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rateconv.cc
/*
* $Id: rateconv.cc,v 1.5 2014/04/07 15:32:10 robert Exp $
*
* RATECONV.C
*
* Convert sampling rate stdin to stdout
*
* Copyright (c) 1992, 1995 by Markus Mummert
*
*****************************************************************************
* MODIFIED BY Alan W Black (awb@cstr.ed.ac.uk)
* Make it compilable under C++
* and integrate into Edinburgh Speech Tools (i.e. no longer
* reads from stdin / writes to stdout)
* Removed interface functions
* ansified function calls
* made it work in floats rather than ints
* I got the original from a random linux site, the original
* author's email is <mum@mmk.e-technik.tu-muenchen.de>
*****************************************************************************
*
* Redistribution and use of this software, modification and inclusion
* into other forms of software are permitted provided that the following
* conditions are met:
*
* 1. Redistributions of this software must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. If this software is redistributed in a modified condition
* it must reveal clearly that it has been modified.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS''
* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A
* PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
* OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE
* USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
* DAMAGE.
*
*
* history: 2.9.92 begin coding
* 5.9.92 fully operational
* 14.2.95 provide BIG_ENDIAN, SWAPPED_BYTES_DEFAULT
* switches, Copyright note and References
* 25.11.95 changed XXX_ENDIAN to I_AM_XXX_ENDIAN
* default gain set to 0.8
* 3.12.95 stereo implementation
* SWAPPED_BYTES_DEFAULT now HBYTE1ST_DEFAULT
* changed [L/2] to (L-1)/2 for exact symmetry
*
*
* IMPLEMENTATION NOTES
*
* Converting is achieved by interpolating the input samples in
* order to evaluate the represented continuous input slope at
* sample instances of the new rate (resampling). It is implemented
* as a polyphase FIR-filtering process (see reference). The rate
* conversion factor is determined by a rational factor. Its
* nominator and denominator are integers of almost arbitrary
* value, limited only by coefficient memory size.
*
* General rate conversion formula:
*
* out(n*Tout) = SUM in(m*Tin) * g((n*d/u-m)*Tin) * Tin
* over all m
*
* FIR-based rate conversion formula for polyphase processing:
*
* L-1
* out(n*Tout) = SUM in(A(i,n)*Tin) * g(B(i,n)*Tin) * Tin
* i=0
*
* A(i,n) = i - (L-1)/2 + [n*d/u]
* = i - (L-1)/2 + [(n%u)*d/u] + [n/u]*d
* B(i,n) = n*d/u - [n*d/u] + (L-1)/2 - i
* = ((n%u)*d/u)%1 + (L-1)/2 - i
* Tout = Tin * d/u
*
* where:
* n,i running integers
* out(t) output signal sampled at t=n*Tout
* in(t) input signal sampled in intervals Tin
* u,d up- and downsampling factor, integers
* g(t) interpolating function
* L FIR-length of realized g(t), integer
* / float-division-operator
* % float-modulo-operator
* [] integer-operator
*
* note:
* (L-1)/2 in A(i,n) can be omitted as pure time shift yielding
* a causal design with a delay of ((L-1)/2)*Tin.
* n%u is a cyclic modulo-u counter clocked by out-rate
* [n/u]*d is a d-increment counter, advanced when n%u resets
* B(i,n)*Tin can take on L*u different values, at which g(t)
* has to be sampled as a coefficient array
*
* Interpolation function design:
*
* The interpolation function design is based on a sinc-function
* windowed by a gaussian function. The former determines the
* cutoff frequency, the latter limits the necessary FIR-length by
* pushing the outer skirts of the resulting impulse response below
* a certain threshold fast enough. The drawback is a smoothed
* cutoff inducing some aliasing. Due to the symmetry of g(t) the
* group delay of the filtering process is constant (linear phase).
*
* g(t) = 2*fgK*sinc(pi*2*fgK*t) * exp(-pi*(2*fgG*t)**2)
*
* where:
* fgK cutoff frequency of sinc function in f-domain
* fgG key frequency of gaussian window in f-domain
* reflecting the 6.82dB-down point
*
* note:
* Taking fsin=1/Tin as the input sampling frequency, it turns out
* that in conjunction with L, u and d only the ratios fgK/(fsin/2)
* and fgG/(fsin/2) specify the whole process. Requiring fsin, fgK
* and fgG as input purposely keeps the notion of absolute
* frequencies.
