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/**********************************************************************
resamplesubs.c
Real-time library interface by Dominic Mazzoni
Based on resample-1.7:
http://www-ccrma.stanford.edu/~jos/resample/
License: LGPL - see the file LICENSE.txt for more information
This file provides Kaiser-windowed low-pass filter support,
including a function to create the filter coefficients, and
two functions to apply the filter at a particular point.
**********************************************************************/
/* Definitions */
#include "resample_defs.h"
#include "filterkit.h"
#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <math.h>
/* LpFilter()
*
* reference: "Digital Filters, 2nd edition"
* R.W. Hamming, pp. 178-179
*
* Izero() computes the 0th order modified bessel function of the first kind.
* (Needed to compute Kaiser window).
*
* 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
*/
#define IzeroEPSILON 1E-21 /* Max error acceptable in Izero */
static double Izero(double x)
{
double sum, u, halfx, temp;
int n;
sum = u = n = 1;
halfx = x/2.0;
do {
temp = halfx/(double)n;
n += 1;
temp *= temp;
u *= temp;
sum += u;
} while (u >= IzeroEPSILON*sum);
return(sum);
}
void lrsLpFilter(double c[], int N, double frq, double Beta, int Num)
{
double IBeta, temp, temp1, inm1;
int i;
/* Calculate ideal lowpass filter impulse response coefficients: */
c[0] = 2.0*frq;
for (i=1; i<N; i++) {
temp = PI*(double)i/(double)Num;
c[i] = sin(2.0*temp*frq)/temp; /* Analog sinc function, cutoff = frq */
}
/*
* Calculate and Apply Kaiser window to ideal lowpass filter.
* Note: last window value is IBeta which is NOT zero.
* You're supposed to really truncate the window here, not ramp
* it to zero. This helps reduce the first sidelobe.
*/
IBeta = 1.0/Izero(Beta);
inm1 = 1.0/((double)(N-1));
for (i=1; i<N; i++) {
temp = (double)i * inm1;
temp1 = 1.0 - temp*temp;
temp1 = (temp1<0? 0: temp1); /* make sure it's not negative since
we're taking the square root - this
happens on Pentium 4's due to tiny
roundoff errors */
c[i] *= Izero(Beta*sqrt(temp1)) * IBeta;
}
}
float lrsFilterUp(float Imp[], /* impulse response */
float ImpD[], /* impulse response deltas */
UWORD Nwing, /* len of one wing of filter */
BOOL Interp, /* Interpolate coefs using deltas? */
float *Xp, /* Current sample */
double Ph, /* Phase */
int Inc) /* increment (1 for right wing or -1 for left) */
{
float *Hp, *Hdp = NULL, *End;
double a = 0;
float v, t;
Ph *= Npc; /* Npc is number of values per 1/delta in impulse response */
v = 0.0; /* The output value */
Hp = &Imp[(int)Ph];
End = &Imp[Nwing];
if (Interp) {
Hdp = &ImpD[(int)Ph];
a = Ph - floor(Ph); /* fractional part of Phase */
}
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
{ /* ...then we've already skipped the */
Hp += Npc; /* first sample, so we must also */
Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
}
}
if (Interp)
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t += (*Hdp)*a; /* t is now interp'd filter coeff */
Hdp += Npc; /* Filter coeff differences step */
t *= *Xp; /* Mult coeff by input sample */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t *= *Xp; /* Mult coeff by input sample */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return v;
}
float lrsFilterUD(float Imp[], /* impulse response */
float ImpD[], /* impulse response deltas */
UWORD Nwing, /* len of one wing of filter */
BOOL Interp, /* Interpolate coefs using deltas? */
float *Xp, /* Current sample */
double Ph, /* Phase */
int Inc, /* increment (1 for right wing or -1 for left) */
double dhb) /* filter sampling period */
{
float a;
float *Hp, *Hdp, *End;
float v, t;
double Ho;
v = 0.0; /* The output value */
Ho = Ph*dhb;
End = &Imp[Nwing];
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
Ho += dhb; /* ...then we've already skipped the */
} /* first sample, so we must also */
/* skip ahead in Imp[] and ImpD[] */
if (Interp)
while ((Hp = &Imp[(int)Ho]) < End) {
t = *Hp; /* Get IR sample */
Hdp = &ImpD[(int)Ho]; /* get interp bits from diff table*/
a = Ho - floor(Ho); /* a is logically between 0 and 1 */
t += (*Hdp)*a; /* t is now interp'd filter coeff */
t *= *Xp; /* Mult coeff by input sample */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while ((Hp = &Imp[(int)Ho]) < End) {
t = *Hp; /* Get IR sample */
t *= *Xp; /* Mult coeff by input sample */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return v;
}
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