Behringer DCX2496 digital X-over

Hi,

That was my point, no de-emphasis in the analog output section. This means that it is done on the digital domain.

If the schematic is right, (and I supose it is, if not, what would those components do there?), a trannie does not look like a good idea for the analog input stage.

Miguel mentero
 
AX tech editor
Joined 2002
Paid Member
Mentero said:
Hi,

That was my point, no de-emphasis in the analog output section. This means that it is done on the digital domain. [snip]

Miguel mentero

I'm not sure that there IS de-emphasis. The technique of giving a slight mid boost often gives perceptually more 'likeable' sound. I did notice that the roll-off of the output section is rather fast, as shown in the posts referred in my above post.

Jan Didden
 
Hi Jan,

It is some time ago that I did simulations and measurements, but I remember that the transfer function of the whole unit is pretty flat.Your measurements show the same.

So I supose there shall be some kind of de-emphasys somewhere and it is not on the analog output section.

Few options left.

Anyway, it doesn't matter unless you are trying to substitute the input section with a transformer.

Miguel mentero
 
diyAudio Member
Joined 2004
Re: Improoved Sound Stage and 3D-Imaging

oehlrich said:
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 :)))

Very interesting stuff.

I'm a little confused as to how you enter your values and then calculate the new ones.

Could you give a little more detail please?
 
diyAudio Member
Joined 2004
OK I've looked into it in a little more detail and read the original paper by JM Lecleach.

I've now modified the original spreadsheet so that the all you have to do is enter the upper and lower crossover frequencies in your 3-way design. These are highlight in yellow on the excel sheet. Change these and the values as a whole change accordingly.

The ones your interested in are those highlighted by the purple box. You add these to your delay values in your DCX, these are in mm, the same unit as the DCX.

I was impressed by the extra spaciousness afforded by this little tweak. Definitely worth a go for the very small effort involved.

BTW: You need a well sorted DCX setup before you do this. Make sure you've time aligned using the auto setup and have a pretty flat frequency response around the XO points and the frequency response in general, for this you'll need a measurement mic and a real time analyser such as TrueRTA.

EDIT: I cannot attach the file :( Its too large at 220k even when zipped. If anyone is interested give me your email address and I'll pass it on.
 
Hi ShinOBIWAN,

Although I am Thierry Martin (the bloke Oelhrich mentions) I'd be interested in having a look at your XL sheet.
My email is accessible in my profile :)

One important thing for the DCX is that you must NOT use "link" mode between LF MF and HF, but choose "free" instead.
One can very easily tumble into this trap - I did :rolleyes:



Thierry
 
diyAudio Member
Joined 2004
I also forgot to mention that you must invert the phase of your mid range in the DCX and add change the crossover values to that of the spreadsheet. Finally make sure your using Butterworth 3rd Order as the XO types. As well as adding the the time delay to your existing values.

You can ignore the gain values and leave them at 0 no matter what your setting is on the DCX.

The folks that asked me to email them via DIYAudio need to pass on there email address since the forum hides them and uses its own script to send a simple text, message meaning no attachments.
 
diyAudio Member
Joined 2004
Re: Modified Spreadsheet

oehlrich said:


Hi,
be so kind and mail the modified file to me. I will put it on my webserver for public access

Charly

Oehlrich,

Could you pass on your email address and then I can email the file to you.

If possible could you take a look at it and make sure everything is working OK and that I havent made and errors. You have experience with this and I'm just going on what I've managed to understand from the white paper that Lechleach wrote.

Cheers,
Ant
 
diyAudio Member
Joined 2004
Thmartin said:
One important thing for the DCX is that you must NOT use "link" mode between LF MF and HF, but choose "free" instead.
One can very easily tumble into this trap - I did :rolleyes:

Yep I know what you mean, if you use LINK then the crossover points move over the same points. Only with FREE can you use truly independant XO points on all the drivers.
 
diyAudio Member
Joined 2004
I've just noticed an error on the sheet, the section that says:

Value to add on low pass (bass) section in (mm)

Should be:

Value to add on high pass (treble) section in (mm)

Otherwise if you add the value to the bass your moving that further away or delay the signal even more which completely defeats the point of the filter.

What should happen is that the value calculated for the distance on the bass driver should actually be added to the treble which moves the treble further away, which is the same as moving the bass driver closer.
 
ShinOBIWAN said:
I've just noticed an error on the sheet, the section that says:

Value to add on low pass (bass) section in (mm)

Should be:

Value to add on high pass (treble) section in (mm)

Otherwise if you add the value to the bass your moving that further away or delay the signal even more which completely defeats the point of the filter.

What should happen is that the value calculated for the distance on the bass driver should actually be added to the treble which moves the treble further away, which is the same as moving the bass driver closer.

The error, is it in the original spreadsheet or you modified one?