Behringer DCX2496 digital X-over

I found that the auto delay feature on mine does not perform correctly

Question: do you select aouto delay short ONLY? If you do not unselect LONG in a home environment, it sets totally wrong delay times. Long is only meant for multi speaker setups in PA systems.
My short delays are repeatable and make sense - i.e. the tweeter ist delayed by app. 3 cm - correct for the mounting position in the ls baffel.
 
Transceivers

Lyckman said:
Ok. 3m is way longer than I was aiming for. More like 30cm or so :) It would be nice if I could "be without" transievers.

Hello Lyckman,
I guess without transceivers I2S is not such a good idea. Some days ago I received a flyer from analog devices. There I found a new chip with differential output which is exactly suited for this application. It has three channels (bitclock, lrclock, data). It is very fast and can also be used for RGB TV-Signals. It's name: AD8133.

http://www.analog-devices.com/en/prod/0,2877,AD8133,00.html

This chip decreases the effort for drivers a lot!
Charly
 
Yes, I understand about the long and short delay. The problem is with the short delay. The auto function does not work , at least on my unit. Its easy to measure this with an MLS burst to each driver in turn - the dlays are way off more, taking in to full account the difference in acoustical path lengths.

On a related point, late last year, I purchased a Behringer SHARK 110 and found by measurement that with the delay set to zero the box was giving nearly 100msec of delay. I chased up Behringer, but gave up after the third attempt of not being able to get through to a design engineer. Then, I returned the unit and got a refund.
 
AD 249 SOIC NOT pincompatible with JRC 4580

I chased up Behringer, but gave up after the third attempt of not being able to get through to a design engineer. [/B]



HA ! DESIGN ENGINEERS ! They are too busy looking at other companies stuff. ;)
BTW, please don't make the mistake that I made - I replaced all of the surface mount 4580 OPAMPS with AD 249s just to find out that the AD's SOIC pin configuration is DIFFERENT than those of their AD 249 DIP packages - I just wanted to kill myself. I have been successfully replacing all 4580 DIPS with AD 249 thusfar. Serves me right.
:mad:
 
Improoved Sound Stage and 3D-Imaging

Dear folks out there,
I would like you to participate at one of the most interesting filter designs I ever tested on my Behringer.

Some time ago a French gui called Thierry Martin mailed me some information about a new kind of filter design for speaker crossovers engineered my Mr. Le Cleach. The material was ONLY in French so it was quite hard for me to follow his approach. Finally I found a French person which translates the article to German (uuh, what an ugly German!). After that I started translating the article to real German and also to English for you. The whole material you’ll find here (8am to 10pm german time):

Document and Speadsheet (doc, xls): http://freerider.dyndns.org/anlage/LeCleach1.zip
Theory of operation (ppt): http://freerider.dyndns.org/anlage/LeCleach2.zip


Well, in-between I tested the approach of Le Cleach with my DCX-2496 and got so astonishing results that I thought I had to let you know about his theory. Believe me or not: With his filter design you get a soundstaging and imaging you never heard before on your system!!

So what has to be done?

First adjust your DCX-2496 the way you have a flat frequency response and time aligned speakers which can be achieved by the auto align function of the DCX 2496 (yes, you need a microphone this is VERY important!). I guess you did it before ;-)

Now read the paper (.doc-file) I submitted in the ZIP, open the spreadsheet and enter the values according to your current DCX-2496 configuration. Do not enter the current time delay of your DCX in the spreadsheet! Now in the spreadsheet modify the values as described in the paper. If you got a good result in the spreadsheet adjust your values in the DCX-2496 so they match the spreadsheet.
A bit care has to be taken at the delay. The auto align function of the DCX calculates values for the delay. To these values the values from the spreadsheet have to be added. This means the distance between the chassis will become bigger. Le Cleach says the speakers (bass, mid) have to be moved towards the listener. With the DCX-2496 you have to move the mid and tweeter away from the listener which is the same!! The distance gets bigger.

Now save this adjustment under a new preset of your DCX, sit down and listen while switching between your old preset and the new one. What you will hear is really astonishing. The whole characteristic of the sound will stay the same (same frequency response) but with the new settings the soundstage expands and you will get a never heard 3D-imaging.

A nice test for the new setup is the SACD “Scared Love” from Sting. In my opinion it is a bit over instrumented and it is sometimes difficult to differentiate between the voices and the instruments. With the new filter parameters the whole soundstage unfolds and ….. you know, I now really love this SACD.

And best of all: The filter modification is for free just try :)))
 
Re: Improoved Sound Stage and 3D-Imaging

oehlrich said:
.... Some time ago a French gui called Thierry Martin mailed me some information about a new kind of filter design for speaker crossovers engineered my Mr. Le Cleach. .....


Hi peeps,

I'm the frenchy Oelrich mentions. Thought it could be of interest to have a look at Jean Michel Le Cléac'h following those links (he's the grey beared chap in the middle of the picture).

http://www.melaudia.net/etf03cr1.php

This was shot at the "European Triode Festival" in 2003.
In 2004 (same period) this yearly event took place in Langenargen, near the Constance lake. Jean Michel demoed his method in front of a bunch of passionate Diyer's. I wasn't there but I will attend a similar demo in Paris at the end of March this year. Maybe Oelrich will join ;)

Cheers,

Thierry Martin
 
Ex-Moderator
Joined 2003
I suspect what's being asked is whether the phase control adjusts driver delay by a very fine amount expressed as phase at the crossover frequency, or does it adjust phase at the crossover frequency but have little effect at other frequencies (ie not pure delay). I'd be interested to know the answer to that question too.
 
