# Equalizer update



## gerG

Sort of rushed, but the info is there (I hope). Crooked pictures too!

Behringer deq2496 update 

 Acrobat reader required.


 gerG


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## zeplin

gerg,
 some day i will get this EQ. how much was it? and when you say you are using a panasonic 963sa, don't you mean the philips? i have decided that my next purchase, and i won't even be buying it (my b-day), will be a new source, which hopefully will give me a little better bass extention than with my rather "low end" sony dvd/cd. oh yeah, and sorry i never got the chance to stop by while in AZ, i was just too busy. so again, after i get the new source, i'm definitely going for an eq like the new one you just bought. so what do ya think of my decision to go for a new source?


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## TravelLite

Looking really nice, gerG! Excellent!

 TravelLite


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## j-curve

gerG, 
 Thanks for this much anticipated review! You have confirmed my hopes that this would be a killer product. The ergonomics provided by the display (despite the barfy colour) and rotary controls alone would be a welcome upgrade to my DSP1124P, for sure. Optical input is just the icing on the cake. A few questions though, if I may, mostly because I'm unclear on how sampling rates are handled inside this gizmo:-

 1. What does the "Sample Rate" screen do? Can you select the sample rate or is it just telling you what the input rate is?
 2. When driving the optical input from a CD player (16 bit, 44.1kHz) do the internals run at 44.1kHz or is the sampling rate converted first? 
 3. What about the optical output? Does it mimic the format of the input or can you select the rate and resolution?
 4. What about the analog outputs? Are they always running at 96kHz or do they mimic the input?
 5. When running from analog inputs do the internals always run flat strap at 96kHz or is that selectable too?
 6. You mentioned making balanced XLR to RCA connectors for the analog output - why not use the 6.3mm auxilliary outputs with RCA adaptors?
 7. Does your XLR->RCA cable ground the negative output and take the feed from the positive?
 8. How would you rate the transparency of the all-analog path?

 Sorry for all the tricky questions. If the DEQ2496 is calculating (parametric & graphic) filters at different sampling rates then that means a full set of filter coefficients for each rate, which would be quite impressive. The alternative is resampling, which is easy for 48->96 or 44.1->88.2 but more challenging for 44.1->96, which I believe requires a 276 pole digital (interpolating) filter.

 Although I'm wildly enthusiastic about the DEQ2496, I do think it's a bit expensive for something which by now could (and should) be part of any mid-market CD player. On the other hand, that may never happen, and if you compare the Behringer to similar products, it looks like an incredible bargain. Maybe Mr. Uli will give me a discount, since if I am not mistaken he owes me a commission on the sale of one supa-fly EQ to a total freq-freak here on Head-Fi. 
	

	
	
		
		

		
			





 Once again, thanks gerG for your interesting review.

 PS: Any chance of a photo of its gizzards? I need to know if my MSCA Amp is going to fit in there. 
	

	
	
		
		

		
		
	


	



 BTW, the gaping holes behind the rack mounts can have advantages if you dare to fit a headphone amp inside the EQ. [Check my MSCA link for piccies.]


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## kwkarth

Very nice gerG, that eq is still calling my name.


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## RussKon

just a word of caution.....

 behringer has had many quality control issues over the past 18 months....

 i personnally had to return three behringer units for the same customer....

 you find the same comments on the pro audio boards concerning behringer products....

 make sure you buy from a company with a good return policy!!!!


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## kwkarth

Good cousel RussKon, I've seen /heard of QC problems with Behringer too. Nowadays, I personally feel it's good practice to buy everything from reputable dealers. I seem to attract QC problems. If anything can fail, it will when I get it. That said, I've NEVER had any quality problems with my HeadRoom MAX. I can't think of another commercial product/brand that I have not had quality problems with, somewhere along the line including Sennheiser, AKG, Grado, Meier Audio, Sony, Carver, etc. If the manufacturer/dealer is reputable and reachable, they've always taken care of the problem, thank goodness! Sometimes I can fix the problem myself and I do, but often I've had to rely upon the dealer or manufacturer to resolve the problem.


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## gerG

Hi gents, I'm back (for a while at least).

 Musician's Friend has a 45 day return policy.

 So, kwk, have you got one of these toys yet?

 Hi j-curve, good to hear from you! Answers (or guesses) based on current limited knowledge:

 1. In the mode that I am running it the rate syncs up with the input. 96 khz is the max, so it will make funny noises if you feed it a direct stream from sacd or dvda at 192 khz (I tried). If I plug in a normal CD transport it switches to the lower rate.
 2. afaik, the unit runs at whatever the feed rate is. I will have to check to see if streams are converted for calculation.
 3. Not know, have to check.
 4. Analog is infinite (tee hee). ok, I know that you are refering to the data rate prior to conversion. Again, more digging required.
 5. Similar to the above questions. 96khz from what I read.
 6. I used a similar approach on the older eq, and didn't like it. It is still a jump from balanced trs to unbalanced rca, so it boils down to a connector preference. Using an adapter also adds another junction. You will just have to start saving for that balanced amp 
	

	
	
		
		

		
		
	


	



 7. Yes, per the Behringer manual.
 8. I haven't tried the analog input of the eq yet. A fair question, but skipping the extra d/a and a/d steps just seemed like the natural way to go. My baseline "analog" loop bypasses the eq altogether.

 Very nice work converting the eq to a surrogate amp. I will happily do the endoscopy of the 2496 if you can figure out how to fit a balanced amp inside there. Feel free to use the trs outputs as the balanced headphone sockets. I get giddy just thinking about it. 

 I will see what I can find out about the internal rates. First, though, I am going to power the toy up and have a listen. Next trip it is going with me!


 gerG


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## j-curve

Hi gerG, thanks for your response to my numerous questions. I flicked through the Behringer manual once and it has plenty of nice diagrams but very little about how the insides work.

 BTW, any attempt to rewire the rear panel plugs as headphone outputs would be _very_ risky indeed. At least in the DSP1124P, those plugs and sockets are mounted directly to a double sided, surface mount PCB. It may be possible to add an extra (shallow) connector on the rear panel though. Good luck drilling the hole for that one! 
	

	
	
		
		

		
		
	


	




 I'm sure there's plenty of empty space in the DEQ2496 for a headphone amp though.


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## gerG

Hi j.

 There is not much room in there at all. There is a space that an amp board would fit, but it is awfully close to a fenced structure labeled "high voltage section". Probably not a good environment. The entire front panel is a front plane module, so no access there.

 It is such a small package that even with a couple of full sized amps sitting on top it still takes up less airspace than my old Ultracurve. What I really want now is a balanced amp in a matching rack mount case. Anybody want to build me one?

 I played with the dbx Driverack over the weekend setting up a speaker system. Although the dbx is a great sounding unit with amazing capabilities, the user interface and flexibility are pathetic compared to the Behringer. I would have swapped them, but I need the crossover function of the dbx. Results were still amazing, but it just took longer than it should have. 

 I have pics of the Behringer guts if you want them. Drop me an email.


 gerG


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## Steve999

I just got a Behringer DEQ2496 and love it. There's not much to add past gerGs definitive review. It's difficult to describe exactly how much FUN this thing is!

 I would add one thing, the "wide" feature makes a very nice bare-bones tonally neutral crossfeed. It not only expands the stereo image, it can compress it too (crossfeed). The compression settings run in ten steps from 0 to .9, with .9 being quite subtle crossfeed and 0 being mono. I compared the mono to the mono button on my NAD receiver and the tonal balance was exactly the same. Then the stereo expansion steps runs from 1 up to 3, with some interesting effects that would be fun to try with speakers.

 It's got a nice dynamic range expander and compressor, too, if anyone is into that type of thing. Nice for Berlioz at 2 AM.

 The sound quality seems unimpeachable to me. 

 Thanks so much to the head-fi community for bringing this product to my attention! It's among the most extraordinary audio products I've ever run across.


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## Orpheus

you know, if you guys wanna check this eq out, you could always get it from Guitar Center (who owns Musicians Friend by the way). they offer a complete refund within 30 days, for ANY reason. i've returned many items before... even sealed packages like cables and stuff. they will ALWAYS allow the return. they ask the reason, and i go, "I don't want it." seriously. the only exception are microphones... can't return those for any reason i believe, if the box is opened.

 also, never pay the asking price there. their normal price is already lower than competitors, but usually you can get at least 10% off extra. sometimes i've even talked them out of 1/2 their asking price. really.
  Quote:


 Good cousel RussKon, I've seen /heard of QC problems with Behringer too. 
 

 i've owned 3 behringer products... a mixer, and two smaller rack units. and i've not really had any QC issues. one channel did go bad in the mixer, but well, things break down. the major beef with Behringer stuff is that they simply suck. they offer way too many features, look spectacular on the outside, but does nothing well, and sometimes does nothing even good enough.

 but it sounds like Greg really likes this unit. so, it's worth a try i say. i mean, you can return it after all. EQ's might be the only thing behringer makes that's good! heh he...


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## Steve999

Yeah, Orpheus, that's accurate, the Rockville, MD, Guitar Center is where I got mine. I am absolutely in love with it. This thing is INSANE. It's like something out of some equalizer fantasy dream and then a whole lot more on top of that. FWIW, the sales clerk said they run through them pretty fast, take it with a grain of salt of course, they had 3 in stock. 

 I had some Christmas money I had been trying to decide how to spend for, well, 3 months now, and finally decided to get the Behringer DEQ2496. 

 Guitar Center noted the 30-day return policy for the Behringer in a text field on the receipt, as I was quite surprised and a little skeptical about it, but it's real. I got lucky, the sales clerk was extremely knowledgeable and enthusiastic about it and talked to me for quite a while about the features and walked me through my disbelief as he told me everything it could do. They gave me a pretty big discount on the XLR to RCA cables, too, I spent just 4 dollars more than gerG spent on the raw materials for making his cables. 
	

	
	
		
		

		
		
	


	




  Quote:


 _Originally posted by Orpheus _
*you know, if you guys wanna check this eq out, you could always get it from Guitar Center (who owns Musicians Friend by the way). they offer a complete refund within 30 days, for ANY reason. i've returned many items before... even sealed packages like cables and stuff. they will ALWAYS allow the return. they ask the reason, and i go, "I don't want it." seriously. the only exception are microphones... can't return those for any reason i believe, if the box is opened.

 ...it sounds like Greg really likes this unit. so, it's worth a try i say. i mean, you can return it after all. *


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## Orpheus

Quote:


 They gave me a pretty big discount on the XLR to RCA cables, too, I spent just 4 dollars more than gerG spent on the raw materials for making his cables. 
 

 actually, they would probably have giving you all the cables for free if you bugged them enough, + a discount on the piece itself. (it kinda depends on how little commission the guy's willing to accept on the deal. different people will give you different deals... and managers can give you the best deals--i used to be friends with all the managers... everytime i walk in, we get right down to business. i say, "what can you give me?" and they know i want the bottom line.)

 heh he...

 but hey, you guys are kinda getting me curious now. i usually look down on the behringer stuff, but you guys seem to really like it....

 have you compared it to other EQ's? how 'bout some higher-end analog EQ's? does it stack up?


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## Steve999

I have an old Yamaha analog EQ with a 108 db signal to noise ratio (purportedly) that I had been really happy with.

 With the Behringer DEQ 2496 it's a whole different ballgame, though, you get 64 memory presets for your beloved settings and you are making all the changes in the digital domain so you're not adding any noise. You get 10-band parametric EQ (graphically represented) and 30-band (1/3 octave) graphic EQ, and shelving EQ options with which I have custom-made my own treble and bass controls suited for particular headphones (I feel the balance changes from recording to recording so these are extremely handy) in addition to the custom curves you can make. 

 I also plan on making my own loudness control with the dynamic EQ settings. The real time analyzer is way cool (I had no idea my recordings had so much full-spectrum 20 hz to 20 khz activity in them!), too bad I don't do acid.
	

	
	
		
		

		
		
	


	




 I also use the stereo image option for extremely effective and adjustable bare-bones crossfeed on old jazz and Beatles recordings and the like. 

 As far as sound quality, I just can't find anything wrong with it (and I am obsessively picky) on the best modern jazz and classical recordings. 

 The real time analyzer is nice too for showing you the noise floor on your equipment, I suspect it would help you hunt down any hum or noise defects in your equipment as well. I was a little surprised at the noise floor on my NAD receiver (-90 db or so from what I can figure).

 I had returned a pair of AKG 240S's at Guitar Center and they sold me a guitar for my son at cost (my son was pretty brutal on my guitar) recently so I was feeling pretty grateful, so I didn't bug them about the price. $300 is the going rate on the internet, so that was good enough for me. And I saw similar cables selling for twice the price at B&H photo, and I remembered reading what gerG had spent on the raw materials, so I wasn't in the mood to nitpick, especially since the guy was so helpful and was so insistent about the return policy. It was all a good deal. The thing is a steal at the market price anyway. I've only scratched the surface as to its capabilities. As far as getting the most out of it, gerG is the man.
	

	
	
		
		

		
		
	


	




  Quote:


 _Originally posted by Orpheus _
*but hey, you guys are kinda getting me curious now. i usually look down on the behringer stuff, but you guys seem to really like it....

 have you compared it to other EQ's? how 'bout some higher-end analog EQ's? does it stack up? *


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## ooheadsoo

I've heard of a lot of DIY'ers who are using behringer products for cheap active crossovers and equalization. This sounds pretty sweet. Maybe there's even a cheaper model that still has the important features...anyone know?


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## Steve999

You can get this for about $180 - $200:

Review of Behringer DEQ8024 

 The reviewer obviously knows a lot more about it than I do, and it makes a very interesting read. Apparently it was selling for $600 - $900 dollars three years ago. Unlike the DEQ2496, it doesn't come with digital ins and outs, although you can pay extra for a digitial I/O add-on.

 I'd strongly recommend the DEQ2496 for $120 more though, with digital ins and outs and a dizzying array of options. Obviously, 'cause that's what I opted for. 
	

	
	
		
		

		
			





  Quote:


 _Originally posted by ooheadsoo _
*I've heard of a lot of DIY'ers who are using behringer products for cheap active crossovers and equalization. This sounds pretty sweet. Maybe there's even a cheaper model that still has the important features...anyone know? *


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## agile_one

Great writeup and photos, gerG. Thanks for all the helpful info. Now I'm thinking I've gotta have one of these in front of the Grace 901 - just when I thought my wallet was safe for a while - thanks a lot


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## gerG

Yeeks, I forgot about this thread!

 Steve, I have an 8024 at work, and it simply cannot compare with the 2496, either in sound quality, or in user friendliness. The user interface on the 2496 is almost like something Apple would come up with, it is just plain telepathic.

 By way of update, I consider the DEQ2496 my only really essential piece of headphone gear. The differences between sources, amps, and cables, and even headphones, becomes a secondary issue with this thing around.

 Personally I have had terrible experiences with analog equalizers. The bands interfere with each other, and the phase shifts will destroy the sound of good cans. The DEQ is nothing even similar. As an additional benefit, mine is not even in the signal path. It resides completely in the bitstream. Imagine, a system without a single interconnect or (friggin) RCA connector. Hmmm... I am tired of typing that. From now on I will just refer to them as FRCA connectors, k?

 Dean, I have the dbx Driverack Pro as well. It can do most of the things that the Behringer can do, but not as easily, and it will not allow digital input or output (a HUGE oversight). otoh it is a very nice digital crossover, so it does triamp duty, as well as the occasional K1000 + sub gig. If you are still wavering, consider that you can feed a digital input into the DEQ, then run a balanced analog out to one setup, AND a digital output to the Grace, at the same time! This leaves the analog output fro the player free for yet a third system. Fun!


 gerG


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## gradofan

gerG helped convince me to by the DEQ (actually, I bought it solely bc he has good judgment and recommended it), and it's a knockout piece of gear! I've had it for a week or so, and it's amazing how much better my Senns sound when I eq them flat. Tight, deep bass helps me get deeper into recordings. There's a veil in the midrange, and gerG's eq settings fix that, plus they fix the highs. All in all, awesome stuff.

 I can't wait to try the Grado SR-325s with the DEQ -- if the results are as good as they've been with the HD-600s and Ety 4Ps, I'll probably end up keeping them! (I was *really* about to dump those cans, and now they have a shot at redeeming themselves.)

 Everybody who's reading this post, you owe it to yourself to try some good eq. And with a good return policy, it might not even damage your wallet. But I can't promise that, because you'll probably give the DEQ a permanent place in your system, like I have. And with the amount of gear I'm going through right now, that's a pretty big compliment.


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## ooheadsoo

Great, gerG! Now I'm conflicted over whether to get a tube preamp first and get rid of my passive preamp or to get this DEQ! 
	

	
	
		
		

		
		
	


	




 Why is my life this difficult?!


