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? >
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RE: [motm] motm100 tech ?
2000-05-17 by Brousseau, Paul E (Paul)
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