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/*
* 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.
*/
/*
* Sound Tools rate change effect file.
* Spiffy rate changer using Smith & Wesson Bandwidth-Limited Interpolation.
* The algorithm is described in "Bandlimited Interpolation -
* Introduction and Algorithm" by Julian O. Smith III.
* Available on ccrma-ftp.stanford.edu as
* pub/BandlimitedInterpolation.eps.Z or similar.
*/
#include <math.h>
#include "st.h"
/* resample includes */
#include "resdefs.h"
#include "resampl.h"
#define IBUFFSIZE 1024 /* Input buffer size */
#define OBUFFSIZE (IBUFFSIZE*MAXFACTOR+2) /* Calc'd out buffer size */
/* Private data for Lerp via LCM file */
typedef struct resamplestuff {
double Factor; /* Factor = Fout/Fin sample rates */
double rolloff; /* roll-off frequency */
double beta; /* passband/stopband tuning magic */
short InterpFilt; /* TRUE means interpolate filter coeffs */
UHWORD Oskip; /* number of bogus output samples at start */
UHWORD LpScl, Nmult, Nwing;
HWORD *Imp; /* impulse [MAXNWING] Filter coefficients */
HWORD *ImpD; /* [MAXNWING] ImpD[n] = Imp[n+1]-Imp[n] */
/* for resample main loop */
UWORD Time; /* Current time/pos in input sample */
UHWORD Xp, Xoff, Xread;
HWORD *X, *Y; /* I/O buffers */
} *resample_t;
/*
* Process options
*/
resample_getopts(effp, n, argv)
eff_t effp;
int n;
char **argv;
{
resample_t resample = (resample_t) effp->priv;
resample->rolloff = 0.85;
resample->beta = 2.120;
/* I don't know why this fails! */
if ((n >= 1) && !sscanf(argv[0], "%f", &resample->rolloff))
fail("Usage: resample [ rolloff [ beta ] ]");
else if ((resample->rolloff < 0.01) || (resample->rolloff > 1.0))
fail("resample: rolloff factor (%f) no good, should be 0.01<x<1.0",
resample->rolloff);
if ((n >= 2) && !sscanf(argv[1], "%f", &resample->beta))
fail("Usage: resample [ rolloff [ beta ] ]");
else if (resample->beta < 1.0)
fail("resample: beta factor (%f) no good, should be >= 1.0",
resample->beta);
/*
fprintf(stderr, "resample opts: %f, %f\n",
resample->rolloff, resample->beta);
*/
}
/*
* Prepare processing.
*/
resample_start(effp)
eff_t effp;
{
resample_t resample = (resample_t) effp->priv;
int i;
if ((long)effp->ininfo.rate > (long)effp->outinfo.rate)
resample->rolloff = ((double)effp->outinfo.rate /
(double)effp->ininfo.rate) * 0.97; /* empirical */
else
resample->rolloff = 0.85;
resample->InterpFilt = 1; /* interpolate filter: slower */
resample->Factor =
(double)effp->outinfo.rate / (double)effp->ininfo.rate;
/* Check for illegal constants */
if (Np >= 16)
fail("Error: Np>=16");
if (Nb+Nhg+NLpScl >= 32)
fail("Error: Nb+Nhg+NLpScl>=32");
if (Nh+Nb > 32)
fail("Error: Nh+Nb>32");
resample->Imp = (HWORD *) malloc(sizeof(HWORD) * MAXNWING);
resample->ImpD = (HWORD *) malloc(sizeof(HWORD) * MAXNWING);
resample->X = (HWORD *) malloc(sizeof(HWORD) * IBUFFSIZE);
resample->Y = (HWORD *) malloc(sizeof(HWORD) * OBUFFSIZE);
/* upsampling requires smaller Nmults */
for(resample->Nmult = 37; resample->Nmult > 1; resample->Nmult -= 2) {
/* # of filter coeffs in right wing */
resample->Nwing = Npc*(resample->Nmult+1)/2;
/* This prevents just missing last coeff */
/* for integer conversion factors */
resample->Nwing += Npc/2 + 1;
/* returns error # or 0 for success */
if (makeFilter(resample->Imp, resample->ImpD,
&resample->LpScl, resample->Nwing,
