Behringer DCX2496 digital X-over

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You're welcome! There is about +/- 10dB of digital level adjustment, but you want to use as little of that as possible. I often use it for fine tuning on the bass section, but not much more. On mids and highs I tend to boost about 6dB just to get the level up. Otherwise after the filters the output is rather low.
 
There is a limit specified in the dcx2496 for how much total EQ can be applied.
If all 6 channels have different EQ, then I suspect that "limit" on processing power will come in quite early.

Some who have experimented with multiple EQs should be able to give guidance on how the processing limit affects DCX2496 performance.
 
Ben,
9 EQs for each input (3) and for each output (6).

Just download the "Editor Software" (Remote Software v1.16a) from the Behringer Site - Here under the "Downloads" tab. You can then see and control all the features. There in no need for a DCX unit to be attached to use the software. You will find it to be very capable.

I use the stock unit full range and my personal take is:
Its noise level is good at LF and MF, but rising at HF.

[It's at pro level and they also seem to have set nominal level low to allow for all the worst case EQ boost and level adjustments. As a result it is often noisy at HF at least when using non-pro amps (particularly bad with compression drivers due their high sensitivity).]

With careful gain management noise can be avoided - at least using my typical domes (92 dB sensitivity). LF and MF should be fine as is, but some care in gain management is still probably a good idea.

There is a limit to processing capacity that can be reached as AndrewT indicates, but it is pretty generous. Using the Remote Software you can test the limits as the remaining processing percentage is reported as the settings are adjusted.
 
jtalden - thanks for very helpful reply.

Gain management always was an issue in old days when S/N of components was worse. Good reason to avoid efficient speakers. I bet a lot of complaints about this Behringer arise from working with weak inputs.

But now-a-days, when we love high power amps and worry about headroom, hard to conceptualize how to set levels, just in case all the sopranos in a choir sing together and there are brief 500 watt peaks.

In preparing for a Behringer with its pro-line-level S/N spec, I measured my Kenwood C2 pre-amp, rated for 1 v output, will output 5 vrms (and more) with a "0"-db CD test signal and the volume control was about 75% (sorry, I didn't scope it to confirm wave form quality).

Ben
 
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Thanks for replies to my earlier post (#3532). But I'm still wondering how many EQs I can put into each of the 6 bandpassed outputs?

It all depends on the number and types of EQs. Put in a lot of them, and/or high Q/steep slopes and you will see the available processing power drop fast. But as long as the percentage is above zero you can continue to put EQs in.

At any rate, all these questions also on config are well explained in the manual, it's worth to download it and read it, much easier than picking up (unrelated) bits and pieces from here (no offense to those who provide the answers - it just makes more sense to read it in context).

Jan
 
Your pre would seem capable enough then. The other concern is the sensitivity of your amps. If they have input trim then you can reduce the sensitivity as needed. If not, as with most audiophile amps, you may need cut the sensitivity with some 10-20 dB attenuators. Again, probably only needed when HF is involved.
 
Your pre would seem capable enough then. The other concern is the sensitivity of your amps. If they have input trim then you can reduce the sensitivity as needed. If not, as with most audiophile amps, you may need cut the sensitivity with some 10-20 dB attenuators. Again, probably only needed when HF is involved.

If lacking a volume control, first thing I add!

BTW, this is one application where transformers (on the XLR connectors) are feasible for quality and cost. (Good thing I have a dozen RCA-XLR cords used in my present Behringer analog CX3400 crossover.)

... apropos the CX3400: cheap pots on a Behringer analog crossover are a big headache (and effort to adjust accurately). But I assume, when you set a digital unit for 150-2000 passband, it counts accurately, even if the printed circuit board is cheaply made.

Ben
 
Hello!

This is a real dream box for me. I have it as hi-fi crossover with my two-way front speakers. When you know how active crossover sounds you can quite easily hear the unfixed sound in a passive system.

An amazing thing is that you can pull interconnected tweeter + tweeter for example, to the "desired cutting point" from your computer. And not only that, you can also drag all outputs connected so that global cutting points are moved to the ultimate place. This is hugely grateful when calibrating last touch with the help of hearing. (image_1.jpg)

I have a question however about the existing headroom for the treble outputs. As you can see, when I maximize 24bit into Ultra Drive so it is a lot of headroom left. Should I touch this setting to get more bitrate or how do you do? (image_2.jpg)

Thanks
 

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Yes Ben, it's all in the manual. And the software will help you understand it. Be careful of your gain structure.

Bought one. Am I ever a happy DIY audiophile!!!

I found an obscure post somewhere that provided some yardsticks for available computational power to do "X" EQ, etc.... but nowhere in the manual.

For me (and I recommend to others) you can use DSP to cut below say 22 Hz at 48dB/8ve which is barely feasible any other way. However, that eats a lot of computational power (an amount you can only guess at).

The issue of gain management is no different with the kind of peak limits in DSP than many other components elsewhere in HiFi. Unless you listen to pop music (where levels barely vary), you have to play at levels that allow soprano choir peaks to pass through. Most of the time whether playing DSP or analog, noise is say, 30dB below what you are hearing (but masked) in order that peaks 60dB louder can pass through.

Now where did I put my harmonic distortion tester... got to give this puppy a test. And eyeball some square waves.

Ben
 
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Viper; it is a question I struggle with, too. If I understand what you mean.
Once you divide the spectrum up into 2 or 3 parts, each of those parts will be lower than the level of the whole signal, at least on music and pink noise. What do we do about that?

I've been boosting the outputs by 6dB to get a little more level out of the DAC chip. For music this has been OK with no clipping, but for test signals like sine sweeps, you have to be careful.
 
Pano >> Thanks!

Viper_user - could you talk about your digital input modification? I am going to do this soon to stop the digital input bug that plagues this unit. Is yours stable? Have you changed anything aside from running coax into the transformer? Thanks!

That is a good question. I have had some problems with drop outs. Lately though, I lost them by direct hook my player in windows to the soundcards spdif(24bit). Don't know if it is the new soundcard driver or the "direct connection"? (Image of Media Player Classic Home Cinema)

Maybe you could temporary swap to an adjustable resistor at the spdif-bridge and see if you can find a good value. I swapped mine with a 75 <--> 110.. Not fully sure on that one.
 

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