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Auto EQ Algorithm
Auto EQ Algorithm
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Old 15th March 2005, 07:50 PM   #1
dswiston is offline dswiston  United States
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Join Date: Feb 2004
Location: Illinois
Default Auto EQ Algorithm

Hi guys. I'm working on a project for my lab class (Digital Signal Processing). Being an audio guy, I of course wanted to do something w/audio and stay away from image processing. So I decided auto-eq'in (like the DEQ2496) would be a fun, interesting idea.

Does anyone have some nice links to theory/algorithms for how to go about implementing auto-eq? I'm planning on bringing my DEQ2496 to provide pink noise to our DSP chip we are programing (TMS320C54). I have a wide variety of libraries avaliable to me (FFT,IFFT,autocorr. . .) so please give me some links!

Or perhaps impart your massive amount of digital knowledge on me.

I've been googling but nothing seems like a "gold mine" of good advice, I want some info I can trust.

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Old 16th March 2005, 07:16 AM   #2
knutinh is offline knutinh  Norway
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Join Date: May 2004
Location: lillestrÝm
You are entering a difficult field :-)

First, you need to read up on papers in this subject. JASC might be a good start.

Then a good measurement means getting the IR or frequency/phas eresponse. Pink noise measurements are not good for this. Use either WinMLS or another acoustic impulse response measurement program.

You should consider if you want a single sample, or measure the response over a volume (typical listening positions), then doing some kind of averaging.

The inverse response is a problem as well. A straight inversion of frequency response results in an overly bright (and excessive clipping) sound. My best bet would be modelling the woofer as a 2nd/4th order highpass system, and not correcting it.

with regards
Knut Inge
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Old 31st March 2005, 11:22 PM   #3
apollyon25 is offline apollyon25  New Zealand
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Join Date: Sep 2004
Location: Auckland, NZ
As Knut has correctly pointed out, doing auto/non-auto room EQ properly is rather a difficult field of endeavour.

The other approach to room eq is to use tones stepped in minute amounts (look up numerically controlled oscillators (NCO) and direct digital synthesis (DDS)) and examining how the room/speakers respond to different frequencies... this is simple from a signal processing point of view and overcomes to some extent the problems inherent with calculating impulse responses...

As an alternative, perhaps create a DSP system that makes a 3 way crossover entirely in the digital domain?
This of course would involve conversion of SPDIF to IIS format data (requires PLLs so you may want to do this in hardware initially) and then the extraction of the serial audio data into parallel data for left and right channels, then the three filters (6 in total for stereo). You can then extend this system to more frequency bands...

Alternatively maybe examine and vary the phase relationships of the L and R signals to show how surround sound 'images' and sound stages are created by varying the phase of the two signals... if you are really keen, maybe re-create 5.1??

Perhaps look into audio compression formats and algorithms (Wavelets/DCT etc)? WiFi?

From past experience - pick a topic you are interested in and start simple, get that going, then add bits to it... that way when the deadline is upon you, you can just hand in the previous version, or squeeze the next one in...

Good luck! It is an interesting, challenging and often frustrating topic to delve into. Just measuring stuff to make sure you have it right is a pain in the ...

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Old 1st April 2005, 09:03 PM   #4
sync40 is offline sync40  Mexico
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Join Date: Mar 2005
Location: Mexico
If you are in a hurry or with a deadline limit for delivering your work, it may be not enough time to sucessfully develop the algorithms (because measuring, checking, error-checking, measuring again, and finally write code...).

If you like audio and do not want to do image processing may be you could consider to build a multieffects unit (chorus, flanger, delay, reverb, eq, etc.).

Good luck.
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Old 22nd April 2005, 03:00 AM   #5
dswiston is offline dswiston  United States
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Join Date: Feb 2004
Location: Illinois
Ok so the project is well underway and SEEMS to be going well. I have a quick question though. Perhaps I should have asked this in the planning stage, but why do RTA equalizers, such as the DEQ2496, flatten the frequency response of the loudspeakers with respect to pink noise? I understand that pink noise has equal energy per octave, but why send pink noise through the speakers and then equalize the response of the speakers so that the mic input is receiving pink noise (or at least close to it)? Couldn't white noise, or any noise for that matter, be used to do the same thing? Is it because upon breaking the frequency spectrum into octaves it's easier to compare the power spectrum since with pink noise there's equal energy per octave?
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Old 22nd April 2005, 01:32 PM   #6
macboy is offline macboy  Canada
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Join Date: Oct 2003
Location: Ottawa, Canada
Is it because upon breaking the frequency spectrum into octaves it's easier to compare the power spectrum since with pink noise there's equal energy per octave?
That's exactly it. Don't second-guess yourself.
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