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## Steve999

Wow, I'm glad I opted for the DEQ2496 instead of the 8024!

 gerG, have you explicitly posted your graphic EQ 1/3 octave settings for the HD580s (or HD600s), or would one be left just to infer from the graphs you have posted what the settings are?

 I agree, the interface of the DEQ2496 is incredibly intuitive. You just have to get your hands on it to understand how brilliantly the controls are laid out.

 After the kids went to sleep, I played with the DEQ2496 for about three hours straight last night. I ran opticals in and out in addition to the analog cables and used it both ways. I spent a good amount of time fiddling with my HD580s. It's amazing, after a while, I a/b'ed the results using the "bypass" feature, and I liked the sound of the EQ tweaking so much better, it was like, who would listen to THAT, when they could listen to THIS? But I guess that's inevitable when you are suiting to taste with really the only obstacle being the learning curve of understanding how to use an incredibly flexible equalizer with infinite possibilities at your disposal, and what the different frequency ranges do to the sound. It's a fascinating learning experience.

 I also value highly what I use as the "crossfeed" (variable 10-step image compression that seems to be tonally neutral) too, I am finding it very rewarding. I notice if you want to get really hyper about it, this feature allows you to treat the bass frequencies differently than the upper frequencies in a second menu, because of differences in imaging of bass frequencies. I haven't gotten to that one yet, I kind of like it without messing with the bass management feature anyway, I think, and I'm not sure if the bass settings are meant for use with image compression as much as for use with the image expansion features.

  Quote:


 _Originally posted by gerG _
*Yeeks, I forgot about this thread!

 Steve, I have an 8024 at work, and it simply cannot compare with the 2496, either in sound quality, or in user friendliness. The user interface on the 2496 is almost like something Apple would come up with, it is just plain telepathic.

 By way of update, I consider the DEQ2496 my only really essential piece of headphone gear. The differences between sources, amps, and cables, and even headphones, becomes a secondary issue with this thing around...

 gerG *


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## gerG

Steve and Gradofan, I am glad you guys are having fun. It really is a high value component, considering that it can make changes anywhere from subtle to gonzo. Unlike most components, if you don't like a particular sonic character, you are not stuck with it.

 Honestly I had sort of given up on advertising equalization as a solution. It wasn't catching on, and it gets really annoying hearing someone singing a one note tune. However, now that you guys have heard the light, I will post some curves. I also have a few guidelines that I have worked up. The main one is to keep the curve smooth, and avoid large steps band to band. Most good headphones have reasonably smooth response, and the eq curve should reflect that.

 ooheadsoo, sorry to add to your wallet's misery. Talk with gradofan about the merits of tubes vs eq.

 This is going to make for some fun discussions!


 gerG


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## Steve999

gerG, I sort of stumbled on that main guideline of yours last night, I created bass and treble controls with the parametric shelving, and two broad (I think the setting was "2" or "3" octaves) symmetrical parametric bands centered at about 1 khz and at about 250 hertz, to smoothly heighten or lessen the midrange or midbass. These four adjustable parametric curves make only broad, smooth adjustments and seemed suited to improve the sound of most any headphone in my collection. I just used them like four tone controls. Obviously, the possibilities are limitless, so I just sort of had to jump in.

 Anyway, I would be keenly interested in your general guidelines and in as specific settings as possible that you use for your HD580s or HD600s, as the case may be (I think this is probably the only headphone we have in common).

 Edit: gerG, I also have the V6s, as you apparently do. Gradofan, I have the SR60s if you want to compare notes on Grado EQ'ing.
	

	
	
		
		

		
		
	


	




 Double edit: I believe I have read in Bangraman's posts that he has and frequently uses a DEQ2496, and WmAx has recommended the DEQ2496 so he may have one as well (I came across their posts while I was researching whether to get one)...Their posts were in the headphones forum. They might not hang out in this part of Head-fi much and might not know about this thread. Perhaps we should invite them to party?

  Quote:


 _Originally posted by gerG _
*
 Honestly I had sort of given up on advertising equalization as a solution. It wasn't catching on, and it gets really annoying hearing someone singing a one note tune. However, now that you guys have heard the light, I will post some curves. I also have a few guidelines that I have worked up. The main one is to keep the curve smooth, and avoid large steps band to band. Most good headphones have reasonably smooth response, and the eq curve should reflect that...

 This is going to make for some fun discussions!


 gerG *


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## Steve999

Just an update --

 After playing with it some more, the "crossfeed" (image compression) on the DEQ2496 actually is very good and very flexible. You get eleven steps from no crossfeed to full mono. There's a bass setting, in case the imaging changes affect the perceived bass, of plus or minus 3 decibels. There's also another slider called "shuffle", that "intensifies" the image on a continuous scale, so you can have your crossfeed perfectly dry and pure or a little juiced up. There's also a slider with which you choose the frequency below which the changes in the bass take place. The manual recommends 600 to 700 hertz, but you can pretty much choose any frequency. Of course, this was all designed with speakers in mind, but it seems to work quite well as a headphone crossfeed.


 Also, I jotted down some settings I like for my HD580. I've been using the parametric 10-band EQ because I find it easier to work with than the 30-band graphic EQ. So for my HD580s here's what I have, in frequency-gain-bw/oct format:

 FREQ GAIN BW/OCT
 44.8 : +3.0 : L 6db (increases bass progressively below 44.8 hertz)
 251 : -3.5 : 3/2
 1261 : +3.0 : 3
 6036 : +2.5 : H 12db (increases treble progressively above 6036 hertz)
 6039 : +1 : 1

 This was done by ear and I'm sure there's room for improvement. Any comments, suggestions, etc., are welcome.


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## Orpheus

about this "crossfeed"... i would like to note, if it's a feature on your EQ piece, it's not the same as your headphone crossfeed. there is no signal being delayed to the other side, to give you a sense of depth. it is merely squashing the stereo image. so, essentially it's just putting things even close to the center of your head...


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## Steve999

About the delay, that may be true, I've thought about that, probably way too much. Here's what I think. We just don't know exactly what it's doing. It has image expansion, too, it's all part of the same continuous "imaging" adjustments. Fact is, I'm more pleased with the DEQ2496 crossfeed than with my Corda HA-1 crossfeed, which I also like very much, mainly because the DEQ2496 features are so adjustable. OTOH, it could be a form of infatuation with my new gear. The HA-1 crossfeed is still ready and able and available.
	

	
	
		
		

		
		
	


	




 With the DEQ2496 "crossfeed," you can optionally adjust the bass to be treated differently than the rest of the spectrum (I believe it creates relatively less or more crossfeed [depending on the other settings] in the bass to maintain a believable perception of the bass), and you can adjust the bass plus or minus one to three decibels depending on how the change in imaging affects your perception of the bass, and select the frequency below which you want these changes to occur. This is a problem better overcome by the DEQ2496 than by either the Meier crossfeed (no bass tweaking so bass can seem a little thin when it's crossfed, but the sound is objectively neutral and subjectively pristine) or the Headroom crossfeed (not adjustable and considerably too much extra bass and rolled off treble for my taste). 

 As to whether the image compression includes any delay, I just don't know -- certainly the unit has the ability to delay a signal, as demonstrated by other features. Further, with the image expansion, there is clearly more going on than just expanding the stereo separation -- this would be obvious to you if you gave it a listen. In fact, if you take the image expansion out past a certain level, you can hear some crossfeed being introduced even as the perceived image becomes wider and wider. Some people might like this, for me it's a bit too dramatic with headphones. So I really don't know what exactly goes on. Additionally, who knows exactly what the "shuffle" (what a weird name) feature does -- whether it includes adding a delay is anybody's guess. The manual says it intensifies the imaging, which might very well include some cross-channel delay to add a sense of depth, no? The shuffle feature is continuously adjustable in intensity, if I remember correctly.

 The huge saving grace is it's very flexible and all in the digital domain, suit to taste or don't use it at all, using your ears. Hearing is believing. You can make a very subtle adjustment and it is an improvement to me, or you can make dramatic, even silly, changes.

 All this to say, you could be right. It's something I've actually thought about and tried to figure out. Thanks for the feedback. Fun stuff. 
	

	
	
		
		

		
		
	


	




  Quote:


 _Originally posted by Orpheus _
*about this "crossfeed"... i would like to note, if it's a feature on your EQ piece, it's not the same as your headphone crossfeed. there is no signal being delayed to the other side, to give you a sense of depth. it is merely squashing the stereo image. so, essentially it's just putting things even close to the center of your head... *


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## Orpheus

Quote:


 We just don't know exactly what it's doing. 
 

 i can almost guarantee you it doesn't delay. it's a stereo compression processor that's designed for live use probably. it has nothing to do with headphone use.

 so, no, it's not a cross-feed, not the headphone type anyway.

 a headphone crossfeed is designed to move the imaging ahead of you, so you don't get the "in your head" feeling as much. this doesn't do that at all, or shouldn't anyway.

 there may be instances when setting up a PA system that you might want to push the stereo image in a bit, and that's probably what this function was designed for.

 you can get the same effect on your mixer by having the panning pots turned a bit in from what would otherwise be hard-left and hard-right.


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## Steve999

Stereo separation is dramatically exaggerated with headphones. This can get especially tiresome in recordings that already exaggerate separation. Unless you've had a few beers or whatnot. Then it's pretty cool.
	

	
	
		
		

		
		
	


	




 In real life, and with speakers, nearly all of what goes to one ear, also gets heard by the other ear. The complete opposite occurs with headphones, the ears share no information. Thus, to me at least, a little left to right crossfeed makes things sound a lot more natural and is a lot more important than a very small delay. (A larger delay would sound goofy and would be highly unnatural -- sound travels very fast, as I'm sure you know, and the distance from one ear to another ear is not very far at all.) 

 Personally, I'm not too concerned about delay. A little delay can make the crossfeed less dry, IMHO (and Meier executes it beautifully and tastefully) but not much more. Meier also alters the frequency response of the cross-fed signal just a little, again, beautifully done, IMHO. The out-of-the head stuff seems a little far-fetched to me. That's more the realm of much more complex signal processing, don't you think? But we are both are admirers of the Meier crossfeed so there's not much to argue over in the end.

 Of course the equalizer image compression is meant for speakers rather than headphones. But it's so flexible I am trying it out and I like it. I'll play with it some more tonight. The DEQ2496 lends itself to a myriad of unintended uses because it is so flexible. 

 All this to say, I think you are probably right that there is no delay. Especially when you compress it all the way to mono.
	

	
	
		
		

		
		
	


	




 You are probably entirely correct in your assertions. I'll check it out again tonight. 
	

	
	
		
		

		
		
	


	




  Quote:


 _Originally posted by Orpheus _
*i can almost guarantee you it doesn't delay. it's a stereo compression processor that's designed for live use probably. it has nothing to do with headphone use.

 so, no, it's not a cross-feed, not the headphone type anyway.

 a headphone crossfeed is designed to move the imaging ahead of you, so you don't get the "in your head" feeling as much. this doesn't do that at all, or shouldn't anyway.

 there may be instances when setting up a PA system that you might want to push the stereo image in a bit, and that's probably what this function was designed for.

 you can get the same effect on your mixer by having the panning pots turned a bit in from what would otherwise be hard-left and hard-right. *


----------



## Orpheus

you ever try the meier cross feed or even the pinkfloyd version?--or the headroom version? i have the meier feed here, and it does pop that image in front. works well. the pinkfloyd one is very subtle, and you almost don't hear the difference, which is good for long-term listening. the headroom version is not as dramatic as the meier version, at least the one i heard, but it works too. you might want to check those out if you're getting good use out of the built in "crossfeed" on your EQ.

 but yes, like you said, stereo separation is exaggerated when comparing to loudspeakers. though there used be a processor called "Q-Sound?" something like that.... it's incredible when you use it with your louspeaker monitors. wonder why it never got big... i once mixed a song i wrote with q-sound (if that's what it's called), and it's really cool hearing different sounds coming from outside the area of the speakers. really neat.


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## Steve999

I've used the Meier crossfeed on my Corda HA-1 daily for over a year now. I don't perceive the image outside of my head, but I do find it pleasing. It may well be we just hear differently or we're arguing over semantics, no biggie. I like it a lot better than the Headroom crossfeed (which I've spent time with at head-fi meets) because the it sounds to me like the headroom crossfeed messes with the frequency respone too much for my taste, and I can't get past that. 

 I messed with the DEQ2496 image compression again. It does give you the "shuffle" thing which does something or other to the bass. The manual says it intensifies the image of the bass frequencies (meant for use with speakers of course). Ironically, it seems to attennuate the bass a bit as it intensifies the image, thus the slider to adjust the bass up or down by one to three decibels, I suppose. It's a little over my head as to how or why all this is done. But it's fun to try it out. It's adjustable by .10 increments from 0 to 3. Let me know if you ever try it. BTW, IIRC, with the DEQ2496 you can also rotate the image plus or minus 45 degrees and shift the image to the left or right 180 degrees with two separate sliders (it's two separate effects), silly with headphones but fun for kicks anyway.

 I have one of the three DEQ2496 outputs going to my JVC minisystem, maybe I'll mess with the speaker effects with the JVC speakers and see what happens. The other two outputs go to my minidisc deck and my Corda HA-1.

 Q-sound sounds interesting, I'll keep my eyes open.
	

	
	
		
		

		
			





 The DEQ2496 has image "expansion" ranging from 1 to 3 in .10 increments, it'd be interesting, maybe I'll have to take it downstairs and try it all on a nice stereo setup. From the manual I think it's supposed to sound like it's coming from beyond the speakers, I doubt it's as striking as Q-sound though.

  Quote:


 _Originally posted by Orpheus _
*you ever try the meier cross feed or even the pinkfloyd version?--or the headroom version? i have the meier feed here, and it does pop that image in front. works well. the pinkfloyd one is very subtle, and you almost don't hear the difference, which is good for long-term listening... 

 but yes, like you said, stereo separation is exaggerated when comparing to loudspeakers. though there used be a processor called "Q-Sound?" something like that.... it's incredible when you use it with your louspeaker monitors... *


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## Orpheus

Quote:


 The DEQ2496 has image "expansion" ranging from 1 to 3 in .10 increments, it'd be interesting, maybe I'll have to take it downstairs and try it all on a nice stereo setup. From the manual I think it's supposed to sound like it's coming from beyond the speakers, I doubt it's as striking as Q-sound though. 
 

 actually, this should be the opposite of your "crossfeed" feature. it will pan sounds farther out to the sides, essentially making your loudspeakers sound more like headphones (which you noted exaggerates the stereo distance.) it does not place any sound outside the normal stereo field. Q-Sound however is a processor that is created to place sounds outside the stereo field, which would not be possible with conventional setups. this is done by some DSP which also does some crossfeed-delay-like type processing. and it works REALLY well. unfortunately, i think the company went out of business or something, cause i haven't seen the feature in the past couple years.


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## Steve999

Well I got curious so I did the mandatory google search. Looks like they are still around and have gotten heavily into computer stuff and have branched out a little. I fished around on their website, is this what you were talking about? Maybe they used to make more involved home stereo components?

Q-Sound UltraQ, only 30 bucks 

 They appear to have much more involved and complex software and computer equipment, BTW.

 That's the only non-PC audio equipment I could find on their web site.
	

	
	
		
		

		
		
	


	




 Hmmm.... just to create a facade of keeping this on topic.... I'm waiting anxiously for gerG's HD600 DEQ2496 settings. Sincerely, I am.
	

	
	
		
		

		
		
	


	




 I spent another two or three hours on the DEQ2496 last night. I've got my revised headphone rig all set up now. It goes from a Sony DVD player (for CDs) and a Sony minidisc deck and XM radio and my JVC minisystem (for AM/FM radio) and an input for portables to the DEQ2496 (I use the DEQ2496 optical input for the Sony DVD player, and I have a 5-way switchbox for the analog inputs)) with the main analog output of the DEQ2496 going to my Corda HA-1. The DEQ2496 optical output goes to my minidisc deck so I can record anything on the system, and the DEQ2496 auxillary analog output goes to the JVC minisystem so I can take off the headphones now and then. Sweet.

  Quote:


 _Originally posted by Orpheus _
*Q-Sound however is a processor that is created to place sounds outside the stereo field, which would not be possible with conventional setups. this is done by some DSP which also does some crossfeed-delay-like type processing. and it works REALLY well. unfortunately, i think the company went out of business or something, cause i haven't seen the feature in the past couple years. *


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## gerG

I get a day off tomorrow, so I will tweak some curves. Steve, you need a microphone now 
	

	
	
		
		

		
		
	


	




 btw, if you are listening to XM and have noticed that nasty HF temporal distortion, throw in a narrow parametric of -12 db at 8.5 khz, and another of -6db at 6.5 khz. It cleans things up a lot. This is the reason that the 8024 migrated to my office, so that I could listen to Luna all day on my xm pcr 
	

	
	
		
		

		
		
	


	





 gerG


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## Steve999

Thanks, I'm gonna try it! 
	

