Well the DEQ2496 and ECM8000 measurement mic have arrived and I managed to slot the machine into the system today.

http://img.photobucket.com/albums/v637/zippy670/IMGP0544.jpghttp://img.photobucket.com/albums/v637/zippy670/IMGP0547.jpg
http://img.photobucket.com/albums/v637/zippy670/IMGP0536.jpghttp://img.photobucket.com/albums/v637/zippy670/IMGP0538.jpg
http://img.photobucket.com/albums/v637/zippy670/IMGP0540.jpg


Inputs are all made using XLR connectors and fortunately I had two pairs of Chord Silver Siren cables in the spares box that I was able to use by fitting the XLRs to one end. The Behringer manual explains how to configure for single ended use and I experienced no problems at all inserting the unit into the tape monitor circuit of my WAD Pre amp. All levels were fine, possibly because the WAD has pretty high gain (X5) and I even managed to get the DEQ2496 meters to show input clipping a couple of times!

For £200 this is a pretty amazing box of tricks. Those used to handling pro gear may be less overwhelmed than me, but for someone more used to hair shirt domestic Hi-Fi the amount of power and technology in this little box is unbelievable. It will perform its many tricks on an analogue input or you can connect a digital source, e.g. a CD transport and simply use the internal converters. This seems sensible as whatever DAC you choose to run into the analogue line inputs is going to get crunched by the internal converter so why process and convert the same signal twice?

The dot matrix display has good resolution (320x80 rest) and is nice and clear. You access the various functions by selecting pages which are then navigated using the three dials on the front panel.

The first test was to insert the DEQ2496 and determine if there was any negative impact with the thing running flat. Level matching to the source with the unit inserted into the tape monitor was perfect and switching the processor in and out did show some differences but they were surprisingly small - a slight loss of depth and air was noticed but I do mean slight and of a magnitude only apparent with a fast A-B switch. This may seem hard to believe especially as this complex box of chips powered by a SMPS was sitting in the circuit of a tube amp - but that's what I heard.

The ECM8000 measurement mic has an impressively flat frequency response from 20Hz-20kHz and has omni directional sensitivity. I simply placed the mic on a pile of CD's at the listening position to get the correct height and hit the Auto EQ button the magic box...........and this is what it told me...........

http://img.photobucket.com/albums/v637/zippy670/IMGP0525.jpg

Not pretty is it. This is the correction applied by the unit so the actual in-room response before correction will be a mirror image of these results.

It tells me that I have a (not unexpected) problem at 40-50Hz, nothing much below 30Hz and a few less minor peaks and dips further up the range. But look at those bass readings - 10-15db of cut being required to flatten the room response. Just imagine what a 15db boost at those frequencies does to bass guitar fundamentals or synth bass.

Switching between the corrected response and the raw signal is a shock. Not only does bass on certain recordings time much better but the corrected response clearly shows up some previously unnoticed honks and quacks in the mid.

These are very much early impressions and I need to spend a lot more time playing with the new toy but at this stage it clearly seems to work well and doesn't screw up the signal.

http://img.photobucket.com/albums/v637/zippy670/IMGP0533.jpg
http://img.photobucket.com/albums/v637/zippy670/IMGP0531.jpg
http://img.photobucket.com/albums/v637/zippy670/IMGP0529.jpg

And just for Joel - virtual VU meters :D

http://img.photobucket.com/albums/v637/zippy670/IMGP0534.jpg

Good for you, Robert...just played a gig last night in a small club; toned out the room beforehand using the DEQ and it sounded excellent.

The eye candy looks better than the SB2's display, BTW, and is more meaningful. Who's up for a DEQ back-end on a Squeezebox? (requires coax-optical conversion)

I would advise against adding 15db lift below 30hz unless you are running kw power amps. In fact, you may well find that the best results are acheived by merely using attenuation, but that's half the fun of units like these, the discovery and learning.

Enjoy, I'd also shelve down th presence area personally, and you will be surprised the difference +-1db makes in that region.

Robert,

Glad your first impresions are so favourable. I've been doing a bit of reading over the last few days since you flagged this and reckon it might just be what I need to sort the system out. A small room re-arrangement would allow me to bring back the Heresys and the DEQ ought to allow me to tame their response in this room.

Looks as though there is a hell of a lot this little box can do once you know how to drive it. Have you play around with target curves yet or have you simply tried for the flat response?

Regards,

Stuart.

I would advise against adding 15db lift below 30hz unless you are running kw power amps. In fact, you may well find that the best results are acheived by merely using attenuation, but that's half the fun of units like these, the discovery and learning. Enjoy, I'd also shelve down th presence area personally, and you will be surprised the difference +-1db makes in that region.

Agreed, though in effect I do have kw amps because the speakers are 95dbw which makes the 40w amp go a hell of a long way!
I do agree that there is little point in having such raised LF in a small room and I've yet to try the TT which given LF arm/cart resonances will probably be a bad thing.

Looks as though there is a hell of a lot this little box can do once you know how to drive it. Have you play around with target curves yet or have you simply tried for the flat response?

Yes, an extremely versatile box box of tricks. So far I've simply asked it to produce a flat response but as you rightly say any preset target rersponse can be set.

Listening to music and speech radio through the 2496 this morning leaves me feeling that the adjusted response is a little extreme - flat and accurate yes, but the impact on the bass end is dramatic. However, switching the unit into bypass after a few hours of listening reveales the some quite distinct quacks and 'cuppy' colourations from the mid horn of the H2 speakers, something I'd not noticed to any real degree before.
The unit has 60 memories so diffeent responses can be set and recalled instantly. This is great if you have a large room with different seating positions.

What has shocked me is the transparency of the unit. I'd expected the insertion of such a complex circuit to add some hash and grain to the signal, particualrly as I'm used to a straight line class A tube amp, but its effect on the signal when run 'flat' is pretty benign.

One other thing I should mention is that the unit has a sensitivity selector unlike the older unit and if you wire the balanced in/outs for single ended use you halve the output level which makes the unit more useable in a domestic audio system.

I ordered the kit from http://www.dolphinmusic.co.uk who offer next day delivery for £6. The cost of the whole kit - DEQ2496/ECM8000/6M XLR Mic Cable was £275. Don't forget to order the XLR mic cable for the ECM8000.

Robert,

Thanks for the update. I've just been pricing the unit and accessories up this evening. I doubt your supplier will deliver here but I've found a well priced dealer so, depending on getting the right (RCA - XLR) cables, I ought to be up and running for about the same (currency converted) cost you mention.

Just about bed time so I'll leave ordering for tomorrow. Don't suppose you could spin a few records with the DEQ in circuit in the mean time? Just to help dispel any last fears I may have about digitising the whole bloody system!

Thanks,

Stuart.

Robert,

Thanks for the update. I've just been pricing the unit and accessories up this evening. I doubt your supplier will deliver here but I've found a well priced dealer so, depending on getting the right (RCA - XLR) cables, I ought to be up and running for about the same (currency converted) cost you mention.

Just about bed time so I'll leave ordering for tomorrow. Don't suppose you could spin a few records with the DEQ in circuit in the mean time? Just to help dispel any last fears I may have about digitising the whole bloody system!

Thanks,

Stuart.

I'll get the discs spinning Stuart :)

I have the same worries about the dreaded digit which is why I decided to spend £275 on the Behringer before jumping in and blowing £3K on Tact.
The good thing about the Behringer is that it can be completely removed from circuit at the flick of the tape monitor toggle, so in a worst case scenario you could just use it to process problem recordings that set the listening room singing.

Also, if it makes you feel more comfortable about digitising the system remember that the ADC/DAC in the Behringer is 24/96 and the response processors are 32/40 bit, so we're not talking about flakey 16 bit Red Book stuff. And remember, Ivor T couldn't reliably detect an ancient 80's Sony PCM processor in circuit ;)

If only the audio accessory market had gone down this route rather than daft stands and cables and Shakti Stones and other such nonsense.

Surely, this is the future of hi-fi. And all for under £300.

If only the audio accessory market had gone down this route rather than daft stands and cables and Shakti Stones and other such nonsense.

Surely, this is the future of hi-fi. And all for under £300.

Herbert, I thought long and hard about even posting anything about the Behringer on PFM. Like you I feel this is the way to go but I feel incredibly guilty about using such a device and recommending it to others because it goes against accepted wisdom.

What we need to remember is that much of the bad press 'digial' processing received was during the infancy of the technology - twenty - five - years - ago. Just stop and think about that time period for a moment and ask yourself, is it fair to dismiss the benefits of digital processing given that it has received 25 years if development?

The technology inside the Behringer is light years ahead of that used in a 1980's graphic equalizer and if it means that you can use your favoured speakers in less than ideal surroundings, where is the harm?

This is the way to go. I felt I had a 'Jimmy Hughes Moment' yesterday :) - you know the sort of thing, you try something which challenges decades of belief and it works, and works rather well.

Welcome to the New Flat Earth :D

I really must get round to using mine in the system soon. I use it regularly in a live PA context and find it useful at getting the best out of tricky venues. And I agree it seems pretty transparent. The DEQ2496 is meant to be a big step up from its predecessor unit (8024, I think).

I still have a thought about using it with a digital output from the CD transport into the DEQ then taking a digital out from the DEQ into the Benchmark DAC1, to gain the benefit of the latter's D to A convertors.

Andrew

Sorry to say I have not heard a digitally EQ'ed system that retained that elusive X factor that appears to be the corner stone of old fashioned BritFi thinking - nor have I heard a decent vinyl performance from one. And I owned all this stuff a fair while ago now.

Sorry to say I have not heard a digitally EQ'ed system that retained that elusive X factor that appears to be the corner stone of old fashioned BritFi thinking - nor have I heard a decent vinyl performance from one. And I owned all this stuff a fair while ago now.

That depends if you believe that the 'elusive X factor' exists or that it is simply something we convince ourselves exists in order to justify the large sums of money that we've spent on purist BritFi. Much of the X factor is in the mind IMHO.

Can't agree Robert. I really wanted to love my digitally EQ'ed setup, and in many ways I did, but I have to admit I have listened to a lot more music and enjoyed doing so more, since reverting to old technology.