Account Closed
Joined 2001
The phase control is frequency dependent delay just like a second-order all-pass function. The high-pass xover frequency set in the appropriate channel (whether it's activated or not) defines the point where the phase response rotates through 180 degrees.

The delay control implements a pure delay.

I have a SoundEasy screen capture of a test I did quite awhile back that illustrates how it operates. I used 1khz for the turnover frequency. I hope this link works.

http://f4.grp.yahoofs.com/v1/8AodQk...E7sTAHGmaFzAURRk0SE-G1zQ/phaseversusdelay.JPG

Cheers,

Davey.
 
Thanks Davey. I can't see the image, but I know exactly what you mean. I'm impressed that they actually implemented this correctly; that's another tool in the arsenal.

My second question wasn't related to delay, I just wanted to confirm that you can have both digital and analog inputs active at the same time. (A+B digital, C analog) The software lets me do it, but I couldn't find anywhere in the manual that explicitly said it was kosher.
 
Ex-Moderator
Joined 2003
Davey said:
The phase control is frequency dependent delay just like a second-order all-pass function. The high-pass xover frequency set in the appropriate channel (whether it's activated or not) defines the point where the phase response rotates through 180 degrees.

Thanks for that. Now all I have to do is to knock up a microphone amplifier so that I have a loudspeaker measurement system and I can play and see what I'm doing...
 
DCX2496 minus 16 dual opamps + 8 transformers

hi guys,

Just finished major surgery to my DCX, I removed the I/O pcb and
mounted input and out transformers. The transformers are microphone input transformers of good quality but of unknown origin, ratio is 1:3.16.

The '1' winding has a centertap that I use to supply a 2.5V offset
voltage (from two TL431's) to the ADC's.

Result is a lot more 'air', space and detail. The sonic impact of
this modification is about the same as mounting the Tentlabs XO2.
I mostly listen to CD through the digital input, but the analog
inputs improved quite a lot also.

kind regards, Peter
 

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power supply bypass

hello

for those who want to mod less radicale i can recommend to bypass the 100nf condensators for ps decoupling for the 4680 opamps (c1-4, 69, 70, 92-97) beside the mods on the analogue i/o suggested in this thread earlier.

i used 10uf smd tantal (c - footprint) and it really is surprising what a big improvement this is. especially on low and very low levels. which means first input led flashing or below you gain great in tonal smothness and soundstage.

one would think that at such very low levels the imperfections of digital converters would dominate the sound but as far as i can hear, there is still a lot of improvement that can be done at the analog side.

http://www.kinotechnik.edis.at/pages/dcx2496/DCP02068.jpg

http://www.kinotechnik.edis.at/pages/dcx2496/DCP02067.jpg

for peter
from your picture i would guess that your bypass condensators for the ad/da converters are still original. try to replace them with 10u tantal - if you dont have anything better in mind

http://www.kinotechnik.edis.at/pages/dcx2496/DCP02031.jpg

its a very cheap mod easy to do and gives very good results as well
 
re: power supply bypass

hello

tantalum condensators seem to have bad reputation among audophils, though they are industry recommended even for very high speed opamps.


http://www.linear.com/pc/downloadDo...Ls72LkVaHwmTWnVCzqqsb0Qp8DcmgF06xQCZLfcHCpxia!-1570396240?navId=H0,C1,C1154,C1009,C1022,P1109,D1367
(linear technology lt1361)

but i have to admit that i havn't compared against BG's for example. i am just saisfied with the tantals until now.

the clock upgrade you recommend is something i will keep in mind, thanks. i probably will give it a try with the cheap xo clock module only.
http://www.tentlabs.com/

greetings michael
 
Our member Bill Dipoala has constructed volume control board, and op-amp replacement board that all fits in Behringer unit. I got from him these boards and the quality is outstanding. Not just that but he also supplies parts list, big schematics printout and parts placement map. Here is the link.
http://www.diyaudio.com/forums/showthread.php?s=&threadid=51519
The boards are allowing you to use optical in instead of AES/EBU if you need, remote control, your choice of any popular dual op-amps, separate board for single ended output, or balanced board...
Circuit boards are in Bill's own words "All PCB’s are constructed using double sided FR4 laminate with plated through holes, solder mask and silk screen (better quality than what is in the DCX). "
If you do not need volume control, op amp board is stand alone and you could just order that. In my mind any modding on existing output stage is just a waist of time and money. In original construction analog stage first converts balanced signal to unbalanced, than it goes to servo balanced output circuit and all that makes up 24 op amp stages.
The fact that I like the most, with Bill's boards you do not need to destroy original Behringer. Everything is connected between boards by ribbon cables, and if you change your mind all that you have to do is unplug. Even more, the power is coming from original Behringer power supply so you do not need to worry about separate case, because everything fits within original housing.
I have built volume control and tested it. The sound is great with minimal influence. I am still to build and test op-amp board, and I will report on it. I am also testing Lundahl 1674 transformers as output instead of op-amps. It is much costlier solution, but the sound is awesome.
I am not associated with Bill at all, just a happy user. I think that Bill did such a great job that anyone wanting to modd Behringer should look into it.
Good luck with your modding
AR2