julius smith has a nice collection of basic & advanced music dsp stuff http://www-ccrma.stanford.edu/~jos/jospubs.html

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

http://www.nullpointer.co.uk http://www.dividebyzero.org ________________________________________________________ C:\AUTOEXEC.BAT
C:\REM [Header] C:> C:>PATH=C:\PERL\BIN;C:\WINDOWS\COMMAND;\C:jdk1.2.2\bin ________________________________________________________

(all suffixes enabled)

-----Original Message----- From: evan sugerman sugie@cats.ucsc.edu To: pdlist pd-list@iem.kug.ac.at Date: 18 June 2001 04:29 Subject: [PD] (DP)

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