Tyler Eaves wrote:
> On Tue, 09 Mar 2004 22:37:18 +0000, scott wrote:
>
> > laurent.artaud[AT]free.fr" <"laurent.artaud[AT]free.fr wrote:
> > > > I will have to try that one out when I get some sound editing
> > > > software installed (just reinstalled Windows).  What CD player
> > > > did you use to play back the audio CD?
> > > >
> > >
> > > I did not test it myself, sorry if I may have implied it
> > > unknowingly. Anyway, my explanation of the process was simplified
> > > for a text demo.
> > >
> > > Note that (if I wasn't clear) a function generator is an
> > > electronic device (I don't have any, it's quite expensive) used
> > > in laboratories to test other electronic devices. It have nothing
> > > to do with any software in the sense that the signal stays
> > > analogous all the way and is in no way digitalized.
> > >
> > > On the other end, using a software to generate the waveforms to
> > > create the CD is a far better way than trying to sample the
> > > function generator's output.
> >
> > Indeed, I was thinking about digital generation of signals, sonic
> > Ray Tracing :-)
> >
> > > Testing the 24/96 signal would be harder:
> > > 1) you have to find a software who can generate it ;
> > > 2) you have to play it.
> >
> > Yeah, I have software and hardware to play 24/96 from my PC, it's
> > just I doubt my speakers could keep up [just looking], apparently
> > they are down -3dB at 22kHz, so who knows what they'll be down to
> > at 48kHz.
>
> That's not the point.

It is the point, I was saying that my speakers couldn't reproduce
frequencies higher than 22kHz accurately.

> It's not like 44khz gives perfect signal at
> anything below 22khz, and nothing above it.

Not perfect, but perfect in terms of most peoples' ears.  You *cannot*
sample a frequency higher than 22kHz at a 44kHz sampling rate - any analogue
to digital converter will filter higher frequencies out before it samples.

> A 22khz wave form at 44khz looks like this:
>  * * * * * * * * * * *
> ------------------------------
> * * * * * * * * * * * *
>
> Regardless of the actual shape of the wave.

That's not entirely true, the 22kHz must be put through an anti-alias filter
first before it is sampled.  And guess what?  That cuts off anything above
22kHz.  Hence, the actual analogue wave you are sampling will be exactly the
same whether it was a square wave, triangle wave, sine wave or whatever to
start with.

When you convert it back to analogue, it will come out as a 22kHz sine wave,
which is identical to what was _actually_ sampled.

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