*
* Numerical design:
*
* Samples are expected to be 16bit-signed integers, alternating
* left and right channel in case of stereo mode- The byte order
* per sample is selectable. FIR-filtering is implemented using
* 32bit-signed integer arithmetic. Coefficients are scaled to
* find the output sample in the high word of accumulated FIR-sum.
*
* Interpolation can lead to sample magnitudes exceeding the
* input maximum. Worst case is a full scale step function on the
* input. In this case the sinc-function exhibits an overshoot of
* 2*9=18percent (Gibb's phenomenon). In any case sample overflow
* can be avoided by a gain of 0.8.
*
* If u=d=1 and if the input stream contains only a single sample,
* the whole length of the FIR-filter will be written to the output.
* In general the resulting output signal will be (L-1)*fsout/fsin
* samples longer than the input signal. The effect is that a
* finite input sequence is viewed as padded with zeros before the
* `beginning' and after the `end'.
*
* The output lags ((L-1)/2)*Tin behind to implement g(t) as a
* causal system corresponding to a causal relationship of the
* discrete-time sequences in(m/fsin) and out(n/fsout) with
* respect to a sequence time origin at t=n*Tin=m*Tout=0.
*
*
* REFERENCES
*
* Crochiere, R. E., Rabiner, L. R.: "Multirate Digital Signal
* Processing", Prentice-Hall, Englewood Cliffs, New Jersey, 1983
*
* Zwicker, E., Fastl, H.: "Psychoacoustics - Facts and Models",
* Springer-Verlag, Berlin, Heidelberg, New-York, Tokyo, 1990
*/
#include <cmath>
#include <cstdio>
#include <fcntl.h>
#include <cstring>
#include "rateconv.h"
/*
* adaptable defines and globals
*/
#define BYTE char /* signed or unsigned */
#define WORD short /* signed or unsigned, fit two BYTEs */
#define LONG int /* signed, fit two WORDs */
#ifndef MAXUP
#define MAXUP 0x400 /* MAXUP*MAXLENGTH worst case malloc */
#endif
#ifndef MAXLENGTH
#define MAXLENGTH 0x400 /* max FIR length */
#endif
/* accounts for mono samples, means */
#define OUTBUFFSIZE (2*MAXLENGTH) /* fit >=MAXLENGHT stereo samples */
#define INBUFFSIZE (4*MAXLENGTH) /* fit >=2*MAXLENGTH stereo samples */
#define sqr(a) ((a)*(a))
#ifndef M_PI
#define M_PI 3.14159265358979
#endif
/* AWB deleted previous byte swap globals, byteswap is done external to */
/* this function */
#ifdef STEREO_DEFAULT
static int g_monoflag = 0;
#else
static int g_monoflag = -1;
#endif
/*
* other globals
*/
static double g_ampli = 0.8; /* default gain, don't change */
static int
/* g_infilehandle = 0, */ /* stdin */
/* g_outfilehandle = 1, */ /* stdout */
g_firlen, /* FIR-length */
g_up, /* upsampling factor */
g_down /* downsampling factor */
;
static float
g_sin[INBUFFSIZE], /* input buffer */
g_sout[OUTBUFFSIZE], /* output buffer */
*g_coep; /* coefficient array pointer */
static double
g_fsi, /* input sampling frequency */
g_fgk, /* sinc-filter cutoff frequency */
g_fgg /* gaussian window key frequency */
; /* (6.8dB down freq. in f-domain) */
/*
* evaluate sinc(x) = sin(x)/x safely
*/
static double sinc(double x)
{
return(fabs(x) < 1E-50 ? 1.0 : sin(fmod(x,2*M_PI))/x);
