rateconv.cc

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/*
 *  $Id: rateconv.cc,v 1.4 2004/09/29 08:24:16 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);
        }
        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;

}