ref: 5c49ab960f58e6f80c4270326b08357b6edc942d
dir: /src/deemphas.c/
/*
* July 5, 1991
*
* Deemphases Filter
*
* Fixed deemphasis filter for processing pre-emphasized audio cd samples
* 09/02/98 (c) Heiko Ei�feldt
* License: LGPL (Lesser Gnu Public License)
*
* This implements the inverse filter of the optional pre-emphasis stage as
* defined by ISO 908 (describing the audio cd format).
*
* Background:
* In the early days of audio cds, there were recording problems
* with noise (for example in classical recordings). The high dynamics
* of audio cds exposed these recording errors a lot.
*
* The commonly used solution at that time was to 'pre-emphasize' the
* trebles to have a better signal-noise-ratio. That is trebles were
* amplified before recording, so that they would give a stronger
* signal compared to the underlying (tape)noise.
*
* For that purpose the audio signal was prefiltered with the following
* frequency response (simple first order filter):
*
* V (in dB)
* ^
* |
* | _________________
* | /
* | / |
* | 20 dB / decade ->/ |
* | / |
* |____________________/_ _ |_ _ _ _ _ _ _ _ _ _ _ _ _ lg f
* |0 dB | |
* | | |
* | | |
* 3.1KHz ca. 10KHz
*
* So the recorded audio signal has amplified trebles compared to the
* original.
* HiFi cd players do correct this by applying an inverse filter
* automatically, the cd-rom drives or cd burners used by digital
* sampling programs (like cdda2wav) however do not.
*
* So, this is what this effect does.
*
* Here is the gnuplot file for the frequency response
of the deemphasis. The error is below +-0.1dB
-------- Start of gnuplot file ---------------------
# first define the ideal filter. We use the tenfold sampling frequency.
T=1./441000.
OmegaU=1./15E-6
OmegaL=15./50.*OmegaU
V0=OmegaL/OmegaU
H0=V0-1.
B=V0*tan(OmegaU*T/2.)
# the coefficients follow
a1=(B - 1.)/(B + 1.)
b0=(1.0 + (1.0 - a1) * H0/2.)
b1=(a1 + (a1 - 1.0) * H0/2.)
# helper variables
D=b1/b0
o=2*pi*T
H2(f)=b0*sqrt((1+2*cos(f*o)*D+D*D)/(1+2*cos(f*o)*a1+a1*a1))
#
# now approximate the ideal curve with a fitted one for sampling
frequency
# of 44100 Hz. Fitting parameters are
# amplification at high frequencies V02
# and tau of the upper edge frequency OmegaU2 = 2 *pi * f(upper)
T2=1./44100.
V02=0.3365
OmegaU2=1./19E-6
B2=V02*tan(OmegaU2*T2/2.)
# the coefficients follow
a12=(B2 - 1.)/(B2 + 1.)
b02=(1.0 + (1.0 - a12) * (V02-1.)/2.)
b12=(a12 + (a12 - 1.0) * (V02-1.)/2.)
# helper variables
D2=b12/b02
o2=2*pi*T2
H(f)=b02*sqrt((1+2*cos(f*o2)*D2+D2*D2)/(1+2*cos(f*o2)*a12+a12*a12))
# plot best, real, ideal, level with halved attenuation,
# level at full attentuation, 10fold magnified error
set logscale x
set grid xtics ytics mxtics mytics
plot [f=1000:20000] [-12:2] 20*log10(H(f)),20*log10(H2(f)),
20*log10(OmegaL/(2*
pi*f)), 0.5*20*log10(V0), 20*log10(V0), 200*log10(H(f)/H2(f))
pause -1 "Hit return to continue"
-------- End of gnuplot file ---------------------
*/
/*
* adapted from Sound Tools skeleton effect file.
*/
#include <math.h>
#include "st.h"
/* Private data for deemph file */
typedef struct deemphstuff {
LONG lastin;
double lastout;
} *deemph_t;
/*
* Process options
*
* Don't do initialization now.
* The 'info' fields are not yet filled in.
*/
int st_deemph_getopts(effp, n, argv)
eff_t effp;
int n;
char **argv;
{
if (n)
{
fail("Deemphasis filtering effect takes no options.\n");
return (ST_EOF);
}
if (sizeof(double)*ST_MAX_PRIVSIZE < sizeof(struct deemphstuff))
{
fail("Internal error: PRIVSIZE too small.\n");
return (ST_EOF);
}
return (ST_SUCCESS);
}
/*
* Prepare processing.
* Do all initializations.
*/
int st_deemph_start(effp)
eff_t effp;
{
/* check the input format */
if (effp->ininfo.encoding != ST_ENCODING_SIGN2
|| effp->ininfo.rate != 44100
|| effp->ininfo.size != ST_SIZE_WORD)
{
fail("The deemphasis effect works only with audio cd like samples.\nThe input format however has %d Hz sample rate and %d-byte%s signed linearly coded samples.",
effp->ininfo.rate, effp->ininfo.size,
effp->ininfo.encoding != ST_ENCODING_SIGN2 ? ", but not" : "");
return (ST_EOF);
}
else
{
deemph_t deemph = (deemph_t) effp->priv;
deemph->lastin = 0;
deemph->lastout = 0.0;
}
return (ST_SUCCESS);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
/* filter coefficients */
#define a1 -0.62786881719628784282
#define b0 0.45995451989513153057
#define b1 -0.08782333709141937339
int st_deemph_flow(effp, ibuf, obuf, isamp, osamp)
eff_t effp;
LONG *ibuf, *obuf;
LONG *isamp, *osamp;
{
deemph_t deemph = (deemph_t) effp->priv;
int len, done;
len = ((*isamp > *osamp) ? *osamp : *isamp);
for(done = len; done; done--) {
deemph->lastout = *ibuf * b0 +
deemph->lastin * b1 -
deemph->lastout * a1;
deemph->lastin = *ibuf++;
*obuf++ = deemph->lastout > 0.0 ?
deemph->lastout + 0.5 :
deemph->lastout - 0.5;
}
return (ST_SUCCESS);
}
/*
* Drain out remaining samples if the effect generates any.
*/
int st_deemph_drain(effp, obuf, osamp)
LONG *obuf;
LONG *osamp;
{
/* nothing to do */
return (ST_SUCCESS);
}
/*
* Do anything required when you stop reading samples.
* (free allocated memory, etc.)
*/
int st_deemph_stop(effp)
eff_t effp;
{
/* nothing to do */
return (ST_SUCCESS);
}