ref: 19079619533a80049dfde894d7f7d1ff5fa31819
dir: /src/filter.c/
/* libSoX Windowed sinc lowpass/bandpass/highpass filter November 18, 1999
* Copyright 1999 Stan Brooks <stabro@megsinet.net>
* Copyright 1994 Julius O. Smith
* Copyright 1991 (?) Lance Norskog (?)
*
* This library is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or (at
* your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
* General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this library; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*
* REMARKS: (Stan Brooks speaking)
* This code is a rewrite of filterkit.c from the `resample' package
* by Julius O. Smith, now distributed under the LGPL.
*/
#include "sox_i.h"
#include <string.h>
#include <stdlib.h>
#define ISCALE 0x10000
#define BUFFSIZE 8192
#define MAXNWING (80<<7)
/* Private data for Lerp via LCM file */
typedef struct {
sox_rate_t rate;
double freq0;/* low corner freq */
double freq1;/* high corner freq */
double beta;/* >2 is kaiser window beta, <=2 selects nuttall window */
long Nwin;
double *Fp;/* [Xh+1] Filter coefficients */
long Xh;/* number of past/future samples needed by filter */
long Xt;/* target to enter new data into X */
double *X, *Y;/* I/O buffers */
} priv_t;
/* lsx_makeFilter is used by resample.c */
int lsx_makeFilter(double Fp[], long Nwing, double Froll, double Beta, long Num, int Normalize);
/* LpFilter()
*
* reference: "Digital Filters, 2nd edition"
* R.W. Hamming, pp. 178-179
*
* LpFilter() computes the coeffs of a Kaiser-windowed low pass filter with
* the following characteristics:
*
* c[] = array in which to store computed coeffs
* frq = roll-off frequency of filter
* N = Half the window length in number of coeffs
* Beta = parameter of Kaiser window
* Num = number of coeffs before 1/frq
*
* Beta trades the rejection of the lowpass filter against the transition
* width from passband to stopband. Larger Beta means a slower
* transition and greater stopband rejection. See Rabiner and Gold
* (Theory and Application of DSP) under Kaiser windows for more about
* Beta. The following table from Rabiner and Gold gives some feel
* for the effect of Beta:
*
* All ripples in dB, width of transition band = D*N where N = window length
*
* BETA D PB RIP SB RIP
* 2.120 1.50 +-0.27 -30
* 3.384 2.23 0.0864 -40
* 4.538 2.93 0.0274 -50
* 5.658 3.62 0.00868 -60
* 6.764 4.32 0.00275 -70
* 7.865 5.0 0.000868 -80
* 8.960 5.7 0.000275 -90
* 10.056 6.4 0.000087 -100
*/
static void LpFilter(double *c, long N, double frq, double Beta, long Num)
{
long i;
/* Calculate filter coeffs: */
c[0] = frq;
for (i=1; i<N; i++) {
double x = M_PI*(double)i/(double)(Num);