	
	
		
		

		
		
	


	




  Quote:


 _Originally posted by gerG _
*btw, if you are listening to XM and have noticed that nasty HF temporal distortion, throw in a narrow parametric of -12 db at 8.5 khz, and another of -6db at 6.5 khz. It cleans things up a lot. This is the reason that the 8024 migrated to my office, so that I could listen to Luna all day on my xm pcr 
	

	
	
		
		

		
		
	


	




 gerG *


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## rsaavedra

Reading this thread, this equalizer has caught my interest. I wonder how its DAC+analog output performance is. Let's say imagine you use completely flat equalization bypassing any processing, connect the 963's digital out to it, and then feed your headphone amp with the analog outs of the equalizer, does it sound as good as the 963's DACs? How about comparing with some other high end DACs, maybe the Grace or BelCanto?


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## gerG

My subjective impression was that the DEQ D/A was an improvement over the one in the 963sa, and the Grace was another improvement yet. That is the configuration of the system now. The only drawback is that I can't play the 1 or 2 decent SACDs that I own (out of a couple dozen crappy ones) on that system. I like the sound of my Sony better for SACD anyway, so it works out fine. The other issue is that the DEQ is limited to "only" 96 khz (by 24 bit depth). I can only upsample to 96 khz, but I will take that compromise for now. Some day there will be a faster digital eq, as well as a faster Grace amp around my house. Not sure when, though.

 One experiment that I haven't tried is to run the digital output of my dvda player to the DEQ. Although dvda can theoretically run at 192 khz, I don't think that I have any recordings that take advantage of it. Maybe I will give it a shot this weekend.

 Funny, when you accumulate enough audio gear, it starts to get like a giant box of Lego(tm) blocks. Endless combinations, and endless fun. otoh I have never fried a tweeter in a Lego block 
	

	
	
		
		

		
		
	


	




 gerG


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## Orpheus

you know greg, since you are so fond of EQ's, maybe you should try investing in a mid-end pro or mastering grade EQ. here are some:

 here's one that's really overkill for us (it's mastering grade--$6000.... very interesting too as its function are completely passive, yet it has tubes inside... dunno how that works):




 here's another high-end mastering EQ:




 but both of those are probably way over us. here's a much more down-to-earth but still good EQ (this one i believe is $2000-3000)... they have a mastering grade version too which is more expensive:




 or you can even turn your computer into a processor with plugins:




 so many good choices!!!


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## rsaavedra

Thanks gerG, to say the least it sounds like a component with great value and many hours of play time guaranteed. All these RTA functionalities can take the hobby to a different level for me I think. As far as measurement and analysis, right now I'm only at the Radio Shack SPL meter + Sterophile test CD's level 
	

	
	
		
		

		
		
	


	




 I've thought about getting some good audio analyisis software and just use a normal computer/laptop, but now my next main goal is a good source, so have postponed that. Now this device is yet another temptation.


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## ooheadsoo

What kind of mic do you guys recommend? And I wouldn't need a preamp would I?


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## gerG

Actually the analysis software is very accessible. I think that the full up version of TrueRTA is $100, and it works great.

 Mics are treacherous. The Behringer is only $50, but mine had some issues in the deep bass realm. I bought an Earthworks M30 as a standard so that I can modify something for my needs, then calibrate against it. Unfortunately it is a bit spensive. I want to try the dbx mic, if I can find one. I will let you know how it compares to the M30.

 For response measurements using a computer you will also need a decent mic preamp, and even more critically, a decent A/D converter. The ones in most soundcards are nonlinear as hell. The one in my laptop changes response with level!

 If you just want something to read phones or speakers with the DEQ, the Behringer is not a bad solution. You won't need a preamp for this, but you will need an XLR mic cable.

 Dean, thanks for the audioporn! Are those all digital? I would put the DEQ up against any analog eq at any price. My reasoning is that analog equalization puts a whole ****load of components in the signal path. The digital eq just does some more math with the numbers prior to conversion to a signal. Nothing in the signal path. Not to say that there can't be errors. No algorithm is perfect. It just seems a safer approach to me.

 gerG


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## Steve999

Okay, for the XM radio, I set up two curves, one at 6471 hz, 1/10 octave, minus 6 decibels; and one 8531 hz, 1/10 octave, minus 12 decibels. It did indeed get rid of a swishing type sound in the low treble on my XM radio... is that what it's supposed to do? Things sounded a lot cleaner. But I noticed a reduction in the perceived brightness.

 So...

 I changed the 8531 hz band to -15 decibels, and the 6471 hz band to - 9 decibels, and set up a one-octave band centered at 7096 hertz at +3 decibels. The curves interacting pulled up the 6471 hz and 8531 hz band to -6 and -12 db, respectively, and created a little treble increase centered at 7.25 khz, so that the perceived brightness when I a/b it is about the same, but the sound is cleaner now. 

 Does that make sense? I'm just learning....Lemme know if I'm violating some profound principle of EQ.
	

	
	
		
		

		
		
	


	




 BTW, I was REALLY surprised to see on the DEQ2496 RTA that the XM radio music has content from 20 hz out to 20 khz.

  Quote:


 _Originally posted by gerG _
*btw, if you are listening to XM and have noticed that nasty HF temporal distortion, throw in a narrow parametric of -12 db at 8.5 khz, and another of -6db at 6.5 khz. It cleans things up a lot. This is the reason that the 8024 migrated to my office, so that I could listen to Luna all day on my xm pcr 
	

	
	
		
		

		
		
	


	




 gerG *


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## Orpheus

Quote:


 Dean, thanks for the audioporn! Are those all digital? 
 

 actually, they're all analog. the Manley one is very special... it's actually a purely passive signal path, though it uses tubes. and i have yet to understand how that works.

 well, i guess theoretically once the signal gets into the digital realm, the mathematical part of the EQ should be perfect. so, i guess if the ADC/DAC is perfect, then perhaps the digital EQ you have might outperform a $6000-10000 mastering EQ. but i would like to think there's a reason why we need $10000 EQ's....


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## ooheadsoo

It just occurred to me...how would I hook this up for a 2.1 system?

 Would I run the preout from my pre to the eq, and then use one output from the eq to the speakers and the auxiliary out to the sub? Is there a better way?


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## gerG

ok, found some spare time.

 HD600 eq curve:





 Before and after response (offset 5 db for clarity). Responses are averages of left and right channels:





 HD600 vs HD650 (both unequalized, the 650 is the one with higher response in the deep bass):





 HD650 eq curve:





 Make sure that your eq is in true response mode. You may need to adjust the gain offset (utility menu) down just a few db to stay off the ceiling.

 A word of warning to anyone using the pink noise generator on any of these equalizers: the noise signal goes through the EQ stage as well. This is good for checking effect of adjustments, but can throw off your baseline measurement if you forget to set it flat.

 I will post more curves as I get them generated. My old mic was just a bit off, so I am reworking them all. Next up will be the DT931 and Grado SR325.

 Have fun!

 gerG


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## gerG

ooheadsoo, it depends on your gear. The ideal place for the DEQ would be between the source and the preamp, upstream of the preamp. That allows the DEQ to run at full bit depth. Connecting it downstream will have you treading the line between reduced resolution (not using full bit range) and digital clipping (ugly). Since I use my dbx as a crossover, as well as an equalizer, it has to reside downstream of the preamp. I have to set the unit up to take advantage of the full signal range at my listening levels. Fortunately it has provisions to do that. I haven't found any such option in the Behringer.

 A general note about 2.1 setups: be sure that you have some sort of high pass filter on the satellites. Most of the subs with built in amps have the low pass side covered, but the sats are allowed to run full range, or with a generic passive high pass that may not be at the frequency that you want.


 gerG


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## ooheadsoo

I added a "home made 
	

	
	
		
		

		
			





 " "diy" sub to my system, so the monitors do indeed run full range. I think I may hold off on the eq until I decide if I'm happy enough with my system after I get the preamp I'm commited to getting.

 I'm also committed to using more than one source (pc, cdp for sacd, and radio) so if i want to use the deq with all 3, it pretty much has to be downstream of my preamp. 

 How's the top end sound after you eq'd the senns? I'd always heard that problems in the FR of omission generally sound better than peaks, but I noticed that you eq'd the very top end of the spectrum up a bit.


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## Orpheus

ooheadsoo, i think nhtpro also makes a sub that goes with the a-10. you might want to check that out, since it'll be more tonally similar to your current system. not to mention being matched in power capabilities. plus it'll have all the correct crossover values... not sure if your DIY has the proper crossovers.


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## ooheadsoo

Hey Dean. Well...I'd get the nht sub if I had the dough...you know very well how much it costs! It's the same sub as yours. That's the only one they make. I'm fairly happy with my sub. It cost me a fraction how much the nht sub costs and the amp I bought to go with it has full phase control, so I think I have it matched fairly well. Sounds good to me, at least. I'm running 2 10" peerless $50 drivers in parallel stuck in a 130 liter or so cabinet.

 Nht sells the same sub for both your a-20 and my a-10, so I don't see how it could match perfectly to both our systems since we supposedly have different F3 values. I think your F3 is 45hz and mine is 55hz.

 I'll probably "upgrade" my sub in a few years. Probably peerless xls drivers and a ton of cabinet bracing and stuffing. New amp...you know, the works. I'm really pretty happy with the way it 's blending right now with the a-10's. It's not quite as tight as the bass on my senns, but it's good enough, and better in some ways.

 I'd offer to bring it to the meet coming up, but it's just practically impossible, seeing as the sub cabinet is 3 feet tall and weighs like 100 lbs with the amp.


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## Steve999

THANK YOU, gerG. The DEQ2496 is amazing.

 I'm printing out the HD600 settings and headin' upstairs. Well, maybe after my 3-year-old son quits making believe he is a gopher hiding in the closet.

 I haven't tried the pink noise generator yet... I'll have to give that a shot when I'm not too busy listening to my favorite music with the best sound I've ever heard over a pair of headphones.
	

	
	
		
		

		
		
	


	




 I've found my HD580s (I don't have the HD600s) seem to have the most unlimited post-EQ sound quality potential out of all of my headphones.

 I'll use your SR325 settings as a rough guide for my SR60s.

 What I have been doing is setting up graphic EQ curves particular for individual headphones and separate parametric curves for things like fine smooth improvements to bass, mid-bass, midrange, and treble, for particular recordings, cleaning up the XM radio distortion, and things of that nature.

 Just my own little factoid, if it might be of use to someone, the DEQ2496 main analog output passes the main analog input through even when the power is turned off.... nice when you're just going to go to sleep or something.

 Thanks again. I really appreciate it.


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## Orpheus

Quote:


 Hey Dean. Well...I'd get the nht sub if I had the dough...you know very well how much it costs! It's the same sub as yours. That's the only one they make. 
 

 heh he, fortunately for you, you're wrong. here's the one you want:

http://www.nhtpro.com/2004/products/...=1&ProductID=7

 here's the one i use:

http://www.nhtpro.com/2004/products/...=1&ProductID=8

 notice that yours is 1/3 the asking price.

 start saving. 
	

	
	
		
		

		
		
	


	




 but you know, i dunno if gerG mentioned it before, but both this EQ and his Furman version both have automatic room adjustment functions right?--Greg? ...so, if you got this EQ, it'll let you tailor your setup to be truly flat. so, ooheadsoo, now that you got good monitors... this EQ could bring it to another level of listening experience.


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## ooheadsoo

That sub is probably pretty cool for what it is. It goes with the M00 series satellites. Great names, huh? 
	

	
	
		
		

		
		
	


	




 M00 and S00. It's their "multimedia" speakers that go for about $1000 for the satellites and sub ($250 per satellite.) The sub that you got was designed for both of our monitors 
	

	
	
		
		

		
		
	


	




 My sub goes down to basically 33hz before rolling off, so it's a better performer than the 8" sub that they have for $500! It's also like...7 times bigger 
	

	
	
		
		

		
		
	


	




 I'm guesstimating this number by listening using a tone generator run off my m-audio revo, which admittedly has light bass. It might be better with a better soundcard.

 Ultimately, unless I buy another system when I have my own place for the tv or some nonsense like that, I probably plan on upgrading to your a-20's and maybe going with the b-20 subwoofer too.

 But yeah, the eq definitely sounds like something that would be really cool to play with!


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## Orpheus

oh oops!!!--sorry man, i kept thinking you bought the m00's.... you actually got the a-10. hmm... yeah, k. heh he. sorry, my mistake.

 but yeah, i'd like an eq this... it would help with acoustics.


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## ooheadsoo

Hehe, it's easy to get confused cuz they discontinued the A-10's. My speakers basically look identical to yours except it uses a crappier woofer with foam surrounds and it doesn't have the foam tweeter waveguide. The amp has half the controls too.


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## Steve999

gerG --

 I input your values for the HD600s into the DEQ2496 and gave a listen to my HD580s. I started with some Mozart piano concertos (I'm going through a Mozart piano concerto phase) and my feeling was "whoa." The woodwinds had such a natural sound it was uncanny. The sound overall sounded extremely natural. The sound was obviously much less colored than what I had come up with by ear when I A/Bed the two.

 Next I listened to Joshua Redman's self-titled first CD (an acoustic jazz quartet). For this, your settings were about 2.5 db too bright in the mid-treble for my tastes. Specifically, the cymbals, high hats, etc., were a bit too strong. I took the treble down 2.5 db at 10,000 hertz with my paremetric EQ treble control (a 3-octave/bw control centered at 10,000 hertz has become my treble control of choice) and I had..... perfection (or something very close to it).

 I am as a general rule finding I like a baseline curve for each headphone on the Graphic EQ, with separate additional variable tone controls set up on the parametric EQ, because I am finding it desirable to make obvious (to my ears) adjustments from recording to recording, or based on different types of music, with the parametric EQ.

 Thanks again. This is as good as it gets.


----------



## gerG

Oh man, there is so much fun stuff here that my head is spinning! (More than normal, I mean).

 First, subwoofers. ooheadsoo, diy subs are the way to go. I have lost count of how many I have built over the years, and I design scads of them just for fun. The commercial subs will squeeze the most out of a small enclosure, but if you have the space, you can easily do better for less money. The Peerless XLS is on my to-do list, right after the JBL "Nuclear Sub" project that I am working on. I need something better than the Shoguns (the ones that broke my water lines and shifted one end of my house, do a search). It will be fun, maybe even fatal to any small animals in the area.

 Another option is to connect the DEQ in a tape loop of your preamp. This will allow multiple sources, but will still run the processor at a relatively constant and high level.

 Steve, I am glad that you are tweaking the levels. My curves are only a starting point. Also, I forgot to mention, my HD600 are not stock. They have a modification which has probably changed the response slightly. That will be the subject of another thread.

 I chose to shelve the treble to match the average response of the Sens. All of those major fluctuations are due to interactions with my ears, as well as to the dimensions of the cans themselves. Although I know several of the sources, I don't try to compensate for those effects yet. As an experiment I would excourage equalizer pilots to change the level of the shelf to see what level of detail you like. Steve made a good call using cymbals as a reference.

 gerG


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## ooheadsoo

Thanks for the vote of confidence, gerG! I think my sub sounds pretty decent, it just doesn't extend super low. But it DOES extend as low as I want it to extend. Rythmikaudio.com sells an inexpensive servo amp, perfect for you to try out if you have the inclination to use sealed in a small box. I don't think there's much doubt it'll sound better than a sunfire.

 I'm eager to hear any impressions you may have if you ever get around to trying out your deq with your speaker rigs!


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## gradofan

Thanks for the EQ, gerG!


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## Steve999

gerG --

 Are you posting a Grado curve anytime soon? I would dearly like to have one.

 Your HD600 curve has been just wonderful for my HD580s. After taking down the treble a little, the level of fidelity I am getting is pretty much to the point where I just wouldn't know how to improve it any further. The sound is magnificent. My HD580s are now my most-used headphones by far.

 I've also got my Bose QC2s sounding like real hi-fi (for when this noisy household gets, well, noisy). I'm pretty confident in the curve, I've been tweaking it for weeks, though it's just by ear. If anyone is interested, I'll be happy to post it, but I doubt anyone else has both the QC2s and the Behringer. I'm hesitant to admit I have both, because I don't want to give the Behringer a bad name by association. 
	

