[Pd] Complex audio signals

[Piotr Majdak]

Chuckk Hubbard wrote:

> On 6/21/06, Piotr Majdak <piotr at majdak.com> wrote:
> 

> For some reason I hadn't checked this before, and my system is indeed
> adequate.  I wasn't sure if Pure Data's FFT has built-in windowing or
> not, apparently it doesn't.

Yes, it has! It windows the signal with a rectangular window by taking only the 
blocksize amount of samples from an infinite long signal. This corresponds to 
convolution of the spectrum(!) with sinc-function (sin(x)/x) for every frequency 
bin. That's the leakage effect...

> Cool, thanks, you might hear from me soon.  I wonder what's the
> simplest way to do convolution natively in Pd?

Georg described already the "fast convolution" method (Grüsse an Georg). Even 
simplier convolution is using [FIR~] from the iem lib. Just load an impulse 
response to a table and convolve it with a signal. [FIR~] implements _the_ 
direct discrete convolution.

BTW: I think that it's hard to debug and to understand these problems using a 
high-optimized real time software like pd. I propose you to use 
Matlab/Octave/SciLab or another numeric software package with many plot 
possibilities and signal control/editing sample by sample in an non-real time 
environment. Then, use the new knowledge to implement your ideas in pd :-)

br, Piotr



-- 
Piotr Majdak
Institut für Schallforschung
Österreichische Akademie der Wissenschaften
Reichsratsstr. 17
A-1010 Wien
Tel.: +43-1-4277-29511
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E-Mail: piotr at majdak.com
WWW: http://www.kfs.oeaw.ac.at