c[i] = sin(x*frq)/x;
}
if (Beta>2) { /* Apply Kaiser window to filter coeffs: */
double IBeta = 1.0/lsx_bessel_I_0(Beta);
for (i=1; i<N; i++) {
double x = (double)i / (double)(N);
c[i] *= lsx_bessel_I_0(Beta*sqrt(1.0-x*x)) * IBeta;
}
} else { /* Apply Nuttall window: */
for(i = 0; i < N; i++) {
double x = M_PI*i / N;
c[i] *= 0.3635819 + 0.4891775*cos(x) + 0.1365995*cos(2*x) + 0.0106411*cos(3*x);
}
}
}
int lsx_makeFilter(double Imp[], long Nwing, double Froll, double Beta,
long Num, int Normalize)
{
double *ImpR;
long Mwing, i;
if (Nwing > MAXNWING) /* Check for valid parameters */
return(-1);
if ((Froll<=0) || (Froll>1))
return(-2);
/* it does help accuracy a bit to have the window stop at
* a zero-crossing of the sinc function */
Mwing = floor((double)Nwing/(Num/Froll))*(Num/Froll) +0.5;
if (Mwing==0)
return(-4);
ImpR = lsx_malloc(sizeof(double) * Mwing);
/* Design a Nuttall or Kaiser windowed Sinc low-pass filter */
LpFilter(ImpR, Mwing, Froll, Beta, Num);
if (Normalize) { /* 'correct' the DC gain of the lowpass filter */
long Dh;
double DCgain;
DCgain = 0;
Dh = Num; /* Filter sampling period for factors>=1 */
for (i=Dh; i<Mwing; i+=Dh)
DCgain += ImpR[i];
DCgain = 2*DCgain + ImpR[0]; /* DC gain of real coefficients */
sox_debug("DCgain err=%.12f",DCgain-1.0);
DCgain = 1.0/DCgain;
for (i=0; i<Mwing; i++)
Imp[i] = ImpR[i]*DCgain;
} else {
for (i=0; i<Mwing; i++)
Imp[i] = ImpR[i];
}
free(ImpR);
for (i=Mwing; i<=Nwing; i++) Imp[i] = 0;
/* Imp[Mwing] and Imp[-1] needed for quadratic interpolation */
Imp[-1] = Imp[1];
return(Mwing);
}
static void FiltWin(priv_t * f, long Nx);
/*
* Process options
*/
static int sox_filter_getopts(sox_effect_t * effp, int n, char **argv)
{
priv_t * f = (priv_t *) effp->priv;
f->beta = 16; /* Kaiser window, beta 16 */
f->Nwin = 128;
f->freq0 = f->freq1 = 0;
if (n >= 1) {
char *p;
p = argv[0];
if (*p != '-') {
f->freq1 = lsx_parse_frequency(p, &p);
}
if (*p == '-') {
f->freq0 = f->freq1;
f->freq1 = lsx_parse_frequency(p+1, &p);
}
if (*p) f->freq1 = f->freq0 = 0;
}
sox_debug("freq: %g-%g", f->freq0, f->freq1);
if (f->freq0 == 0 && f->freq1 == 0)
return lsx_usage(effp);
if ((n >= 2) && !sscanf(argv[1], "%ld", &f->Nwin))
return lsx_usage(effp);
else if (f->Nwin < 4) {
sox_fail("filter: window length (%ld) <4 is too short", f->Nwin);
return (SOX_EOF);
}
if ((n >= 3) && !sscanf(argv[2], "%lf", &f->beta))
return lsx_usage(effp);
sox_debug("filter opts: %g-%g, window-len %ld, beta %f", f->freq0, f->freq1, f->Nwin, f->beta);
return (SOX_SUCCESS);
}
static int p2(int n)
{
int N;
for (N = 1; n; n >>= 1, N <<= 1);
return max(N, 1024);
}
/*
* Prepare processing.
