Digital EQ

Was wondering if anyone found success with running a digital EQ? I am looking at 2 Behringer models: the 2596 and the 1024. The appealing thing, as it looks from the design is, I can run the S/PDIF output from my digital source, into the Digital EQ, then back out (in either AES/EBU or S/PDIF) to my Bel Canto Dac3. In theory, the equalisation takes place in the digital domain without the need for conversions. The only conversion takes place from the Bel Canto....
Anyway this seems like an inexpensive and effective way to get high quality equalisation into play.

I am really only interested in keeping this in the digital domain. I don't want to use the analog circuits of the would be EQ unit.

I did exactly as you describe, with very good results. Digital out from a Squeezebox, to a Behringer 2496, then out to an external DAC. I've since upgraded to a TacT RCS 2.0 (digital room correction/pre/dac), but the Behringer was a great start. You'll be amazed how good your system sounds once you correct for the room.
Is 2596 a new model? The 2496 is excellent. The 1024 is obsolete.
You might also consider a Rane RPM 2m digital processor. You'll need another system for measuring with appropriate resolution.
Ecruz, I use a Tact as well, but have been looking at the 2496 for a friend with Thiels (too darn bright!) . In reading the 2496 manual I got the impression that it does not correct the bass, or at least there is a problem when using the correction below 100HZ. Any suggestions?
I read that as well, but I corrected down to 30hz and didn't notice any issues. There are others here who know a lot more about the Behringer than I do. I think Eldartford is one of them, maybe he'll chime in.
typo on my part...meant 2496.

You guys would not believe this one:
Last night I configured my squeezebox/server to run that DSP/EQ function (InGuz Audio)...amazing help to my problem with crappy recordings.

I had to remove SqueezeCenter 7.2 (btw I DO NOT LIKE the new resource hog, slow squeeze centers) and install 6.5.4 i think it run the EQ....

I don't think i need the 2496 now, as all i was trying to do was equalize my poor recordings...and so far its a success. I still have lots of playing to do with the config file and what not, but WOW what a Godsend. This dsp engine actually lives on your server PC, and processes the filter function on the files before it sends to the Squeeze. There is about a 10 second delay b4 you hear the changes. You can save many different settings and load as you see fit. Totally killer as this approach is the most simple, cost effective and seems to work well.
You can configure 2band, 3, 5, and 9 band eq functions. Perhaps there is a parametric setting that i don't know about yet, but it seems to work fine the way it is for me.
Thank you Inguz Audio!!!!

PS: if you attempt this yourself make sure you have at least .NET 2,0 installed ;)
Ok, now thanks to you I looked at the TACT site...starting to drool at the possibilites. I also realize i've been fighting room resonances and what not over the years and if I would have got into this method years ago with a proper implementation, I would have likely saved several $K on the merry go round...just a hunch but given what the 9 band Inguz Audio DSP engine did via the Squeezebox i'm sure the TACT would take this many levels further.
How are you running the a DAC, or running all digital then doing your conversion with some other DAC?

If the TACT has a DAC that is on the level of my Bel Canto DAc3 then I guess I can lose the DAC3, but I like this piece. I guess I could use the TACT this way:

SB3 -> TACT -> DAC3 -> AMP
or this
SB3 -> TACT -> AMP

how is the TACT as analog preamp? Any comparisons with typical
"high end"? I've used ARC Ls15, LS25 mki/mkii, Ref1, BAT Vk5i, Pass X2.5, Pass X1...pretty much all slight degrees of change, all very neutral (duh) and transparent. What I NEED is to address the room modes and coupling to my speakers. I have Dynaudio Contour S5.4s, a mid size room, and a Plinius SA102. I know the speakers are extremely capable; I am positive the amp needs to be about double the power, esp if i'm going to correct the response with shelved up lower end (the speakers can easily accomodate the curve). So, my game plan is to sell the amp, replace with a pass x250.5.
What I heard so far (with the inguz audio) has made it clear that I have issues with proper room/speaker coupling that I was ignoring; but, stupidly continued on the merry go round swapping components to make me happy-- "buzzz" wrong answer.
Samujohn...It's the AUTOMATIC equalization process that Behringer suggests may not work well below 80Hz. The equalization itself works fine. In my experience with my room and system the autoeq function also works all the way down to 20 Hz.

I feel the same about room correction. If I had figured it out years ago, it would have saved me a lot of $$$ too.

I don’t use any analog inputs, so I can’t comment on how it compares to other (analog) pre-amps.

I connect the digital outputs of my Squeezebox and my Oppo 970 to the TacT. The TacT’s internal DAC does the D/A conversion. I’ve considered running the digital out of the TacT into a better DAC, but I’m concerned about adding more cables, connections, etc and all of the issues that go with that. There is a company that mods TacT products, one of the upgrades is to the DAC. I’ve heard of it’s a worthwhile upgrade, but I have not tried it myself.

