Behringer DCX2496 digital X-over

Hi chaps,

Been thinking of getting one of these Behringer units for my PA/hifi system, in place of a 31-band graphic and active crossover (all Behringer).
Could even be tempted to convert my mid-high speakers for active crossover use on one of the inputs - I've loads of amps lying around.

I'd be running a 3.5mm jack to XLR lead in, and the outputs would go direct to the amplifiers. Volume would be controlled by the device on the input.
I know this isn't best practice with digital units, as I'd be losing resolution, but >95% of the time, the system's used for movies and background music. Its set up in the living room of a student house, so critical listening is basically a non-issue.

So, in its stock form, what do y'all think of it?
Is there another, similar unit that some people prefer, or is the Behringer unmatched at this price?

TIA
Chris
 
Hmmmm...

I suppose I could put something on the front end up nudge the signal up, but I'd rather avoid that - more rack space etc etc.

Given an amp that clips at 200w (into 8ohm) at 0dB input, an iPod running at -10dB and speakers of ~98dB@1w...

Setting the Behringer to unity would give 111dB per speaker with the iPod and amp turned all the way up.

This is probably a terrible way of thinking of it**, but that's 121dB system gain.
The noise on the Behringer is (according to the spec sheet) -90dB. Put through my system, that's 31dB of noise at 1 metre. A side.

That seems reasonably quiet. I suspect my old QSC amp's fan is noisier than that.

** while the maths seems very crude, I think this works out right: -10dB at the input yields 111dB per side, so 0dB at the input ought to give 121dB per side. You get the idea. The 121dB number seems constant here, so it seems kind-of sensible to apply it to the noise figure of the Behringer unit.

Chris
 
my solution was to feed spdif in (avoiding the whole a/d section and level matching problem) and also build a board that gives spdif-out for the channels I want (2 out of the 3, for high/low pass), and the spdif-out goes to 2 outboard dacs, then into a multi-channel vol control and onto amps.

its a sneaky way to avoid the analog sections entirely and mostly not care abou the psu quality in the dcx, as well.

output is too hot and input wants it too hot. analog with consumer gear is a mess with the dcx. its not even +4db level, its higher than that! for me, it was always going to be a 'numbers box', only, anyway.
 
Hmmm...

I have various boxes that I could use to get some gain in there (20dB wouldn't be much trouble), but I'd rather not use them - that's more rack space, which is at a bit of a premium: I'm using big old class AB amps everywhere, most of them 3U.

Feeding spdif in isn't really an option: this is the stereo that anyone should be able to use, which is why I made sure there was a 3.5mm jack input, and no knobs to operate.
You plug your phone/laptop in, switch the stereo on at the wall, and all the controls you need are in your hand.
I'd really like to keep this ease of use, as I suspect nobody would bother to use the stereo if it was complicated to get working.

So, does anyone have any experience with this kind of set-up?

I've played around with the software and have fallen in love with the functionality of the thing, but if there'll be 70dB of background noise, and everything sounds like a 56kbps MP3, I'll give it a miss.

TIA

Chris
 
You mean almost as bad as analog tape or vinyl? :)
Whatever the devil that final smiley face is suppose to mean (and I really don't know what smileys ever mean really), the only difference between how we think about analog noise and digital noise is the fact that digital noise has an additional metric (in bits) to scare us.

Unless you or an automaton is going to stand with one hand permanently on the gain control (and thus inherently compress the sound), we are always balancing dynamic range versus residual noise.

If you value classical music, you have to use gain management so as to allow a large dynamic range. So my Behringer displays just one or two LEDs most of the time. I don't have a full idea how those LEDs behave and I don't know if there are very brief peaks which aren't visible on the LEDs but which I do not want clipped... or maybe the opposite problem.

For a given technology, you can't have it both ways. For sure, no need to cry about missing the last "bit" of noise suppression when you already have it so good.

(BTW, I earlier pointed out that my 1v pre-amp puts out a clean 5 volts, if I recall correctly. Some of us just have to get used to cranking the volume control up more.)

Ben
 
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Whatever the devil that final smiley face is suppose to mean (and I really don't know what smileys ever mean really), the only difference between how we think about analog noise and digital noise is the fact that digital noise has an additional metric (in bits) to scare us.

Noise is noise - and it tends to be measured as SNR, in dB, be it digital or analog.

