Behringer DEQ2496 Ultracurve Pro Equalizer

[Colin]

thanks, that will help I am sure, what exactly does -60 mean on that display?

[Colin]

I've studies showing that many people prefer compressed music (proably on their mediocre systems)...

[timbley]

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On 3/18/2005 9:32:54 AM Colin wrote:

thanks, that will help I am sure, what exactly does -60 mean on that display?

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Hi Colin,

-60 means that the pink noise sound level is attenuated 60db from the maximum the unit can output. That's quite a lot of attenuation, since the power level doubles as you go up every 3db. I turn it all the way up to 0db and adjust the volume using my receiver. However, if you are using the analog outputs the level could be too high for your pre-amp, causing it to clip. In that case, you'd want to turn it down a bit. I'm not sure how you'd know if that was happening with pink noise, other than that the pink noise would start to sound bad. The only problem with that is pink noise already sounds bad.

[timbley]

That's interesting that people would prefer the sound of compressed music. The Behringer manual says that compression can add some "punch" and fullness to the music. Maybe it makes the quieter overtones and reverbs easier to hear.

[mikebse2a3]

Colin ask:

What does the red LEDs on the clipping meter really mean? is the amp clipping?

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Hi Colin

The red clipping indicater LEDs means the input signal is set to high for the Behringer. If this is happening to you check on the back of the Behringer next to the XLR connectors and you will find a push switch which will adjust the sensitivity of the Behringer and see if this helps in your setup.

If you still have trouble then tell us how you have the Behringer installed in your system.

mike

[flamminifra]

Hi there! I'm looking for some advices from the ones who experienced a lot with this unit. I'm using it in the digital path of my hi-fi system (no ADC/DAC). I've read through the forum, but several posts are in conflict with each other. What I understood is that it is often a matter of tastes, impressions and the results are dependant on the specific system and environment. But I still would like to find an optimal and stable equalization for my system, which, after several AUTO EQ's, I was not yet able to find out. In general, the overall obtained sound is more crisp and defined, even if too bright with some CDs. The optimal equalization, then, is to be obtained as a compromise of different factors (maybe smoothing somehow the eq results?).

Now, I would like to know what are the factors that significantly influence the AUTO EQ function of the DEQ2496 and how they should be set:

1) listening volume: is it influential and, if yes, how must it be set?

2) dithering and sampling frequency of the pink noise

3) lower frequency (<100Hz) equalization: is it always affected by measurement errors or can it be activated?

4) mic positioning: does the mic have to be positioned vertically at about the ear level of the listening position or at the midrange level near the loudspeakers?

Im asking this because I obtained slightly different equalizations for the same system in the same room in different measurement sessions, and I would like to undestand what parameters that I changed (except for some, I think negligible, picture or carpet repositioning) have influenced the output of the AUTO EQ.

Moreover, does the speed (slow, medium or fast) influence the results of the equalization? And what about the other RTA options?

Thank you.

[mikebse2a3]

flamminifra said:

Now, I would like to know what are the factors that significantly influence the AUTO EQ function of the DEQ2496 and how they should be set:

1) listening volume: is it influential and, if yes, how must it be set?

2) dithering and sampling frequency of the pink noise

3) lower frequency (<100Hz) equalization: is it always affected by measurement errors or can it be activated?

4) mic positioning: does the mic have to be positioned vertically at about the ear level of the listening position or at the midrange level near the loudspeakers?

Im asking this because I obtained slightly different equalizations for the same system in the same room in different measurement sessions, and I would like to undestand what parameters that I changed (except for some, I think negligible, picture or carpet repositioning) have influenced the output of the AUTO EQ.

Moreover, does the speed (slow, medium or fast) influence the results of the equalization? And what about the other RTA options?

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First welcome flamminifra

This is my experience/opinion with some of the questions you have asked. Hopefully others will offer their thoughts/knowledge on this also as they get time.

No 1: I believe its a good idea to set the Pink Noise Level a minimum of 20db to 30db above the ambient noise level of the room your testing in to prevent the ambient noise level from interfering with the measurement. My thinking is though that the closer to my normal listening levels that I set the Pink Noise the better because any changes in the speaker/room response at these levels could be taken into account by the Auto EQ Test.

No 2: I'm not sure what your asking here but will look into it more when time allows me to. Maybe you can elaborate more on this question?