Sorry to say I have not heard a digitally EQ'ed system that retained that elusive X factor that appears to be the corner stone of old fashioned BritFi thinking - nor have I heard a decent vinyl performance from one. And I owned all this stuff a fair while ago now.
As Robert says, I think it's more in the mind than anything. Although as far as hifi goes, the mind is surely the single most important factor...

Fascinating stuff. Just out of interest ... seeing as this looks mostly positive (have you tried it with an analogue input yet?) ... what exactly does a £3k Tact do that this little box of tricks doesn't?

The processing of the Tact looks decidedly basic compared to the DEQ2496 but you do get a 200w digital amp and a nice case thrown in. Someone correct me if I'm wrong but I don't believe the Tact stuff uses 'audiophile' components - looks to be lots of SMD inside those amps.

I've now had the chance to feed it some vinyl and would say that its just as difficult to hear when the unit is in circuit as when using CD or FM radio. Good old analogue vinyl still sounds analogue. The next step will be a digital connection to the CD transport but that means making up more cables so will have to wait.

Someone needs to buy a container full of these units at £200 a go, fit the gubbins into swanky casework, change the XLR socketry to phono, dissable some of the 'pro' features and charge a couple of grand for each unit.

Someone needs to buy a container full of these units at £200 a go, fit the gubbins into swanky casework, change the XLR socketry to phono, dissable some of the 'pro' features and charge a couple of grand for each unit.
And I give you this:
http://www.accuphase.com/dg-38.jpg
http://www.accuphase.com/dg-38_e1.htm
It works, too.

And I give you this:
http://www.accuphase.com/dg-38.jpg
http://www.accuphase.com/dg-38_e1.htm
It works, too.

WOW!

Excuse me for a moment while I wipe the dribble from the keyboard.......

I shudder to think how much that thing costs.

For the equivalent of £200/275 I'm very tempted to try one out. If it doesnt 'retain that elusive X factor' I haven't thrown away any large sum of money.

BTW, can it be used to measure loudspeaker frequency response in a simulated free field environment? Then it's a useful DIY tool to.

JohanR

Accuphase DG-38:
OMG! I don't know if it works but by jove I want one!

The processing of the Tact looks decidedly basic compared to the DEQ2496 but you do get a 200w digital amp and a nice case thrown in. Someone correct me if I'm wrong .

Robert you are wrong. Way wrong. But I'll leave Oedipus to explain why. Have a look at the Tact web site first though - that might help.

I still have a thought about using it with a digital output from the CD transport into the DEQ then taking a digital out from the DEQ into the Benchmark DAC1, to gain the benefit of the latter's D to A convertors.

Andrew


I'd be very interested in that.

Stephen

Rob,

Any further thoughts on this thing? I've got one on order, should be here next week (fingers crossed for Monday).

Regards,

Stuart.

Rob,

Any further thoughts on this thing? I've got one on order, should be here next week (fingers crossed for Monday).

Regards,

Stuart.

Hi Stuart,

Yes, much fiddling and experimenting has taken place over that past week and I've nothing but respect for this device.

One thing I would urge is to read the manual thorouhly - twice.
There are so many settings on this unit that you need to be sure you havn't missed anything and the screen navigation takes some practice before it sinks in.

I made a bit of a boob when I first measured the room response because I had the Behringer set to 'stereo L+R' mode when it should have been set to 'dual mono' as this measures each speaker seperately and produces two correction curves. The in room response of my speakers is starlingly different below 200Hz and getting both reasonably flat has produced excellent results. I can now play albums which really caused problems before - reggae and 'clubby' stuff is now so much better that much head shaking has been taking place. The improvements directly effect PRaT because, for e.g. I can now hear bass guitar notes cleanly and evenly with the 'buzz' of the bass string no longer obscured by room drone.

These are the correction curves for each speaker:

Right:

http://img.photobucket.com/albums/v637/zippy670/IMGP0551.jpg

Left:

http://img.photobucket.com/albums/v637/zippy670/IMGP0548.jpg

I've no doubt those curves give a flat response at the listening position but in all honesty the correction sound a little extreme so I've taken those setting as a base and produced less extreme variations which work really well. I also dont correct below 25Hz and up at 20Khz since there is little point and boosting below 30Hz in a vinyl based system is asking for problems.

I hope you enjoy using the Behringer. You'll wonder how the hell they do it for £200.

i thought we had left graphic equalisers behind in the seventies

large amounts of eq can be acheived with speakers positioning and room treatment.

so many very high q filters must screw the sound up.

try a couple of parametrics to compensate for major room modes.

and see if that retain some of your x factor.

i thought we had left graphic equalisers behind in the seventies

large amounts of eq can be acheived with speakers positioning and room treatment.

so many very high q filters must screw the sound up.

try a couple of parametrics to compensate for major room modes.

and see if that retain some of your x factor.

Yes we did and that was a big mistake IMO.

Speaker positioning can only take things so far and small rooms give limited positioning options.The room is a major influence on the performance of a h-fi system, far, far more important than the differences between competent transistor amplifiers, or cables, or supports.

Yes, the insertion of a box of chips into the signal chain is not in itself desirable but the benefits can be huge.



The Behringer performs all of the EQ in the digital domain and sounds nothing like an old 70s analogue EQ.

What digital inputs does this unit have? From pics I've seen it appears to have AES/EBU input only.... and most CD players have coax digital ouput so can't work with unit. Is there a way round this?

What digital inputs does this unit have? From pics I've seen it appears to have AES/EBU input only.... and most CD players have coax digital ouput so can't work with unit. Is there a way round this?

Yes, according to the manual the 2496 'knows' what type of digital device it is connected to and adjusts. This from the manual:

"Your DEQ2496 features a digital optical interface for the transmission/reception of data in both AES/EBU and S/PDIF formats.

The digital AES/EBU interface (XLR connectors) also sends/receives AES/EBU or S/PDIF signals."

An XLR - Phono cable would be required.

digital filters work in exactly the same way as analogue ones

(just in a different domain)

and digital graphic equalisers have many of the same drawbacks

as 70's analogue ones. Try modelling the phase response given by even three

of these filters.

digital filters work in exactly the same way as analogue ones

(just in a different domain)True, but ... FIR filters are not easily implemented in the analogue domain because they require analogue delay lines. (FIR filters can be implemented without phase distortion.) Digital implementations also don't suffer from the component tolerances of analogue implementations of the same filters (although they do suffer the problems associated with analogue to digital and digital to analogue conversion, of course).

What digital inputs does this unit have? From pics I've seen it appears to have AES/EBU input only.... and most CD players have coax digital ouput so can't work with unit. Is there a way round this?

It has ordinary optical inputs and outputs too which can accept the standard CD output of S/PDIF as well as the professional AES/EBU standard. Most CD players have both optical and coaxial digital outputs.

The AES/EBU interface on the DEQ2496 (on XLR) also accepts S/PDIF format signals such as would be sent from the coaxial digital output of a CD player. Rather confusingly, AES/EBU is the name of a standard as well as being commonly used to describe a type of physical interface, but the electronics within the DEQ2496 allow you to send a consumer format S/PDIF signal through the professional AES/EBU physical interface - this is just an option you select in the I/O menu of the device. This means that you could make up a phono socket to XLR (male) lead to interface directly between a CD player which only has a coaxial S/PDIF output (on phono socket) and the DEQ2496.

Andrew

Andrew,

Have you tried it between your transport and DAC1 yet?

Stephen

After some delay, I should have my hands on a DEQ, microphone and cables tomorrow. Soldering iron at the ready for a quick re-termination.

Wish me luck..

Stuart.

Let us know how you get on Stuart.

Mine sits on the *floor under one of the Sound org stands and is concealed from view. I simply flick it into circuit using the tape monitor when needed.

*Much as it pains me to say this, I still cannot bring myself to place the unit in full view next to my lovely straight line, purist amps - thats decades of ingrained audio snobbery for you!

Well this is interesting.

Got it up and running this afternoon and after a couple of quick checks to ensure that the Rega amps. weren’t going to get mutinous about a flashy new box being put in circuit I started playing.

My first attempt at auto EQ took a massive slice out of the mid range. I’d expected the DEQ to make a cut or two in this area as these ‘speakers are somewhat forward, but not of such a magnitude (approx. -12db). Turns out this was something of a measurement error. I’d stuck the mic. through a hole in the end of a shoe box as a temporary stand, which turned into a nice little resonate structure pumping energy back into the mic.

With mic. stand mk2, made from cushions, the auto EQ results were somewhat more successful, bringing a greater sense of balance to the music. Given the success of this curve, I ploughed forward, switching the DEQ into dual mono mode and set about another auto EQ routine. This has given two markedly different curves, which is to be expected I guess as one ‘speaker is in a corner whilst the other is about 75cm out from it’s corner. There are some significant cuts and gains across the mono curves so I’ll experiment further with ‘speaker positioning later in the week.

The other feature I’ve had a quick play with is the Width function. Given I’ve now got the ‘speakers reasonably far apart, giving a fairly wide image, I was able to compress it into the centre a little (after going silly and turning it mono for a few tracks). Unfortunately, this menu isn’t available in dual mono mode.

I’ve a lot more playing, learning and, most importantly, listening to do yet; however, at this very early stage I’m mighty impressed with this thing. I’ll follow up with some more comprehensive thoughts in a week or so when I’ve got a better understanding of what I’m doing.

Regards,

Stuart.

PS – Rob, I know what you mean about hiding the Behringer from your main boxes. I couldn’t come up with a good hiding place for it so stuck some rubber feet on it and stuck it on top of the pre. I’ll get used to it. The VU meters certainly help :-)

Glad you like it Stuart. For the money this is one amazing little box.
Mic position is critical and I found getting it exactly at ear height and as free standing as possible helps.

I found the Behringer remarkably difficult to detect when in the signal path and set to flat which I guess means that the basic electronics and PSU are pretty sound. What did you think?

I found the Behringer remarkably difficult to detect when in the signal path and set to flat which I guess means that the basic electronics and PSU are pretty sound. What did you think?
Hi,

Yes, surprisingly transparent. I’ve thus far not been able to notice it being added to the circuit when set flat; however I’ve not gone out of my way listen for any minute signal variation.

It can be something of a shock, having listened to a corrected signal for a while, to switch the DEQ out of circuit and hear the ‘speakers in their normal state. My system sounds as though it’s had a significant ‘speaker upgrade with the DEQ. For an all up cost of $650 (including mic. and cables) it is astonishingly good value.

