Re: [LAU] jack/oversampling

On Sun, Mar 16, 2014 at 07:50:57PM +0100, Jörn Nettingsmeier wrote: Bi-Linear Transform, a mathematical trick used to transform the transfer function of an analog filter (in the s-domain) into that of a digital one (in the z-domain).

It introduces a 'warping' of the frequency axis. If A(f) is the frequency response of the analog filter, and D(f) that of the digital one, they will be different but there is a function F(f) such that D(F(f)) = A(f). And it's always possible to arrange things such that the 'defining' point of the FR (the -3 dB point, or the center frequency) is correct - by applying the inverse of F(f). For low frequencies F(f) = f, so the two filter are the same. But as f->inf, F(f)->Fs/2. So filters in the upper part of the frequency range will have a different shape of the FR. For example a standard 2nd order parametric will have a symmetric shape when plotted on a log frequency scale, but the equivalent digital one won't if the center frequency is high. But there is no reason to say that one is 'better' than the other, there is no reason why any digital filter used in audio processing should be an exact copy of an analog one.