}
/*
* evaluate interpolation function g(t) at t
* integral of g(t) over all times is expected to be one
*/
static double interpol_func(double t,double fgk,double fgg)
{
return (2*fgk*sinc(M_PI*2*fgk*t)*exp(-M_PI*sqr(2*fgg*t)));
}
/*
* evaluate coefficient from i, q=n%u by sampling interpolation function
* and scale it for integer multiplication used by FIR-filtering
*/
static float coefficient(int i,int q,int firlen,double fgk,double fgg,
double fsi,int up,int down,double amp)
{
float val;
double d;
d = interpol_func((fmod(q*down/(double)up,1.) + (firlen-1)/2. - i) / fsi,
fgk,
fgg);
val = amp * d/fsi;
return val;
}
/*
* transfer n floats from s to d
*/
static void transfer_int(float *s,float *d,int n)
{
memmove(d,s,sizeof(float)*n);
}
/*
* zerofill n floats from s
*/
static void zerofill(float *s,int n)
{
memset(s,0,n*(sizeof(float)));
}
/*
* FIR-routines, mono and stereo
* this is where we need all the MIPS
*/
void fir_mono(float *inp,float *coep,int firlen,float *outp)
{
float akku = 0, *endp;
int n1 = (firlen / 8) * 8, n0 = firlen % 8;
endp = coep + n1;
while (coep != endp) {
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
akku += *inp++ * *coep++;
}
endp = coep + n0;
while (coep != endp) {
akku += *inp++ * *coep++;
}
*outp = akku;
}
static void fir_stereo(float *inp,float *coep,int firlen,float *out1p,float *out2p)
{
float akku1 = 0, akku2 = 0, *endp;
int n1 = (firlen / 8) * 8, n0 = firlen % 8;
endp = coep + n1;
while (coep != endp) {
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
}
endp = coep + n0;
while (coep != endp) {
akku1 += *inp++ * *coep;
akku2 += *inp++ * *coep++;
}
*out1p = akku1;
*out2p = akku2;
}
/*
* filtering from input buffer to output buffer;
* returns number of processed samples in output buffer:
* if it is not equal to output buffer size,
* the input buffer is expected to be refilled upon entry, so that
* the last firlen numbers of the old input buffer are
* the first firlen numbers of the new input buffer;
* if it is equal to output buffer size, the output buffer
* is full and is expected to be stowed away;
*
*/
static int inbaseidx = 0, inoffset = 0, cycctr = 0, outidx = 0;
static int filtering_on_buffers
(float *inp,int insize,float *outp, int outsize,
float *coep,int firlen,int up,int down,int monoflag)
{
if (monoflag) {
while (-1) {
inoffset = (cycctr * down)/up;
if ((inbaseidx + inoffset + firlen) > insize) {
inbaseidx -= insize - firlen + 1;
return(outidx);
}
fir_mono(inp + inoffset + inbaseidx,
coep + cycctr * firlen,
firlen, outp + outidx++);
cycctr++;
if (!(cycctr %= up))
inbaseidx += down;
if (!(outidx %= outsize))
return(outsize);
}
}
else {
/*
* rule how to convert mono routine to stereo routine:
* firlen, up, down and cycctr relate to samples in general,
* wether mono or stereo; inbaseidx, inoffset and outidx as
* well as insize and outsize still account for mono samples.
*/
while (-1) {
inoffset = 2*((cycctr * down)/up);
if ((inbaseidx + inoffset + 2*firlen) > insize) {
inbaseidx -= insize - 2*firlen + 2;
return(outidx);
}
/* order?
