# OK I have a Ultra Curve DSP8024, a Mic, SPL and all gear now what?



## superchad

Hi again, newish member here.
I have the Ultra-Curve Pro DSP8024, the matching Mic, Rat Shack SPL, Stereophile test disc with tones, Rives disc and all gear up and in use in a dedicted room that I have tuned with treatments for optimal Hi-Fi sound but am curious to see if I can take it to another level. A friend gave me the Behringer unit along with Mic and all cables but the user manual is written in cryptic text I cant seem to wrap my head around. My system works great from the Projector to Sub and everything in between but the only real issue I have is unknown potential performance but as noted no experience with this Behringer unit, my friend said I could use it to adjust my outboard active cross-over in my Horizontal Bi-amp speaker set-up but agian I cant figure the manual out as it looks to be designed for engineers...which I am not.
Maybe this thread is old news but I cant find info I need so if anyone could clue me in I would be grateful to even know a small amount about this product and procedure.....clueless! Thanks, Chad


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## Sonnie

I do not know much about the 8024... we have not seen it used too much for the same application we use the BFD for... sub equalization.

That unit has an auto equalization feature, but not being familiar with it, I would not know how to set it up and use it.

Maybe some others will know or can interpret the manual for you. :reading:

What exactly do you want to use it for?


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## Wayne A. Pflughaupt

The 8024 is a digital 1/3-octave EQ primarily used for full-range equalizing. The manual would be written for someone with some pro audio experience who had some idea how such a processor would be used, so I can understand how a newbie would find it perplexing. I’ll wait until you answer Sonnie’s question about how you intend to use it before delving into things further, but personally I would prefer good analog EQs for full range, that won’t add another AD/DA conversion step to the signal chain.

Here is a review of the 8024. Google “Ultra-Curve Pro DSP8024” for more info and reviews.
http://www.audiophilia.com/hardware/hw1.htm

Regards,
Wayne


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## superchad

Thanks for the reply guys but to be honest I dont know what I can do with it or what it can be used for so I was hoping for clues, my friend said I could use it to measure my cross-over point for woofer level but I think what I want to do means I need a different unit. What I would like to do is maximize my room and systems interaction and performance for music and movies, I have no real issues as of now mind you but sometimes you dont know what your missing till you find it. All in all I guess I need to admit I am lost and have no idea of how to use this or even if this is what I need, thanks for any and all additional help you can give including bit not limited to the proper method of frisbee tossing into dumpster:surrender:


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## Wayne A. Pflughaupt

I haven’t specifically used this unit, but I’m familiar with the type. 

Basically it’s a 1/3-octave real time analyzer / equalizer in one chassis. “1/3-octave” means it has three EQ bands for each octave of the audio spectrum (although the 8024 does include a few parametric filters available).

To give you some perspective, maybe you’ve seen those 10-band equalizers they often used in stereos back in the 80s. Those were one-octave equalizers, so a 1/3-octave EQ has two additional sliders in between each of those, for a total of three per octave, or about 30 sliders. The 8024 doesn’t have “real” sliders, naturally, but the concept is the same, even if the execution is a little different.

The built-in real time analyzer (aka RTA) will show you what the SPL level is at each frequency where you have an EQ slider (real or virtual). Imagine the vertical LED level meters on the BFD:










Now, imagine having a row of 30 “level meters” showing the level of each of those 30 frequencies, and you should get the idea of what the display does. Typically, only the top (peak) reading of each “meter” is shown, for the sake of simplicity. Often the resolution of the RTA window can be switched so that the difference between each vertical meter segment represents a 4-dB, 2-dB, 1-dB, etc. change. The display will look smoother with coarser resolution, and more ragged with finer resolution.

Here’s how the RTA works: You connect the RTA/equalizer’s outputs to an input on your stereo. You plug the calibrated mic into the proper input on the rear panel and situate it at your listening position. The RTA generates a pink noise signal that plays through the speakers (it sounds like the noise between radio stations from an old analog FM tuner). The mic picks up the signal generated from the speakers and shows a display on the screen of the RTA. 