resample->rolloff, resample->beta))
continue;
else
break;
}
if(resample->Nmult == 1)
fail("resample: Unable to make filter\n");
if (resample->Factor < 1)
resample->LpScl = resample->LpScl*resample->Factor + 0.5;
/* Calc reach of LP filter wing & give some creeping room */
resample->Xoff = ((resample->Nmult+1)/2.0) *
MAX(1.0,1.0/resample->Factor) + 10;
if (IBUFFSIZE < 2*resample->Xoff) /* Check input buffer size */
fail("IBUFFSIZE (or Factor) is too small");
/* Current "now"-sample pointer for input */
resample->Xp = resample->Xoff;
/* Position in input array to read into */
resample->Xread = resample->Xoff;
/* Current-time pointer for converter */
resample->Time = (resample->Xoff<<Np);
/* Set sample drop at beginning */
resample->Oskip = resample->Xread * resample->Factor;
/* Need Xoff zeros at begining of sample */
for (i=0; i<resample->Xoff; i++)
resample->X[i] = 0;
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
resample_flow(effp, ibuf, obuf, isamp, osamp)
eff_t effp;
long *ibuf, *obuf;
long *isamp, *osamp;
{
resample_t resample = (resample_t) effp->priv;
long i, last, creep, Nout, Nx;
UHWORD Nproc;
/* constrain amount we actually process */
Nproc = IBUFFSIZE - resample->Xp;
if (Nproc * resample->Factor >= OBUFFSIZE)
Nproc = OBUFFSIZE / resample->Factor;
if (Nproc * resample->Factor >= *osamp)
Nproc = *osamp / resample->Factor;
Nx = Nproc - resample->Xread;
if (Nx <= 0)
fail("Nx negative: %d", Nx);
if (Nx > *isamp) {
Nx = *isamp;
}
for(i = resample->Xread; i < Nx + resample->Xread ; i++)
resample->X[i] = RIGHT(*ibuf++ + 0x8000, 16);
last = i;
Nproc = last - (resample->Xoff * 2);
for(; i < last + resample->Xoff ; i++)
resample->X[i] = 0;
/* If we're draining out a buffer tail,
* just do it next time or in drain.
*/
if ((Nx == *isamp) && (Nx <= resample->Xoff)) {
/* fill in starting here next time */
resample->Xread = last;
/* leave *isamp alone, we consumed it */
*osamp = 0;
return;
}
/* SrcUp() is faster if we can use it */
if (resample->Factor > 1) /* Resample stuff in input buffer */
Nout = SrcUp(resample->X, resample->Y,
resample->Factor, &resample->Time, Nproc,
resample->Nwing, resample->LpScl,
resample->Imp, resample->ImpD,
resample->InterpFilt);
else
Nout = SrcUD(resample->X, resample->Y,
resample->Factor, &resample->Time, Nproc,
resample->Nwing, resample->LpScl,
resample->Imp, resample->ImpD,
resample->InterpFilt);
/* Move converter Nproc samples back in time */
resample->Time -= (Nproc<<Np);
/* Advance by number of samples processed */
resample->Xp += Nproc;
/* Calc time accumulation in Time */
creep = (resample->Time>>Np) - resample->Xoff;
if (creep)
{
resample->Time -= (creep<<Np); /* Remove time accumulation */
resample->Xp += creep; /* and add it to read pointer */
}
/* Copy back portion of input signal that must be re-used */
for (i=0; i<last - resample->Xp + resample->Xoff; i++)
resample->X[i] = resample->X[i + resample->Xp - resample->Xoff];
/* Pos in input buff to read new data into */
resample->Xread = i;
resample->Xp = resample->Xoff;
/* copy to output buffer, zero-filling beginning */
/* zero-fill to preserve length and loop points */
for(i = 0; i < resample->Oskip; i++) {
*obuf++ = 0;
}
for(i = resample->Oskip; i < Nout + resample->Oskip; i++) {
*obuf++ = LEFT(resample->Y[i], 16);
}
*isamp = Nx;
*osamp = Nout;
resample->Oskip = 0;
}
/*
* Process tail of input samples.