Have you settled on an EQ curve you like, or are you still adjusting?

Regards,

Stuart.

All,

This thread has convinced myself and a friend to invest in the Behringer and try it for ourselves, so it's "in the post" at the moment as a Xmas pressie.

Looking through the manual (phew!), I'd be interested what settings fellow PFM-ers used at some of the key points. For example, what did you choose for :

- GEQ Mode? (True Response or not?)
- Dither? (Off, 16-bit?)
- Noise Shaper? (On, Off?)
- Speed of EQ (Fast, Mid, Slow?)
- Room Correction (On, Off?)

And also, did you process all frequency bands or exclude the low-frequency range (up to 100 Hz) as suggested in the manual?

I'm sure some of these make significant differences .... so your pooled knowledge would no doubt be invaluable in setting this all up when it arrives.

Many thanks in advance for your help and keep posting your findings - it's all good stuff :)

Happy listening.

Chris

A very interesting thread, what system are you using it with?

Keep us updated

PS have you tried mounting the mic on Mana :D

Looking through the manual (phew!), I'd be interested what settings fellow PFM-ers used at some of the key points. For example, what did you choose for :

- GEQ Mode? (True Response or not?)
- Dither? (Off, 16-bit?)
- Noise Shaper? (On, Off?)
- Speed of EQ (Fast, Mid, Slow?)
- Room Correction (On, Off?)

And also, did you process all frequency bands or exclude the low-frequency range (up to 100 Hz) as suggested in the manual?

Chris,

It's early days so I've not really had a chance to play with a lot of this stuff yet, but thus far:
- GEQ mode = True response - don't see the point of letting the software emulate some of the (mis)behaviour of old analogue EQs :-)
- Dither = 24 bit (not tried anything else yet)
- Noise Shaper =
- Speed of EQ = slow
- Room correction = off - IIRC this applies a tilt to the Auto EQ correction, I'll leave that up to my target curve thanks!

Thus far have been processing all bands for auto EQ

Hopefully Rob has experimented with this stuff a bit more. You are likely to loose many many hours with the DEQ, tweaking here and there. Much fun to be had.

Regards,

Stuart.

All of this is utterly fascinating. A few questions if I may.

Firstly, does the Behringer have the ability to 'average' a number of different readings, such as might be useful if you wanted to programme in a setting which would accommodate a number of different listening positions?

Secondly, you appear to be using it between source and preamp. Have you tried between preamp and power amp (I don't know your system so maybe that's not possible).

How easy is it to select 'programmes'? FOr instance, I do a fair bit of listening from the sofa, and a fair bit from the recliner. Both have different sounds, and I'd ideally want it to be easy to swap between the two.

Can you get it without the mike? I already have an sE large diaphragm condensor which is flat to some ridiculously high frequency (40KHz+ I believe) and if I can buy one of these units without the mike and save a few bob then great. Having just written this, I then started to ponder whether it may introduce a variable into the operation that's perhaps not wise...

Lastly, I'd like to try one out. Is there anyone relatively nearby who has one who might be willing to loan me theirs for a day or two? My new speakers are a huge advance on previous speakers, but being big, there's more bass than I'm used to and I am concerned that it's a room interaction thing. Something like this would surely help more than days of mucking about in Cara or tweaking this, that and the other...

Ta,

John

I have an idea to let us other have an impression of what the box can do:

If one of the owners have the possibility to record digitally?

- With a microphone (it can be mono) record the sound of music being played, in room and from the speakers, whithout EQ/room correction.
- The same thing with EQ/room correction.
- post the two files here. They only have to be, say, 20 seconds each.

Would be nice.

JohanR

Firstly, does the Behringer have the ability to 'average' a number of different readings, such as might be useful if you wanted to programme in a setting which would accommodate a number of different listening positions? Not that I'm aware of; however I guess you could record x number of curves and manually average and enter this as a new target curve.

Secondly, you appear to be using it between source and preamp. Have you tried between preamp and power amp (I don't know your system so maybe that's not possible). Mine is currently in one of the tape loops for easy in/out of circuit comparisons. I may eventually put it between pre and power but I don't currently see this as a priority. A possible benefit of placing the DEQ between pre and power would be to take advantage of dynamic EQ settings - eg. pre defined 'loudness' compensation at low volumes and/or 'save my 'speakers/ears' adjustments for high volume.

How easy is it to select 'programmes'? FOr instance, I do a fair bit of listening from the sofa, and a fair bit from the recliner. Both have different sounds, and I'd ideally want it to be easy to swap between the two. You can store up to 64(?) curves in memory and re-call them as you see fit. I'm currently playing with two different set ups (only on day 3) and have plans for a few more as I get used to it. A remote would be really nice for this!

Can you get it without the mike? I already have an sE large diaphragm condensor which is flat to some ridiculously high frequency (40KHz+ I believe) and if I can buy one of these units without the mike and save a few bob then great. Having just written this, I then started to ponder whether it may introduce a variable into the operation that's perhaps not wise... Each of the stores I looked at sold them seperately, infact I don't recall anyone doing a deal combining the two. Check the specs of your mic against the DEQ manual and take it from there. The ECM 8000 mic is not expensive.

Lastly, I'd like to try one out. Is there anyone relatively nearby who has one who might be willing to loan me theirs for a day or two? My new speakers are a huge advance on previous speakers, but being big, there's more bass than I'm used to and I am concerned that it's a room interaction thing. Something like this would surely help more than days of mucking about in Cara or tweaking this, that and the other... Bloody long drive from there to here so 'fraid I can't help you there. Check out pro audio stores in your area, you may find one that hires them out.

If one of the owners have the possibility to record digitally? Eeek! Good idea but may be a wee bit beyond me at the moment.

Regards,

Stuart.

A very interesting thread, what system are you using it with?
LP12/Valhalla/Ittok/DV17d2 mk2/Trichord Dino
Squeezebox 3 / Non OS dac
Old Pioneer tuner
Rega Cursa and Maia
Klipsch Heresy II (treble and mid horns with 12" bass driver in a small (relatively) box)

PS have you tried mounting the mic on Mana :D None of that girly stuff here - only the finest pillows for my microphone!

Regards,

Stuart.

Chris,

It's early days so I've not really had a chance to play with a lot of this stuff yet, but thus far:
- GEQ mode = True response - don't see the point of letting the software emulate some of the (mis)behaviour of old analogue EQs :-)
- Dither = 24 bit (not tried anything else yet)
- Noise Shaper =
- Speed of EQ = slow
- Room correction = off - IIRC this applies a tilt to the Auto EQ correction, I'll leave that up to my target curve thanks!

Thus far have been processing all bands for auto EQ

Hopefully Rob has experimented with this stuff a bit more. You are likely to loose many many hours with the DEQ, tweaking here and there. Much fun to be had.

Regards,

Stuart.

Thanks very much for that Stuart (I assume that Noise Shaper is off since you left it blank) - you've certainly given me a head start in setting it up myself.

Rob - if you're still around on this thread, any further insights into settings would also be much appreciated - even just reading the manual I can't believe how much configuration is possible :)

Happy listening.

Chris

The Behringer is getting a lot of good user reviews on other forums as well,as a decent,cheap,and easy to use method of getting digital room EQ,and certainly beats my own solution on price grounds.

My own impression of well implemnted digital room EQ is that it can make an otherwise unsuitable room into something that becomes a joy to use,and having used a variety of strictly minimalist systems over the years,it's been one of the biggest single upgrades I've heard and used in ages.

What does this contraption sound like being put in bypass & inserted by a third party (one's She Who Must Be Obeyed mebby?) as you move about??? My experience is that the 'speaker/room interaction is best tweaked by adjusting the positioning of the LS's then rearranging the rest of the contents of the listening space, as the soundfield recreation relies on the frequency/polar diagram of the transducers. The designer probably did a better overall job than yonder gadget does. Mind you, Beringer kit (designed in Germany & screwed together in China) gives fanbloodytastic value for money. H.

As I've said, I find this device pretty hard to detect when in bypass mode and yes Harry it is bloody fantastic VFM.

I use in dual mono mode but ended up using the measured results as a guide and used less extreme correction - about half of the indicated results and only below 200Hz. Full-on correction, while I'm sure it measures superbly, sound a bit gutless and flat. Mine runs in the tape monitor circuit and I only place it in circuit on 'difficult' recording that give the room problems - reggae, OTT EQ'd rock/pop etc, which means its in circuit about 30% of the time.

I only use the room correction module and have not dabbled with any of the other setting.

Harry, while I agree that the best way to tackle room sound is at source, ie the speaker positioning within the room, there are limits to what that can achieve. If a small room gives 13db of lift at 50Hz you have to resort to more drastic measures to get correction.

All I would say to those who turn their noses up at the EQ solution is that if it allows you to listen to music that is otherwise pretty much unplayable, how on earth can that be a bad thing?

Something that has been causing members of the BFD Fan Club to stick it in the bin is the recent arrival of the stand-alone unit that is built into Velodyne's DD range, called the Velodyne SMS-1. It does a similar thing, only more and better :)

As I've said, I find this device pretty hard to detect when in bypass mode and yes Harry it is bloody fantastic VFM.

I use in dual mono mode but ended up using the measured results as a guide and used less extreme correction - about half of the indicated results and only below 200Hz. Full-on correction, while I'm sure it measures superbly, sound a bit gutless and flat. Mine runs in the tape monitor circuit and I only place it in circuit on 'difficult' recording that give the room problems - reggae, OTT EQ'd rock/pop etc, which means its in circuit about 30% of the time.

I only use the room correction module and have not dabbled with any of the other setting.

Thanks for that info - most useful. I'll be picking mine up soon, so I've now got plenty of suggestions to help me. Thanks to all for that.

One last point - when you say that you haven't dabbled with the other settings, does that mean you've just left things like the GEQ mode, Dither, Noise Shaper, Speed of EQ etc. settings as their factory defaults?

Happy listening.

Chris

Having owned a TACT RCS 2.0 , recently sold, I have to add a few points:

use in dual mono mode but ended up using the measured results as a guide and used less extreme correction - about half of the indicated results and only below 200Hz. Full-on correction, while I'm sure it measures superbly, sound a bit gutless and flat.