Unless you or an automaton is going to stand with one hand permanently on the gain control (and thus inherently compress the sound), we are always balancing dynamic range versus residual noise.
But if your source has less SNR than your system, there is no way any "balancing" or gain management can help you.
 
Noise is noise - and it tends to be measured as SNR, in dB, be it digital or analog.

But if your source has less SNR than your system, there is no way any "balancing" or gain management can help you.

Thanks for teaching me what a db is (insert smiley meaning I am being sarcastic).

The issues remain similar whatever the preceding SNR is. While you can't (in theory) make it better*, you can limit how much you make it worse.

Ben
*in fact, you can process sources in various ways to make them sound and perhaps even measure better, even if that doesn't work for Shannon's theory. My treasured vinyl recordings sound a lot better after processing and transfer to CD.
 
in fact, you can process sources in various ways to make them sound and perhaps even measure better, even if that doesn't work for Shannon's theory.

Sure. That kind of processing is what recording and mastering engineers do all day. But what sounds "better" to you might not sound better to me - and might me less faithful to the original recording. But then some people even like Autotune... And I don't see why that would contradict Shannon in any way.

My treasured vinyl recordings sound a lot better after processing and transfer to CD.
Are you saying CD sounds better than vinyl?

We are diverging quite a bit from the OP's question...
 
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Chris, you are going to have to boost the input level from your source, or modify the DCX. That's all there is to it. As stock, the inputs need a very high level.

A little line preamp wouldn't be huge (a dual opamp could do) or you can bypass the analog input circuit in the DCX with your own. I've done that and it works great. I will see if I can dig up a schematic. You just need single ended in, the balanced out to feed the ADC chip. Feeding it with a low impedance source is the key.

The DCX is unity gain. If you feed 1 volt into the ADC, you'll gt 1 volt out of the DAC chip (all filters off, of course). The stock analog interfacing circuitry is meant for higher, pro levels and adjust the signal come into and out of the ADC - DAC.
 
either boost via analog (which I don't recommend) or boost via spdif.

ie, get an outboard a/d converter that accepts 'normal' 2v inputs, unbalanced, and produces spdif. from there, you can import spdif directly and you have then avoided the problem of too low a level for input.

I'm sure you can find a low cost a/d box. midiman used to sell them years ago, I bet you can find used versions on the bay. here's one, for example:

M Audio Midiman Flying Calf 24 Bit A D Analog to Digital s PDIF Converter Cow | eBay

its not high end, but it WILL give you a reasonable digital-out from any kind of consumer line-level input.
 
everyone tries their best to 'live with' the analog sections and psu sections of the dcx, but there are ways to avoid it all.

spdif-out is a bit of hard work; but spdif-in is trivially easy and you avoid the entire a/d section and the high voltage requirement this way. you can get to 0db easily this way and the midiman box has an adjustable input level, iirc, so you can trim to be just under 0db on your peaks.

so, you avoid the high voltage input AND the unbal->bal conversion. the effective bit loss you'd get by under-driving the dcx's analog-in is probably worse than using a cheap consumer grade a/d box; and there's a chance the midiman box would produce better analog-in once all is said and done.
 
eBay "24/96 USB > RCA SPDIF + I2S SA9023 converter for DAC" may help, I think the SA9023 may be driver-free… probably works with a phone or …computers-wise I think
Linuxworks would confirm that at least some variants of Linux and certainly Mac OSX about 10.6 have native USB audio kernel streaming support [on a Mac I think you look under Applications/Utilities/Audio MIDI Setup] [not under preferences].
 
I can recommend this if you are using linux: M2Tech HiFace Two

it is usb/coax-spdif and works fine for asynch usb UAC2 audio. on windows, it needs drivers and is a pain to deal with for non-ascio or win7 equiv i/o. for watching movies, I use an older 48k usb/spdif dongle, but for high quality audio, the hface is a nice little box that works well.

that can directly drive this thing:

$(KGrHqQOKo0FH,d4zjjMBR41gl511!~~60_3.JPG


add some rg59 bnc-bnc coax and you're done with the input side of things.

IF you are ok with the pc being the source, that is. and for most people, it would be (or a set-top streamer box, same thing, for our purposes).

you could mount (as a data directory) your mp3 player to the nearest pc, then have the pc play the audio out via usb/spdif ;)