musicdsp.org source code archive - WSInf

}
}
float x = (phase-0.5)/freq;
sum -= exp(pole*x)-1.0f;
phase -= 1;
return sum.real();
There's big speedup potential in this i think.
Since the filtering is done "before" sampling,
aliasing is reduced, as a free bonus. with high cutoff
and high frequencies it's still audible though, but
much less than without the filter.
If aliasing is handled in some other way,
a digital filter will sound just as well, that's what i think.
-- Magnus Jonsson