	
	
		
		

		
		
	


	




 Well, thanks once again. It really is hard to get the word out about the DEQ2496. I'm just about to give up myself. It's so hard to get across the nature of what you can do with it. I was at a point where my equipment was nice enough so that the stratoshperically expensive stuff was leaving me cold. Now I have a system that is infintely flexible and vastly improved in sound quality. If you've got nice equipment and you want the ultimate headphone system upgrade, this is it, IMHO. And it's FUN! It sure would be nice to have more people to share the experience with. 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *gerG* 
_I will post more curves as I get them generated. My old mic was just a bit off, so I am reworking them all. Next up will be the DT931 and Grado SR325.

 gerG_


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## gerG

Hi Steve. Sorry, I got sidetracked by speaker projects. I think that I can do more with the Grados than just EQ. They have a resonance in the hf range that I think is caused by the reflected wave off the grill. If I can kill that it should eliminate that seashell effect that I always get with Grado headphones. The pads are the other problem. I have to get these things off my ears to clean up the highs, and my only method at the moment is to shim them with a set of Beyer DT931 pads. I will go ahead and test with that, but the curve will require adjusting for your setup.

 Incidentally, I only have Grados around because Gradofan loaned me his to try out. My motivation is to figure out why I don't get along with their products.

 A agree on your assessment of the evolution in performance offered by the digital EQ. I have tried to listen to my other systems, but they all sound either dead, peaky, bassless, or all of the above. I need to punch some of the curves into the dbx so that I can use my Sony player. Unfortunately no upsampled digital output.


 gerG


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## Steve999

Thanks for the reply. I just have a pair of Grado SR60s with comfy pads, so I'm just looking for general guidance, and I can fly by ear from there. 

 The HD600 curve has been so awesome for me with the HD580s that I was anxious to try one of your curves with another headphone. I actually like the SR60s, but I don't doubt your insights one bit, and of course the EQed HD580s smoke the SR60s pretty badly so they are not getting so much use anymore. I guess what you are saying is to the effect that the hardware of the Grados, with its resonances and earpad design and whatnot, is more limiting in terms of what EQ can do for it -- the EQ upside of the HD580s is likely much higher.

 If I recall you have Gradofan's SR325s? I am persuaded by others' posts that those may have a peaky midrange resonance due to the extended length of the chamber as compared to the SR225 and lower models, which don't seem to have the same large midrange resonance. The headroom FR graphs seem to corroborate this pretty strongly. If you have a different analysis that would be interesting. 

 And it's always fund reading about your subwoofers. 
	

	
	
		
		

		
			





  Quote:


  Originally Posted by *gerG* 
_....Incidentally, I only have Grados around because Gradofan loaned me his to try out. My motivation is to figure out why I don't get along with their products....

 I agree on your assessment of the evolution in performance offered by the digital EQ. I have tried to listen to my other systems, but they all sound either dead, peaky, bassless, or all of the above. I need to punch some of the curves into the dbx so that I can use my Sony player. Unfortunately no upsampled digital output.

 gerG_


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## gerG

dig dig dig dig CLANK..... whoa, look at this prehistoric thread!

 Actually this is a serious update. The DEQ 2496 is still the most important component in my headphone system. It is so essential that I bought another one. This one is a permanent resident one of the downstairs systems. It will provide conditioned digital signal (via AES/EBU cables) to the patio, deck, and garage. Source is a Sony 333es. Some day I hope to add upsampling to the mix, but it is pretty good right now. This setup allows me to select a headphone curve, load some CDs, and connect to the other end of the cable with a Grace 901 amp.

 Enough trivia, and on to a very important discovery. I knew that the DEQ would stream all settings as a data block over the midi interface. It made sense that connecting a twin unit via midi and setting it up to receive would make direct transfer of all 20 some EQ curves possible. Amazingly it was even easier than that. I connected the "out" port of the old unit to the "in" port of the new one. In the utilities menu there is a "dump all" function. I pressed that one, took one sip of scotch, and the units were synchronized. It probably helped that the scotch was a single malt from Islay, so be advised.

 Now all I do is choose the cans, select the curve, head outside with Grace and cans, hit play on the RF remote, and watch the sun set to tunes. Not a bad way to end a rough week. Highly recommended.


 gerG

 edit: boooring, needs a picture






 and







 edit: downsized photos again. Sorry about that.


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## Duncan

Wow... I don't know whats better, the setup, or the view...

 Ha! I do really... who needs headphones, with a view like that? nice 
	

	
	
		
		

		
		
	


	




 The trouble for me, with EQ, is that its hard to be consistent, i'm one of these really annoying people that keeps tweaking, and tweaking and tweaking ad infinitum...

 Although, you say that it can adjust itself... hmm... tempting


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## gerG

Hi Duncan. The views are one of the main reasons that I live here. Pictures can't do it justice. Well, my pictures can't anyway.

 The auto eq function does not work well with headphones. I have never gotten good results. The simplest approach is to pick up the Behringer mic, tweak levels for near flat response, and adjust by ear from there. I try for smooth curves, so there are no bands way out of line with their neighbors. That eliminates most of the fiddling right there.

 As for relentless tweaking, well, there is a bit of that, but as you close in on the sound that you are after, it becomes less and less. The one thing that I still do with most of my headphones is listen for a while without the eq, then switch it into the loop. I am looking for a positive effect across the entire spectrum. Headphones do change with time, and this is a way of checking it. I also do the reverse on occasion for the same reason. The improvement with the eq is always amazing.


 gerG


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## kwkarth

Excellent view gerG and excellent info. One of these days, I'm going to have to pick up a 2496 for home. BTW, the view looks familiar. My brother and Dad both live in North Scottsdale off of Pinnacle Peak Rd.
 Cheers!


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## gerG

KW, good to hear from you. Excellent spotting on the view. Those are the McDowells, and North Scottsdale is on the other side. That is my view to the west. Here is what it looks like to the east:






 Or a less gearcentric shot:






 gerG


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## jhenderson010759

I just ordered a DEQ2496. I am planning placing it in the signal path before the Grace901/HD650 (digital) as well as a pair of Stax 4040s (analog). 

 My question is, what is the best procedure to perform headphone equalization? I though I read somewhere that you actually place a small mic next to your ear while listening to the target headphones? Is there actually a suitable mic with an acceptable form-factor for that use?


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## gsferrari

Why do you want to trust a microphone for EQ? I can understand the reason to do it out of academic interest but really - use your ears and set it so that it sounds good and sounds right.

 No offense but you really have to stop looking at the technical aspect of things and get on with enjoying the music...isnt that why we are all here? 
	

	
	
		
		

		
		
	


	




 I have the Behringer too (thanks gerg 
	

	
	
		
		

		
		
	


	




 ) and I love it. Used my ears and use my ears everytime I need to tweak it.

 Fantastic piece of gear!!


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## jhenderson010759

Why measure? 
 "When you can measure what you are speaking about, and express it in numbers, you know something about it; but when you cannot measure it, when you cannot express it in numbers, your knowledge of it is of a meager and unsatisfactory kind..." - Sir William Thompson, Lord Kelvin (1824-1907) 

 "..isn't that why we're all here?" 
 That's not (the only) reason I'm here. I enjoy the tinkering as much as the music. I need my technology "fix". 

 Jim


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## Jahn

I'm poor so i grabbed a BSR EQ3000 for fun on the side - do you think this will do the trick? i hope it wont degrade the signal too much when i put it in the chain. it has a pink noise filter and a spectrum analyzer - eye and ear candy!


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## gsferrari

How does it sound? Got a link to this piece of gear?

 Cheers and welcome to the EQ group. This is the honest to goodness truth -

 Spend a LOT of money buying expensive headphone cables, speaker cables, interconnects, amplifiers, headphones, power cables, power conditioners, damping blah blah blah - frustrating, expensive and you will notice that you are enjoying the music less and less.

 Instead - get a good digital EQ like the behringer deq2496 and forget about everything else. Just nail an amp you like, headphones that you like, source that you like and put them all around the EQ.

 The EQ has become the centerpiece of my setup. Nothing is changed...just minor tweaks to the EQ settings or selecting from several presets that I saved in memory according to the CD, Headphone, Amp 
	

	
	
		
		

		
		
	


	




 This is the way I want my music to be


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## Jahn

hmm cant find much about it - here's a reference to it, it seems to still be in use in someone's rig so take that for what it's worth -

http://www.audioholics.com/forums/sh...7190#post17190

 i got it for $37 shipped last week, hopefully it will come tomorrow. we'll see how it sounds!


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## gerG

Hey Jim, welcome to team DEQ. I have to start with measurements. I have a very unreliable component in my listening chain (my brain) so measurements are the only way of getting consistent and reliable results. There are details of that process farther up in this thread, as well as that thread 

 I would love to get into a discussion on the topic, but I am cramped for time at the moment (in the middle of collecting engine data and trying to figure out What is going on in there). I will be back when I get some time (and sleep).


 gerG


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## mkmelt

Regarding the BSR 3000 equalizer, I recently picked one up at Goodwill for $20. I did some research and found it was a decent 10 band equalizer that provides for +/- 15db at discrete frequencies, with the added benefit of having a built-in spectrum analyzer and pink noise generator. The pink noise generator can be used with the factory-supplied microphone * for positioning loudspeakers in a room for flattest bass response, equalizing each channel at a given listening position for flattest response, or even setting the recording bias adjustment on a cassette deck.

 The BSR 3000 was also sold as the ADC model 315X after BSR was merged with ADC, or so I have read.

 The BSR 3000 originally had a suggested retail price of $349, but I have an invoice for a second one I purchased off of eBay that shows that it was sold by DAK for $149. Anyon remember this mail order house that would advertise in audio and electronics magazines with closeout specials on electronics and other toys?

 To my ear the BSR 3000 is a very clean sounding unit, so I was not surprised to read the specifications that it has a wide bandwith of 5 ~ 100,000 Hz. When it is switched out of the EQ mode, I can not detect that it is in the circuit of my tape monitor on my amplifier.

 A couple of words of caution: First, the owner's manual warns against accidentally plugging in a headphone into the front panel microphone jack.

 * Second, the only microphone that is safe to use with the BSR 3000 is the calibrated electret condenser microphone that came with the unit. Do not attempt to use a regular dynamic microphone with this equalizer. If you do, the owner's manual states that unit can be damaged.

 Prices are all over the place for the BSR 3000. I paid $20 for a well used but working example at Goodwill, and $35 plus shipping for a very clean unit including the manual and the original microphone. I have seen these go on eBay for as little as $25 and as high as $155, so don't be fooled into paying too much.


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## Jahn

Hmm not a good sign when it ends up being found on a vintage site -

 http://www.oaktreeent.com/Stereo_EQ'S_And_Signal_Processors.htm


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## gsferrari

I dont know if there is a way to make sure that left and right get the same EQ'd data...I can do separate L/R and combined L/R EQing with the Behringer.

 This should not be a MAJOR issue with speakers but with headphones it can be frustrating to set each channel EQ just right...maybe even impossible.

 For whatever it is worth $40.00 is not a big hit and I am being optimistic about the chances that this could do well in your setup. there is no reason why it should sound bad...balancing is the key point to take into consideration here...channel balancing.


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## mkmelt

Do you really think your ears have identical sensitivity at all frequencies? This is highly unlikely.


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## gsferrari

Quote:


  Originally Posted by *mkmelt* 
_Do you really think your ears have identical sensitivity at all frequencies? This is highly unlikely._

 


 Exactly - which is why I am not so sure using the autoEQ is a good idea. better to use your ears right? What they can hear means more than what a microphone can hear.

 But if you are saying that the Stereo EQ is a good idea because each ear might be different in sensitivity - THEN I AGREE...

 Never thought about that...It would frustrate me psychologically to know that the EQ is pumping out different signals on each channel...
	

	
	
		
		

		
			





 I like everything to be perfect.

 Cheers!


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## jhenderson010759

Even assuming an identical signal is sent to both speakers and the reproduction capabilities of each speaker is also identical, each would almost certainly need to synthesize a different output in order to properly reproduce that signal at your listening station. This is because the overall transfer function of each speaker is influenced by it's environment. For example a drapery may be nearer to one speaker than the other. 

 So, you should recoil in psychological horror at the prospect of _identical_ signal processing from both equalizer channels. 

 OTOH, this effect should be substantially mitigated for headphones (except for Van Gough).


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## jhenderson010759

Two questions: 
 1) Which PC-based MIDI board(s) have been successfully used to interface with the Behringer DEQ 2496? Is software interoperability an issue with MIDI boards, or will any run-of-the-mill midi board do the trick? I'd like to upgrade the firmware on my DEQ and be able to transceive configuration info and equalization settings. 

 2) I'd like to connect the balanced differential analog outputs of the DEQ directly to my Stax 006t amp. However, I noticed that Stax and Behringer wire the differential hot and cold signals opposite polarity from one another (Behringer wires 2 hot, Stax 3 hot). If a stock, straight-through XLR cable is used, this would result in phase reversal (both channels). Is it worthwhile to rewire the cable/make one from scratch, or is this sonically insignificant? Seems like it wouldn't be a problem, since both channels would be phase-inverted, thus it would not be analogous to a loudspeaker phase incoherancy (which is readily discernable). 

 Thanks in advance.


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## ooheadsoo

I think the jury is out on that with re: to absolute phase because it's said that nearly all recordings are phase random. That said, some people say they can hear the difference. Go figure.


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## ooheadsoo

Quote:


  Originally Posted by *mkmelt* 
_Do you really think your ears have identical sensitivity at all frequencies? This is highly unlikely._

 

gsferrari, mkmelt's point is that the ear hears differently at each frequency, not from left ear to right ear. Fletcher-Munson curves describe an average of how average humans hear at different levels. For example, in practice, if I EQ the bass of my speakers to be the same level as the mids of my speakers _using my ears_ and I play some music, the bass sounds way out of proportion, much louder than it should be. This is because the ear is less sensitive to bass. The sensitivity of the human ear changes at different volume levels as well. At low volumes, bass is even harder to hear compared to mids and highs, than at loud volumes, etc. A decent microphone with decent FR would be better in this case.


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## gerG

I have used an M-audio firewire interface, as well as a basic Edirol usb midi interface. I really like the latter solution because it is cheap, has built in midi cables, and is highly portable (fits in a back pocket). I use midi-ox as the interface software. The only real issue is getting the speed low enough for the Behringer to cope, since the PC will dump data too fast. The DEQ to DEQ direct trick works incredibly well, making it a snap to synchronize units.

 I would not worry about absolute phase too much. Too many things in the system reverse the phase to keep track of it (even which way they put the drivers in the headphones). Since they are AC signals, + is only a phase reference, and it may have been reversed by the time it gets to the terminals anyway. My usual trick in a dynamic system is to feed a DC signal into the input, and see that the driver steps in the positive direction. Although this works like a charm on speakers and dynamic cans, I would not try it on electrostatics, unless you have a laser interferometer handy. If you do have a LI handy, then you have the complication of the signal, which we would prefer to keep digital. I would suggest generating a timed square wave of 1 second duration stepped from zero to positive with an amplitude well under digital clipping. If you don't have a LI, go with soldering up both sets of cables and see if you can hear the difference. Alternatively you can build a box with a phase reverse switch. Put one on each channel and you can even see what they sound like with a side to side phase mismatch (it is nothing like speakers).

 I am not going into ears not matching (one of mine is lower than the other, but fortunately my brain is in crooked, so it all works out). It does not matter for my purposes. I am using the DEQ to compensate for defficiencies in the headphones, not in my ears.


 gerG


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## Jahn

Thanks for the history on my equalizer by the way! Hmm, it can be used to even out the bias in a tape deck - i wonder if it can do the same for my turntable, which might pick up some odd stuff from its needle. Well in that case I won't have to get a mic for it, since i dont have speakers - I'll just use the pink noise to help even out the turntable signal.

 good call on the mic - no dynamic mics eh? just a condenser. i wonder if a non-factory condenser mic can be used for recording purposes through the equalizer - not just to pick up signals for pink noise use.


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## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_ <excerpt>... If you do have a LI handy, then you have the complication of the signal, which we would prefer to keep digital. I would suggest generating a timed square wave of 1 second duration stepped from zero to positive with an amplitude well under digital clipping.
 gerG_

 

I guess I'll compromise and use some stock XLRs for now. But, I'll ask my wife for a laser interferometer for Christmas, so that I can resolve this issue once-and-for-all (www.lasermotion.com appears to have some good deals on used equipment). I'm sure one of the kids would be willing to sleep under the Mohave granite optical isolation table, instead of their bed, as their bedroom morphs into a suitable laboratory annex 
	

	
	
		
		

		
		
	


	




.


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## mkmelt

[I wonder if it can do the same for my turntable, which might pick up some odd stuff from its needle. I'll just use the pink noise to help even out the turntable signal.]

 If you had a stereo test record that included a band of pink noise, you could see the playback response, but the resolution is fairly coarse because there are just 10 center frequency indicator bands for the entire audio spectrum, and the smallest increase or decrease that can be displayed is approx. 2db. 

 [good call on the mic - no dynamic mics eh? just a condenser.]

