Re: [LAU] jack/oversampling

On 03/16/2014 05:45 PM, tim wrote:

> a) any non-linearity introduces harmonics, some non-linearities
> introduce an infinite amount of harmonics, which will cause foldover
> distortion. the large the sampling-rate, the lower the foldover.

ok, so you are trying to do weird synthesis that can produce
non-bandlimited output? i can see how you might want to use high
sampling rates there, but then again there will always be another
processing step that causes yet higher harmonics - addressing that with
high sample rates seems like a somewhat blunt approach that is bound to
fail eventually.

> b) delay-lines have a higher precision at higher sampling-rates

that statement is definitely not correct. granted, if you only do delays
with sample granularity (which has the big advantage of not requiring
any computation), there is some benefit in using higher rates. but you
can produce sub-sample delays with arbitrary precision easily. for IIR
feedback, i sure see the point, but then the question becomes "why do
you need to expose this to the outside world?" - just upsample in your
processing application and leave the rest of the jack graph running at a
sane rate.

> c) the tuning of digital filters is more precise at higher
> sampling-rates due to the frequency warping in the blt

i don't understand this. can you elaborate? what is "blt"?

> note on a:
> if your signal processor introduces the Nth harmonic, you have to
> upsample your signal by a factor of N. or apply a pre-filter on your
> signal by nyquist/N.

true. it's a funny and somewhat strange thought exercise for me to try
and achieve the highest possible "fidelity" with brutal distortion
algorithms - obviously, since i don't work with distortion, i try to
keep my signal chains as linear as possible.
but i can see how somebody well trained in distortion synthesis would
want to eliminate aliasing artefacts, since those would conceivably
interfere with systematic exploration of sounds based on prior
experience, and make the sonic outcome even more erratic than it already
is...

but in any case, there is no point in taking the internal higher
sampling rates out into the real world, so the zita resampling approach
might be your best bet.

> question for the reader: in order to completely prevent foldover
> distortion, how much do you have to upsample for a tanh waveshaper (a
> processor that introduces infinite harmonics)?

incidentally, just returned from musikmesse, and i've had my share of
DXD/DSD loonies... if you want to go there, there is people who want to
sell you 256-times oversampled single-bit delta sigma gear, and they
will happily talk megahertz with you.
it would be a ton of fun to discuss with them the best way to handle a
tanh waveshaper, and what new ultimate fidelity frontiers are required
for the distortion synthesis crowd. just make sure you avoid the term
distortion, call it "spectral enhancement processing" instead. >;->

best,

jörn

-- 
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487
Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT
http://stackingdwarves.net
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