That display shows your in-room frequency response with 1/3-octave resolution. The pink noise signal is flat, so in a perfect world with perfect speakers in a perfect room, the display would also be perfectly flat. But you’ll see on the display won’t be perfectly flat. Not to worry. As long as transition from one level indicator to the next is close to the ones on either side of it, you’re okay. What’s bad is if you see big jumps up or down from one to the next. 

The RTA can also show you generally how good your high frequency extension is, e.g. if it’s flat all the way out to 20 kHz or not (it’s not unusual to see a bit of sag above ~12 kHz at the listening position), and how low your speakers go on the bottom end before they start rolling out. It’s called a “real time analyzer” because any adjustments you enact with the equalizer section will instantly show up on the screen. It’s pretty handy, actually.

As far as telling you what’s happening with your electronic crossover, you would pay attention to what you see on the display relative to your crossover frequency. For instance, if you’re crossing over between the tweets and mids at 3 kHz and the RTA shows everything above 3 kHz is elevated compared to what’s below 3 kHz, then the tweeter amp needs to be turned down.

As far as actually using the equalizer, you’ll have to connect it between your pre-amp and electronic crossover. As mentioned, this means you will have an extra AD/DA conversion in your signal chain.

You might want to take a look at this thread, where a fellow member measured his in-room response and equalized improvements. He was using REW as his measuring tool and parametric equalizers, but the concept is the same as what the 8024 will do, if not quite as precise.

Hope this helps to get you started. The 8024 being a digital device is going to be a bit more complicated to operate than an analog RTA like the one I have, but hopefully with this background you can figure things out in the 8024 manual. Get back to us if you have anymore questions.

Regards,
Wayne


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## superchad

Wayne thanks alot for the detailed reply, I know alot more about this unit than I did before talking to you and I will take a look at that other thread you linked...thank you VERY much!


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## cixelsid

The 8024 is the little brother of the DEQ2496. 

The Auto EQ function uses only the 1/3rd octave filters. Independent of the AutoEQ function, all the 1/3rd octave filters and the 6 bands of parametric EQ can be dialed in by the user. If the AutoEQ function is used the parametric filters can be user set to fine tune the AutoEQ created plots. Similar to the BFD the 8024 has 10 separate stereo memories.

In addition the 8024 allows for setting time delay distances between the mains and the sub. It also functions as an accurate SPL meter

BTW here's a good review of the 8024
http://www.enjoythemusic.com/magazine/equipment/0101/behringer8024.htm


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## superchad

Thanks for the additional info!


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## superchad

Ok so I am thinking of running this between bass amp and pre amp to try and see what bass may sound like if corrected, my bass amp has adjustable gain so I should be ok there right? I dont know what other options I could do with this being I am running Bi-amp, the woofers cross at 120Hz so correcting from there to 20 cycles might be of great benefit with little to no risk of introducing its sound signature into my system but still am a bit leary of trying this all and wonder if anyone may be up for a phone call to help me out? If anyone would be willing to talk with me I would surely appreciate the help! Thanks
 Chad


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## cixelsid

Hi,

I've actually not used my DEQ yet so I can't be of much assistance other than to post all the thread posts I've collected about it....