This is exactly the problem. I also started from full correction, then moved to half-correction, then removed it because it was too bland as well. (and also to simplify the system for use by family).

I don't know the Behringer, but my problem with the TACT was that the things I liked about my setup were also corrected. Maybe I just couldn't find a good enough correction "graph", maybe it's my inability to force the TACT to leave everything unchanged except for the 80hz problem (something possible with the expensive DEQX), maybe ?

One thing, when I moved from Naim amplification, the resonance problem got much better. Later after upgrading to my current big amp, the problem is now 80% solved and is not the big issue it's been as everything sounds so much involving.

Omer.

Having got the DEQ2496 now and had a brief play, I have got a couple of questions for others who are more familiar with it ....

1. The Auto EQ only seems to work through the Analogue Outputs - I just get Pink Noise through the XLR Digital output - any ideas why this might be the case?

2. Should the Auto EQ "finish" after a while, or do you just need to push the "Stop EQ" button once it has settled down on an equalisation?

3. I have selected Dual-Mono but when I do the Auto EQ, I still get noise out of both channels for each equalisation - is this right, or am I supposed to set the balance on my pre-amp to one channel at a time?

I may have more questions as I get more into the functionality, but any help that fellow PFMers could give would be much appreciated.

Thanks again and happy listening.

Chris

1. The Auto EQ only seems to work through the Analogue Outputs - I just get Pink Noise through the XLR Digital output - any ideas why this might be the case?Not tried the digi. outs myself however it ought to work. Have you checked the configuration in the I/O menu?

2. Should the Auto EQ "finish" after a while, or do you just need to push the "Stop EQ" button once it has settled down on an equalisation?You stop it when you either get bored or reckon it looks about done.

3. I have selected Dual-Mono but when I do the Auto EQ, I still get noise out of both channels for each equalisation - is this right, or am I supposed to set the balance on my pre-amp to one channel at a time?Hmmmm, not my experience. Once again, well worth checking the I/O configuration. You have to run the dual mono eqing twice, once for each channel, using the 'A' key (I think) to swap channels before setting the target curve (again). I'm not convinced that dual mono is the way to go, at least in my system and with less than a week's worth of nerding under my belt.

Take your time to learn the unit as there's an awful lot you can do with it, and different approaches to take. One of the key things to keep in mind in your first experiments is to pay attention to the gain structure. For example, if you are adding up to 6db of eq in places then you ought to reduce the output by 6db to offset this (keep an eye out for the clip indicators as this can alert you to a potential gain issue). You probably don't want to exceed the power handling limits of your 'speakers.

Regards,

Stuart.

Chris, Stuart,

- Isn't the pink noise needed to actually do the Auto EQ? That's all that comes out of my digital o/p too - what else would you want?

- Mine never seems to finish, but I've realised that you can set the display to show the mic input so you can watch what it's picking up as the Auto EQ makes the changes. Seems to take up to a minute, but you'll see when it finally goes flat.

- I've tried the separate left/right auto EQ but it looked like I was getting odd results - the two adjustment curves seemed to be almost the opposite of each other. This got me thinking about the pink noise coming from both channels while EQing one, i.e. when the second channel is being done you might also be hearing the first channel in its already EQ'd state. Can't quite get my head around what difference that would make.....


Also, what's the thinking behind the 'Room Correction' option? It adds a tilt in favour of the bass but why? Surely the Auto EQ is actually the room correction.

Great piece of kit though. I can finally play the title track from Massive Attack's Protection without the suspended floor joining in.....

Stephen

I have yet to use the digital outputs so cannot comment on what is on the output.

As has been said above, the Auto EQ just keeps running until you stop it so just wait about a minute until the displayed curve stops moving around and settles into a nice flat line.

In my system dual mono definitely works best though both options seem to apply too much correction. I'm sure the final response is flat and technically correct but it sound a little overdone. I use the auto generated curve as a base and then back of to a less extreme curve, though following the pattern of the original.

Great piece of kit though. I can finally play the title track from Massive Attack's Protection without the suspended floor joining in.....

Stephen

I know exactly what you mean Stephen.
I bought the new Madonna single yesterday and it is unplayable without the EQ on. Great record BTW :)

My room honks a bit on a low A so I put a row of coat hooks (oak and brass, 14.99 from Robert Dyas) in the corner next to the door and a selection of anoraks cleaned things up nicely...Nodes and Antinodes; Eigentones & all that bollox, what. For lousily EQ'd source material the equiliser in yon PowerBook G4 is no slouch either. H.

Not tried the digi. outs myself however it ought to work. Have you checked the configuration in the I/O menu?

Hmmmm, not my experience. Once again, well worth checking the I/O configuration. You have to run the dual mono eqing twice, once for each channel, using the 'A' key (I think) to swap channels before setting the target curve (again). I'm not convinced that dual mono is the way to go, at least in my system and with less than a week's worth of nerding under my belt.


Thanks again for all the PFMers help - it does sound like my unit is not doing the right things - I'm slightly concerned it may be faulty, to be honest. The fact that it's outputting noise on both channels for the dual mono mode doesn't bode well, in my opinion, and I can't see anything obvious in the I/O configuration that would fix this. Stuart - sounds like you get noise in just the channel you're correcting - what are your I/O settings?

In fact, I can't find anywhere in the I/O menu where you configure the digital output - so how does it know you want it to output 44.1Khz S/PDIF format? It works when I'm playing my CD player into it - since it's copying the 44.1Khz S/PDIF input for the output - but otherwise how would it know? For example, when I'm doing the Auto EQ, the CD player isn't on (and presumably it automatically selects the input as the internal pink noise generator), so how does it know what digital output I want?

Do I need to select "Pink Noise" in the I/O section before I do Auto EQ? And then, perhaps, manually select the sampling frequency? I have to say, none of this is really clear in the manual and I've done quite a lot of trial and error (which lead me to the faulty unit concerns).

Oh, and I appreciate the confusion I could when I said that the "Auto EQ only seems to work through the Analogue Outputs - I just get Pink Noise through the XLR Digital output" - I meant to say that I just get this distorted crackling sound that the unit doesn't recognise - it sounds totally different to the actual pink noise that I get out of the Analogue Outputs and the unit itself recognises to do the analysis.

The trouble is wanting to call Behringer and get some technical support during the day when I'm at work and not with the unit - I'm sure they'll want to walk through things with me. Maybe I'll just e-mail them and see what they say .... but if anyone here has some words of wisdom, as ever they'd be much appreciated.

Thanks again.

Chris

Hi Chris,

I'll have a crack at your questions but in so doing, please keep in mind I've only had the unit a week today and still have a lot to learn. So with that caveat out of the way -

...The fact that it's outputting noise on both channels for the dual mono mode doesn't bode well, in my opinion, and I can't see anything obvious in the I/O configuration that would fix this. Stuart - sounds like you get noise in just the channel you're correcting - what are your I/O settings?
I/O settings as follow:
pg1 - Input - Main In, Clock =96Khz, Noise Gain -60 (meaning it has to be turned up when in the Auto EQ menu)
pg2 - Aux/Dig out - GEQ+PEQ+DYN+Width to Aux out & Dig out, S/PDIF selected (A key), Dither off (upper small knob), Noise shaper off (B key)
pg3 - RTA Input - Main in (so has to be switched to Mic in RTA/Auto EQ menu)
pg4 - dig. delay - not used/default

In fact, I can't find anywhere in the I/O menu where you configure the digital output - so how does it know you want it to output 44.1Khz S/PDIF format? It works when I'm playing my CD player into it - since it's copying the 44.1Khz S/PDIF input for the output - but otherwise how would it know? For example, when I'm doing the Auto EQ, the CD player isn't on (and presumably it automatically selects the input as the internal pink noise generator), so how does it know what digital output I want?
Page 1 of the I/O menu allows you to set the sample rate, however if you are using dig. input then the output will sync to the input sample rate). Page 2 of the I/O menu allows you to set where in the chain of possibilities the dig. output will pick up it's signal. Set it as I have mine above ie. stick it at the end of the chain.

Do I need to select "Pink Noise" in the I/O section before I do Auto EQ? I don't, it ought to switch in when you go to Auto EQ menu; however, if you have the noise gain in page 1 of the I/O menu set to -60 as I do, then you will get no sound when going into Auto EQ mode and will have to increase the gain via the lower (?) smaller knob.
And then, perhaps, manually select the sampling frequency? I have to say, none of this is really clear in the manual and I've done quite a lot of trial and error (which lead me to the faulty unit concerns).The sample frequency should (can?) only be set from the I/O menu. I'm sure that if you have the unit configured to dual mono (via the Utility menu) that when you enter Auto EQ and increase the noise gain, you ought to be able to switch between left and right channels using either the A or B key. I'll happily try it out tomorrow evening after work (bit late now to fuck around) to confirm this if you are still having difficulty.

Oh, and I appreciate the confusion I could when I said that the "Auto EQ only seems to work through the Analogue Outputs - I just get Pink Noise through the XLR Digital output" - I meant to say that I just get this distorted crackling sound that the unit doesn't recognise - it sounds totally different to the actual pink noise that I get out of the Analogue Outputs and the unit itself recognises to do the analysis. Cool, I get what you mean now. Check the config. as outlined above and try again. If this still doesn't work, you also ought to check the gain switch (Max switch in manual) on the back panel.

May be worth while drawing out the connections and configurations you want on paper, then reading the manual and making notes for step by step configuration, then having a glass (or two) of wine before making the connections and working through the menus. May sound a bit involved but probably worth while to eliminate user error before contacting your vendor or Behringer :-)

Regards,

Stuart.

I/O settings as follow ...

I'll happily try it out tomorrow evening after work (bit late now to fuck around) to confirm this if you are still having difficulty.

May be worth while drawing out the connections and configurations you want on paper, then reading the manual and making notes for step by step configuration, then having a glass (or two) of wine before making the connections and working through the menus. May sound a bit involved but probably worth while to eliminate user error before contacting your vendor or Behringer :-)


Hi Stuart,

Many thanks for getting back to me - your post makes complete sense and your settings look remarkably similar to my own, but I'll have a play around tonight after making some notes as you suggest and report back.

Especially thank you for your offer to try out the dual mono stuff if I'm still having problems - that would be ideal to prove if I have got a problem before I go back to the vendor or Behringer - I'll let you know.