*/
static int sox_filter_start(sox_effect_t * effp)
{
priv_t * f = (priv_t *) effp->priv;
double *Fp0, *Fp1;
long Xh0, Xh1, Xh;
int i;
f->rate = effp->in_signal.rate;
/* adjust upper frequency to Nyquist if necessary */
if (f->freq1 > (sox_sample_t)f->rate/2 || f->freq1 <= 0)
f->freq1 = f->rate/2;
if ((f->freq0 < 0) || (f->freq0 > f->freq1))
{
sox_fail("filter: low(%g),high(%g) parameters must satisfy 0 <= low <= high <= %g",
f->freq0, f->freq1, f->rate/2);
return (SOX_EOF);
}
Xh = f->Nwin/2;
Fp0 = lsx_malloc(sizeof(double) * (Xh + 2));
++Fp0;
if (f->freq0 > (sox_sample_t)f->rate/200) {
Xh0 = lsx_makeFilter(Fp0, Xh, 2.0*(double)f->freq0/f->rate, f->beta, (size_t) 1, 0);
if (Xh0 <= 1)
{
sox_fail("filter: Unable to make low filter");
return (SOX_EOF);
}
} else {
Xh0 = 0;
}
Fp1 = lsx_malloc(sizeof(double) * (Xh + 2));
++Fp1;
/* need Fp[-1] and Fp[Xh] for lsx_makeFilter */
if (f->freq1 < (sox_sample_t)f->rate/2) {
Xh1 = lsx_makeFilter(Fp1, Xh, 2.0*(double)f->freq1/f->rate, f->beta, (size_t) 1, 0);
if (Xh1 <= 1)
{
sox_fail("filter: Unable to make high filter");
return (SOX_EOF);
}
} else {
Fp1[0] = 1.0;
Xh1 = 1;
}
/* now subtract Fp0[] from Fp1[] */
Xh = (Xh0>Xh1)? Xh0:Xh1; /* >=1, by above */
for (i=0; i<Xh; i++) {
double c0,c1;
c0 = (i<Xh0)? Fp0[i]:0;
c1 = (i<Xh1)? Fp1[i]:0;
Fp1[i] = c1-c0;
}
free(Fp0 - 1); /* all done with Fp0 */
f->Fp = Fp1;
Xh -= 1; /* Xh = 0 can only happen if filter was identity 0-Nyquist */
if (Xh<=0)
sox_warn("filter: adjusted freq %g-%g is identity", f->freq0, f->freq1);
f->Nwin = 2*Xh + 1; /* not really used afterwards */
f->Xh = Xh;
f->Xt = Xh;
if (effp->global_info->plot == sox_plot_gnuplot) {
int N = p2(2 * Xh + 1);
double * h = lsx_calloc(N, sizeof(*h));
double * H = lsx_malloc((N / 2 + 1) * sizeof(*H));
for (i = 1; i < Xh + 1; ++i)
h[i - 1] = f->Fp[Xh + 1 - i];
for (i = 0; i < Xh + 1; ++i)
h[Xh + i] = f->Fp[i];
lsx_power_spectrum(N, h, H);
free(h);
printf(
"# gnuplot file\n"
"set title 'SoX effect: filter frequency=%g-%g'\n"
"set xlabel 'Frequency (Hz)'\n"
"set ylabel 'Amplitude Response (dB)'\n"
"set grid xtics ytics\n"
"set key off\n"
"plot '-' with lines\n"
, f->freq0, f->freq1);
for (i = 0; i <= N/2; ++i)
printf("%g %g\n", i * effp->in_signal.rate / N, 10 * log10(H[i]));
printf(
"e\n"
"pause -1 'Hit return to continue'\n");
free(H);
return SOX_EOF;
}
if (effp->global_info->plot == sox_plot_octave) {
printf("%% GNU Octave file (may also work with MATLAB(R) )\nb=[");
for (i = 1; i < Xh + 1; ++i)
printf("%24.16e\n", f->Fp[Xh + 1 - i]);
for (i = 0; i < Xh + 1; ++i)
printf("%24.16e\n", f->Fp[i]);
printf("];\n"
"[h,w]=freqz(b);\n"
"plot(%g*w/pi,20*log10(h))\n"
"title('SoX effect: filter frequency=%g-%g')\n"
"xlabel('Frequency (Hz)')\n"
"ylabel('Amplitude Response (dB)')\n"
"grid on\n"
"disp('Hit return to continue')\n"
"pause\n"
, effp->in_signal.rate / 2, f->freq0, f->freq1);
return SOX_EOF;
}
f->X = lsx_malloc(sizeof(double) * (2*BUFFSIZE + 2*Xh));
f->Y = f->X + BUFFSIZE + 2*Xh;
/* Need Xh zeros at beginning of X */
