I think that would produce a different effect. What I'm interested in is
"sampling" the input at something like 2kHz, or 8kHz, so it gets that crappy
ancient-soundcard or sampler feeling. I suppose the "right" way of doing
this is to send the signal into a A/D->D/A system, clocked at whatever rate
I'm interested in.
However, if we're clocking at audio rates, if there still a popping issue?
I mean, isn't this what those old D/A systems are doing? Every time slice,
they're holding the analog signal at some DC value, until it gets changed in
the next time slice?
Thanks!
--PBr
> -----Original Message-----
> From:Doug Pearson [SMTP:ceres@...]
> Sent:Wednesday, May 17, 2000 11:09 AM
> To:motm@egroups.com
> Subject:RE: [motm] motm100 tech ?
>
> You run the output of your unspecified audio source into the audio input
> of
> a VCF, and the S&H output into the 1V/Oct or FM input of the same VCF.
>
> -Doug
> ceres@...
>
> At 11:02 AM 05/17/2000 -0700, "Brousseau, Paul E (Paul)"
> <noise@...>
> wrote:
> >Really? I don't own one (yet), but I was expecting that I could route
> audio
> >into it and listen to the results (at least, if it was sampling at an
> audio
> >rate). Bummer. If I wanted to get a low-rate sampler effect on an
> >unspecified audio source (I.E., something I don't have control over), how
> >would I achieve that?
>