An article in Swedish about digitalization and the sinc function: http://www.lts.a.se/artiklar/sincen.pdf

A discussion in Swedish on a forum about filters and pre-ringing: http://www.faktiskt.se/modules.php?name=Forums&file=viewtopic&p=251444

Basically, I think that most A/D-D/As are conceived with the assumption of the sinc, although there may be exceptions to this. With digital filters, I think the pre-ringing is no problem.

Basically, how can the D/A can know how the A/D was made? Jitter is just one issue of many here, and I’m starting to think this filtering business will prove very interesting.

I was not aware that there was filtering at this step; I have only read up on D/A, not A/D. I have some work to do…

I am somehat confused by the Wikipedia graph though:
http://en.wikipedia.org/wiki/File:Sinc_function_%28both%29.svg

Wouldn’t this indicate that the resulting signal, which is to be digitized, is in effect detuned from the original signal? The red line has different frequencies, and it’s not halved either. I’m in over my head now, but in this exampe it looks as though this particular filter implementation would be detrimental to the music rather than helping the listener approximate signal data in between sample points.

That was my impression from reading this site, otherwise I wouldn’t have bothered. :-)

“But what you are saying is that B is a representation of everything that happened between these two tics, and that a transient at +20 between A and B would have given B another value even if the amplitude was at the equivalent of 5 for that clock tick.

If this is what you mean…”

Well, not quite. The point is that the analog signal is filtered through a sinc filter (and thus transformed) BEFORE it is sampled. Imagine an very (infinitely, you migh say) narrow pulse between two sampling points. This would indeed be “invisible” for the sampling function. But since it is first sinc filtered, it is transformed into another shape (see the curve on the wiki page), and this curve has values ON the sample points, and so it is “visible”.

It is not easy to explain in words only. I have some links to quite informative articles, if you are interested.

Lest say an analog signal results in sample points A, B and C (a very short signal, in order to simplify this argument). These have amplitudes of 0, 5 and -4 respectively. What I was saying is that sample point B, with a value of 5, would be 5 regardless of what happened in the analog domain between the clock tics at sample points A and B. But what you are saying is that B is a representation of everything that happened between these two tics, and that a transient at +20 between A and B would have given B another value even if the amplitude was at the equivalent of 5 for that clock tick.

If this is what you mean, I understand your argument but I’m still very skeptical: A sonic event between two sample points at 44,1kHz would by definition be above 22,05kHz, wouldn’t it? And as such irretrievable from the data, just as the 30kHz tone in my example.

I am unfortunately no mathematician (as I ended up following the desires of the right side of my brain instead), so the Wikipedia article is a little difficult to digest. Nonetheless, I appreciate the discussion and would like to emphasize that I desire to learn the truth, rather than to claim I already know the truth.