for (i = 0; i < Xh; i++)
f->X[i] = 0;
return (SOX_SUCCESS);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
static int sox_filter_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * f = (priv_t *) effp->priv;
size_t Nx;
long i, Nproc;
/* constrain amount we actually process */
/* sox_debug("Xh %d, Xt %d, isamp %d, ",f->Xh, f->Xt, *isamp); */
Nx = BUFFSIZE + 2*f->Xh - f->Xt;
if (Nx > *isamp) Nx = *isamp;
if (Nx > *osamp) Nx = *osamp;
*isamp = Nx;
{
double *xp, *xtop;
xp = f->X + f->Xt;
xtop = xp + Nx;
if (ibuf != NULL) {
while (xp < xtop)
*xp++ = (double)(*ibuf++) / ISCALE;
} else {
while (xp < xtop)
*xp++ = 0;
}
}
Nproc = f->Xt + Nx - 2*f->Xh;
if (Nproc <= 0) {
f->Xt += Nx;
*osamp = 0;
return (SOX_SUCCESS);
}
sox_debug("flow Nproc %ld",Nproc);
FiltWin(f, Nproc);
/* Copy back portion of input signal that must be re-used */
Nx += f->Xt;
if (f->Xh)
memmove(f->X, f->X + Nx - 2*f->Xh, sizeof(double)*2*f->Xh);
f->Xt = 2*f->Xh;
for (i = 0; i < Nproc; i++)
*obuf++ = f->Y[i] * ISCALE;
*osamp = Nproc;
return (SOX_SUCCESS);
}
/*
* Process tail of input samples.
*/
static int sox_filter_drain(sox_effect_t * effp, sox_sample_t *obuf, size_t *osamp)
{
priv_t * f = (priv_t *) effp->priv;
long isamp_res, osamp_res;
sox_sample_t *Obuf;
sox_debug("Xh %ld, Xt %ld <--- DRAIN",f->Xh, f->Xt);
/* stuff end with Xh zeros */
isamp_res = f->Xh;
osamp_res = *osamp;
Obuf = obuf;
while (isamp_res>0 && osamp_res>0) {
size_t Isamp, Osamp;
Isamp = isamp_res;
Osamp = osamp_res;
sox_filter_flow(effp, NULL, Obuf, &Isamp, &Osamp);
/* sox_debug("DRAIN isamp,osamp (%d,%d) -> (%d,%d)",
* isamp_res,osamp_res,Isamp,Osamp); */
Obuf += Osamp;
osamp_res -= Osamp;
isamp_res -= Isamp;
};
*osamp -= osamp_res;
/* sox_debug("DRAIN osamp %d", *osamp); */
if (isamp_res)
sox_warn("drain overran obuf by %ld", isamp_res);
/* FIXME: This is very picky. osamp better be big enough to grab
* all remaining samples or they will be discarded.
*/
return (SOX_EOF);
}
/*
* Do anything required when you stop reading samples.
* Don't close input file!
*/
static int sox_filter_stop(sox_effect_t * effp)
{
priv_t * f = (priv_t *) effp->priv;
free(f->Fp - 1);
free(f->X);
return (SOX_SUCCESS);
}
static double jprod(const double *Fp, const double *Xp, long ct)
{
const double *fp, *xp, *xq;
double v = 0;
fp = Fp + ct; /* so sum starts with smaller coef's */
xp = Xp - ct;
xq = Xp + ct;
while (fp > Fp) { /* ct = 0 can happen */
v += *fp * (*xp + *xq);
xp++; xq--; fp--;
}
v += *fp * *xp;
return v;
}
static void FiltWin(priv_t * f, long Nx)
{
double *Y;
double *X, *Xend;
Y = f->Y;
X = f->X + f->Xh; /* Ptr to current input sample */
Xend = X + Nx;
while (X < Xend) {
*Y++ = jprod(f->Fp, X, f->Xh);
X++;
}
}
static sox_effect_handler_t sox_filter_effect = {
"filter",
"low-high [ windowlength [ beta ] ]",
0,
sox_filter_getopts,
sox_filter_start,
sox_filter_flow,
sox_filter_drain,
sox_filter_stop,
NULL, sizeof(priv_t)
};
const sox_effect_handler_t *sox_filter_effect_fn(void)
{
return &sox_filter_effect;
}