Path: utzoo!attcan!uunet!husc6!m2c!ulowell!cbmvax!augi
From: augi@cbmvax.UUCP (Joe Augenbraun)
Newsgroups: comp.sys.amiga
Subject: Re: Sampling at 29KHz (long)
Message-ID: <3854@cbmvax.UUCP>
Date: 23 May 88 23:29:12 GMT
References: <2845@polya.STANFORD.EDU> <734@eos.UUCP> <53788@sun.uucp> <5637@cup.portal.com>
Reply-To: augi@cbmvax.UUCP (Joe Augenbraun)
Organization: Commodore Technology, West Chester, PA
Lines: 65

In article <5637@cup.portal.com> doug-merritt@cup.portal.com writes:
> A number of questions have been raised about sampled sound by several
> people, and the following posting is long because it answers all of
> the ones that weren't answered previously, as well as correcting some
> errors in statements by previous posters. If you don't care about
> high quality sound generation, don't read it.
> 
> Since Tom got a mild flame for saying he was not an expert, I guess I
> should say that I *am* an expert, as far as the following goes.
> Trust me. 1/2 :-)
> 

I've had a couple of questions that I've been meaning to ask an expert. :-)
The top waveform (let's say) is an 18 KHz signal and the lines underneath
it are the 44 khz sampling signal used by a CD player:



       .                           .                           .
   .       .                   .       .                   .       .
 .           .               .           .               .           .
.             .             .             .             .             .
               .           .               .           .               .
                 .       .                   .       .                   .
                     .                           .                          .

       |          |          |          |          |          |          |

Resulting in the following samples:

       .                                                      . 
       |                                                      |     
       |                     .          |                     |       
       |                     |          |                     |        
       |                     |          |                     |         
       |                     |          |                     |          .
       |          |          |          |          |          |          |   

Which a normal DA converter would probably recunstruct into:

       .                                                      . 
       |.                            .                       .| .   
       |  .               .  .  .       |  .               .  |   .   
       |    .           .    |          |                .    |     .  
       |      .       .      |          |     .        .      |       . 
       |        .   .        |          |        .   .        |          .
       |          |          |          |          |          |          |   

I know that in theory a sampled signal should be recontructed using some kind
of funny exponential, but I assume that this is real hard to do in real time.
So what did I do wrong here?  Is the distortion on the waveform just the
aliased sampling frequency, and its not easily recognizable because I am
drawing in the time domain?  Or is this an artifact of not reconstructing
using the exponential?

(I just used CD parameters to give me reasonable numbers.  I am not interested
in CD issues here, just the theoretical reason of what causes the distortion
that I drew.)

							joe

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