Behringer DCX2496 digital X-over

"What about daisy chaining two DCX's for greater slope?"

Then you go through more D/As, A/Ds with the potential for noise problems and likely more latency in the processing.

Even two 48dB/octave crossovers on top of each other isn't nearly as sharp as the above setup. The initial slope of the low pass for example is around 210dB/octave.

You can actually go higher slope a little more traditionally with the DCXs anyway. The parametric filters can be set to second order high/low pass filters instead of being notch filters. You can add them to the existing filters to get slopes above 48db/octave but still not nearly as sharp as the elliptics response.

Shawn
 
Played around with that briefly too. When I did that though it gives a smaller arc like ripple higher up (around -15dB if I recall it properly) then more attenuation then another arc and then the resulting roll off. It also made the initial attenuation not as steep.

I could put another parametric at around 810hz to knock down the arc further but I wonder if there is a point to that seeing how it is down 30dB already. Something to play with when I get time to see if I notice an audible difference I suppose.

I wanted to do this with as few parametrics as possible to leave DSP resources for a mid/tweeter crossover. Unless I dedicated a DCX per speaker I think it is going to run out of DSP trying to do this for (2) three way speakers in one DCX.

Shawn
 
Ok so here is my conundrum.... the Delta-66 Outputs digital as S/PDIF from an RCA at 75 ohms ...the digital input of the Behring DCX2496 is an XLR AES/??? (cant remember acronym) input which i believe to be
1. a different language and
2. a different impedance. 100 ohm ?

How do i get my digital signals to the unit? When i buy the DEQ i want Digital to digital to analog but i dont want to go an buy a converter just for that 🙁

Sorry to double post but i just realized Digital gets 1/20 th the traffic

Please help me Obi-Wan ...your my last hope.
 
One way to interface SPDIF to AES.
{DVD player digital output RCA to DCX2496}

One of these;

Part number: B-P-3
TecNec Premium BNC Male to RCA Male Flexible Video Cable 3Ft

and

Part number: BCJ-XP-TR
BNC Female In to XLR Male Out Impedence Transformer Plug

Order from;
http://www.markertek.com

Set your DCX to accept AES input instead of analog input.

///

DVD player digital out [coax] -> RCA to BNC cable -> BNC/XLR transformer -> DCX input.
 
akunec said:
Somewhere on the bottom of the DCX manual, i thought it said that it can accept SPDIF signals as well.

Yes, it can. Just. Be prepared for it to crash periodically. Fed from the S/PDIF output of a CD player, mine sometimes decides to lose most of the treble and needs to be reset to restore proper sound. I really must get around to fitting an AES3 output to the CD player...
 
EC8010 said:


Yes, it can. Just. Be prepared for it to crash periodically. Fed from the S/PDIF output of a CD player, mine sometimes decides to lose most of the treble and needs to be reset to restore proper sound. I really must get around to fitting an AES3 output to the CD player...


I think that might have something to do with 75 ohm S/pdif and expecting 110 ohm XLR connection.

Also i am running straight output from my soundcard/entertainment comp . I have no CD player perse.
 
"I think that might have something to do with 75 ohm S/pdif and expecting 110 ohm XLR connection."

I tend to doubt that. The difference in impedance isn't going to change just the samples for the high end... it doesn't work like that.

The non-audio data pre-amble bits in a frame of S/PDIF have different meanings between consumer and pro formats. I think one of the bits that is different is the bit for pre-emphasis. Possibly the Behringer is getting confused and applying/not applying pre-emphasis which would screw up the high end of the signal.

Shawn
 
It could be a problem of mistermination because the pulses will not be totally absorbed in the 110R load, causing a reflection to bounce back to the 75R source where it will be totally absorbed. However, the cable is quite short (<1m), so I doubt that a slight mistermination will have very much effect on pulse shape. I think the problem is far more likely to be that S/PDIF is typically only 0.5Vpk-pk unbalanced whereas AES3 is typically 5Vpk-pk balanced, so AES3 enjoys greater immunity from induced noise spikes.

Agreed (consumer) S/PDIF has a bit for invoking 50/15 equalisation, but it's highly unlikely that the (professional) Behringer is designed to act upon it. No, the occassional loss of HF is a symptom of it getting its knickers in a twist, no more, no less.
 
"Agreed (consumer) S/PDIF has a bit for invoking 50/15 equalisation, but it's highly unlikely that the (professional) Behringer is designed to act upon it. "

Pro format has pre-emphasis available on it to and has more available settings for pre-emphasis then consumer S/PDIF.

For Pro format S/PDIF transmissions it is configured with pre-amble bits 2,3 and 4 of byte 0.

Bit 2 is emphasis not indicated so it defaults to no emphasis but allows over-riding that setting based on bits 3 and 4.

In the consumer format pre-emphasis is also in Byte 0... however it is encoded in bits 3,4 and 5. Bit 2 (the start of the pre-empasis bits in Pro) is Copy Protect status.

It very well may be the Behringer is just flipping out but I have heard other times converting from Consumer to Pro caused a loss of treble in totally different equipment.

Shawn
 
Thanks Thy but i am going to buy a used Ultramatch they have at the pro shop for 63 bucks. that will kill two birds with one stone as it has a gain control on it so viola' i now have a pre amp 😀

Only bad part is that it is NOT 192 capable 😡
 
Madmike2 said:
Thanks Thy but i am going to buy a used Ultramatch they have at the pro shop for 63 bucks. that will kill two birds with one stone as it has a gain control on it so viola' i now have a pre amp 😀

Um, what you actually have is an expensive means of converting S/PDIF to AES3. The volume control is a means of increasing distortion.

The ideal solution is to convert your source to produce AES3 and to use stepped attenuators at the outputs of the crossover to ensure that the DACs are operated at their optimum level.
 
hmmmm ... i dont know about expensive. The adapters he is speaking of come out to pretty much the same thing. And how would i introduce distortion to digital feed ?

Right now Winamp is my volume control 🙁 ANYTHING will be better then that.
 
Quite apart from whether the gain calculations and subsequent dithering are done correctly, if you send a signal to the DACs that has been adjusted to be a produce a quieter analogue signal, then you must have increased the distortion compared to operating the DACs at full resolution. Given that DACs aren't perfect, the optimum place for a volume control is after the DACs.