*/
resample_drain(effp, obuf, osamp)
eff_t effp;
unsigned long *obuf;
unsigned long *osamp;
{
resample_t resample = (resample_t) effp->priv;
long i, Nout;
UHWORD Nx;
Nx = resample->Xread - resample->Xoff;
if (Nx <= resample->Xoff * 2) {
/* zero-fill end */
for(i = 0; i < resample->Xoff; i++)
*obuf++ = 0;
*osamp = resample->Xoff;
return;
}
if (Nx * resample->Factor >= *osamp)
fail("resample_drain: Overran output buffer!\n");
/* fill out end with zeros */
for(i = 0; i < resample->Xoff; i++)
resample->X[i + resample->Xread] = 0;
/* SrcUp() is faster if we can use it */
if (resample->Factor >= 1) /* Resample stuff in input buffer */
Nout = SrcUp(resample->X, resample->Y,
resample->Factor, &resample->Time, Nx,
resample->Nwing, resample->LpScl,
resample->Imp, resample->ImpD,
resample->InterpFilt);
else
Nout = SrcUD(resample->X, resample->Y,
resample->Factor, &resample->Time, Nx,
resample->Nwing, resample->LpScl,
resample->Imp, resample->ImpD,
resample->InterpFilt);
for(i = resample->Oskip; i < Nout; i++) {
*obuf++ = LEFT(resample->Y[i], 16);
}
*osamp = Nout - resample->Oskip;
}
/*
* Do anything required when you stop reading samples.
* Don't close input file!
*/
resample_stop(effp)
eff_t effp;
{
resample_t resample = (resample_t) effp->priv;
free(resample->Imp);
free(resample->ImpD);
free(resample->X);
free(resample->Y);
}
/* From resample:filters.c */
/* Sampling rate up-conversion only subroutine;
* Slightly faster than down-conversion;
*/
SrcUp(X, Y, Factor, Time, Nx, Nwing, LpScl, Imp, ImpD, Interp)
HWORD X[], Y[], Imp[], ImpD[];
UHWORD Nx, Nwing, LpScl;
UWORD *Time;
double Factor;
BOOL Interp;
{
HWORD *Xp, *Ystart;
IWORD v;
double dt; /* Step through input signal */
UWORD dtb; /* Fixed-point version of Dt */
UWORD endTime; /* When Time reaches EndTime, return to user */
dt = 1.0/Factor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
endTime = *Time + (1<<Np)*(IWORD)Nx;
while (*Time < endTime)
{
Xp = &X[*Time>>Np]; /* Ptr to current input sample */
v = FilterUp(Imp, ImpD, Nwing, Interp, Xp, (HWORD)(*Time&Pmask),
-1); /* Perform left-wing inner product */
v += FilterUp(Imp, ImpD, Nwing, Interp, Xp+1, (HWORD)((-*Time)&Pmask),
1); /* Perform right-wing inner product */
v >>= Nhg; /* Make guard bits */
v *= LpScl; /* Normalize for unity filter gain */
*Y++ = v>>NLpScl; /* Deposit output */
*Time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
/* Sampling rate conversion subroutine */
SrcUD(X, Y, Factor, Time, Nx, Nwing, LpScl, Imp, ImpD, Interp)
HWORD X[], Y[], Imp[], ImpD[];
UHWORD Nx, Nwing, LpScl;
UWORD *Time;
double Factor;
BOOL Interp;
{
HWORD *Xp, *Ystart;
IWORD v;
double dh; /* Step through filter impulse response */
double dt; /* Step through input signal */
UWORD endTime; /* When Time reaches EndTime, return to user */
UWORD dhb, dtb; /* Fixed-point versions of Dh,Dt */
dt = 1.0/Factor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
dh = MIN(Npc, Factor*Npc); /* Filter sampling period */
dhb = dh*(1<<Na) + 0.5; /* Fixed-point representation */
Ystart = Y;
endTime = *Time + (1<<Np)*(IWORD)Nx;
while (*Time < endTime)
{
Xp = &X[*Time>>Np]; /* Ptr to current input sample */
v = FilterUD(Imp, ImpD, Nwing, Interp, Xp, (HWORD)(*Time&Pmask),
-1, dhb); /* Perform left-wing inner product */
v += FilterUD(Imp, ImpD, Nwing, Interp, Xp+1, (HWORD)((-*Time)&Pmask),
1, dhb); /* Perform right-wing inner product */
v >>= Nhg; /* Make guard bits */
v *= LpScl; /* Normalize for unity filter gain */
*Y++ = v>>NLpScl; /* Deposit output */
*Time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
void LpFilter();