 Here are the original microphone specs:
 Element type: Electret condenser
 Directivity: Omnidirectional
 Impedance at 1Khz: 600 ohm
 Sensitivity 0db = 1V/microbar, -70db
 Frequency Response, Compensated: 50 - 13,000Hz
 Bias: 1.5V DC supplied by EQ 3000

 Note: Connect the supplied microphone only, use of other microphones, i.e. dynamic type etc., will damage your system.

 [I wonder if a non-factory condenser mic can be used for recording purposes through the equalizer - not just to pick up signals for pink noise use.]

 The mic is not stereo, but I don't see why you could not feed the output from the mic to a tape deck via your amplifier.


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## gerG

Jim, lol. Those granite tables are awfully cold. Do not annoy your wife.

 Another thought, you could do the same square wave trick, but use a mic to pick up the pressure offset (with headphone coupled to something to seal the cup). Of course, now you have to figure out if your mic/preamp/AtoD invert the phase. For something that seems so simple at first, headphones can sure get complicated. Better to follow the advice of the guru and just enjoy the music. While you do that I will think up another absurd way of figuring out phase on ES cans 
	

	
	
		
		

		
		
	


	




 Jahn, good luck with your new toy. I haven't joined in the discussion because I am not familiar with that unit, so I have nothing useful to contribute (although that does not usually slow me down). You may also want to experiment with some of the PC based equalizers out there.

 gerG


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## ooheadsoo

Hm, what good PC equalizers are there out there? If any are free, I'd like to check some out.


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## cosmopragma

Hm, what good PC equalizers are there out there?
 It doesn't have to be free if it's really good.
 My rack is already congested, and the Behringer is so damned ugly.


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## tomek

hey guys, after reading the article on enjoythemusic.com regarding the huge fluctuations speakers have in their actual listening environment, i'm tempted to pick up the behringer unit myself.

 my question: would i only need to have a mic in my listening position and use the auto eq function? would this be an adequate solution for correcting for my room? has anyone done this? has it provided a real change in the FR of the speakers for the better?

 after reading the article, it seems like listening to speakers without correction for the room is *really * underutilizing their potential.


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## cosmopragma

Bump and a chance for the west coast guys and fellows down under to share your knowledge about audiophile grade PC based equalizers.
 I'd prefer software, no matter what it costs.
 I don't need another ugly, power consuming box.


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## ooheadsoo

Nope, eq is definitely not the best solution. It doesn't treat the root of the problem, which is the room. Bass traps and other acoustic treatments are the best way. EQ at best only treats the listening location. Fluctuations caused by the room can vary by the inch. Or less.


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## Flea Bag

gsferrari, I've seen you take a stand for equilizers in other threads before. One of such of your posts have stayed stuck in my head and with the help of the rest the characters here, gerG included, I'll be hoping to buy the DEQ2496 when I return to Singapore this Saturday. This component had better save me lots of money in future! 
	

	
	
		
		

		
		
	


	




 I spent 3 hours reading through the instruction manual this morning and I'm glad I understand most of it(I hope), but what should I look out for before buying?

 I have a cheap DVD player for the time being with a digital optical out. I'll have to check what sampling rate it does but I know it should be either 44.1, 48, 88.2 and 96 kHz or the digital in won't work and I might have to use analog in. However, is there anything else I should look out for? Is the rest just plug and play? 

 My main aim at this time is to enable good low-volume listening with the HD600 and I hope the dynamic equalizer function will help that!
 Thanks for your time!


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## gerG

fleabag, you read the manual??? Heck, I haven't even done that. I am impressed. You should have no problem with sampling rate. I have plugged into various players, and the DEQ just resets itself. Interestingly, the digital output sample rate always equals the input rate. That makes me suspect that it operates at whatever the source frequency is all the way through (no sample rate conversion going on).

 Tomek, I agree with ooheadsoo on the best treatment for room acoustics. Use of acoustic treatment and careful placement is more effective that equalization. The problem with eq is that room response variations are caused by unwanted reflections. By simply adjusting the amplitude you will get the level more uniform, but the level of reflection relative to source (speaker) will remain the same. This plays hell with image and detail.

 HOWEVER, having a mic and rta is extremely useful in helping resolve the room issues. The eq is also useful for broad level matching. For example I have a system with a broad sag in the response from 2 Khz to 7 Khz. I use a wide parametric centered at 5 K to pull it up a few db. It is also safe to eq low frequency peaks. It is not good to try to fix lf valleys (a very strong null may require HUGE amounts of boost, straining drivers and amps, and creating problems elsewhere in the room). Dr. Toole has a good white paper on the subject over at the Infinity website.

 Sorry, I tend to get verbose on this stuff. Bottom line: I think that a DEQ with mic would be a very valuable tool in setting up a speaker system in a room. It has certainly been an ear opener for me 
	

	
	
		
		

		
		
	


	




 I am in the process of setting up a room and speakers. I stumbled across something seemingly remarkable. Sort of a new twist on an old concept (think Roy Allison). It eliminated a lot of room acoustic issues. I hope to write it up this weekend.


 gerG


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## ooheadsoo

Hey gerG, is that dip you're talking about from 2khz to 7khz room related or just a problem with the speaker design? The frequency range you listed makes it look like the tweeter choice wasn't exactly ideal for the woofer and it's a xover/design problem of sorts. Or it could be a diffraction problem I guess since you're talking about an in room measurement, but those usually aren't so broad in range.


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## Flea Bag

Quote:


  Originally Posted by *gerG* 
_fleabag, you read the manual??? Heck, I haven't even done that. I am impressed. You should have no problem with sampling rate. I have plugged into various players, and the DEQ just resets itself. Interestingly, the digital output sample rate always equals the input rate. That makes me suspect that it operates at whatever the source frequency is all the way through (no sample rate conversion going on).

 gerG_

 

Thanks! Regarding sample rate adjustment, on page 13, column 2 of the instruction manual (section 3.6, I/O menu); I came across the wording in bold:

 'If the connected sample rate does not correspond to the rate adjusted on the DEQ2496, this field shows the message UNLOCKED, which mutes all outputs of the DEQ2496.'

 This message was a bit misleading and worried me a bit but then I read more carefully and I read this in the earlier parts of the same section:

 'When the digital input(source) has been selected, the sample rate cannot be changed, because the DEQ2496 tracks the sample rate contained in the input signal.'

 Phew... I now hope to get my hands on the 2496 really soon! 
	

	
	
		
		

		
		
	


	




 However, could I just ask what 'Pink Noise generation' is?


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## gerG

fleabag, pink noise is synthesized broadband noise that has equal energy per octave. As such it best represents how we perceive sound, as well as how most RTAs (real time analyzers) acquire data. A system yielding perfect reproduction will measure flat across the spectrum using an RTA with pink noise. If you are using an FFT rig, you probably need "white noise" which is equal energy per frequency interval. The noise part refers to the fact that it is a random blend of sine waves, without any stationary frequencies. One can also measure room behaviors using sine wave sweeps (can be misleading) or steady state sine waves (tedious, but best for nailing down killer resonances).

 ooheadsoo, I haven't tracked it down yet. The speakers are specified at +/- 1.5 db from 100 hz to beyond 20 Khz. It is Infinity's flagship, so I don't think that they exagerated too much. Nevertheless, I will have them outdoors for a listen in the real listening room, and I will put a mic on them at that time.

 I am a bit ashamed to admit that I actually bought speakers, but I don't have time to build my next ones just now (although the design is about half complete). At least the new subs are DIY, and man can they do the job!


 gerG


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## Flea Bag

Thanks for the time and effort gerG,

 On a different matter, my desired setup seems to be very similar to yours (963sa upsampled to digital 96kHz to Behringer to an amp (which I'm not sure which one to upgrade to yet) and then to HD600 with Oehlbach cable! Seems like you've got a good cost-effective rig!


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## ooheadsoo

Your subs make me drool 
	

	
	
		
		

		
		
	


	




 I would be really interested to know if the Infinity flagship has big xover related dips in it or not 
	

	
	
		
		

		
		
	


	




 You never know, if they went for a first order xover on it, maybe they had reason to not care about having it dip.


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## gerG

Actually they are supposed to be fourth order crossovers, set at 300 hz and 2 Khz. The dip would be solidly in the tweeter range. Unfortunately there are no provisions for bi or triamp, so I can't try an active xover.

 flea, I am very happy with my Grace amp. Having the built in D/A convertor makes it a very flexible system. I just solder up long digital cables to reach wherever I feel like listening around the house. When I get through with my speaker/room project I hope to pick up a benchmark dac. Then I can use it for d-to-a, and use the 2 Graces as fully balanced amps for the left and right channels. The ultimate in system complexity!


 gerG


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## jhenderson010759

gerG - 

 The DEQ has done a fantastic job with my Stax and Senn headphone setup - love it. I just picked up another DEQ2496, with the intention of equalizing my living room stereo. Can you point me to a thread summarizing some "tricks of the trade" relating to loadspeaker equilization? Speakers are Dalhquist DQ10s with amplified Dalhquist subwoofer - old, but I still like it for my chamber and classical guitar music. 

 Jim


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## gerG

Jim, you may want to consider this toy for speaker management:
digital crossover 
 Most of the benefits of the DEQ, plus full up 3-way digital crossover capability. I have a similar dbx unit, and it is amazing, even if you are only working with subs and satellites.

 Man, I haven't heard DQ-10s for awhile. I don't know why I don't have a pair. The DQ subs were pretty nice as well.

 I don't have many good resource links. There is an awful lot of incorrect information out there, so I mostly go the basic physics + experimental route. It is also a lot more fun as well. Here is a good paper by Floyd Toole that covers a lot of aspects of room acoustics, and some eq remarks:

loudspeakers and rooms 

 I will rummage through bookmarks when I get home. I was actually planning to start a "Room Acoustics" thread this weekend. Maybe we can lure in some experts.

 Have fun.

 gerG


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## ooheadsoo

Here are some links to Ethan Winer's articles. He makes Real Traps as a commercial product. David Chesky liked them, for one 
	

	
	
		
		

		
		
	


	




 His articles are all in plain english, which helps.

 Some general articles with some really good acoustics articles mixed in: http://ethanwiner.com/articles.html

 A direct link to his excellent acoustics FAQ: http://ethanwiner.com/acoustics.html

 Here's a link to his Real Traps affiliated articles, including one written for Rives: http://www.realtraps.com/articles.htm


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## pbirkett

A little comment regarding the onboard DAC, a member of zerogain has the DEQ2496 and he says its comparable to his Arcam Alpha 9 CD player in quality, which is no mean feat....

http://www.zerogain.com/forum/showth...hlight=deq2496


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## jhenderson010759

I think the DEQ2496 DAC is excellent also. I use it to drive my Stax 006t amp via the XLRs. Very engaging, musically. 

 The interesting thing about that particular setup is that I can't seem to find an eq setting that makes the Stax sound better to my ears than _unmodified _ response. I have the frequency response measurement from Stax, and I've used it to create very subtle eq patterns which provide, for instance, flattened bass response. But upon critical listening sessions, I still prefer the unmodified response. Curious, because the measured Stax response cries out for a +3 dB shelving filter below 1 kHz. OTOH, 3 dB isn't much of an amplitude variation. 

 Now, this is not true for the DEQ -> Grace -> HD650 setup. In that case, I much prefer the flattening of the bass hump, as outlined in gerGs post (above), although I tend to prefer leaving the midrange and treble unmodified.


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## Steve999

Hey jhenderson, I'm a 5-month-plus DEQ2496 user, this was the piece of gear that made everything worth it for me. I use mine every day and do something a little different with it nearly every day. I'm really enjoying your posts. Are you a digital sound processing engineer? (I thought I read that or something like it in your profile.) I feel like the true lucky one, I get the benefit of the discoveries and expertise of folks like you and gerG. I would never have stumbled on this stuff or known how best to use it on my own.

 You guys are so far over my head I really feel I have nothing to contribute! And since I got the DEQ2496, the great headphone quest has been over for me. If I want a new sound, I make one! What to say?!?!? Other than, thanks for all the great info, it's wonderful to read the new posts and have a way of sharing experiences with what for me is a very treasured piece of gear.

 It's exciting to see more people trying this out and experiencing it for themselves.

 All the best!!!


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## jhenderson010759

Steve - 

 Thanks for the kind words, but I am far less knowledgeable about audio as gerG and many others lurking around here. 

 Our DSP boards are primarily used for industrial control and data acquisition applications, not audio. But, sometimes the technology is similar. For example, we make a card with 32 channels of 192 kHz, 24-bit A/D for sonar apps. Developing a card like that is a humbling experience - the smallest layout or component selection error degrades performance a few dB. Maintaing 100 dB of SNR and 95+ dB of SINAD with minimal crosstalk across a large channel count is a hellacious task. That <$100 sound cards produce numbers greater than this (though with fewer channels) is... fantastic. 

 Being and engineer and an audiophile is oxymoronic. On the one hand, I have to dismiss as nonsense all of the posts I have read about being able to discern sound quality differences between two different TOSLink cables or someone discussing "jitter" over a USB cable. ABX test anyone? 

 I favor "auro-nihilism". But, I don't think we understand everything that must be measured yet. 

 I realize now that, I don't always _want_ accuracy in my audio chain. I think I've probably attended over one hundred classical guitar concerts in my life. But, if my gear sounded exactly like some of those concerts, I'd sell it. No, what I am seeking is enjoyable, musical hyper-realism. 

 So, the quest continues....

 Jim


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## tomek

hey gang, question about using the Behringer.

 in the manual, it says to only flatten the room for 100hz and up when using the auto EQ feature with pink noise and a mike.

 i've done this now, and have fine tuned my system to an amazing level. however, i notice that when the machine is adjusting pink noise levels, it does have everything flat at 100hz and up, but below that, there is a dip in my system at 80hz, a significant one, then it is back on level at 40hz.

 my speakers are quite large, and are supposed to be flat to 32hz. is hole in the bass part of the room, and show i try to adjust for it with the behringer unit, or is there a reason the manual recommends against it. is this something a 'bass trap' solves? it seems quite a big dip when the rest of the speaker is now perfectly flat.

 ps: i love this thing. i can't listen to my speakers with the EQ off now. i have trouble imagining i ever did in fact.


----------



## kwkarth

Quote:


  Originally Posted by *tomek* 
_hey gang, question about using the Behringer.

 in the manual, it says to only flatten the room for 100hz and up when using the auto EQ feature with pink noise and a mike.

 i've done this now, and have fine tuned my system to an amazing level. however, i notice that when the machine is adjusting pink noise levels, it does have everything flat at 100hz and up, but below that, there is a dip in my system at 80hz, a significant one, then it is back on level at 40hz.

 my speakers are quite large, and are supposed to be flat to 32hz. is hole in the bass part of the room, and show i try to adjust for it with the behringer unit, or is there a reason the manual recommends against it. is this something a 'bass trap' solves? it seems quite a big dip when the rest of the speaker is now perfectly flat.

 ps: i love this thing. i can't listen to my speakers with the EQ off now. i have trouble imagining i ever did in fact._

 

The reason you only want to auto calibrate 100Hz and up us because of the wavelengths of sound below 100Hz and how those frequencies interact with your room geometry to produce standing waves and rarefactions. Moving the mic or the speakers even one inch can produce drastically different results. Put on a 50Hz sine wave tone and walk around the room. It helps illustrate the problem if you also augment your listening journey with an audio sound level meter. Move your ear and the sound level meter near a side wall, move into the corners of the room. You will most likely notice drastic differences in sound level. You can't properly equalize nodal room modes. You have to treat the room to remove those problems.

 Yes, this is where tube traps and the like come into play.


----------



## ooheadsoo

Hey tomek, what mic did you use with the behringer?

 For some Jon Risch SQ&D (simple quick and dirty) bass traps, go down to whereever you can find bales of fiberglass for sale. Not rigid fiberglass, just the regular stuff stuffed into clear plastic bags. Buy 3 bags for each corner of your room. Leave the stuff inside the bags, stack them in the corners of your room. Presto. Decorate/dress them up as necessary. Most stuff won't stop the bass traps from being effective.

 Or you could buy rigid fiberglass tubes. Jon Risch doesn't think it works ideally, but it does work according to many people who have tried it. You might even stuff the tubes with the fiberglass mentioned above 
	

	
	
		
		

		
		
	


	




 Again, decorate the tubes as necessary.


----------



## gerG

tomek, which model of deq did you get?

 Be careful with the auto-eq function. It is really intended for large spaces, and yields funky results in a normal sized room. You may end up with sliders yanked all over the place. Although such an adjustment may measure flat on the rta, it probably is not. Work for smoothness.

 Man, lf irregularities are frustrating. You should get a cancellation trough and a reinforcement peak associated with each dimension of the room. The trough will be at a frequency associated with a wavelength twice the characteristic dimension. The associated peak will be at twice that frequency. The equation is simple, f = a/l, where f is frequency, a is the speed of sound, and l is wavelength. As an example, my room is 14' wide by 8.5' high, and I get this (subs only):






 First trough is about 41 hz, speed of sound is 1134 ft/sec, which tells me that my room has a dimension of 13.8 ft. There is also an associated peak at 82 hz. Pretty cool, huh? I am lucky that I have no rear wall, which is the real beast in these situations. The next trough is the ceiling.

