991 Lance Norskog And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Lance Norskog And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
#include <math.h>
#include <string.h>
#include "st_i.h"
#include "FFT.h"
/* Private data for STAT effect */
typedef struct statstuff {
double min, max, mid;
double asum;
double sum1, sum2; /* amplitudes */
double dmin, dmax;
double dsum1, dsum2; /* deltas */
double scale; /* scale-factor */
double last; /* previous sample */
st_size_t read; /* samples processed */
int volume;
int srms;
int fft;
unsigned long bin[4];
float *re_in;
float *re_out;
unsigned long fft_size;
unsigned long fft_offset;
} *stat_t;
/*
* Process options
*/
int st_stat_getopts(eff_t effp, int n, char **argv)
{
stat_t stat = (stat_t) effp->priv;
stat->scale = ST_SAMPLE_MAX;
stat->volume = 0;
stat->srms = 0;
stat->fft = 0;
while (n>0)
{
if (!(strcmp(argv[0], "-v")))
stat->volume = 1;
else if (!(strcmp(argv[0], "-s")))
{
double scale;
if (n <= 1)
{
st_fail("-s option: invalid argument");
return (ST_EOF);
}
if (!sscanf(argv[1], "%lf", &scale))
{
st_fail("-s option: invalid argument");
return (ST_EOF);
}
stat->scale = scale;
/* Two argument option. Account for this */
--n; ++argv;
}
else if (!(strcmp(argv[0], "-rms")))
stat->srms = 1;
else if (!(strcmp(argv[0], "-freq")))
stat->fft = 1;
else if (!(strcmp(argv[0], "-d")))
stat->volume = 2;
else
{
st_fail("Summary effect: unknown option");
return(ST_EOF);
}
--n; ++argv;
}
return (ST_SUCCESS);
}
/*
* Prepare processing.
*/
int st_stat_start(eff_t effp)
{
stat_t stat = (stat_t) effp->priv;
int i;
stat->min = stat->max = stat->mid = 0;
stat->asum = 0;
stat->sum1 = stat->sum2 = 0;
stat->dmin = stat->dmax = 0;
stat->dsum1 = stat->dsum2 = 0;
stat->last = 0;
stat->read = 0;
for (i = 0; i < 4; i++)
stat->bin[i] = 0;
stat->fft_size = 4096;
stat->re_in = stat->re_out = NULL;
if (stat->fft)
{
stat->fft_offset = 0;
stat->re_in = (float *)malloc(sizeof(float) * stat->fft_size);
stat->re_out = (float *)malloc(sizeof(float) * (stat->fft_size / 2));
if (!stat->re_in || !stat->re_out)
{
st_fail("Unable to allocate memory for FFT buffers.");
return (ST_EOF);
}
}
return (ST_SUCCESS);
}
/*
* Print power spectrum to given stream
*/
static void print_power_spectrum(unsigned samples, float rate, float *re_in, float *re_out)
{
float ffa = rate / samples;
unsigned i;
PowerSpectrum(samples, re_in, re_out);
for (i = 0; i < samples / 2; i++)
fprintf(stderr, "%f %f\n", ffa * i, re_out[i]);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
int st_stat_flow(eff_t effp, const st_sample_t *ibuf, st_sample_t *obuf,
st_size_t *isamp, st_size_t *osamp)
{
stat_t stat = (stat_t) effp->priv;
int len, done, x;
short count = 0;
len = ((*isamp > *osamp) ? *osamp : *isamp);
if (len==0)
return (ST_SUCCESS);
if (stat->read == 0) /* 1st sample */
stat->min = stat->max = stat->mid = stat->last = (*ibuf)/stat->scale;
if (stat->fft)
{
for (x = 0; x < len; x++)
{
stat->re_in[stat->fft_offset++] = ST_SAMPLE_TO_FLOAT_DWORD(ibuf[x], effp->clippedCount);
if (stat->fft_offset >= stat->fft_size)
{
stat->fft_offset = 0;
print_power_spectrum(stat->fft_size, effp->ininfo.rate, stat->re_in, stat->re_out);
}
}
}
for(done = 0; done < len; done++) {
long lsamp;
double samp, delta;
/* work in scaled levels for both sample and delta */
lsamp = *ibuf++;
samp = (double)lsamp/stat->scale;
stat->bin[(lsamp>>30)+2]++;
*obuf++ = lsamp;
if (stat->volume == 2)
{
fprintf(stderr,"%08lx ",lsamp);
if (count++ == 5)
{
fprintf(stderr,"\n");
count = 0;
}
}
/* update min/max */
if (stat->min > samp)
stat->min = samp;
else if (stat->max < samp)
stat->max = samp;
stat->mid = stat->min / 2 + stat->max / 2;
stat->sum1 += samp;
stat->sum2 += samp*samp;
stat->asum += fabs(samp);
delta = fabs(samp - stat->last);
if (delta < stat->dmin)
stat->dmin = delta;
else if (delta > stat->dmax)
stat->dmax = delta;
stat->dsum1 += delta;
stat->dsum2 += delta*delta;
stat->last = samp;
}
stat->read += len;
*isamp = *osamp = len;
/* Process all samples */
return (ST_SUCCESS);
}
/*
* Process tail of input samples.
