hey there,

does anyone know of sites where i can find some (pseudo) code about how one can filter specified frequency from a wav file (from its samples) ?? i am looking for how such things basically work

i cant find anything like that ...

cheers,
ello

edit: seems i found something on wikipedia, anyhow mor input on this is welcome

hmpf, i dont really understand what i found on wikipedia..
how can i adapt that pseudo code (http://en.wikipedia.org/wiki/Low-pass_filter) so i can check for a specific frequency of a sample?

that pseudocode looks not too complex, but what for example does that := means??
and is this thing representing the whole sound: real[0..n] x ??

thanks in advance,
ello

if i was you i'd have a look into fourier transformation, especially fft:
http://en.wikipedia.org/wiki/Fast_Fourier_transform

like that you can transform your data from time space to frequency space. there you can process the data and after that you transform it back into time space. that's how most digital filters work.

edit: also have a look at
http://en.wikipedia.org/wiki/FFTW

edit2: and here
http://www.captain.at/howto-fftw-spectrograph.php

thank you very much.. this gives much more examples on the net. i'll check if i can get something to work

one question which i cant answer, whats the diference between dft and fft ??

discrete fourier transformation just means that the signal is not given as a continuous function but at given intervals, "discrete". continuous fourier transformation (or transformation of infinite series) is not relevant for computer applications but rather for scientific purpose. and fft is a method how to calculate such a discrete transformation.

thank you for clarifying this...

you're welcome.