Hi Frank,
the solution Werner solved is working the same way. The multiplier and delta
gets computed once when receiving all necessary data. For each sample you then
only have to run one multiplication and one addition plus reassignment. The
math behind it looks a bit complicated but for very
> current = previous * multiplier + delta.
Hi Frank,
we developed the same idea six years ago for www.audiotool.com
Werner Van Belle worked the math out:
http://werner.yellowcouch.org/Papers/fastenv12/index.html
Cheers
~
André Miche
ties is so small that it rarely
> matters if you are running at 88.2 / 96 khz.
I am missing to many aspects of your suggestion. Any hints where to learn about
this would be appreciated.
~
André Michelle
https://www.audiotool.com <https://www.audiotool.com/>
oblems -
so easy they look at the first glance - ends up really complicated if you want
to do the first thing that comes into your mind. Hope I can present the
solution for you to fiddle around with online soon.
Cheers!
~
André Michelle
https://www.audiotool.com <https://www.audiotool.com
> I could phrase my question more general:
> Given any f(x): Is it possible to sample f(x) with a given sample-rate
> ignoring all frequencies (slopes) higher than SF/2?
>
>
> Couldn't you just implement a brickwall FIR filter in your signal chain,
> after whatever signal you're deciding to
so wave-shape morphing. In fact it is a solution
(thanks for that) but I was hoping for something more elegant.
I could phrase my question more general:
Given any f(x): Is it possible to sample f(x) with a given sample-rate ignoring
all frequencies (slopes) high
over time) waveform
function f(x) excluding frequencies over Nyquist?
Any suggestions are highly appreciated!
~
André Michelle
https://www.audiotool.com
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