Where would I be without helpful PFMers? :)

Chris

Where would I be without helpful PFMers? :)


Hi Stuart,

Well, your advice was perfect - I copied your settings and it worked straight away. I then started adding back my previous settings to find out what the problem was.

Turns out that if you've got the Digital XLR input selected as the input on the I/O, it messes up the Auto EQ function via the Microphone. Possibly something to do with the Microphone being Analogue, not Digital - who knows - but as long as you change that setting when you want to do an Auto EQ, it works a treat.

This also sorts my left and right issue (the pink noise from the Auto EQ now only comes out of one channel at a time) and also my Digital Output issue (I could hear the pink noise properly through my Digital Output during the Auto EQ).

So, I'm now able to start playing with the unit properly and trying different equalisations and settings etc., safe in the knowledge that my unit is working properly.

Thanks once again for all your help - really very much appreciated.

Happy listening.

Chris

Chris,

I get pink noise out of both channels on mine while in dual mono mode, although as I mentioned above, I'm not sure all is well with this method.

I don't know if the analogue outs are any different as I'm only using it digitally.

Does your system sound right after the changes it's made to the EQ?

Stephen

Chris,

Hang on, I was writing that as you posted.

As I seem to be having this same problem, although am not really sure what it is, can you explain to me what I need to do? (And what the problem is!)

Cheers,

Stephen

As I seem to be having this same problem, although am not really sure what it is, can you explain to me what I need to do? (And what the problem is!)


Stephen,

I found that the problem is I was using it all digitally too and that when you're got a digital input selected on the first page of the I/O menu, then it doesn't measure things properly on the Auto EQ (with pink noise coming out of both channels).

So, the thing to do is to go to the first page of the I/O menu and select "Main In" rather than one of the "Dig. In" options. Then go through the Auto EQ process one channel at a time. You should now notice that it only has pink noise coming through one channel at a time (hence not messing up the results).

Once you've finished the Auto EQ process, you go back to the I/O menu and re-select the Dig. In input of your choice so you can actually use the unit digitally.

I hope that makes sense - feel free to post again if you need more explanation or still have problems - I'm happy to be able to help someone rather than just post my own problems this time :)

As to the results .... much improved once you're doing it right!

Happy listening.

Chris

Hi Chris,

Thanks for the info - I will try that.

Just to check: despite changing the input, are you still running the Auto EQ through the digital output?

Cheers,

Stephen

Just to check: despite changing the input, are you still running the Auto EQ through the digital output?


Yes - once I'd changed the input to "Main In", I was able to listen through the digital output with no problems (something that hadn't worked before).

Let me know how you get on.

Chris

Excellent! Glad you got it sorted.

One final question to all DEQ2496 users - how loud do you have the noise coming through the speakers when you do the Auto EQ?

I know there are settings on the unit itself to adjust the volume, but ultimately it depends how loud you've got the volume up on your system - so how loud are people going? Is louder better?

Chris

One final question to all DEQ2496 users - how loud do you have the noise coming through the speakers when you do the Auto EQ?

I know there are settings on the unit itself to adjust the volume, but ultimately it depends how loud you've got the volume up on your system - so how loud are people going? Is louder better?

Chris

I had it pretty loud. The unit will warn if the sound level is too low.

I had it pretty loud. The unit will warn if the sound level is too low.

Yes - I started out too low and then cranked it up until the unit stopped warning me. However, I wondered if there was benefit from going louder still - I'll give it a go anyway and see what happens!

Also, are there any users of the ECM8000 microphone for this? I'm assuming I should be pointing the microphone horizontally at the gap in between my speakers where I would normally sit? I shouldn't be having the microphone vertical or anything daft like that, should I? (you sometimes see microphones in recording studios being mounted vertically - just wondering if that was the case here).

Chris

The ECM 8000 is an omnidirectional mic, so it doesn't matter which way you point it (provided it is kept away from boundaries/surfaces). Just put it where you would normally sit at something close to your seated ear height.

You might afterwards try alternative mic positions (where the other seats are) just to see the effects of room modes etc.

Andrew

The ECM 8000 is an omnidirectional mic, so it doesn't matter which way you point it (provided it is kept away from boundaries/surfaces). Just put it where you would normally sit at something close to your seated ear height.

You might afterwards try alternative mic positions (where the other seats are) just to see the effects of room modes etc.

Andrew

Thanks Andrew - suspected that was the case, but good to have it confirmed.

Chris

One final question to all DEQ2496 users - how loud do you have the noise coming through the speakers when you do the Auto EQ?



I have the DEQ set at 0db and the volume on my (your old) Kairn at 60. Quite loud for a minute or so of pink noise. Feels like a plane is about to land on the house. Room issues are greater at higher volumes so I didn't see the point of auto EQing at lower volumes.

The ECM package seems to demonstrate how wonderfully omnidirectional the mic is so I've been using it pointing upwards.

Stephen

Room issues are greater at higher volumes so I didn't see the point of auto EQing at lower volumes.


Although just thinking about that for another moment it does mean that I would be over-compensating at lower volumes. Hmmm...........

Although just thinking about that for another moment it does mean that I would be over-compensating at lower volumes. Hmmm...........

I suspect I'm going to measure it at "medium" volumes then :)

Did my advice re: switching the I/O from Dig.In to Main In fix the "pink noise from both speakers" issue Stephen?

Chris

Chris,

I haven't had a chance to try it yet but will do as soon as I get a moment to play. Thanks for the info - I'll report back.....

Stephen

One final question to all DEQ2496 users - how loud do you have the noise coming through the speakers when you do the Auto EQ?
Hi,

I set the gain to zero on the DEQ and use the vol. control on the pre to adjust. I simply try to find a median listening level and use that for the EQing. Certainly not pushing out 100db of pink noise whilst I'm in the room!

As an aside, I thought the honeymoon period had ended last night. I was playing some records and expecting but not getting the same magic I'd heard on Sunday. I was working on all sorts of theories about why it sounded so ordinary and was contemplating re-setting the unit for a fresh attempt at getting a good result and mentally berating myself for buying such a cheap solution.

Then I noticed that the tape loop wasn't switched in...

I'd been listening to the system in its un-corrected state. It is a measure of the positive effect the DEQ has in my system that I now find the un-corrected balance of my system a bit off putting.

Still amazed by the DEQ.

Stuart.

Harry's room eq NB-work in progress. Old calrec desk EQ module (i.e. paralytic eq as we oiks call it) followed by width control (i.e. differential gain of sum and difference signal). Tweak 'till happy. If bothered, put through Dolby Pro-Logic for 100% ANALOGUE surrond. Do no pass go. Do not collect 200 euros Patent pending when I can be arsed to butcher an old Hi-Cap & get the whole caboodle up and running. AtoD & DtoA; I spit on you both... Cheers HBS.
:)

How long are people finding it takes the Auto EQ to settle down? I'm finding its still moving the dots around after 10 mins! which seems a bit extreme!

I set the gain to zero on the DEQ and use the vol. control on the pre to adjust.


Stuart,

I found that setting the gain to zero on the DEQ caused it to occasionally "clip" (with the red clip light coming on a bit) - you might want to check that and re-calibrate with the gain set to, say, -10dB and just boost the pre-amp.

I've been using -10dB gain on the DEQ2496 and Volume 70 on the 5103.

Also, I turned off all the things I wasn't using on my DEQ2496 e.g. I selected the "GEQ/PEQ" option on Page 2 of the I/O settings (instead of the "Width" option) and then went to the Bypass menu and switched everything off except the GEQ module.

This definitely improved the sound for me - I'd certainly recommend that all you DEQ2496 users out there give this a go. Of course, it shouldn't make a difference - but it does.

The experiment continues .....

Chris

How long are people finding it takes the Auto EQ to settle down? I'm finding its still moving the dots around after 10 mins! which seems a bit extreme!

I've found that it settles much quicker if you are using a higher volume during the Auto EQ phase.

That said, it does seem to keep moving the dots around, even after 10 minutes, although it's a generally settled curve. I guess given that the whole algorithm it's using is a trial and error method, this is unavoidable. But after a couple of minutes, most of the job seems to be done and the response curve seems broadly flat.

Hope that helps.

Chris

After 2 hours fiddling the difference this is making is incredible.... I can get a much cleaner sound with different voices much more defined in the soundstage than before. I have my CD player connected to the DEQ from its balanced outputs and also direct to my amp from its single-ended outputs - I've never noticed much difference between them before (apart from levels which I can adjust on my amp anyway).

Switching back to the direct feed from the CD player... the sound seems smeared untidily across the soundstage, although it does have a bit more warmth and bloom.

I have EQ'd simply setting it for Auto EQ (not Room correction) and setting a completely flat default target curve, not touching below 100Hz.

I only got the unit today, I'm fairly gobsmacked at the moment!

I seem to have a hole in my normal response where there is a sharp dip at about 120Hz, the correction isn't doing much to fix this.

Alos the results for left and right speaker are quite markedly different, the left side needs a lot more EQ applied..... may try to get some pictures.

After 2 hours fiddling the difference this is making is incredible.... I can get a much cleaner sound with different voices much more defined in the soundstage than before. I have my CD player connected to the DEQ from its balanced outputs and also direct to my amp from its single-ended outputs - I've never noticed much difference between them before (apart from levels which I can adjust on my amp anyway).

Switching back to the direct feed from the CD player... the sound seems smeared untidily across the soundstage, although it does have a bit more warmth and bloom.

I have EQ'd simply setting it for Auto EQ (not Room correction) and setting a completely flat default target curve, not touching below 100Hz.

I only got the unit today, I'm fairly gobsmacked at the moment!

I seem to have a hole in my normal response where there is a sharp dip at about 120Hz, the correction isn't doing much to fix this.

Alos the results for left and right speaker are quite markedly different, the left side needs a lot more EQ applied..... may try to get some pictures.

<sits with smug "I told you so" grin on face> :)

Good isn't it :)

Let the EQ run down to 30Hz - it does work despite the warnings in the manual.

I found that setting the gain to zero on the DEQ caused it to occasionally "clip" (with the red clip light coming on a bit) - you might want to check that and re-calibrate with the gain set to, say, -10dB and just boost the pre-amp.
Thanks, I'd not noticed that. Will check it out later.

