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