*/
int st_stat_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp)
{
stat_t stat = (stat_t) effp->priv;
/* When we run out of samples, then we need to pad buffer with
* zeros and then run FFT one last time to process any unprocessed
* samples.
*/
if (stat->fft && stat->fft_offset) {
unsigned int x;
for (x = stat->fft_offset; x < stat->fft_size; x++)
stat->re_in[x] = 0;
print_power_spectrum(stat->fft_size, effp->ininfo.rate, stat->re_in, stat->re_out);
}
*osamp = 0;
return (ST_EOF);
}
/*
* Do anything required when you stop reading samples.
* Don't close input file!
*/
int st_stat_stop(eff_t effp)
{
stat_t stat = (stat_t) effp->priv;
double amp, scale, rms = 0, freq;
double x, ct;
ct = stat->read;
if (stat->srms) { /* adjust results to units of rms */
double f;
rms = sqrt(stat->sum2/ct);
f = 1.0/rms;
stat->max *= f;
stat->min *= f;
stat->mid *= f;
stat->asum *= f;
stat->sum1 *= f;
stat->sum2 *= f*f;
stat->dmax *= f;
stat->dmin *= f;
stat->dsum1 *= f;
stat->dsum2 *= f*f;
stat->scale *= rms;
}
scale = stat->scale;
amp = -stat->min;
if (amp < stat->max)
amp = stat->max;
/* Just print the volume adjustment */
if (stat->volume == 1 && amp > 0) {
fprintf(stderr, "%.3f\n", ST_SAMPLE_MAX/(amp*scale));
return (ST_SUCCESS);
}
if (stat->volume == 2)
fprintf(stderr, "\n\n");
/* print out the info */
fprintf(stderr, "Samples read: %12u\n", stat->read);
fprintf(stderr, "Length (seconds): %12.6f\n", (double)stat->read/effp->ininfo.rate/effp->ininfo.channels);
if (stat->srms)
fprintf(stderr, "Scaled by rms: %12.6f\n", rms);
else
fprintf(stderr, "Scaled by: %12.1f\n", scale);
fprintf(stderr, "Maximum amplitude: %12.6f\n", stat->max);
fprintf(stderr, "Minimum amplitude: %12.6f\n", stat->min);
fprintf(stderr, "Midline amplitude: %12.6f\n", stat->mid);
fprintf(stderr, "Mean norm: %12.6f\n", stat->asum/ct);
fprintf(stderr, "Mean amplitude: %12.6f\n", stat->sum1/ct);
fprintf(stderr, "RMS amplitude: %12.6f\n", sqrt(stat->sum2/ct));
fprintf(stderr, "Maximum delta: %12.6f\n", stat->dmax);
fprintf(stderr, "Minimum delta: %12.6f\n", stat->dmin);
fprintf(stderr, "Mean delta: %12.6f\n", stat->dsum1/(ct-1));
fprintf(stderr, "RMS delta: %12.6f\n", sqrt(stat->dsum2/(ct-1)));
freq = sqrt(stat->dsum2/stat->sum2)*effp->ininfo.rate/(M_PI*2);
fprintf(stderr, "Rough frequency: %12d\n", (int)freq);
if (amp>0)
fprintf(stderr, "Volume adjustment: %12.3f\n", ST_SAMPLE_MAX/(amp*scale));
if (stat->bin[2] == 0 && stat->bin[3] == 0)
fprintf(stderr, "\nProbably text, not sound\n");
else {
x = (float)(stat->bin[0] + stat->bin[3]) / (float)(stat->bin[1] + stat->bin[2]);
if (x >= 3.0) /* use opposite encoding */
{
if (effp->ininfo.encoding == ST_ENCODING_UNSIGNED)
fprintf (stderr,"\nTry: -t raw -b -s \n");
else
fprintf (stderr,"\nTry: -t raw -b -u \n");
}
else if (x <= 1.0/3.0)
; /* correctly decoded */
else if (x >= 0.5 && x <= 2.0) /* use ULAW */
{
if (effp->ininfo.encoding == ST_ENCODING_ULAW)
fprintf (stderr,"\nTry: -t raw -b -u \n");
else
fprintf (stderr,"\nTry: -t raw -b -U \n");
}
else
fprintf (stderr, "\nCan't guess the type\n");
}
/* Release FFT memory */
free(stat->re_in);
free(stat->re_out);
return (ST_SUCCESS);
}
static st_effect_t st_stat_effect = {
"stat",
"Usage: [ -s N ] [ -rms ] [-freq] [ -v ] [ -d ]",
ST_EFF_MCHAN | ST_EFF_REPORT,
st_stat_getopts,
st_stat_start,
st_stat_flow,
st_stat_drain,
st_stat_stop
};
const st_effect_t *st_stat_effect_fn(void)
{
return &st_stat_effect;
}