Also, I turned off all the things I wasn't using on my DEQ2496 e.g. I selected the "GEQ/PEQ" option on Page 2 of the I/O settings (instead of the "Width" option) and then went to the Bypass menu and switched everything off except the GEQ module.
Interesting; however, I use all 3 modules you mention!

Does anyone know the correct procedure for adding two saved curves from memory? I've somehow muddled through it but not sure I could repeat.

Thanks,

Stuart.

[QUOTE=RobertLet the EQ run down to 30Hz - it does work despite the warnings in the manual.[/QUOTE]

I've tried this but it doesn't seem to work well for me at all - the measurements I get for the 6-7 lowest frequencies oscillate wildly between large +ive and -ive gains and when you listen to the result, most of the bass seems to have been sucked out of the room. Most disappointing.

To be fair, I don't have a problem with boomy bass in my room, so that wasn't the point of getting the unit, but I'm confused as to why it's doing so badly with correcting the bass.

Chris

Interesting; however, I use all 3 modules you mention!

Does anyone know the correct procedure for adding two saved curves from memory? I've somehow muddled through it but not sure I could repeat.


Stuart - interesting that you're using the other modules. How have you set these up? It would be great if you could post your settings like you did before for the I/O section so I could give them a go :)

The "adding curves" bit - isn't that where you go to the Memory menu, go to page 2 and then use the bottom small data wheel to choose the "G" option and then the top small data wheel to choose the "Add" option? (that's all from memory - let me know if you'd like me to check this when I'm back home).

Chris

Let the EQ run down to 30Hz - it does work despite the warnings in the manual.

I tried this on my first attempt - but ended up with a cold dead dessicated sound that was no good at all. Will try it again though, now using the EQ on slow setting and limited delta and span, seems to produce better results for me.

The differences between un-EQ'd and EQ last night were so big I decided there must be something wrong with my speakers (ATC SCM20). have just tried it again with ATC SCM12s and the difference is just as pronounced. The main benefit subjectively seems to me to be in the treble. Without EQ its just plain wrong - though before I thought it was fine.

Stuart - interesting that you're using the other modules. How have you set these up? It would be great if you could post your settings like you did before for the I/O section so I could give them a go :)

Basically I've used the para. eq to correct for the big stuff. To do this I ran a basic Auto EQ to flatten the response and looked for any big -ve adjustments and transferred (roughly and manually) correction to the para. eq. With my room/'speakers this was a peak around 63Hz where I've applied -10db @ 1/4, ideally this would be broader but smacks up against a null looking bit at 100Hz. Also a broad swathe from the mid (over eager horn) where I've applied a -3db 1/4 BW cut centred at 1002Hz and -6db 3/4 BW cut centred at 1261 Hz. Then I've re-run auto eq to return balance to flat (well, flatish. Gentle top and bottom roll off programed in)

The theory is that this, when refined a bit more, will form my base correction to which I can add/subtract whatever overlay curve takes my fancy eg. BBC dip. Should allow me to play around a fair bit when I feel like it without having to regularly re-run the Auto eq function.

I use the Width control to simply shift everything to the right a bit as I usually sit a little off centre. Sometimes I use it to shift to mono. Just basic stuff that a simple balance control and mono switch on the pre would otherwise have sorted.



The "adding curves" bit - isn't that where you go to the Memory menu, go to page 2 and then use the bottom small data wheel to choose the "G" option and then the top small data wheel to choose the "Add" option? (that's all from memory - let me know if you'd like me to check this when I'm back home). Thanks, I'll play again tomorrow and approach it with a clear mind and the manual in hand!

Regards,

Stuart.

I thought I'd post my latest findings while getting to grips with the DEQ2496.

Firstly, I've found that it sounds better using the Analogue Input and Outputs compared to the Digital ones.

I guessing that this might be due to my digital set-up being all 16 bit / 44.1 Khz and the DEQ2496 preferring to just start from a 24 bit / 96 Khz signal internally - but there's no doubt that the Analogue Input/Output gives the sound much of its "life" back - it was sounding somewhat dead through the digital set-up.

Secondly, when doing Auto EQ, you should definitely reduce the "Delta Max" parameter on page 3 of the AEQ menu from +15dB to something more sensible, like +6dB. This is particular of relevance if you are equalising the bass frequencies, where otherwise the unit will swing from -15dB to +15dB too quickly across the frequencies and then the unit is programmed to stop analysing these frequencies to get it right. I suspect this might be the cause of some people complaining that when they tried to correct the bass frequencies, they ended up with a cold / dead sound.

Finally, it's important to reduce the gain offest in the Utility menu to ensure you don't get clipping. For me, -10 dB was plenty.

I'll post more as I continue to discover how best to use this unit .... but it's just getting better and better the more time I invest in trying all the combinations :)

Happy listening.

Chris

I still have not tried EQ'ing below 100Hz.... maybe tomorrow. I too found that reducing the delta max to 5db in my case helped get a better result.

As things stand at present, the benefits are enormous. I simply cannot now listen to my system without this being in the loop, otherwise it sounds wrong. Even SACDs played through the analogue inputs are enhanced, I worried that that A-D conversion and back to PCM digital would kill proper DSD SACDs but it hasn't, which augurs well for vinyl. I have tried the same on a Tact TDA2200 - playing an SACD into the A-D inputs - and the Tact killed it.

Anyone tried feeding this into a 96KHz capable DAC yet? Any improvement?

Mine's going into a DAC1 via AES. Haven't had much time to play with it the last week or so though so must admit to not getting much further with it. But isn't the output sampling rate automatically the same as the input rate? Or can this be changed?

Stephen

Secondly, when doing Auto EQ, you should definitely reduce the "Delta Max" parameter on page 3 of the AEQ menu from +15dB to something more sensible, like +6dB...
I manually take frequency boost back to a max of +6db post Auto EQ. Doing it this way allows the system to cut by greater than 6db where necessary; although if you've got the para eq set appropriately this should, I suspect, be very rare.

As things stand at present, the benefits are enormous. I simply cannot now listen to my system without this being in the loop, otherwise it sounds wrong.
Yep, I know exactly what you mean. I have actually found the improvement most noticeable when playing vinyl, which is nice as I was somewhat worried about the effect AD and DA conversion would have on it - infact I refrained from ordering the DEQ until Rob confirmed earlier in the thread that vinyl sounded fine through it! Getting a more balanced sound in the room has allowed the record player to really show what it can extract from the groove.

I'm so happy with the sound I'm currently getting that I've stoped nerding around with the DEQ. I'm pretty sure there is more to come through further experimentation but it currently sounds so good I just keep listening to music.

Regards,

Stuart.

Mine's going into a DAC1 via AES. Haven't had much time to play with it the last week or so though so must admit to not getting much further with it. But isn't the output sampling rate automatically the same as the input rate? Or can this be changed?

Stephen,

I think you're right - if you've got a digital input, then the output mirrors the same frequency.

You could try taking an analogue input into your DEQ2496 and then setting the output frequency to the higher sampling rate - as I've said, I've found using the analogue inputs to be superior to the digital ones.

Worth trying when you've got a spare few hours to play with :)

Chris

I've got one of these, in between my Squeezebox 2 and Panasonic digital amp, hooked together with optical cables. After a bit of head scratching I've set it up with a basic automatic correction, down to (I think) 50Hz.

Worth having. It sounds much warmer and valvey, which I suppose is the opposite of what I expected. Texture galore, no bass weirdnes.

Anyone else Thorsten Loesch's suggested curve? In the last third of this article:

http://www.prijsindex.net/tmp/room%20acoustics%20and%20eq.html

Not quite sure what he means by "place a -3db dip at 2.5KHz with a 3db/octave slope." Either side? One way?

It mixes the classic BBC curve and boosts the bands that contain a lot of spatial information. I really like it, it makes the sound lusher and detaches the soundfield from the speakers, so it seems to hang free in a cloud.

To spare my speakers death by bass, I've Auto EQ'd below 125Hz flat then adjusted the graphic EQ to attenuate only.

interesting read that was, dont NOS DACS roll off naturally as well?

On your question i'd guess he means both ways as an immediate dip wouldn't be very natural. So i guess that means flat at 1.25k, -3 at 2.5 and flat again at 5K ?

I would think your speakers would be ok going flat to 70/80 hz but i dont know what they are of course. Anyone trying to go flat to 20hz needs some serious amps and serious speakers.

Remember if you attenuate to make flat you'll turn your volume up a bit probably - therefore having the same effect - more bass power to get the flat response - so you still need to be careful.

Kit,

Thorsten posts under the name ''3dsonics'' at zerogain.

He is always happy to talk about hifi if you PM him there or start a thread.

Cheers
Chris

Kit,

I'v got some of that stuff noted down to try, but still haven't got 'round to it. Just been so happy with the settings I've got I simply haven't had the desire to experiment any further.

Perhaps next week. No, definately next week, must try harder!

Regards,

Stuart.

I've fixed a problem with some audible midrange disortion I was having.

Turning the volume down to -15dB didn't help, but reducing the maximum boost at any individual frequency to +6dB did the trick. I then set the volume to -6 dB, without clipping problems.

Not sure where I got 6 dB from. I recall Royd saying this was the minimum varaiation in a speaker's frequency response that ear was sensitive to.

I have now connected the Behringer to my CD player's digital output - going through a Monarchy 48/96 Upsampler to upsample the signal from 44.1Khz to 96Khz.... its an improvement on the analogue inputs so far, and a big improvement on digital connection at 44.1Khz (but need to spend more time listening).

However - getting a bit of a problem.... most discs are going into digital clipping fairly regularly, and I can't seen anyway of reducing the sensitivity of the AES/EBU input on the Behringer. Anyone got any ideas? How do you attenuate a digital signal anyway?

You shouldn't need to attenuate a digital signal, unless your Monarchy device is adding gain (which may be why you think it sounds better than 44.1 straight, which it shouldn't really).

You can attenuate a digital signal mathematically - lots of devices have volume controls in the digital domain (eg Squeezebox).

I'd check the settings on your Monarchy thingy.

Andrew

However - getting a bit of a problem.... most discs are going into digital clipping fairly regularly
THe reason for this is too much gain somewhere. As per Andrew, I'd look closely at what exactly this Monarchy thing is actually doing.

