Behringer DCX2496

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stef1777 you and every other tweaker hearted owner of this unit is out there with the same quest. Until now it seems that there isn't even a hope to find such a thing.

Noone even seems to know anyone who has one :(

Ergo

PS. So backengineering is the only way to go and it's under work.
 
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It's very silly, especially as the clever stuff is in the code in the PROM, not in a little twiddly bit of analogue interfacing.

Musical Fidelity took the same silly approach when I wanted a circuit diagram of their Digilog DAC years ago. I traced the circuit out and found out why they didn't want people to see it. It was a hotch-potch of semiconductor manufacturer's application notes stuck together with a few odds and ends, plus a pair of shunt regulators.
 
Software for the DCX2496

Have anyone tryed the software from sound easy?
SoundEasy V9.00 has been released. Now shipping with Modular Crossover and programming.

http://www.interdomain.net.au/~bodzio/

32. Remote flashing of MC parameters into Behringer DCX2496 (software version 1.14). Uses RS232 port.
Warning: Bodzio Software makes every effort to maintain compatibility between SoundEasy and DCX2496 crossover. However, due to inevitable software upgrade variations in DCX2496, we can not guarantee the compatibility with all versions of DCX software.
 
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I have SoundEasy version 9.0 and it works fine with my DCX2496.
The crossover CAD module in SE has what's called a "modular component" which has configurable parameters that duplicate the DCX2496 capabilities. I haven't yet done an actual design using this process but I see no reason why it shouldn't work perfectly.
In the context of all the other terrific capability SE has I think the DCX programming is a rather minor feature, but it is very cool. :)

Cheers,

Davey.
 
EC8010 said:
The digital input of the DSP chip looks for a balanced AES3 signal. AES3 is generally 10x the size of S/PDIF (the unbalanced stuff that dribbles out on RCA sockets). ....


I did not notice this discrepancy from AES/EBU to SPDIF to cause a problem with my DSP8024. I plug the SPDIF from my CD player direct into the 8024's AES input, play a test CD at 0dB, and it shows 'digimax' (0dB digital) input level.
 
Can i expect signal degradation from converting?

Im thinking of connecting one
of those cheap digital converters between CD player and Behringer for optical to coax conversion.

http://www.partsexpress.com/pe/showdetl.cfm?&DID=7&Partnumber=180-961

I want to maintain the maximum in SQ ofcourse.

If there is any increased jitter will the Behringer DCX2496 correct this later in the chain? (reclocking)

Any better ways to doing this?

Thanks,

Coolin
 
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nowater said:
I did not notice this discrepancy from AES/EBU to SPDIF to cause a problem with my DSP8024. I plug the SPDIF from my CD player direct into the 8024's AES input, play a test CD at 0dB, and it shows 'digimax' (0dB digital) input level.

The audio level coded into the digits is an entirely different thing. The significance of S/PDIF being only 1/10 the amplitude of AES3 is that the input slicing circuit of equipment expecting to see AES3 may be able to distinguish between the 1s and 0s of S/PDIF, but it will certainly recover the data with more jitter than if presented with AES3.

Coolin: The best way would be to go inside the CD player, pick up the CMOS signal feeding the optical driver, and feed it instead to a RS485 line driver chip. (RS485 is effectively the electrical specification of AES3.)
 
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ergo said:
stef1777 you and every other tweaker hearted owner of this unit is out there with the same quest. Until now it seems that there isn't even a hope to find such a thing.

Noone even seems to know anyone who has one :(

Ergo

PS. So backengineering is the only way to go and it's under work.


I deciphered some of the signals in the DCX2496 going through the flatcables and the ps. Anybody has any additions?

Comment: it seems one half of either the AK5383 or the AK5393 ADC is unused, which could be used with a switching option to use the Behringer as a preamp for analog sources.

Secondly, since the input channels in the ADC's (as well as the output channels in the DACs) are centered around +2.5/2.75 common mode voltage, how do they couple the analog sources to the ADCs? Capacitors?? There ARE 6 caps (47uF/25V electrolytics) just opposite where the flatcable lands on the input board. Then again, it seems that the same cap is used liberally throughout the unit, also for supply decoupling.