Good luck!
I think that is an outstanding solution. Now that i have full range and wide dynamic range speaks the amp needs to be beefed up to say, the 250.5. Let the software correct the acoustics, and I should be all set.
FWIW, im going to take this Inguz Audio DSP stuff further and sweep the room and generate a correction filter. Im just learning about this niche stuff as we 'speak'. Its definitely a tweaker's paradise. Hugh, who wrote the EQ plugin, has quite a bit of info on the technology and filtering methods, and the stuff is very flexible. You can choose longer filters at the expense of CPU time, but most machines can keep up fine. The cool thing about this approach is that is all done to the actual audio file prior to the squeeze even getting it (and on less component/cable) ...its the same thing as the tact, in that its in the digital domain; but at diff points in the stream. The Tact prob is truly on the fly but the Inguz has a ~10 sec latency due to the squeeze buffering the input.

I would like to learn more about the DSP approach taken by TacT. I think for now im happy learning this way, and then eventually ill end up with the Tact for max flexibility. Does yours have a truly parametric eq, or single band?

"how is the TACT as analog preamp? Any comparisons with typical"high end"?"
I use a Tact 2.0S as my analog preamp. My CJ EF1 phono preamp goes into the analog inputs along with my tuner.
Before I bought the Tact, I heard my old CJ PFR line preamp's output run through a high quality AD/DA converter (Meridian) on its way to the power amp (I think at 96KHZ). To my amazement the bypass vs. the double conversion was hardly noticeable. The Tact 2.0S A/D converter is even higher resolution at 192KHZ and the older Tact 2.0 converts at 96KHZ.
All that said, the obvious for all to hear advantages of digital equalization and room effects compensation, swamp any slight effect of the AD conversion.
I can see your point. I'm learning alot more about this now. The Inguz DSP engine for the Squeezebox is a powerful piece of software. Anyway I am customizing my own EQ curves, and plan on using the EQ Wizard to generate my correction filter. Right now i'm just tuning by ear (trial and error) but the results are very positive. Unlistenable recordings are now quite enjoyable...IMPOSSIBLE with the typical high end "purist" approach (despite having passive acoustic panels and decent room for starters) i've been taking over the years. You can literally tune in different 'voicings'. I'm going to use this stuff to learn (deq/drc) with but seriously consider using the Tact and that's that.
As for the room response side: once the hardware is in place you use, what, a single omni mic located at listening position, and let the system play the tones, while the controlling pc captures the response? What about the use of an SPL instead of, or in conjunction with a dedicated mic?

Yes, the TacT uses a single mic and it sends out it's own signal, that it uses to "read" the room. You can save different mic positions. For example, I have one setting for the sweet spot for "serious listening" and another for when I'm sitting in a different spot when reading, under the light. Theoretically, you could have a setting for each seat in the room.

I've only scratched the surface of what the TacT can do. I only use the built in curves, for those two positions. There's much more that it's capable of.
I have had my Tact for about two years. There are nine presets so we have been trying all sorts of equalization curves and switching them on the fly. For my ears, there is only one "correct" curve for each source. What I mean is that no matter how many choices I start with, over time only one sounds correct so I abandon all others.
I progress by erasing all but the current "correct" curve and making slight variations on the theme. This way I slowly close in on the "most correct" curve.
Silly I know, but I am just describing the process. Peter Walker of Quad fame maintained that each recording had a correct volume. I expect that if I had a quick and easy means of equalizing each individual recording and saving that setting, that would be the best system, but my Tact is not that advanced. Some of the current models are more flexible and easier to adjust on the fly.

That's exactly what I've done. I put 9 curves in, only used 2 or 3 and now really only use the one that sounds "right".

Sometimes, just to remind myself, I bypass the correction and cannot believe how awful everything sounds. I can't imagine ever going back to listening without some form of digital room correction.
Ecruz/Samujohn...right on!!!!...already with this Inguz DSP/Server side correction (it processes the music files as they are read off the HDD and before it is streamed to the squeeze) I have gone back to direct/flat and the sonic result actually sounds BROKEN. I can't belive this is the sound I've been building for years. No wonder I could not escape the merry go round!!!

The thing that has blown me away is that it's not just hearing bass boost or treble cuts, it's that the whole spectrum is WAYYY more natural and seperated...the stacked images i've been fighting are coming apart and easy to place. This is amazing stuff. Last night I literally dialed in liquidity to female vocalist, that reminded me of when I had not 1 but 2 ARC VT100MKIIs over the years.
--I now fully realize that i've been fighting phase/amplitude/room reactions that end up obscuring the recorded message. Stuff that cable swaps (and likewise componenet swaps of similiar caliber) won't/can't fix.