[LAU] jack/oversampling

[tim]

>> I would mix the project  at 48k or 96k
> 
> Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best
> sound quality.

if that would only be true ...


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.

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

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

iir filters may have a higher quantization noise, but that is the
reason, why a good filter implementation is done in double-precision.


frankly, 48k may be a good enough for distribution, but it is
sub-optimal not for production ... and it is horrible for digital synthesis.

fwiw, for digital synthesis (non-standard or distortion synthesis) i
ended up rendering my compositions at 3mhz ... which was a good
compromise between computation time and sound quality.


best, tim

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.
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)?