No 3: The manual definitly warns of possible measurement errors if these frequencies are left operational and I really prefer to adjust these manually while looking at the RTA Pink Noise Response and by Listening to Music that I feel is recorded well in these ranges. It is also advisable to know your speakers electrical/mechanical limits before boosting in these low ranges. For Example if a speaker is not capable of response at 30Hz but you used the Auto EQ Function with this Frequency Band active it would try to use its maximum boost to bring the 30Hz range up which would actually not help since the driver couldn't handle it and the Amplifier would be wasting power trying to reproduce this boosted signal. While talking about this I also personally prefer to control the frequencies above 5000Hz Manually since again of possible abnormal amounts of boost by the Auto EQ Program could cause damage to a tweeter and also like the bass I feel this range is better adjusted by ear.

No 4: The answer to this question really depends on what your trying to measure and correct with the EQ. If your trying to do some equalization of the individual drivers of a speaker then close MIC(YES SHOULD BE VERTICALLY MOUNTED) Placement makes good since to help minimize interferance from close objects to the MIC. If your goal is to equalize the sound at your area/areas of listening then My Thinking is you must measure where your ears are going to be because this is the only way to know what corrections are needed for the area/areas we will be listening from. Also some speakers are designed to be listened at from a far enough distance for all the drivers responses to blend properly. In the case of the Khorn I have read and noticed test have been run at a distanse of 3 meters for this reason I believe.

As far as the speed (slow, medium or fast) if your talking in the Auto EQ Test I prefer the slow mode because this speed seems to give me the results I want best which is more of an average of the wildley swinging peaks and dips of some of the frequency bands due to room interferance.

These have been some quick thoughts because I'm kind of pushed for time this week but I did want to give you some feedback to your good questions.

mike

[mikesp1]

I have three dips of -10 to -15db in my frequentiespectrum at 30, 60 and 115hz

Will the behringer and my audiosystem be ablo to flat that out?

If yes this is my solution!

[PrestonTom]

I have three dips of -10 to -15db in my frequentiespectrum at 30, 60 and 115hz

Will the behringer and my audiosystem be ablo to flat that out?

If yes this is my solution!

An EQ probably is not the best choice in this instance. Others may disagree but please consider the following.

Without knowing your system & set up I am taking a few guesses.

The dips you are seeing are most likely due to the speaker and room interacting (modes). You can verifying this by noting whether the dips change in frequency as you measure at greater or lesser distances.

The dips are large (10-15dB) then EQ'ing is not great way to solve this (they should be used sparingly).

At these low frequencies, trying to provide that much gain may be asking for too much from your amp (especially if the speakers are not efficient and if they present a low impedance to the amp at this frequency range).

If it is a room problem (my guess) it is quite possible that you have some low frequency "ringing" as well. Again, EQing is not the preferred treatment and will have little impact on this "time domain" problem.

Again, I am guessing at what the probelms are and what equipment you might be using.

Good Luck,

-Tom

[mikesp1]

Equipment: Sunfire tgIV, Sunfire cinema grand power amp, DVD25 H/K, FL8380 H/K cd player,
RF5 mains, RC7 center, RS7 surrounds, 2x RSW12 subs, Beamax micro perforated screen, Benq pb6210 projector.

Yes i know it's a room problem. to much absorption at the low frequencies! But rebuilding my room is very costly, so that's why i was thinking at the behringer!

[mas]

Mike, with regards to your problem, it sounds like the web site is almost functioning properly! Now...if it would only do that to Coytee![][][][] ...Just funnin'...Couldn't resist! []

I have been interested in playing with one of these low cost alternatives ( the other Behringer - the Behringer DCX2496 Ultra-Drive Pro Digital Crossover System) for quite awhile, especially as it makes a 2 channel 4-way xover with delay economically feasible (you need 2 or them - but at $250 each, its cheaper than one more expensive but more limited 2 or 3 way/bandpass high end units or the Rane AC27). And I guess they are supposed to be available again soon, if not already available.

Low freq auto functions are always going to be squirrelly, just as LF measurements are due to the long sampling periods required. And you are not going to be able to isolate the direct from the reflected signals...

That being said, the delay and active crossover functions alone are worth the money if it performs accurately.

The only issue I have heard from several in the SR environment is that it occasionally loses its mind (stored program) which, while it could be a real 'inconvenience in a large show, is not much more than an occasional hassle in home use.

Edit: And just how are you achieving too much LF absorption? Even with Helmholtz resonators, the best we can hope for in tuned rooms and studios is to knock 3dB off of the mode peaks! And that is considered a success! And I would suggest not using EQ as a source of gain - but only as a notch device. Instead I would simply amplify what Tom said...sounds like a mode issue and your listening position is in a null.

Edit 2: Those frequency multiples point to comb filtering &/or room modes.