Path: utzoo!attcan!uunet!cbmvax!augi
From: augi@cbmvax.UUCP (Joe Augenbraun)
Newsgroups: comp.sys.amiga
Subject: Re: Sampling at 29KHz (long)
Message-ID: <3891@cbmvax.UUCP>
Date: 27 May 88 22:04:03 GMT
References: <2845@polya.STANFORD.EDU> <734@eos.UUCP> <53788@sun.uucp> <5637@cup.portal.com> <3854@cbmvax.UUCP> <5872@cup.portal.com>
Reply-To: augi@cbmvax.UUCP (Joe Augenbraun)
Organization: Commodore Technology, West Chester, PA
Lines: 27

First of all, for those who have pointed out that my D/A convertor is
predicting the future by drawing the nice diagonal lines to the next
sample, you are right, I was not thinking clearly.

I also agree that you need a 22khz low pass filter, and that it will
make things better.  And I have gone through all of the math proving that
in an ideal system that this is adequate (although it was several years
ago when I was an undergrad).

The thing that I don't quite see is how in the real world type system
that I drew that a low pass filter would actually output the original
signal.  Generally a low pass filter will take the 'edge' off of corners
in the time domain, and there are places in my reconstructed signal
that never reach zero, but were zero in the original signal.  Those parts
will Fourrier transform into a low frequency, and the low pass won't
touch them.  I think if you would actually low-pass my sampled signal
(or actually the correct sampled signal that someone posted just a
couple messages ago) you would find that the amplitude and phase of the
output will waver.  Does anyone have a program that would let you actually
do this?

							joe

-- 
Joe Augenbraun                          ucp: {uunet|ihnp4|rutgers}!cbmvax!augi
System Engineering                     	arpa: cbmvax!augi@uunet.uu.net
Commodore Business Machines		Phone: 215-431-9332