Alan,

Is this extra gain caused by the DEQ's GEQ/room correction? I think I'm having the same problem, if that's the case.

Also, what's 'dither', and should I set it to 'off', '24 bit', '20 bit', etc...?

Stephen

Did my advice re: switching the I/O from Dig.In to Main In fix the "pink noise from both speakers" issue Stephen?

Chris

Yes! Had a chance to set it up again and this is much, much better. Thanks!

There's a gain control in the utility menu, which you can use for digital volume control. THat should stop the meter on the left going into clipping.

I found that too much frequency boosting causes distortion. It must be pushing the levels on the CD past 0dB.

I s it wrong to have pink coming from both speakers when calibrating? I mean, you get sound from both speakers when listening, and the sound bounces across both sides of the room.

I s it wrong to have pink coming from both speakers when calibrating? I mean, you get sound from both speakers when listening, and the sound bounces across both sides of the room.

Yes, it's definitely wrong. If this is happening to you, see my earlier comments in this thread about switching the Dig. In to Main In for the calibration.

If you think about it, the DEQ2496 can't know which speaker the pink noise is coming from, so can't possibly calibrate properly.

(This is assuming you have the unit in "Dual Mono" mode - which you should also do - unless your room is perfectly symmetrical!).

HTH.

Chris

Yes! Had a chance to set it up again and this is much, much better. Thanks!

Good to hear that :)

Chris

I s it wrong to have pink coming from both speakers when calibrating? I mean, you get sound from both speakers when listening, and the sound bounces across both sides of the room.
Kit,

As Chris has said above, if you are in dual mono mode you ought only get sound from one 'speaker at a time. If you're not in this mode then yes, you will get sound from both.

Personally, I'm not sold on dual mono equalisation. It works very well in the spot you've equalised for but not elsewhere in the room. As I tend to move 'round the room a bit I find the dual mono config. sounds wrong in certain areas. No doubt this is in large part a function of my room (L shape) but it is worth experimenting with both methods to work out which works best for you.

Regards,

Stuart.

Personally, I'm not sold on dual mono equalisation. It works very well in the spot you've equalised for but not elsewhere in the room. As I tend to move 'round the room a bit I find the dual mono config. sounds wrong in certain areas. No doubt this is in large part a function of my room (L shape) but it is worth experimenting with both methods to work out which works best for you.

L-shaped room here to, and I'm surprise how even the sound seems to be across the room, as you say.

Do try the Thorsten Loesch target curve. It doesn't sound tweaky, it's quite flat earth really. The improved sensation of height is cool, perhaps this is the flat earth equivalent of left/right fron/back imaging?

I've also EQ'd <=125Hz to +4dB, which gives quite a warmer sound. Hopefully my bass drivers will forgive me. My room boosts the entire second octave by about 15dB (!), so I think I'm okay.

Kit,

I've been running the suggested curve for a couple of days now and am enjoying it. Also have the bass boosted as you describe, but only by 2db. Might have to re-eq later tonight for tomorrow as I've been moving some furniture around.

Regards,

Stuart.

Here's another thing to try, especially if you've smaller speakers. It works really well with my Royd RR2s in a 3m x 6m x 2m room (not counting the L-shape bit).

I've EQ'd for Thorsten Loesch's target curve, leaving out <=63Hz. I used Delta Max +3dB, which is a function I don't honestly understand. Pink noise from both speakers at once. Calibration only took a few seconds for each channel.

Then set all of <=63Hz to -15dB.

Very punchy, big and airy. The cymbals on The Inner Mounting Flame have that somewhat vertiginous crasssssh! quality to them. The pitch is all there, you can hear deep notes and feel them, but the bottom layer sounds very stripped and exposed.

I think there's quite a bit of spatial information in the bass, as now it's not rolling around the room I find the image a lot more stable.

It really plays to the RR2s main strength, which is to be able to create an absolutely massive sound from a dinky box. You must put these speakers absolutely as far apart as possible if you get to try them.

I have the volume at -7.5dB, which is the maximum boost on my EQ curve. Turning the volume up seems to make the sound fatter and warmer even on discs that are nowhere near clipping.

Can anybody take a picture of Thorsten's target curve on the DEQ's display and post a link to it? Ta.

I've been running the suggested curve for a couple of days now and am enjoying it. Also have the bass boosted as you describe, but only by 2db.

I'm reading the article and I think I end up with the following curve settings on the DEQ2496 - could someone let me know whether I've got the right end of the stick here? It would be great if other people could post their curves here too - I think that's why Alanbeeb wanted a screenshot .....

Happy Xmas listening :)

Chris

Frequency - Boost/Cut

20 +2.0dB
25 +2.0dB
31.5 +2.0dB
40 +2.0dB
50 +2.0dB
63 +2.0dB
80 +2.0dB
100 +2.0dB
125 +2.0dB
160 +1.5dB
200 +1.0dB
250 +0.5dB
315 +0.0dB
400 +0.0dB
500 +0.0dB
630 +1.0dB
800 +2.0dB
1000 +2.0dB
1250 +2.0dB
1600 +1.0dB
2000 -1.0dB
2500 -3.0dB
3150 -1.5dB
4000 +0.0dB
5000 +1.5dB
6300 +3.0dB
8000 +4.5dB
10000 +5.0dB
12500 +5.5dB
16000 +6.0dB
20000 +5.5dB

I've EQ'd <=63Hz out altogether, but that's what I've got, except for the upward slope from 3k125. I went downwards.

I haven't tried the resonant frequency adjustment yet.

Hi,

Similar, however my current setup differs at the following points:
630 = 0
800 = +2
1,000 = +3
1,250 = +2
1,600 = 0
2,000 = -3
2,500 = -4
3,150 = -3
4,000 = -2
5,000 = -1
6,300 = +3
8,000 = +3
10000+ = 0 however the base curve tails down to -3 at 20,000 and also has a gentle bass roll off at the bottom two points.

Merry Christmas,

Stuart.

Merry Xmas everyone.

As 'tis the season to be tweaking, I've applied Thorsten's resonant frequency correction thingummyjig with -6dB/octave slope at each end of the EQ. With my Royd RR2s this means -10db at 20Hz and -11dB at 20kHz. I have the full range switched one when AutoEQ-ing.

This sounds pleasant just on it's own, and is probably a good starting point for when the Behringer is fresh out the box.

I've redone everything to properly match Thorsten's target curve, though I've left the <315Hz boosting out to avoid annoying the neighbors.

Very pleased. A warmish, clear, twangy flat earth-ish sound. I'm surprised at how much the high frequencies can be boosted without making the the sound bright. Distortion on loud CDs is what you need to keep an eye out for.

The main issue with a device such as this is that it can infinitely tweaked. This means that you are rapidly moving away from the interest in the music to the interest in the hifi.

Come on admit it lads, curves this, EQ that, pink noise the other. What are you on about?

This is a device to instil fear into any audiophile. What sounds good to your ears might be reading as bad on the EQ device, before you know it you are never "Truly Happy™"

Still have a happy Christmas and let us know when you sell these devices on, because I have a bet with someone.

Quite right. I've nailed my speakers to the floor for exactly that reason :)

If you own a nice hifi, you are interested in sound, not pure music. Sounding good so far, the voices are less present but more intelligible, and percussion is very thwacky and twangy.

It does seem to make reflection problems more noticeable. Because of my L-shaped room one speaker has much more space behind it than the other, and this side is better defined spatially.

And nothing can make you happy, you have to be happy, man. That said (http://www.amazon.co.uk/exec/obidos/ASIN/072253325X/qid=1135528143/sr=1-1/ref=sr_1_2_1/202-1054894-8611042)...

Woo, cheap Accuphase impression!

If you press the button on the far left you cycle through 3 decibel meters, and the 3rd one is a pair of VU meters! Very cool.

There's another one that let's you see precisely when you're clipping. Playing a few of my favorite records, -3.5dB is the highest I can go without clipping.

Might it be an idea to put an upsampler before the EQ? I'm working in 44.1kHz/16 bit, so I don't know if attenuating means losing information. Would 24 bits give more room for manuever?

TIA

And nothing can make you happy, you have to be happy, man. That said (http://www.amazon.co.uk/exec/obidos/ASIN/072253325X/qid=1135528143/sr=1-1/ref=sr_1_2_1/202-1054894-8611042)...

Alex, please open a new thread and tell us about your experience of this amazing new discovery.

Might it be an idea to put an upsampler before the EQ? I'm working in 44.1kHz/16 bit, so I don't know if attenuating means losing information. Would 24 bits give more room for manuever?

TIA

That's what I'm doing with the Monarchy 48/96 Upsampler - its upsampling from 44.1Khz to 96Khz, and also converting from RCA SPDIF to XLR AES/EBU which is handy..... however, its giving me the problem of pushing the digital signal into clipping on a lot of recordings, looks like its adding gain. The manual says it simply pads out the 16 bit word with zeros to make it a 24 bit signal, but I'm wondering if its adding eight 1s instead..... Monarchy are suggesting a modification to it and are sending parts and instructions but I wonder if its more complex that that.

Anyway.... when the music is not clipping - its only really modern pop/rock recordings that are really suffering, well recorded classical and accoustic is fine, I honestly can't tell the difference between using the digital input in this way at 96khz, or using the analogue inputs and the Behringer's onboard A/D conversion. The Behringer A/D really seems to be that good.

Next step - try a good 96Khz DAC on the output....

That's what I'm doing with the Monarchy 48/96 Upsampler - its upsampling from 44.1Khz to 96Khz, and also converting from RCA SPDIF to XLR AES/EBU which is handy..... however, its giving me the problem of pushing the digital signal into clipping on a lot of recordings, looks like its adding gain. The manual says it simply pads out the 16 bit word with zeros to make it a 24 bit signal, but I'm wondering if its adding eight 1s instead..... Monarchy are suggesting a modification to it and are sending parts and instructions but I wonder if its more complex that that.

Next step - try a good 96Khz DAC on the output....

The Perpetual Technologies P1A upsampler would probably be a much better option for this sort of thing....it doesnt add any gain,but does upsample/interpolate at a variety of word depths from 16 to 24 bit,and upsamples to 96kHz.