fir_stereo(inp + inoffset + inbaseidx,
coep + cycctr * firlen, firlen,
outp + outidx++, outp + outidx++);
*/
fir_stereo(inp + inoffset + inbaseidx,
coep + cycctr * firlen, firlen,
outp + outidx, outp + outidx+1);
outidx += 2;
cycctr++;
if (!(cycctr %= up))
inbaseidx += 2*down;
if (!(outidx %= outsize))
return(outsize);
}
}
}
/*
* set up coefficient array
*/
static void make_coe(void)
{
int i, q;
for (i = 0; i < g_firlen; i++) {
for (q = 0; q < g_up; q++) {
g_coep[q * g_firlen + i] = coefficient(i, q, g_firlen,
g_fgk, g_fgg, g_fsi, g_up, g_down, g_ampli);
}
}
}
/***********************************************************************/
/* Serious modifications by Alan W Black (awb@cstr.ed.ac.uk) */
/* to interface with rest of system // deleted various io functions */
/* too. */
/***********************************************************************/
static WORD *inbuff = NULL;
static int inpos;
static int inmax;
static WORD *outbuff = NULL;
static int outpos;
static int outmax;
static int ioerr(void)
{
delete[] g_coep;
return -1;
}
static int gcd(int x, int y)
{
int remainder,a,b;
if ((x < 1) || (y < 1))
return -1;
for (a=x,b=y; b != 0; )
{
remainder = a % b;
a = b;
b = remainder;
}
return a;
}
static int find_ratios(int in_samp_freq,int out_samp_freq,int *up,int *down)
{
// Find ratios
int d;
d = gcd(in_samp_freq,out_samp_freq);
if (d == -1) return -1;
*down = in_samp_freq / d;
*up = out_samp_freq / d;
if ((*up > 1024) || (*down > 1024))
return -1; // should try harder
return 0;
}
static int intimport(float *buff, int n)
{
/* Import n more samples from PWave into buff */
int i,end;
if ((inpos+n) >= inmax)
end = inmax - inpos;
else
end = n;
for (i=0;i < end; i++)
buff[i] = inbuff[inpos++];
return i;
}
static int intexport(float *buff, int n)
{
/* Export n samples from buff into end of PWave */
int i,end;
if ((outpos+n) >= outmax)
end = outmax - inpos;
else
end = n;
for (i=0;i < end; i++)
outbuff[outpos++] = (short)buff[i];
return i;
}
static int init_globs(WORD *in,int insize, WORD **out, int *outsize,
int in_samp_freq, int out_samp_freq)
{
int new_size;
g_monoflag = 1; /* always mono */
if (find_ratios(in_samp_freq,out_samp_freq,&g_up,&g_down) == -1)
return -1;
g_fsi = 1.0; /* ? in_samp_freq ? */
if (g_up > g_down)
{ // upsampling
g_fgg = 0.0116;
g_fgk = 0.461;
g_firlen = (int)(162 * (float)g_up/(float)g_down);
}
else
{ // downsampling
g_fgg = (float)g_up/(float)g_down * 0.0116;
g_fgk = (float)g_up/(float)g_down * 0.461;
g_firlen = (int)(162 * (float)g_down/(float)g_up);
}
if (g_firlen < 1 || g_firlen > MAXLENGTH)
return -1;
g_ampli = 0.8;
g_coep = new float[g_firlen * g_up];
inpos = 0;
inmax = insize;
inbuff = in;
new_size = (int)(((float)out_samp_freq/(float)in_samp_freq)*
1.1*insize)+2000;
*out = new WORD[new_size];
outbuff = *out;
outmax = new_size;
*outsize = 0;
outpos = 0;
/* For filter_on_buffers */
inbaseidx = 0;
inoffset = 0;
cycctr = 0;
outidx = 0;
return 0;
}
/*
* External call added by Alan W Black, 4th June 1996
* a combination of parse args and main
*/
int rateconv(short *in,int isize, short **out, int *osize,
int in_samp_freq, int out_samp_freq)
{
int insize = 0, outsize = 0, skirtlen;
if (init_globs(in,isize,out,osize,in_samp_freq,out_samp_freq) == -1)
return -1;
make_coe();
skirtlen = (g_firlen - 1) * (g_monoflag ? 1 : 2);
zerofill(g_sin, skirtlen);
do {
insize = intimport(g_sin + skirtlen, INBUFFSIZE - skirtlen);
if (insize < 0 || insize > INBUFFSIZE - skirtlen)
return ioerr();
do {
outsize = filtering_on_buffers(g_sin, skirtlen + insize,
g_sout, OUTBUFFSIZE,
g_coep, g_firlen, g_up, g_down,
g_monoflag);
if (outsize != OUTBUFFSIZE) {
transfer_int(g_sin + insize, g_sin, skirtlen);
break;
}
if (intexport(g_sout, outsize) != outsize)
return ioerr();
} while (-1);
} while (insize > 0);
zerofill(g_sin + skirtlen, skirtlen);
do {
outsize = filtering_on_buffers(g_sin, skirtlen + skirtlen,
g_sout, OUTBUFFSIZE,
g_coep, g_firlen, g_up, g_down,
g_monoflag);
if (intexport(g_sout, outsize) != outsize)
return ioerr();
} while (outsize == OUTBUFFSIZE);
delete[] g_coep;
*osize = outpos;
/* The new signal will be offset by half firlen window so fix it */
memmove(*out,*out+g_firlen/4,*osize*2);
*osize -= g_firlen/4;
return 0;
}