 Anyway, you can eq the peaks down, but do not try to boost the cancellations. You may run out of amp before you get there.

 Options for getting rid of the bass troughs are:

 1) Bass traps on the offending walls (one of which is overhead)
 2) Active control (too complex for most setups)
 3) Get rid of the walls
 4) Ignore the dips, eq the peaks, and get on with the music
 5) Fair warning, I could be one of the dips 
	

	
	
		
		

		
		
	


	




 Moving speakers around and shifting listening location will move things a bit, but the basic modes will persist.


 gerG


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## tomek

hi there. i got the deq2496. as for the mic, i'm not using the Behringer one that they recommend because the shop didn't have it, I'm using a substitute, another omnidirectional flat response mic.

 i've had some differing results with the auto EQ function. on a trial to trial basis, with the same target curve i'll get significantly different adjustments.

 however, i've got one setting now that is just fantastic. it really puts a big boost on at 8k+, but is a lot more subtle before that. the only thing that bothers me is the +8.5db adjustments in the 100-160hz range, which is a result of room interactions that i shouldn't really be EQing I guess.

 tonight i'm going to fiddle more to give the vocals a little bit more 'fullness'. right now they're prominent in this setting, but they lack a certain smoothness that i get on other settings. i'm trying to blend the best of some of these settings to get it right where i want it, so i never have to touch it again.


----------



## tomek

Quote:


  Originally Posted by *gerG* 
_. Although such an adjustment may measure flat on the rta, it probably is not. Work for smoothness._

 

what do you mean, aim for smoothness? do you mean smooth out the sliders so they resemble a curve more than the random up and down positions they are in now?


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## gerG

Hi tomek. Sorry for the confusion, but for some reason I thought that you had the older 8024 unit.

 At any rate, yes, that is exactly what I meant. There is some level of averaging going on in the rta, and I am not certain how the edges of the sample bins line up with the eq bins. I get suspicious when the auto-eq comes back with an up/down sequence in an area that is relatively flat to begin with. I like to use the geq for large offsets, and the peq for local stuff, if necessary.


 gerG


----------



## tomek

hi again,

 i'm looking for some help with this thing. when it auto eq's, it does it in that up and down fashion you mentioned. it will have one frequency at normal, and then the one on each side significantly altered.

 the unit has removed some 'tinny' sound from my stereo. the imaging is better and it is smoother, but the vocals seem recessed. how would you suggest i alter the auto eq to remedy this. is the peq better, or should i smooth out the geq? and which frequencies will bring the vocals forward?


----------



## jhenderson010759

During the automatic RTA equalization process, clicking the Page button will toggle you between pages 2 and 3. On page 3, you can adjust the Delta and Max parameters, via the rotary knobs. 

 The Delta parameter controls the maximum permissible amplitude difference between adjacent graphic equalizer frequencies. If you make this a small number, say 2 dB, you are indicating that the resultant equalization curve must contain no "abrupt" adjustments greater than 2 dB. The output curve will be much "smoother". 

 The Max parameter specifies the largest permissible excursion between the highest and lowest equalizer adjustment in the resultant equalization curve. So, by setting this to, say 8 dB, you indicate that the automatic RTA may not make adjustments to the response whose total span exceeds 8 dB.


----------



## Jahn

EQ is really helping with these old recordings I have of the Beach boys. now if i only had some crossfeed...


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## jhenderson010759

If using a PC as a source, Foobar supports a number of crossfeed filters. The standard filter is subtle and musically unobtrusive. Others, such as the 4Front crossfeed filter are more obvious.


----------



## cosmopragma

Quote:


  Originally Posted by *jhenderson010759* 
_If using a PC as a source, Foobar supports a number of crossfeed filters. The standard filter is subtle and musically unobtrusive. Others, such as the 4Front crossfeed filter are more obvious._

 

I've searched for an alternative crossfeed filter for foobar today, no results.
 4Front is based on a reverb algorithm.
 There seems to be some confusion about what crossfeed is.There is a good description on head-fi sponsor Meier-Audio's webpage for those who are interested.
 I'm used to the adjustable crossfeed of my Prehead, and I need a decent crossfeed for my Staxes.No thanks, I don't want another box and cables.
 I really like foobar, but I think I'll have to switch to winamp.
 There's a nice adjustable crossfeed effect for winamp called Headplug.


----------



## jhenderson010759

How about the following links:

Naive Software 
 or
Foobar Distribution Site 

 Additionally, with the appropriate parameters, I believe that the convolver on the latter site may also implement portions of the classical crossfeed algorithm (see also Convolver Impulse Files). However, I've not tried to derive these settings myself. 

 Are these too inflexible for your needs? I personally would like to locate user-programmable crossfeed filter, as the ones listed above are fixed-functionality.


----------



## kwkarth

More good links to good information about crossfeed:
Fixing the Blobs in your Head 
How we hear 
The problem with headphones 
Fixing Headphones using electronics
Using Computers to fix Headphones


----------



## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_Jim, you may want to consider this toy for speaker management:
digital crossover 
 Most of the benefits of the DEQ, plus full up 3-way digital crossover capability. I have a similar dbx unit, and it is amazing, even if you are only working with subs and satellites.

 Man, I haven't heard DQ-10s for awhile. I don't know why I don't have a pair. The DQ subs were pretty nice as well.

 I don't have many good resource links. There is an awful lot of incorrect information out there, so I mostly go the basic physics + experimental route. It is also a lot more fun as well. Here is a good paper by Floyd Toole that covers a lot of aspects of room acoustics, and some eq remarks:

loudspeakers and rooms 

 I will rummage through bookmarks when I get home. I was actually planning to start a "Room Acoustics" thread this weekend. Maybe we can lure in some experts.

 Have fun.

 gerG_

 

Well, my DQ-10s have given up the ghost. While in the process of equalization, I heard distortion from the left tweeter. Since I "just happen" to have a replacement DQ-10 tweeter in my audio garage, I popped the left grill off to effect the repair. 

 Upon inspection, I noticed small cracks in the woofer and midrange surrounds. An inspection of the right speaker shows similar degradation. All five transducers in each speaker, except the supertweeter, should probably be replaced or reworked. But, the cost of obtaining replacements for all four drivers in each speaker is more than an new set of more-modern speakers. 

 So, I'm very intrigued with the notion of using this digital crossover to construct a new speaker system. 

 Any recommendations on an attractive, 3-way enclosure plus three well-matched transducers to be directly-driven by the six 100W channel amps in a Yamaha receiver? Should I buy a second digital crossover in order to drive a powered sub, or should I scale back to a two-way speaker design and use one or two subs as the third frequency band on a single digital crossover? 

 I see many potential benefits to directly driving each transducer in each speaker, ala a studio monitor. And, this project sounds like a lot of fun. But is this approach really likely to sound equal to or better than one of the many high-quality speaker systems out there at the same cost point (~2000)?


----------



## Flea Bag

Greetings!

 I thought the following might proove useful for recent owners of the DEQ2496:

*1) Input clipping?* (With regards to digital optical input into EQ)

 The input level LEDs may occasionally flash red to signify 'clipping'. I thought this to be highly unusual as my input was pure digital from my DVD player's optical out. I used the below procedure and came up with the following findings:

 First, activate the 'METER' button to display METER menu page 1.
 With the CD playing, and with 'Input' as the source reading, even the loudest tracks should only produce a peak reading of '-0.0 dB'.
 This indicates that the peak digital input capicity of the EQ was reached but never exceeded. Only when the peak reading 'Clip' is produced do you have a problem with the unit. Remember the above is only applicable to my experimentation with optical digital input.

 My Bon Jovi 'Keep The Faith' album consistantly produced peak readings of '-0.0 dB' over and over again. On occasion, one or two Def Leppard tracks from 'Vault' woud exceeded '-0.2 dB' peak. While all Dave Brubeck 'Time Out' songs never exceeded '-0.9 dB'.

 The same is also true for peak meter readings for output (as long as there have been ABSOLUTELY NO MODIFICATIONS to any settings like the GEQ, PEQ, Utility Menu etc...) However, once any of the above audio-related-settings are changed from the default, things get a bit impractical.

*2) Inconvenience: output clipping*

 I realised that the red LED flashes I saw in the above situation were actually to do with output clipping. Of course this was not due to incompability with external equipment like my amp but rather, the EQ's ability to produce a higher output to meet my equilisation demands. (The LED *always* measures input signals but when clipping occurs, regardless of wheter it is output or input clipping, the red 'cliping' LED will flash.)

 I discovered that this was due to the changes I had done to the GEQ. I had boosted the bass by a few dB and reduced treble by a max of 8.0 dB. So if I remove the bass boost, leaving only the treble cut, then clipping should no longer occur right? Wrong!

 Output clipping continued to occur despite the GEQ being almost flat except for the drop in treble. I increased the treble again to completely flatten the GEQ and sure enough, clipping stopped. This didn't make sense. It seemed that any change of more than a few dB, wheter be it a boost or cut, would result in output clipping.

 I finally solved the problem by changing the 'GEQ-MODE' (in Utility Menu) from the default 'TRUE RESPONSE' to 'UNCORRECTED'. Clipping stopped and only returned when there was a gain in any sector of the GEQ settings.

 As stated in the instruction menu, the true response modifications help to smooth out the response curve. However, what it probably did not mention was that in order to do so, certain frequencies might actually exceed those seen in the GEQ settings. Either that, or the modified signal output has an accuracy of +/- 0.7 dB (for the case of my music and my relatively mild GEQ settings.)

 So how can I get a true response from my desired equilizer settings without clipping? The solution was to drop the 'gain offset' (again in Utility Menu) to about -1.0 dB. (For my case -1.0 dB is fine. People requiring more extreme GEQ curves might have to go much lower than this. Perhaps past -3.0 dB mark and increase amplifier volume to compensate.)

 Other modifications which resulted in clipping were adjustment made to stereo width (expansion; values larger than 1.0) and the PEQ (I've not experimented with DEQ/DYN settings just yet).

 My impressions of the unit so far:

 The above is my complaint with regards to the DEQ2496 at this point of time. Owners running a weaker analog in signal will probably be less restricted than users running unmodified digital optical inputs into the DEQ2496. Statistically, at least half of all changes to PEQ/GEQ settings will require owners to drop the unit's gain offset from the original factory default level of 0.0 dB. I'm fortunate that a drop of only -1.0 dB is sufficient for my purposes so far. Even if I keep the 0.0 dB offset, the maximum clipping I'll get is a cut of 0.7dB with my loudest songs, which is hardly noticable.

 Finally, I would advise owners of the DEQ2496 (especially gerG who already has a particle stuck on the inner surface of the display) to minimise the possibilty of dust and insects getting near to the equilizer! An ant or two was seen walking around inside the display of the behringer! I saw one crawling out of the buttons of the front panel. Behringer could have done better produce higher quality buttons.

 Other than the above two complaints, this equilizer has turned out to be my most significant audio purchase I'm ever likely to make from now onwards. Even upgrading from a buffered Cmoy to a Rega Ear or even a Talisman wouldn't be as significant.

 Best sacrifice my wallet made for this hobby.


----------



## ooheadsoo

Hey jhenderson, if it's just the foam surrounds that have fallen apart, check out this site: http://speakerrepair.com/

 They're in orange county so you could even drive down there on a weekend if you really wanted to.


----------



## gerG

Jim, I prefer the diy route. I have used the modular approach in my outdoor setup, with a different cabinet for midwoofer, tweeter, and sub. Since you can compensate for efficiency differences with the x-over, matching drivers is not as critical. I will bet that the Behringer has the option of high pass, even on the low channels. This means that you actually have a 3.5 channel xover. For the sub you only need the low pass, and there is an abundance of sub amps out there with variable low pass built in. The very cool thing about the digital x-over is that it is phase correct with any q or slope. There are also time delays to allow fine adjustment for speaker placement and alignment. Does the Yamaha have pre-outs and main ins, or will you be using 6 separate inputs? The latter would resolve my problem of running the x-over at too low a level and losing resolution. At least I am assuming that happens with a digital device.

 Flea, thanks for the great observations, and welcome to team DEQ. Sorry to hear about the "bugs" in your unit. Make sure the Ant option is turned off in the fifth page of the utilities menus 
	

	
	
		
		

		
		
	


	





 gerG


----------



## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_Jim, I prefer the diy route. I have used the modular approach in my outdoor setup, with a different cabinet for midwoofer, tweeter, and sub. Since you can compensate for efficiency differences with the x-over, matching drivers is not as critical. I will bet that the Behringer has the option of high pass, even on the low channels. This means that you actually have a 3.5 channel xover. For the sub you only need the low pass, and there is an abundance of sub amps out there with variable low pass built in. The very cool thing about the digital x-over is that it is phase correct with any q or slope. There are also time delays to allow fine adjustment for speaker placement and alignment. Does the Yamaha have pre-outs and main ins, or will you be using 6 separate inputs? The latter would resolve my problem of running the x-over at too low a level and losing resolution. At least I am assuming that happens with a digital device.

 Flea, thanks for the great observations, and welcome to team DEQ. Sorry to hear about the "bugs" in your unit. Make sure the Ant option is turned off in the fifth page of the utilities menus 
	

	
	
		
		

		
		
	


	





 gerG_

 

getG - 

 The Yamaha has a special 6.1 input mode, which relegates it to a 6 channel amplifier, devoid of any pre-amp signal processing or routing (except for volume control). I was planning on running the six Behringer outputs into these line-level inputs, then adjusting the digital crossover to compensate for the amplitude differences and frequencies required for each individual driver. Once done, I think the Yamaha volume control will act as a pre-amp for all six channels. Think this will work? 

 I don't follow your comment about the "3.5 channel" operation of the digital crossover. If I drive each transducer in a three-way satellite with each of the three digital crossover outputs (for each channel), the woofer channel would be a lowpass, the midrange would be a bandpass and the tweeter would be a highpass, correct? How could I extract only LF signals for the sub without another crossover channel? 

 Wait.. I think I see. You're saying that I route the lowpass line level digital crossover output to the Yamaha amp line level input in parallel with the sub's crossover circuitry, so that they both see woofer LF signal, then the sub crossover rejects above ~90 Hz? Seems like this simply shifts the problem. Don't I still need an active, line-level crossover for use by the sub near the electronics rack? I wouldn't want to run a line-level signal across the room to a powered sub...


----------



## gerG

Jim, what I meant by 3.5 was that all channels have bandpass capability in a digital crossover. That would give you both high and low pass on the woofer channel. Unfortunately I was not considering digital input to the crossover, which complicates things quite a bit (pardon the pun). In that case I think that you either have to stick with 6 channels, or add a crossover to the analog output of the bass channel. You could also add another digital device in parallel (I don't recall if that unit has a passthrough). I suspect that the Yamaha will run the 6 channels through it's preamp, so you should have volume control. Make sure that it does not go through another series of A/D and D/A.

 In this situation I would be tempted to stick with a passive between the mid and tweeter, and go active at the other points.


 gerG


----------



## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_Jim, what I meant by 3.5 was that all channels have bandpass capability in a digital crossover. That would give you both high and low pass on the woofer channel. Unfortunately I was not considering digital input to the crossover, which complicates things quite a bit (pardon the pun). In that case I think that you either have to stick with 6 channels, or add a crossover to the analog output of the bass channel. You could also add another digital device in parallel (I don't recall if that unit has a passthrough). I suspect that the Yamaha will run the 6 channels through it's preamp, so you should have volume control. Make sure that it does not go through another series of A/D and D/A.

 In this situation I would be tempted to stick with a passive between the mid and tweeter, and go active at the other points.


 gerG_

 

gerG - 

 What's your position on scaling back to a two-way satellite, plus one or two subs versus the three-way satellites + 1 sub discussed earlier? I am attracted by the simplicity of this latter approach: Each transducer gets an amp; The digital crossover has enough channels; I could procure higher-grade transducers for the same money, since I'd need only a mid-bass plus a tweeter. I'm thinking of something like a ScanSpeak 7" mid-bass + 1" tweeter for the satellites. Don't know what to use for subs yet. Perhaps I could just use my single, DQ-1 subwoofer initially. 

 I have a stereo Dyna 200 W amp w/ext capacitor for powering the sub(s). So, I am not short on amplifier channels, just crossover channels.


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## gerG

I like that approach. With 2 subs you can push the crossover a bit higher and unload the midwoofers, allowing smaller drivers (better midrange). There are a lot of MTM configurations out there now, which also lets you use smaller woofers without sacrificing dynamics or output. You have seen them, woofer above and below the tweeter, nice narrow cabinet. Parts Express and Madisound have cabinets, kits, drivers.