From what I can remember from their release blurbs at the time I bought mine,it doesn't add zeros until the word length is suitably increased,but interpolates data from an internal algorithm to increase the word length....whatever the software,the results IMHO are very good,and critically it's been generally very well reviewed also,but for someone who already has an upsampler,it would be expensive.

I use it between my CD transport and TAG AV32R/DP,before it's own room EQ software does its bit.

Alan,

most modern recordings I own clip anyway. The Tact will show this on the front panel. The solution was to attenuate everything by 6db.

I get quite different results if I include the <=125Hz range in the AutoEQ. The top end seems to need a lot more boosting if you include the bottom.

Best sound so far is from AutoEQ-ing the two ranges separately, switching the appropriate frequencies off with the big knob (oo-er!) before measuring the pink noise.

The target curve on the GEQ is flat with Blauert bands and bass boost overlayed. Then with the PEQ I've added in the -3dB dip at 2k5, plus the 1st order roll off at each end of the scale to stop my speakers howling/being killed.

But as iGary says, at least half the fun of this thing is mucking about with it like a toy, so that'll probably change.

I wonder if it would be better to attenuate the frequencies out side the Blauert bands and bass boost, rather than boost the ones within?

Kit,

If I'm understanding you correctly, you appear to be building the psychoacoustic curves into the room correction target. If so, try setting everything back to default/flat and re-auto-eq-ing for a flat (or top&tail roll off with flat between). When complete, look at the GEQ corrections and manually transfer any largeish attenuations to the PEQ, then re-run the auto eq targetting the same flat(ish) response. This should give you a base to work from - save it.

You can then create the psychoacoustic curves in the GEQ and save them, adding or subtracting them to your base correction as you please. Makes it a lot easier if you want to experiment with B.curves, differing levels of bass boost for low vol. listening etc.

Of course, if I've misunderstood and you are already doing it this way then sorry, as you were!

Regards,

Stuart.

Alan,

most modern recordings I own clip anyway. The Tact will show this on the front panel. The solution was to attenuate everything by 6db.

I haven't found a way to attenuate on the digital input on the Behringer. The Monarchy is definitely adding gain - using a direct feed from the cd player digital out at 44.1Khz, there is occasional clipping on most pop/rock recordings, but not enough to be noticeable. Going through the monarchy the same recordings are clipping most of the time, and its audible - in fact its totally unlistenable!

Given that I'm not noticing a benefit on recordings that don't clip over using the analogue inputs I'm not going to investigate further for now.

More fannying about.

It seems speaker placement makes as much difference as it always does. Things like port placement still change character, whilst the Behringer cleans up the various excesses. For involvement and immersion, nothing beats my favorite "headphone" speaker arrangement. 6m apart, the listening chair about 50cm off axis and loadsa toe in.

First I've AutoEQ'd for a flat curve with a 1st order rolloff at each end. Then I've added the Blauert bands by attenuating all the frequencies outside the range by -3dB. I've boosted Thorsten's suggested frequencies by +3dB rather than +2dB, and left out the +1db boost at each end of the band.

This works well, you can turn the Behringer's volume up higher without clipping, and the sound is clear and spatially quite stable.

My other amazing new discovery is also going well, but I'll spare you the gruesome details.

Some tips. I find reducing the volume on the Behringer thins the tone quite a bit, so set it to the desired level before you AutoEQ. The first order rolloffs on the target curve seem pretty much essential in my system, if serious weirdness is to be avoided.

I read somewhere else that our hearing is most sensitive in the 1k-4kHz range. I've boosted this range by +3dB, with +1.5dB at each end of the scale for a smoother transition from 0db.

I've also done the same for the 100-400Hz range, as I like a bit more warmth and it seems these sorts of symmetries are generally thought to be good news sonically.

I'm trying these integrated into the target curve before for the AutoEQ, rather overlaying them once the target flat + roll off curve has been AutoEQ'd.

This is a good sound. A lot less bass energy sloshing around, but all the pitch information is there and the sound very punchy, so the sounds only seems clearer rather than thinner or colder. The top end seemed muffled at first, but now my ears have adjusted to make other presets sound metallic.

I'm surprised how much the ears adjust to changes in the sound. Swapping back and forth between presets invariably makes them sound weird and phasey. Taking the Behringer out of the loop makes things sound exceedingly odd indeed. So always give any changes a few minutes before dismissing.

Technical question.

Is there any theoretical advantage in in upsampling the signal to 24-bit before it goes into the Behringer? If the the Behringer is attenuating a 16 bit signal, I assume this means losing information from the recording.

However, I assume this is only noticeable if I'm playing play loud enough to cover the full dynamic range of the CD (plus ambient noise), which I doubt.

Behringer do an upsampler + ADC/DAC for £85, and tragically, I quite fancy another cool looking studio box. With my Squeezebox and the EQ, it'd make quite a light show.

TIA

*tumbleweed*

I've put a -3dB notch at 1.6kHz, and matched it with -3dB at 160Hz and 16kHz to follow the assumed symmetry rule.

Sounds less metallic, imaging is a bit less strong. A keeper.

Eek, am I the only one using any fun, freaky, overboard EQ settings?

My current target for the AutoEQ curve is -9dB 100-400Hz, -9db 1k-4.5kHz and a first order rolloff at each end (-10dB at 20Hz, -11dB at 20kHz). On the left channel, I've then zero'd a large 20-25Hz boost added by the EQ. This last thing makes little if any perceptible difference, but it should keep the speakers and neighbors happy.

This gives a yummy soft and gooey overall sound, with the metal stuff sounding punchy without being edgy and grainy. Notably attenuated compared to the flat curve, but the sound is all the better for it.

I tried overlaying these over my previous EQ to give, in theory, exactly the same in-room response, but this sounds quite a bit softer and less clear and punchy in the treble. There are less artefacts though, aswell as less information. SOund, fiddle about with it if you try this.

I think what's happening with these curves is that they're like a "loudness" button. By increasing the bass level, relative to the treble, you compensate for the fact that the ear is less sensitive to bass at low volumes, and get real sounding sound without having to resort to real life decibel levels.

The Behringer upsampler has been ordered, will report back once I have it.

Eek, am I the only one using any fun, freaky, overboard EQ settings?
The Behringer upsampler has been ordered, will report back once I have it.

I've had a quiet month with the DEQ2496 due to Xmas taking over and the room having to accomodate a tree etc. so everything got moved around somewhat.

However, after this weekend, all should return to normal and so I'll be playing with the EQ settings a lot more - especially now I understand how to overlay one on top of another - it gets much easier to have a "Flat" setting and then play with other curves.

So, Kit, your posts aren't falling on deaf ears at all (no pun intended!) and I'll be trying them out soon. Oh, and very interested to hear about the Behringer upsampler - do let us know how that works out.

Happy listening.

Chris

Behringer do an upsampler + ADC/DAC for £85, and tragically, I quite fancy another cool looking studio box. With my Squeezebox and the EQ, it'd make quite a light show.


I assume you've ordered the ULTRAMATCH PRO SRC2496?

Chris

I'll be playing with the EQ settings a lot more - especially now I understand how to overlay one on top of another - it gets much easier to have a "Flat" setting and then play with other curves.


I'm not sure how to tranfer the GEQ to the PEQ. Do you just port the bands across? How wide should each parameter be, 1/3 octave? TIA

I find my ears can adapt very quickly to some fairly severe softening of the sound very quickly, but an excessively bright sound seems impossible to adjust to. So it's worth bearing with new settings for a few tracks if they seem at bit murky at first.

I wonder if the advantage of attenuating certain frequenices is not so much the change in sound as the ear adapting to them and becoming more sensitive?

I've ordered the Behringer SRC2496, but apparently there are supply problems. Quite a few people are also waiting.

I'm not sure how to tranfer the GEQ to the PEQ. Do you just port the bands across? How wide should each parameter be, 1/3 octave? TIA

I've ordered the Behringer SRC2496, but apparently there are supply problems. Quite a few people are also waiting.

Personally, I've been doing all my work within the GEQ - adding curves to each other - and haven't used the PEQ at all. But there are others on the forum who've been using the PEQ to cope with some of the larger room issues - perhaps they can comment.

I hope the supply problems get sorted and you get your SRC2496 - look forward to hearing about your experiences of upsampling.

Chris

My current target for the AutoEQ curve is -9dB 100-400Hz, -9db 1k-4.5kHz and a first order rolloff at each end (-10dB at 20Hz, -11dB at 20kHz). On the left channel, I've then zero'd a large 20-25Hz boost added by the EQ. This last thing makes little if any perceptible difference, but it should keep the speakers and neighbors happy.

I've added -3db at 2.5kHz to the above, as per Thorsten Loesch's suggested curves. I've also cut the top three bands (can't remember what these actually are) by a further -9dB, as I can hear little if any difference in tweaking these bands. So the less energy into the speakers the better.

Altogether, that seems to chunky things up a bit. Worth doing.

I'm not sure how to tranfer the GEQ to the PEQ. Do you just port the bands across? How wide should each parameter be, 1/3 octave? TIA

Hi,

I've not used a particularly scientific approach to this, but I guess you could copy the GEQ settings to a spreadsheet and mess 'round with formulas. I simply look at the GEQ settings post auto EQ and identify any areas that have particularly deep cuts, note what frequency the cut is centered on and how deep it is and try to aproximate it with the PEQ. As for the bandwidth, I simply turn the knob tilll I get the sort of shape I'm looking for. I have a reasonably broad cut in the mid range which I've used two overlapping thingies to get about right.

Once anything noteable has been translated into the PEQ, re-run the auto EQ to get the flatish response, make any final GEQ adustments and save. Then go mad with all the different overlay curves you wish to try.

Regards,

Stuart.

Here's another target curve to try in the AutoEQ.

In Thorsten's article, he said that for a natural sound it was important to get the -3dB points on the 1st order roll offs to match. I've also had some success in applying a tweak at one frequency to the other decades on the EQ (eg 50Hz, 0500Hz, 5kHz).

So I've applied each rolloffs to the other two decades on the GEQ. The curve follow this pattern up to 20kHz, with 2dB per 1/3 octave in between:

20: -10dB
50: -2dB
63: -1dB
80: -3dB
160: -9dB
200: -10dB
etc

20kHz is -10dB rather than -11dB, for the sake of neatness. Remmber that the correct starting points of the first order rolloffs depend on