After the useful help of forum comunity I'm using the simple 3bands DJstyle EQ (with biquad~), but now I need more complex EQs.

I need a 8 bands parametric EQ, where I can set Frequency, Q and gain range of +12 / -12 dbs for every band.

What do you think is the best option in therms of CPU load and sound quality to do this?

Do you know some example patch?

Thanks again for your help.

Here's where to get The Puckette's Book Pineapple.

~:o)

http://crca.ucsd.edu/~msp/techniques.htm
http://crca.ucsd.edu/~msp/techniques/latest/book-html/

Keep well all ~ SSSSSSSSSSSS

i'm studying. i went from knowing nothing an hour ago to knowing a tiny little bit after reading a bunch of websites.

this one was good:

http://www.dspguru.com/info/terms/filtterm/index2.htm

this one has good diagrams about how phase works:

http://www.soundfirst.com/EQ_Phase.html

this was informative:

http://mixguides.com/studiodesign/basics/equalizer-inequality-eq-0699/

this could be useful if you just want to add filters together like the 3-band examples we already have:

http://www.sengpielaudio.com/calculator-bandwidth.htm

from the first site i linked there, is a diagram like this:

which shows how the original signal is not distorted by using linear phase filters (andy said that only FIR filters have linear phase).

but it has a 40ms 'group delay'.

the equation for calculating group delay is:

filter phase shift

-2pi

and exactly how to calculate filter phase shift i don't know, but that's what i'm gonna try to figure out now.

wwwooowwww! Excellent info.

Thanks again.

i just posted on the mailing list about this. the answer is that i should do fft on the FIR filter to get the phase. but other software eq's don't do this. there is definitely a way to calculate it.

just no-one using pd knows how. otherwise we'd have a good parametric eq.

getting closer though,

Phase shift is just the maximum delay (in samples) multipled by the sampling rate.

Any FIR is a polynomial (maybe a big one with loads of terms) where every term is a factor of the unit sample delay.

So you might have filter like

y(n) = 0.5x(n) + 0.125x(n-2) - 0.301x(x-3).... 0.1x(n-25)

In which case the maximum delay is the biggest term, 25 samples = 0.0005668934 seconds

The idea is to make ALL your filters have the same total delay (by zero padding terms even if they aren't needed), so each filter gives the same shift.

At least that's how I understand it, but I've never made a graphic EQ from FIR filters this way before.

i think i just understood basically how a filter works. cheers andy.

Hi,

I have been working on this problem, with zero phase distorsion, in offline processing for the moment, to perform audio measurements and audio scenes cartography (notably) using speech, music or any complex (either monophonic or stereophonic) stimuli rather than linear or exponential chirps or "pure" sinusoids.

I have been trying to realise a pd external to perform a fast but FFT-free temporal convolution, to be used (in future) to perform a 10-band analysis and partial to complete re-synthesis in real time, if pd and the computer can manage the CPU power necessary to manage 10-band sharp FIR filtering.

With two assitants (doing a huge and great job), we have been working to port my scilab prototype of the perceptive analyser in C++ to be able to offer cross platform sources and binaires by the end of the year. Sorry, there is a lot of stuff to do!

If you need some more information, you can try to access to the following Audio Engineering Society (AES) convention papers (I am not sure I can send the papers or put them on a web site for free downloading... as the AES sell the convention acts or papers):

• Millot L., Pelé G., and M. Elliq, Using perceptive subbands analysis to perform audio scenes cartography, 118th Convention of the Audio Engineering Society, Barcelona, Spain, 2005 may 28-31, Paper 6340.

• Millot L., Some clues to build a sound analysis relevant to hearing, 116th Convention of the Audio Engineering Society, Berlin, Germany, 204 may 8-11 mai, Paper 6041.

Another paper would be available by the 05/21, after the AES Convention in Amsterdam, dealing with the FFT-free but fast temporal convolution algorithm. And, maybe I would also have finished some externals to make real time demonstrations by monday 05/19:
Millot L. and Pelé G., An alternative approach for the convolution in time-domain: the taches-algorithm, 124th Convention of the Audio Engineering Society, Amsterdam, Nederlands, 2008 may 17-20.

Other papers dealing with the use of the perceptive analyser have also been proposed not only within the AES conventions. Let me know if you are interested.

Sincelrly yours,
Laurent Millot

Dear Laurent Millot
I think your (huge) work will be a great tool for people like me.
I have a few, very poor, theoric knowledge about this issue, but I like use software like Puredata to build my own simple audio applications.

But, obviously, I can't work in a "serious" EQ app.

Thanks a lot in advance. Waiting for your news.

Best regards. -> Pineapple

hey pineapple, i just posted a reasonable 7 band eq on another part of this forum.
yoU'll find it if you look.