Re: Sega sound boards...

From: <jwelser_at_ccwf.cc.utexas.edu>
Date: Fri Dec 05 1997 - 08:02:52 EST

On Fri, 5 Dec 1997, mayday19 wrote:

> OK, I figured I should have left it alone.. please post it if you ever
> decide to type it in Joe. :>

        Ugh...someone please tell me why I'm reading my email/newsgroups
when I have a big project due ~3pm :-( Anyways...
 
> > How to deal with this easily? (at least with state of the shelf DACs circa
> > 1985) Over sampling. In a grossly simplified view, oversampling adds
> > additional stair steps to the square signal, making it more "analog"
> >like.
>
> so this is part of Quantizing then?

        No, quantizing is the "Resolution" of those stair steps (i.e. how
many different places they can be. There are only a finite number of
digital values that you can quantize to. Oversampling will not effect
the quantization process at all.

        When you're not dealing with an oversampled converter, I don't see
any inherent benefit of oversampling other than to make the design of your
final, analog low-pass filter after your DAC easier (i.e. lower order.)
Probably not worth schleping around all the additional samples, as I think
the stuff in our chips is 2nd or 3rd order Butterworth (i.e. pretty
straightforward) anyways (No, you don't all have to go sign NDAs, that
stuff is in our datasheets ;-) )
 
> I though oversampling was just for aiding error correction at playback...
> in conjunction with that process that scatters the data acoss the disc so
> a scratch will not wipe out a good chunk of the data (what was this
> process called again?)

        Ahhh, could be. I'm not too familiar with that kind of stuff.

> > Apply a little analog filtering to smooth out the edges, and presumably
> >you
> > have a nice signal. Apply too much, and things sound like a dog shitting in a
> > swimming pool (all muddy).
>
> I thought there was just another low-pass (anti-aliasing) filter after
> the D/A convertor.

        There is, but its purpose isn't to prevent aliasing. Aliasing
can't occur in D -> A conversion. If you go through the math, basically
that low-pass filter replaces all the "sharp corners" of the stairstep
analog waveform out of the DAC with sync functions (basically sin(x)/x),
which, if you sampled the signal corrrectly, will all add up to give you
the proper (i.e. non-aliased) waveform.
 
        I probably just confused you more, so I should probably just keep
my mouth shut in the future.....

Joe
Received on Fri Dec 5 05:05:44 1997

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