int makeFilter(Imp, ImpD, LpScl, Nwing, Froll, Beta)
HWORD Imp[], ImpD[];
UHWORD *LpScl, Nwing;
double Froll, Beta;
{
double DCgain, Scl, Maxh;
double *ImpR;
HWORD Dh;
long i, temp;
if (Nwing > MAXNWING) /* Check for valid parameters */
return(1);
if ((Froll<=0) || (Froll>1))
return(2);
if (Beta < 1)
return(3);
ImpR = (double *) malloc(sizeof(double) * MAXNWING);
LpFilter(ImpR, (int)Nwing, Froll, Beta, Npc); /* Design a Kaiser-window */
/* Sinc low-pass filter */
/* Compute the DC gain of the lowpass filter, and its maximum coefficient
* magnitude. Scale the coefficients so that the maximum coeffiecient just
* fits in Nh-bit fixed-point, and compute LpScl as the NLpScl-bit (signed)
* scale factor which when multiplied by the output of the lowpass filter
* gives unity gain. */
DCgain = 0;
Dh = Npc; /* Filter sampling period for factors>=1 */
for (i=Dh; i<Nwing; i+=Dh)
DCgain += ImpR[i];
DCgain = 2*DCgain + ImpR[0]; /* DC gain of real coefficients */
for (Maxh=i=0; i<Nwing; i++)
Maxh = MAX(Maxh, fabs(ImpR[i]));
Scl = ((1<<(Nh-1))-1)/Maxh; /* Map largest coeff to 16-bit maximum */
temp = fabs((1<<(NLpScl+Nh))/(DCgain*Scl));
if (temp >= 1L<<16) {
free(ImpR);
return(4); /* Filter scale factor overflows UHWORD */
}
*LpScl = temp;
/* Scale filter coefficients for Nh bits and convert to integer */
if (ImpR[0] < 0) /* Need pos 1st value for LpScl storage */
Scl = -Scl;
for (i=0; i<Nwing; i++) /* Scale them */
ImpR[i] *= Scl;
for (i=0; i<Nwing; i++) /* Round them */
Imp[i] = ImpR[i] + 0.5;
/* ImpD makes linear interpolation of the filter coefficients faster */
for (i=0; i<Nwing-1; i++)
ImpD[i] = Imp[i+1] - Imp[i];
ImpD[Nwing-1] = - Imp[Nwing-1]; /* Last coeff. not interpolated */
free(ImpR);
return(0);
}
/* 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 */
double Izero(x)
double x;
{
double sum, u, halfx, temp;
long 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 LpFilter(c,N,frq,Beta,Num)
double c[], frq, Beta;
int N, Num;
{
double IBeta, temp;
int i;
/* Calculate filter coeffs: */
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;
}
/* Calculate and Apply Kaiser window to filter coeffs: */
IBeta = 1.0/Izero(Beta);
for (i=1; i<N; i++)
{
temp = (double)i / ((double)N * (double)1.0);
c[i] *= Izero(Beta*sqrt(1.0-temp*temp)) * IBeta;
}
}
IWORD FilterUp(Imp, ImpD, Nwing, Interp, Xp, Ph, Inc)
HWORD Imp[], ImpD[], *Xp, Inc, Ph;
UHWORD Nwing;
BOOL Interp;
{
HWORD a, *Hp, *Hdp, *End;
IWORD v, t;
v=0;
Hp = &Imp[Ph>>Na];
End = &Imp[Nwing];
if (Interp)
{
Hdp = &ImpD[Ph>>Na];
a = Ph & Amask;
}
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[] */
}
}
while (Hp < End)
{
t = *Hp; /* Get filter coeff */
if (Interp)
{
t += (((IWORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
Hdp += Npc; /* Filter coeff differences step */
}
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON ARRAY BOUNDS */
}
return(v);
}
IWORD FilterUD(Imp, ImpD, Nwing, Interp, Xp, Ph, Inc, dhb)
HWORD Imp[], ImpD[], *Xp, Ph, Inc;
UHWORD Nwing, dhb;
BOOL Interp;
{
HWORD a, *Hp, *Hdp, *End;
IWORD v, t;
UWORD Ho;
v=0;
Ho = (Ph*(UWORD)dhb)>>Np;
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[] */
while ((Hp = &Imp[Ho>>Na]) < End)
{
t = *Hp; /* Get IR sample */
if (Interp)
{
Hdp = &ImpD[Ho>>Na]; /* get interp (lower Na) bits from diff table */
a = Ho & Amask; /* a is logically between 0 and 1 */
t += (((IWORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
}
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON ARRAY BOUNDS */
}
return(v);
}
These are the contents of the former NiCE NeXT User Group NeXTSTEP/OpenStep software archive, currently hosted by Netfuture.ch.