Adjusting reverb algorithms against changes in sample rate
Posted: Fri Jun 23, 2017 7:25 am
I have a customer (using the FV-1) who has a reverb that sounds great at (sample rate X) but really their board is designed to use (sample rate Y). They don't like the sound at (sample rate Y).
Now I figure there are a few things that need to be adjusted.
#1 the length of any all pass filters and delay lines should be proportionally adjusted so that the delay TIME stays the same.
#2 the coefficients of any damping filters need to be adjusted to keep the frequency the same. This requires a little calculation with exp/ln (natural log) but that is not too difficult.
#3 The all-pass coefficient(s). Here's where I get a little lost. My experiments with all passes show that the center frequency (where the phase shift is the greatest) depends on the length (probably TIME) of the delay line, while the sharpness of the phase shift depends on the coefficient. But I'm unaware of any formula which would allow you to make an all-pass equivalent at different sample rates (other than adjusting the length).
Anyone have a clue? Of course I can "adjust it by ear" but as I'm doing this for someone else I'd like to have a solid theoretical basis for getting everything as close as possible.
btw I have used the spectrogram view available in Audacity to visualize all pass phase shifts. It is very handy and I suggest you try it if you are at all interested in these things.
Now I figure there are a few things that need to be adjusted.
#1 the length of any all pass filters and delay lines should be proportionally adjusted so that the delay TIME stays the same.
#2 the coefficients of any damping filters need to be adjusted to keep the frequency the same. This requires a little calculation with exp/ln (natural log) but that is not too difficult.
#3 The all-pass coefficient(s). Here's where I get a little lost. My experiments with all passes show that the center frequency (where the phase shift is the greatest) depends on the length (probably TIME) of the delay line, while the sharpness of the phase shift depends on the coefficient. But I'm unaware of any formula which would allow you to make an all-pass equivalent at different sample rates (other than adjusting the length).
Anyone have a clue? Of course I can "adjust it by ear" but as I'm doing this for someone else I'd like to have a solid theoretical basis for getting everything as close as possible.
btw I have used the spectrogram view available in Audacity to visualize all pass phase shifts. It is very handy and I suggest you try it if you are at all interested in these things.