 For subs I am really impressed by the NHT 1259 drivers (strictly for sealed cabinets). I was going to use 2 per side mounted in opposite sides of each cabinet to cancel vibrations. I ran across the big JBL drivers first and had to try them. Unfortunately the JBLs were a buyout, so I don't think there any more around.


 gerG


----------



## Az B

I've been using a DCX for an active crossover for a while now, and a DEQ for over a year, but I just found this forum. Most audio forums are quick to dismiss these items as "pro gear" and therefor not worthy of fooling with.

 Glad to see you guys like to experiment!

 For Flea, regarding the clipping issue, are you actually hearing the clipping? Unless you're hearing audible distortion, this is a relatively common indication for most digital devices. Though the CLIP LED's will illuminate, a correctly mastered CD will never actually exceed the 0dBFS ("decibel - full scale") limit, even when it is indicated as such. The actual signal level on modern mastered CD's can be literally -.01 dB short of full scale, but the resolution of the meter circuit under real-time conditions may not discern between this level and 0dBFS (max/clip). It is quite common for manufacturer's to implement a circuit that illuminates the CLIP indication PRIOR to actual clipping, so that the engineer can correct any issues before an audible result is heard.

 You will know immediately if the input is ACTUALLY clipping, as this form of digital distortion is particularly uncomfortable to listen to. Also, the ULTRA-CURVE has a limiter built-in to it that should alleviate any possibilities of clipping within reasonable limits.

 Az


----------



## jhenderson010759

Quote:


  Originally Posted by *Az B* 
_I've been using a DCX for an active crossover for a while now, and a DEQ for over a year, but I just found this forum. Most audio forums are quick to dismiss these items as "pro gear" and therefor not worthy of fooling with.

 Glad to see you guys like to experiment!

 For Flea, regarding the clipping issue, are you actually hearing the clipping? Unless you're hearing audible distortion, this is a relatively common indication for most digital devices. Though the CLIP LED's will illuminate, a correctly mastered CD will never actually exceed the 0dBFS ("decibel - full scale") limit, even when it is indicated as such. The actual signal level on modern mastered CD's can be literally -.01 dB short of full scale, but the resolution of the meter circuit under real-time conditions may not discern between this level and 0dBFS (max/clip). It is quite common for manufacturer's to implement a circuit that illuminates the CLIP indication PRIOR to actual clipping, so that the engineer can correct any issues before an audible result is heard.

 You will know immediately if the input is ACTUALLY clipping, as this form of digital distortion is particularly uncomfortable to listen to. Also, the ULTRA-CURVE has a limiter built-in to it that should alleviate any possibilities of clipping within reasonable limits.

 Az_

 

When I first equalized with the 2496, I failed to constrain the permissible step between adjacent frequencies. I tried auto-eq-ing to provide a quick starting point. Well, the auto-derived response changes were fairly severe and when I tried playing some music using this curve, the equalizer clipped very badly. The sound was extremely distorted - very noticable. But, until I read Flea's report, I didn't understand the problem or how to fix it. So, under the right conditions, the 2496 can definitely produce audible clipping. 

 It's great to talk with someone actually using the DCX the way I intend to. I am just now assembling my satellites and subwoofer in preparation for tri-amping using the DCX. I have about five-hundred questions rolling around in my head about this configuration. 

 To start, I am concerned about protecting the tweeter. Though I intend to be careful, during initial configuration, I could steer an inappropriate frequency band to the tweeter. Is there an effective way to mitigate this risk? Will a fast-blow fuse do any good? 

 What procedure do you follow in order to adjust the filters on the DCX to insure optimal flatness at the crossover frequency? Similarly, how about insuring proper phase response for each driver? Is it reasonable to do DCX configuration in the listening room, or must they be performed in a near anechoic environment to achieve any reasonable accuracy? 

 Is there a text resource that I could read that explains some of this?


----------



## Az B

What kind of tweeters are they? I use some old EV T-25s and they're pretty much indestructible so that's not a worry. Dome tweets are probably the most delicate. If you can find the specs on the driver, it will give you a good idea how much extension you can have. OF course, the main killer of tweeter is clipping, and one of the beauties of active bi and tri amping is that your tweeter amp will never be stressed by big bass notes and will clip the tweeters a lot less.

 For creating the crossover points, I used an RTA. I couldn't find specs on all my drivers, so I measured them while the passive crossovers were still in the cabinets. Unhooking all but one driver at a time gave a very clear xover point and curve on the RTA. This gave me a reference point to start from. If you have the specs for your drivers, you can get a pretty good idea where to start from there.

 After that, it's a fair amount of trial and error. I started with a 48db Linkwitz-Riley arrangement at the xover point of the orginal passive crossovers (which were second order Butterworth) Using the RTA, I simply tried several combinations and looked for smoothness, listening to the setups that seemed better as I went. 

 The end result was a far smoother crossover point than was recorded with the speakers using the passive xovers they came with. I ended up with 4th order L-R crossover type, but that's just what worked best with my setup. Theoretically, a steeper slope will most often be best, but not always. It depends on the cabinet, drivers, etc.

 The DCX automatically corrects phase errors, so that's not an issue. (For that matter, L-R slopes above second order have very few phase errors anyway) There are tests for this, but they're long, complicated, and not needed. So unless you really want to know, I won't bother.

 I see no reason to try to get any kind of anechoic response out of your speakers since they'll never be used in an anechoic chamber. Set 'em up for the room you'll be listening in. Another great thing about active crossovers is that it's simple to change things if you change drivers, the room, or whatever. You can compensate somewhat for room problems with the xover setup. Try that with passive crossovers!

 Az


----------



## jhenderson010759

Quote:


  Originally Posted by *Az B* 
_What kind of tweeters are they? I use some old EV T-25s and they're pretty much indestructible so that's not a worry. Dome tweets are probably the most delicate. If you can find the specs on the driver, it will give you a good idea how much extension you can have. OF course, the main killer of tweeter is clipping, and one of the beauties of active bi and tri amping is that your tweeter amp will never be stressed by big bass notes and will clip the tweeters a lot less.

 For creating the crossover points, I used an RTA. I couldn't find specs on all my drivers, so I measured them while the passive crossovers were still in the cabinets. Unhooking all but one driver at a time gave a very clear xover point and curve on the RTA. This gave me a reference point to start from. If you have the specs for your drivers, you can get a pretty good idea where to start from there.

 After that, it's a fair amount of trial and error. I started with a 48db Linkwitz-Riley arrangement at the xover point of the orginal passive crossovers (which were second order Butterworth) Using the RTA, I simply tried several combinations and looked for smoothness, listening to the setups that seemed better as I went. 

 The end result was a far smoother crossover point than was recorded with the speakers using the passive xovers they came with. I ended up with 4th order L-R crossover type, but that's just what worked best with my setup. Theoretically, a steeper slope will most often be best, but not always. It depends on the cabinet, drivers, etc.

 The DCX automatically corrects phase errors, so that's not an issue. (For that matter, L-R slopes above second order have very few phase errors anyway) There are tests for this, but they're long, complicated, and not needed. So unless you really want to know, I won't bother.

 I see no reason to try to get any kind of anechoic response out of your speakers since they'll never be used in an anechoic chamber. Set 'em up for the room you'll be listening in. Another great thing about active crossovers is that it's simple to change things if you change drivers, the room, or whatever. You can compensate somewhat for room problems with the xover setup. Try that with passive crossovers!

 Az_

 

I am building North Creek Rhythms as the satellites and two Poseidons for subs. The Rhythms normally use a ScanSpeak D2905/9900 for the tweeter, but the owner of NC now recommends using the North D25-06S silk-dome tweeter instead, so that's probably what I'll end up with. 

 I am really happy to hear you discount the damage susceptabilty. I thought that I'd use an existing set of full-range speakers to rough-in the crossover settings for each amp output, then migrate over to the new speakers. 

 When you measure the in-situ performance of each of the individual drivers to determine their specific frequency responses, do you position the mic very close to the speaker in order to minimize room effects? Or, should that be done at the normal listening position? 

 I've purchased both a DCX and a 2496 equalizer for use on the system. But, do you find that both are useful, or does the DCX handle the crossover function as well as room equalization by itself? 

 What about subwoofer equalization. Though I plan on building some bass traps and improving the room, presently, it's a regular 15 x 15' living room with hardwood floors, rugs drapes and furniture. It has a vaulted ceiling and the rear wall has very large arches that open to the dining room. I am suspicious of all RTA measurements below 100 Hz. How did you configure for optimal bass response?


----------



## Az B

Wow, those rhythms look awesome! Those are going to sound great actively bi/tri amped.

  Quote:


 When you measure the in-situ performance of each of the individual drivers to determine their specific frequency responses, do you position the mic very close to the speaker in order to minimize room effects? Or, should that be done at the normal listening position? 
 

Yes. Get them flat with close micing, then you can move the mic to the listening position for level matching and room correction.

 I also use the DEQ2694 for RTA and SPL meter purposes. Originally I was not planning on inserting into the loop, but since I can use the digital in, and then the digital out to the DCX, and the DCX has limited processor power for EQ, I have it in the loop and it works great.

 For the sub, you may find it useful to have some sort of parametric EQ for room modes. Although the 1/3 octave RTA doesn't really have the resolution for some low bass peaks and valleys, you can still use it to find them with a little detective work and a test tone CD. A Behringer Feedback Destroyer is a very inexpensive solution for providing some sub EQ since you're using all six channels of the DCX for your mains.


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## jhenderson010759

Az B - 

 What are you using as a digital volume control? I'd like to drive the CD player TOSLink output into some sort of digital pre-amplifier providing volume control adjustment in the digital domain. The only place I intend to go analog is at the outputs of the DCX, so I think I need a digital volume control somewhere in the TOSLink path. 

 Originally, I planned on using a Yamaha receiver as both my 6 channel amp and my digital volume control. But now, I am concerned that the Yamaha's 100 W/ch may be inadequate (for the subs, particularly). Worse, there is a technical snag with the Yamaha: It features an input source mode in which the 6 amp channels may be directly driven by six RCA inputs (perfect for the DCX). But, this is _mode-selectable_. So, if the unit were ever inadvertently switched to any other source (such as the FM tuner), full-range signals would be driven out of _all_ of it's amp channels, damaging my tweeters. A shame, since the only source I intend on using is the CD, but the FM tuner mode can't be locked out. 

 I could use a laptop PC as my source, via Foobar (my CD collection is already ripped lossless), but I'd prefer to avoid this if possible. It'd be nice not to have to 'boot' my stereo.


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## Flea Bag

Az B,

 Thanks for the manufacturer's insight. I've discussed the issue of output 'clipping' with someone last week and felt that I need to clarify or better phrase my wording. The 'clipping' I was originally referring to was the User Manual's definition of clipping which is actually just a form of frequency limiting which cuts only the particular frequency which exceeds the equalizer's output capability. 'Frequency chopping' will be what I'll call it from now onwards or if someone could name the proper technical term.

 With regards to the more common definition of clipping, like jhenderson010759 mentioned, the equilizer does encounter very noticable clipping across the whole frequency spectrum if the frequency chopping reaches more servere levels. My frequency chopping only involves amounts of less than 0.7dB -too small for me to hear any difference. If one were to equalize more serverely than I do, the DEQ2496 will indeed exhibit clipping as opposed to just frequency chopping. This is most obvious when playing around with the DEQ function of the 2496.

 As for good recordings, they never seem to exceed the -0.0dB peak reading. However this does happen to some very well known various-artists-compilations, hastingly put together. This is especially so with a particular pirated CD I tested. Now we have aural and measured proof of bad recordings!

 So for new 2496 owners; If your optical digital input shows 'clip' as a peak reading, its more likely that your recording is bad than your equipment is faulty. However, if you still get 'clip' as a frequent reading even with well mastered CDs, then your DEQ or source equipment could have a problem.

 I forgot to mention that if you are boosting any particular frequency until the out level indicates 'clip', you can:

 a) Reduce gain offset in page one of the utility menu.(as stated in my previous post)

 b) Do it directly from the GEQ page. (THIS IS NOT LISTED IN THE INSTRUCTION MANUAL!!!) Press the upper (small) data wheel to change the FREQ selection to read 'FULL'. Now you can use the large data wheel to horizontally offset the entire frequency range.

 c) If you would like a bass boost for example any range below the 80Hz mark, the PEQ can also serve as an alternate method of preventing output clipping/frequency chopping. In the PEQ page, you could turn the lower small data wheel until it reads 'L6dB', 'L12dB', 'H6dB', 'H12dB', 'LC' or 'HC' and then make the bass more prominent by reducing all other frequencies.

 On the topic of bass boosting, I've got a quick and convenient tip to make the DEQ2496 operate with the convenience of a 'Mega Bass' boost button you see on a portable Discman. I save most of my settings for my HD280 in GEQ mode, keeping them neutral.(still haven't found what I call neutral yet!)

 Next, I make a bass boost in PEQ mode and save both PEQ and DEQ as a single preset which I call preset A. So preset A allows me to listen to my music in a nice neutral tone (or so I would think) and when I'm not happy with the bass, I simply press and hold the PEQ button to activate/deactivate the bass boost. No need to switch between presets.

 That's all I have time for at the moment!

 gerG, do you think it would be good if you put up some kind of review/FAQ for the DEQ2496? It's obviously quite popular a piece of equipment around here and the instruction manual alone just isn't enough. Perhaps it'll be more of an extended instruction manual than just a review. If you would like me to put all my findings about 'clipping' in one post for the review, just let me know.


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## tomek

i'm all for a FAQ!

 the tip i got to lower the delta between adjacent frequencies when using auto EQ really helped. it was a big difference and nothing that i found in that skimpy user's manual.


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## Az B

Quote:


  Originally Posted by *jhenderson010759* 
_Az B - 

 What are you using as a digital volume control?_

 

I have a Carver Reference CD player with a volume control.

 Az


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## jhenderson010759

Quote:


  Originally Posted by *Az B* 
_... Originally I was not planning on inserting <the 2496> into the loop, but since I can use the digital in, and then the digital out to the DCX, and the DCX has limited processor power for EQ, I have it in the loop and it works great._

 

What data format do you use for the digital connection between the DEQ and DCX? I need to use SP/DIF from my CD transport to the DEQ, but it isn't clear from the DEQ manual whether it will simultaneously drive the AES/EBU outputs when receiving SP/DIF inputs via TOSLink. Is that your configuration also?


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## gerG

Jim, sweet speaker project! I use a toslink into the DEQ, and AES balanced outgoing. That line at the bottom of my sig is pretty much the settup. For speakers, I have been using the dbx digital xover with high pass directly to Morel tweeters, with never a blip. I do the same trick in my car, with a dedicated amp just for the tweeters and an active crossover. I was worried about a stray chirp taking out a tweeter, but it has not happened in 2 years of operation.

 I am getting very curious about the DCX. I love the idea of going digital straight into the x-over. I am suspicious of digital volume control, though. Does that not reduce amplitude resolution? I am thinking analog volume controls downstream of the D/A step. As touchy as my amps are, I don't need any extra gain in the loop.

 Flea, I think that a FAQ is an excellent suggestion. This thing has started to catch on lately, and the info should be easier to mine than reading through this whole thread (and the twin thread that is burried at the moment). Thanks for mentioning the trick for changing volume in the GEQ window. That feature is actually unique to the latest firmware update, I think. I always bounce back to the meter window to check my levels (just like the old tape days). When I am finished with adjustments I leave the display in the rta mode.

 Az B, welcome aboard! Have you tried the DEQ with headphones yet? You are right about the resolution of the built in RTA. For people who have a test mic already I recommend picking up a decent PC interface/preamp and a copy of TrueRTA. Take a peek at my room acoustics thread for an example of what it can do. For a $100 software package it is just short of miraculous.

 A parametric eq can help with the LF modes in a room, but it is better to get at the source if you can. Pulling down a reinforcement with a parametric resolves the quantity issue, but now you have a signal that is roughly half source and half 360 deg shifted reflection. The balance will be right but you are giving up transient response. You will still be left with the cancellations.


 gerG


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## gerG

Forgot to mention another subwoofer on my must build list. The Peerless XLS 12" with the 12" PR is an amazing combination in a small cabinet. Peerless has an excellent white paper on their website:

Peerless 12 

 Don't be put off by the disclaimer, it is a good read.


 gerG


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## ooheadsoo

Oh yeah, the peerless is supposed to be sooooooooome really really good driver, maybe one of the best. I had been considering that driver until this one popped up: http://www.gr-research.com/sub.htm

 According to my sims, I can get -3db at 22-24hz or so with the stock PR. 
	

