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 _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
