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From: mjm@abic.UUCP (Mark Medovich)
Newsgroups: net.music.synth
Subject: Re: ircam?
Message-ID: <701@abic.UUCP>
Date: Fri, 13-Dec-85 14:04:04 EST
Article-I.D.: abic.701
Posted: Fri Dec 13 14:04:04 1985
Date-Received: Wed, 18-Dec-85 06:44:05 EST
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>
> Ditto.  Wish we all had TMS32020 based DSP boards in our P.C.'s so we
> could trade sig-proc programs and try various algorithms etc.
> Maybe we could start of by discussing some of the various digital filter
> algorithms, favorites etc.  Sampling techniques (hardware) etc.
>
> Keith Doyle
> #  {ucbvax,ihnp4,decvax}!trwrb!cadovax!keithd
> #  cadovax!keithd@ucla-locus.arpa

Good idea Keith, I'll start!

  Adaptive Delta Modulation seems to be a nice idea, but I've found that
no manufacturer is using this technique with the exception of Delta Lab
Inc. My impression is that the sample frequency is a function of the slope
of the input, (and hence, d(-)/dt?).  Can you tell me more?
Also, if the sample frequency is not constant, are there algorithms(theorms)
to handle frequency variance. The only thing I've seen (formally) is an
article in last quarters IEEE Acoustics, Speech, and Signal processing on
sliding window covariant transversal filtering. Though interesting, the
article didn't go into the covariant algorithm in detail. Further, it only
considered the case of linear variance(in time). I think algorithms of
this sort will reduce processing time and conserve memory.