HI All,
I know that this is off topic in a way... but I
am tinkering with writing dsp code
(some externals for pd ..e.t.c.) and I can't
really understand how to implement a filter
(such as lowpass e.t.c.) How do you assess the
frequencies of a particular signal and therefore attenuate them?
I assume some method of holding and comparing
samples in a buffer or something, but then this would delay output?
Anyway, I am no math expert but I would
appreciate any help.. I took a look at pds filter code and it seemed to be
easier (in amount of code i.e. quite short) than I was thinking and I realise
that i probably need to know more about filter math/operation in general..so any
pointers e.t.c.
Thanks
Tom
(all suffixes enabled)
If a conversation is to continue; 2 things for actual
communication:
/Off of this list to allow for
clarity.
/Extinguish cryptic language forms. It doesnt
function to communicate an
idea which is based in minor
reality.
-granted,..language is arbitrary, but I would like to read
what you say
without translating symbols which are in essence
english
anyway.
soon.evan
p.s I dont hate!
you