ref: 8791cbd4a9f89b06f87ea3f85f8c2d2432bf8903
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) { st_fail("Deemphasis filtering effect takes no options.\n"); return (ST_EOF); } if (sizeof(double)*ST_MAX_PRIVSIZE < sizeof(struct deemphstuff)) { st_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) { st_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); }