[LAU] jack/oversampling

tim tim at klingt.org
Sun Mar 16 19:16:34 UTC 2014


>> 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.
> 
> You should not have any non-linearities, except those introduced
> on purpose, i.e. distortion plugins and the like. And then it
> all depends on how these are designed. If done well, they will
> not add any aliased components. One way to avoid that is using
> higher sample rates internally, but it's not the only one.

i'm curious, what are the other ways?


>> frankly, 48k may be a good enough for distribution, but it is
>> sub-optimal not for production ... and it is horrible for digital
>> synthesis.
> 
> Only if you use 'primitive' algorithms. Unfortunately there's
> a lot of those around.

well, we are living in a world of df2 biquad filters, which tend to blow
up when modulating parameters, most delay lines are 1/2/4-point
interpolations and non-linearities are applied without any oversampling ...


> In summary, 96 or 192 kHz will allow you to use simpler algorithms.

or get better sound quality from existing plugins ;)

tim



More information about the Linux-audio-user mailing list