Jan Didden
 

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I do not know if you guys have a same problem, or if it is just me. I have DEQ and DCX 2496. I feeding them directly from CD ether optical or standard 75ohms coaxial Toslink. It is heavily clipping at the input. I would like to avoid attenuation available at the input, because of loss in resolution - dynamic range. If I feed optical to DEQ and than pass signal through the AES/EBU to DCX it is clipping at input of DEQ. I tried coaxial feed to DCX directly trough regular XLR/RCA adappter and it is overloading it in the same way. I didn't try yet to find some converter from 75 to 110 ohms, but I doubt that that will solve the problem. I found that M audio makes digital converter from any format:
http://www.m-audio.com/index.php?do=products.main&ID=e16b51d69355be3746c413e171976727
but I am little bit worried of spending extra cash on this solution. It seems that ether CD sends too much of a signal, or there is some icompatibility.
I would prefer to go directly to DCX digital input. I was thinking that DEQ would do a good conversion from digital optical signal to AES/EBU, but that is not case since even that is getting overloaded with the signal from CD. In both units there is possible attenuation, but as expressed in this and other treads that affects resolution.
Just to share my experience on remote post DCX attenuation. I am using Marantz receiver with 6.1 inputs and outputs and is doing very fine job.
If anyone could relate to this problem, please let me know what is your solution.
Thank you
AR2
 
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Is this happening with ALL CD's or just some? If there is clipping on the digital inputs, it's not an electrical level problem, but it would mean that the data samples coming off the CD are max value. Some (many?) CDs are recorded with regular digital clipping, I have several of them. There is nothing simple that you can do, other than a digital level control which in fact is a DSP. As far as I can see, the Co3 doesn't do that.

Jan Didden
 
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AR2,

Are you sure the input is really clipping or are just the indicators flashing red? When using the digital input of the DCX the flashing indicators don't reliably tell you if clipping is occuring and they don't follow any input attenuation settings you might have. In fact, they're basically useless.

I've used the digital-out from a couple of different CD/DVD transports using test disks with 0dbFS recorded signals. The red indicators are lit continuously but the output signal may or may not be clipping depending upon the setting of the input sliders. The clipping threshold on my DCX seems to be between the 0db and -1db settings. The only way to tell for sure is to monitor the outputs with an oscilloscope.

I hope I explained that well enough. It can be a bit confusing, but I think your situation is probably okay and not as bad as you might think.

I'm not sure how the DEQ factors into your situation, but I don't think it's making things worse or better.

Cheers,

Davey.
 
Thanks for the fast responce.
My judgement is not as much based on input LEDs, but there is audible
clipping sound from speakers. I would agree that LEDs are not very good indicator, and it is correct that they do not represent signal after attenuation. DEQ has much more elaborate input/ output signal display, and it is visibly clipping there as well. If all is right I do not need DEQ since DCX has all the features I need, and it will be one less unit in the chain. If I run analog input all is fine and signals are fine, but I would prefer to have digital in in order to minimise AD DA conversions. I am checking now on usage of limiter on the input of DCX.
Thanks
 
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I don't know how this is with the DEQ, but the DCX clip LEDs indicate a digital signal of value -1dB, this is (for 16 bit) 11111111111111110 or something. It has NOTHING to do with the electrical signal level, it can handle input (electrical) levels between 0.5 and 10V without clipping. It also has nothing to do with the (digital) input level set, that just converts the samples for the DSP to a lower value, and comes later in the chain.

A signal of 1V amplitude containing the bit stream as indicated above will light the RED LED. As such, the LED (which is turned on/off by the processor) is very exact.

Jan Didden
 
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Jan,

Yep, understood. However, that is upstream circuitry that is out of the users control.

I believe we're trying to understand any operational settings that need to be addressed to prevent clipping or "clipping" that might occur later in the DSP chain as you noted. In that case an adjustment of the input slider may be required. That was what I was trying to explain (not very successfully.) Sorry for the confusion.

Cheers,

Davey.
 
DEQ2496

It's a pain, but it should be evident that you must have as much overall digital cut as you have digital gain at *any* frequency band if you want to guarantee that you will never clip. You may get away with more gain at the frequency extremes but if you have a band that is +6dB in the midrange and your overall level is 0dB, you are playing with fire.

I can say this from personal experience. ;)
 
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Davey said:
Jan,

Yep, understood. However, that is upstream circuitry that is out of the users control.

I believe we're trying to understand any operational settings that need to be addressed to prevent clipping or "clipping" that might occur later in the DSP chain as you noted. In that case an adjustment of the input slider may be required. That was what I was trying to explain (not very successfully.) Sorry for the confusion.

Cheers,

Davey.


Yes, of course. Sorry I didn't pick up your point.

Jan Didden
 
AR2 said:
My judgement is not as much based on input LEDs, but there is audible clipping sound from speakers.

Initially AR2 said clipping was at the DEQ input, but later changed it to at the speakers. The former focuses our attention on the DEQ/source interface (where there is more likely to be a fault than a mismatch). The latter brings all the DEQ settings, all the DCX settings, and even the receiver and power amps into the equation. For example, what is the input overload level for the Marantz receiver? It is accepting a +22dBu pro audio signal from the DCX, and probably expects -10dBV consumer level.

What happens with all DEQ settings zeroed, and all DCX EQ settings zeroed, including all master volume levels of both components?

Grant
 
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