	
	
		
		

		
			





 Sound should be decent, I would think. The only thing I know gerG doesn't like is the foam surround, but I have no problems with them


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## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_...
 I am getting very curious about the DCX. I love the idea of going digital straight into the x-over. I am suspicious of digital volume control, though. Does that not reduce amplitude resolution? I am thinking analog volume controls downstream of the D/A step. As touchy as my amps are, I don't need any extra gain in the loop.

 ...For people who have a test mic already I recommend picking up a decent PC interface/preamp and a copy of TrueRTA. Take a peek at my room acoustics thread for an example of what it can do. For a $100 software package it is just short of miraculous.
 gerG_

 

gerG - 

 I agree that a digital volume would reduce the amplitude of the outbound signal, signal purity should not be affected. Digital volume should be implemented as a scalar multiply, which is a linear operation. True, the dynamic range of the output will be reduced (that's the whole point), but the frequency content will be unaltered. And I'm only interested in unity gain or less (attenuation operation only), which avoids saturation scenarios. 

 If I could locate a digital preamp featuring multiple analog and digital input sources plus a digital output amplitude which tracks it's volume control, then the DEQ and DCX could both lie downstream from this switch. All source matter could readily benefit, rather than just a single source, such as a CD, as in my current hookup. Basically, I want Pre-Out, which traditionally is volume controlled, in digital form. Unfortunately, none of the pre-amps or AV receivers I've seen do this. 

 What do you recommend for an inexpensive mic pre-amp? I have TrueRTA and the Behringer ECM8000 mic, and I'm thinking about either the Behringer Shark DSP 101 or the Behringer UltraVoice Digital Vx2496. The former is cheaper, but the latter has digital out, so I could skip a D/A -> A/D conversion on the way to my EMU 0404 sound card if I get a pre featuring digital out.


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## gerG

Good morning Jim.

 I am using an M-Audio Duo usb interface right now. I would prefer firewire, but my Sony laptop has a rather quirky implementation of firewire, and it only works sometimes. usb is noisier, but it is reliable on that PC.

 There are lots of options out there for a single box to do the preamp and A/D. I had a Behringer Shark, but it was just another piece of claptrap to wire up (and another wallwart). There may be units available that can be powered off the usb port. The firewire box that I had did that trick off my powerbook (true 6-wire firewire) so no extra wallwart/wire. Unfortunately TrueRTA only runs on a PC.

 Another thought, does the Emu 0404 have phantom power and a mic input? IF so all you need is a long mic cable.

 Let me know what you figure out on the digital preamp. I am still concerned that for large attenuation I will be losing resolution. I need to think about the math on that a bit. For small changes, not an issue.

 gerG


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## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_...Let me know what you figure out on the digital preamp. I am still concerned that for large attenuation I will be losing resolution. I need to think about the math on that a bit. For small changes, not an issue.

 gerG_

 

I think you're right about the resolution proportional to volume. I don't think distortion would vary, but that's not the whole picture. And, I certainly expect that I would be using substantial attenuation most of the time, so if it is a problem, it's probably not one that should not be ignored.

 A quality, six-channel stepped attenuator between the output of the DCX and the amp inputs would do the trick, albeit in the analog domain. Do you know of any sources for one of these? XLR I/O would be nice.


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## ooheadsoo

How does this digital preamp stuff work? As far as i can tell, wouldn't you need at least 5 bits of volume attenuation data to get 31/32 steps of definition? That's a lot of data that would get truncated from a 16 or even 24 bit signal.

 It would seem to me that the cheapest way (perhaps only temporary in the long run) would be to find a nice 6.1 receiver and instead of using the receiver for surround duty, use the discrete inputs to send the signal signal to each driver. That way you get unified volume control. I haven't heard of a 6 channel stepped attenuator 
	

	
	
		
		

		
		
	


	




 That would be monstrous! Maybe 10 inches long!


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## jhenderson010759

Quote:


  Originally Posted by *ooheadsoo* 
_How does this digital preamp stuff work? As far as i can tell, wouldn't you need at least 5 bits of volume attenuation data to get 31/32 steps of definition? That's a lot of data that would get truncated from a 16 or even 24 bit signal.

 It would seem to me that the cheapest way (perhaps only temporary in the long run) would be to find a nice 6.1 receiver and instead of using the receiver for surround duty, use the discrete inputs to send the signal signal to each driver. That way you get unified volume control. I haven't heard of a 6 channel stepped attenuator 
	

	
	
		
		

		
		
	


	




 That would be monstrous! Maybe 10 inches long!_

 

ooheadsoo - 

DACT makes a six channel attenuator, but it's about $600. Atop that, I'd have to buy the case, connectors and build it. MSB makes an active attenuator, for about $800. I can't believe that I need to spend that much money for a volume control. There's got to be a better way. 

 Regarding the AV receiver, that is my current setup. But, it suffers from this problem, and is under-powered for the subs. Optimally, I'd like to be able to use a mix of amps as required for the different drivers in the system.


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## gerG

Jim, there are similar units to the DACT, but none are cheap. There is a stepped attenuator thread over in the diy section. I bet that company can stack wafers. That would end up at $180 for 6 channels. Not bad for attenuators of that quality. I actually like the idea of a long narrow housing with the inputs on one side and the outputs on the other. you could use XLR, but forget going fully balanced on the attenuators. You would need 12 rotary switches for that trick!

 Another approach that I am considering is to build a 3 position volume control, then use the gain control within the crossover to fine tune. I am assuming, of course, that there is a global gain option in there.

 You could also use integrated amps and use the built in volume controls to set your baseline, then tweak down with one of the digital boxes.

 I haven't looked, but there are bound to be multi-channel passive preamps out there. In fact, there must be used stuff in circulation.

 I will look for other options this weekend. Even though my current setup is analog in, I have the same problem because the amps are so touchy that the x-over is operating down in the weeds. I thought that it had a gain offset option, but that seems to be on the input, with helps nothing. Resistors or transformers on the output are my only options, other than switching back to an analog crossover (blech). I suppose I could listen with earplugs in and turn loose 2 kilowatts, but I don't feel like repairing water lines again.


 gerG


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## jhenderson010759

gerG - 

 I found a Roland M1000, 24-bit digital mixer for $299 on the web and ordered one for experimentation. This unit can perform the digital attenuator function, operates with 24-bit resolution and allows sourcing my CD player and/or my PC via USB to the Behringer DEQ and DCX. I don't think resolution loss will be an issue. Here's why:

 If I upsample from 16 to 24-bits before sending to the Roland, I should be free to effect an 8-bit attenuation, approximately 45 dB, with zero theoretical resolution loss. This is because upsampling produces samples with valid information in bits 23..8 and either dither or zeros in the least-significant byte. So, when the Roland attenuates the input signal using 24-bit arithmetic, the least-significant eight bits of each sample may be truncated without a loss in resolution. 

 As my entire CD collection is stored on my PC in .ape format, I can use Foobar to upsample to 24-bits during playback - just as I do in my headphone setup. If I can find a a CD transport that up-samples, I should be in good shape. 

 What do you think?


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## gerG

That sounds like a great approach. It was also quite informative. Is this a common approach for upsampling? I had always assumed that they interpolated to get to higher resolution, but this is certainly easier. I should think that a 45 db envelope would be plenty. Plus you get to have yet another digital toy in the loop!

 As an alternative, you could also do the digital volume control on the PC after upsampling.


 gerG


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## Az B

Quote:


  Originally Posted by *jhenderson010759* 
_What data format do you use for the digital connection between the DEQ and DCX? I need to use SP/DIF from my CD transport to the DEQ, but it isn't clear from the DEQ manual whether it will simultaneously drive the AES/EBU outputs when receiving SP/DIF inputs via TOSLink. Is that your configuration also?_

 

AES/EBU and S/PDIF are very similar. AES/EBU allows for manipulation, but it can be set up the same as S/PDIF and they're pretty compatible.

 Here's an adaptor:

http://www.innovativemusic.com.au/Pr...htm?sync.htm&2

 Az


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## jhenderson010759

Quote:


  Originally Posted by *gerG* 
_That sounds like a great approach. It was also quite informative. Is this a common approach for upsampling? I had always assumed that they interpolated to get to higher resolution, but this is certainly easier. I should think that a 45 db envelope would be plenty. Plus you get to have yet another digital toy in the loop!

 As an alternative, you could also do the digital volume control on the PC after upsampling.


 gerG_

 

Even if another technique were used to extend the redbook 16-bit data to 24-bits, that data would not represent genuine, additional resolution. So, how could one argue that truncating it as part of an attenuation would represent an actual data loss? 

 I want an attenuator with a _physical knob_ in my playback loop. Haven't you ever fired up your media player in Windows, only to discover that someone had previously ratcheted the Windows volume control to maximum, then left it there?


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## gerG

Ouch, touche. I am exactly the same way about having a volume control knob, preferably within arm's reach of my listening chair. I hooked up a system once that went from source to x-over to amps, and although it sounded great (dynamic as all he11) I could not deal with the lack of control. Just another character flaw that I have to live with, I guess.

 gerG


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## jhenderson010759

Quote:


  Originally Posted by *jhenderson010759* 
_gerG - 

 I found a Roland M1000, 24-bit digital mixer for $299 on the web and ordered one for experimentation. This unit can perform the digital attenuator function, operates with 24-bit resolution and allows sourcing my CD player and/or my PC via USB to the Behringer DEQ and DCX. I don't think resolution loss will be an issue. Here's why:

 If I upsample from 16 to 24-bits before sending to the Roland, I should be free to effect an 8-bit attenuation, approximately 45 dB, with zero theoretical resolution loss. This is because upsampling produces samples with valid information in bits 23..8 and either dither or zeros in the least-significant byte. So, when the Roland attenuates the input signal using 24-bit arithmetic, the least-significant eight bits of each sample may be truncated without a loss in resolution. 

 As my entire CD collection is stored on my PC in .ape format, I can use Foobar to upsample to 24-bits during playback - just as I do in my headphone setup. If I can find a a CD transport that up-samples, I should be in good shape. 

 What do you think?_

 

Upon reflection I don't think a 16-bit digital source of any kind (including my CD player) should be a problem after all. It is a virtual certainty that 16-bit sources will be arithmetically shifted to 24-bit significance (per above) upon receipt by the Roland. This is required in order to normalize the full-scale ranges (volumes) of all input sources. Bam!


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## gradofan

It's been 1 1/2 years since this thread's been added to. Where are you all now? Have you found new EQ settings, tried different programs, come up with other ways to measure headphone accuracy? Since I last posted, I'm DEQ-less (well, almost), and only have the EQ on my mp3 player. What has changed for you guys (and girls?)?


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## gerG

Jeez, my luck that I browse head-fi right after you dug this thread out of the dumpster.

 I started looking into what was wrong with my technique. One big problem was that I was using a free-field mic for headphone measurements. Another issue was separating cavity effects from the headphone native response. I devised a mic method that gave me the data to separate the effects. I had to shelve the effort in January due to the real world intruding into my little diy lab. Some day I will lock myself in there again and finish up what I started. In the mean time I just enjoy the music and stick with small projects.


 gerG


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## jhenderson010759

Quote:


  Originally Posted by *gradofan* 
_It's been 1 1/2 years since this thread's been added to. Where are you all now? Have you found new EQ settings, tried different programs, come up with other ways to measure headphone accuracy? Since I last posted, I'm DEQ-less (well, almost), and only have the EQ on my mp3 player. What has changed for you guys (and girls?)?_

 

Aside from the poor reliability of the gear, I like the Behringer more and more. I have three DEQ2496s on my surround rig to equalize fronts, center and rears. I use the DEQ and DCX in my den for the K1000s+Sub and actively driving the N.C. Rhythms. And, I use a DCX in the garage for speaker building. Very flexible, good-sounding, well-measuring gear.


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## jpelg

God, you are soooo tempting me, gerG! With it's optical in/outs, I could put it in between my Sony portable & Benchmark, keeping everything in the digital domain until the amp section. Damn you 
	

	
	
		
		

		
			





!


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## gerG

Hey zeplin, quite correct on my mistype. I was exhausted last night, but I wanted to get an update done before I take off to visit my family.

 The Philips is an excellent choice. I really like the upsampling feature. I prefer the sound of my Sony S9000 for SACDs, but I have found damned few SACDs that I think are worth listening to.

 The Behringer sells for around $300 (Musician's Friend, Guitar Center, etc.). I built the cables that go XLR to RCA. Less than $20 in parts there. It would have been less, but the Neutrik RCA connectors are $10 per pair. Very nice, but XLR on both ends would be cheaper and better both mechanically and electrically. EOW (end of whine).

 A point that I forgot to make: if you want to run the 963sa with an SACD, or with a higher upsample rate, you have to go with the analog inputs, or bypass the eq altogether. In this setup the switchable amp inputs make it easy. I have it set up with an analog bypass that goes RCA to RCA and skips the eq. That was not in the photo because I need to make a short cable. The Outlaw is just too huge for a portable system like this.

 jpelg, you will love it! It would also give you an upgrade path to a better player as well. I was quite stunned by the sound quality of the old Panasonic DVD A320 that I hooked up initially. DVD players make great CD transports, and they are getting cheap! Phillips has a slick little slot loading unit that would be wonderful, and Target has the damned thing on sale for $120 or so. I would be using one of those, but the 963sa works too perfectly here. It is reasonably small and light.

 btw, the Behringer has a variable jitter compensator. I don't know what it does yet, so I haven't moved it from the default setting, yet.

 A more gonzo approach would be a small, accurate transport, optical to an upsampling converter, optical to the Behringer, optical to a 96 khz D/A, analog to an amp. If I try that route I will skip the outboard D/A and go balanced analog to a balanced amp. Wouldn't a balanced Gilmore in a rack case be interesting? That was the initial goal, but no time to build. Some day.

 As it is, this is a fun and amazing little system. I would have a tough time beating this sort of sound quality for the mony invested in the stack. At about $1100 for a system that includes everything from source to amp, does upsampling, plays SACDs, and can bring out the best in almost any set of headphones, it is one hell of a combo. Being transportable is just the ice cream on the cake (sorry to abuse an old phrase).

 Another point to mention: my thanks to this board for the suggestions. Every piece here is something that I learned about on Head-Fi, right down to the inexpensive glass cable from MCM. The exception is the mini power strip, which has a flat 90 degree rotating grounded plug at the other end. Home depot has those (Belkin).

 Thanks for keeping me aware guys!


 gerG


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## Bullseye

After three years since the last post, how is the DEQ working for you guys?

 I am reviving this thread because I am going to buy either this week or next week the DEQ2496.

 As a source I will be using either PC with UCA202 external sound card via toslink cable to DEQ or a Cowon D2 mini jack with a mini-XLR male converter.

 I am thrilled now after reading this thread about the use of the Ultracurve Pro with headphones. I have read a lot of this product before, but mostly with the use of speakers.


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## ironmine

Quote:


  Originally Posted by *Bullseye* /img/forum/go_quote.gif 
_ As a source I will be using either PC_

 

Bullseye, if you intend to use a PC as a sorce, why do you need DEQ? You can do all equalization using Foobar's VST-plugins.


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## Bullseye

Quote:


  Originally Posted by *ironmine* /img/forum/go_quote.gif 
_Bullseye, if you intend to use a PC as a sorce, why do you need DEQ? You can do all equalization using Foobar's VST-plugins._

 

Well this question comes a bit late. I bought around a month ago the DEQ2496.

 I need the DEQ because it has an excellent DAC. The sound card from my laptop is bad, and doesn't have optical output. I use an external one now with optical output. 

 Then also because right now I can't use a headphone system, but as soon as I can I will be buying some speakers and will treat my room according to the system. If I want a room to sound good I need proper equalization. If i can use it as a DAC with Headphones and an EQ with speakers I will be spending less money than buying separately.


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## gilency

Bullseye. You mention the DAC in the DEQ2496 as being excellent. I could not find information as to what kind it uses. How do you think it compares with more expensive, dedicated DACs? I am mainly interested in the equalizer feature and in playing with all the digital features.


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## Bullseye

Hey gilency,

 The info I have over the DEQ2496 is here:

Google Translate

 Original here: Matrix-HiFi --> Test Behringer DEQ2496


 Hope it helps


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## gilency

Thank you Bullseye. Very interesting. Now I need to find me a used one at a decent price.


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## a2audiophile

Bumping this thread because I'm still using gerG's EQ for my HD-600s almost a decade since I first discovered Head-Fi (on a long-gone account tied to my AOL email address!).
   
  These days, I'm using foobar2000 with the graphic equalizer plugin, rather than the DEQ2496. Still, gerG's EQ makes a huge difference, and even my non-audiophile friends have been blown away by how much better the cans sound with that EQ curve. I noticed the EQ hyperlink disappeared, though, so here it is again - resurrected by Archive.org:
   
  http://web.archive.org/web/20051102064839/http://members.cox.net/yrogerg/headphones/eq%20update.pdf
   
  Enjoy!


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