Interpolation for samplers

Hi,

I’ve been struggling to find literature or example implementations of interpolation algorithms, for the purpose of reading through an audio buffer at low speeds for a sampler.
I am aware of hermite interpolation which is very speedy:
https://www.musicdsp.org/en/latest/Other/93-hermite-interpollation.html
However this still produces too many artefacts at low playback rates.

I understand sinc interpolation to be the “ideal” way, and I have an implementation but I am not convinced its correct. So I would need a reference to compare against.

any discussion or advice would be appreciated.
Cheers

one way to check the correctness of your interpolation is to run a sine wave through it, then observe the results on a spectrum analyser plugin. OH, note the the frequency of the sine must NOT be a ‘nice’ factor of the sampling rate like 441 Hz, otherwise the artifacts tend to cancel each other out.

e.g. A perfect interpolator should reproduce a sine wave as a single clean ‘spike’ on the analyser. Below is an interpolator with a bit of undesired ‘grunge’.

1 Like