This forum's software won't let me put this all in one post so there's a part #2

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With my room, the Auto EQ room correction function really does wonders. Music sounds completely different - much more realistic, and right there in front of you, particularly acoustic music. 
IMO, the best way to use it is:
1) Run the Auto EQ function on everything above 100hz to start;
2) Then, while playing pink noise, manually adjust everthing below 100hz to get a response as flat as your speakers will allow;
3) If you have a room like mine, that may end up in some fairly aggressive adjustments, e.g. boost/cut differences of up to 15db in neighboring frequencies. This can be problematic, particularly in the bass range, so you may want to soften them up a little to make the differences less dramatic.
4) Finally, adjust to taste (e.g. I like a little more bass, less brightness).
That works pretty well for me. But I'll have to check out what's going on in the rest of the room.
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I've been experimenting with this a fair amount, and I think I finally have it down. I have to say, it is making a huge difference in my sound, and for the better. I really like this thing now that I've learned how to use it.
However, the manual sucks. At some point, I'm going to type out step by step instructions, but here are some general tips for now. Note that this is just what I do; other people with different components, rooms, tastes, etc. may find something else works better.
First, here's my setup: I'm taking a digital signal out of the optical Toslink output from my Squeezebox (essentially equivalent to a CD transport), and running it into the Behringer's optical input. I then take the digital Toslink optical out from the Behringer, and run it into the digital optical input of a Benchmark DAC1. I take the balanced XLR analog outs from the DAC1 and run it into my amp (Pathos Logos driving Magnepan 1.6qr speakers). So all the processing is done in the digital domain.
1) I plugged the ECM8000 microphone into the RTA input, and place the microphone right where my head would be in the normal listening spot. I measured the distances from the tip of the mic to each of the speakers to make sure the mic was equidistant. The mic was pointed straight ahead to the spot directly between the speakers.
2) This is crucial: I put the Behringer in dual mono mode, NOT stereo link; this lets you EQ each speaker independently. When I first got the thing, I didn't do this, and was getting some pretty crazy results. To do this, press the Utility button, scroll down to Channel Mode and change it to Dual Mono. Don't forget to press the B button to accept the change.
3) Now hit the I/O button. Make sure on page 1 (called "Select Input") that the input is set on "Main In." I think this is the only way to EQ each speaker independently. I initially thought this had to be set to Pink Noise, but in fact you'll get pink noise automatically when you go into Auto EQ mode.
4) Still in I/O mode, on Page 2 (called "Aux/Dig. Out"), select GEQ-PEQ. This will make sure the EQ'd sound is coming out of the Behringer. 
5) On Page 3 ("Select RTA Input"), select RTA/Mic. 
6) Now press the GEQ button. Flatten the curve, putting all frequencies on 0db. Make sure you do this for both channels, pressing the A button to switch between them.
7) Make sure you've got the proper input sensitivity and phantom power for the microphone you're using. If you're using the ECM8000, I think the default settings should have it covered.
8) Turn up your amp volume fairly loud, and warn your wife that a jet plane is about to land in the living room. Make sure nobody is walking around, or otherwise making noise.
9) Hit the RTA button, and cycle through so that the Auto EQ label shows up next to the B button. Adjust the Max and Range levels so that you'll get a good window on the frequency read out. You want something narrow enough so that you can easily see differences between the frequency levels, but wide enough so that you can see all the peaks.
10) Start the AutoEQ process. Note that you'll have to do this twice, once for each speaker. The pink noise should only be coming out of one speaker at a time. If for some reason noise is coming out of both, you need to change what you're doing.
I like to have "Room Correction" on, which "tilts" the target frequency spectrum from flat to slightly weighted in favor of the low freqs. Just sounds better to me.
I also didn't do any automatic correction below 100Hz. Instead, I manually adjusted them later, after the AutoEQ had flattened everything else.
I also put the noise level at about -1db, so as to avoid clipping. Keep an eye on the level meter and make sure there's no red flashing. If there is, lower the noise level; if there isn't, raise the noise level until there is, then back off slightly.

Note that during the AutoEQ process you can toggle between pages to compare the RTA readout with the adjustments the Behringer is automatically making to the GEQ. You can also switch between "Fast", "Med" and "Slow" to determine how fast it reacts. I like to start it at Fast, then change it to Med after a minute, then to Slow for a minute or so. 
I set the Delta Max and Span to their maximum values, and didn't have any problems. Others may have to narrow these parameters, e.g. if you've got a big problem with your room or setup somehow. 
At some point, after a few minutes, it should get to the point where the ongoing adjustments are fairly modest, and you can stop the AutoEQ then. Again, note that you have to do this twice, once for each speaker. 
After you've run the AutoEQ, go back to the I/O page and change the input to Pink Noise. Noise should be coming out of BOTH speakers now. Then go back to the RTA, and look at the spectrum. You may notice -- as I did -- that it is no longer flat, now that you have both speakers going. I had a dip right around 12khz. I went back into GEQ, and adjusted the EQ (for each channel -- so you have to switch between Left and Right, making the same changes in each), then toggled back to the RTA, back and forth, until it was fairly flat. 
Next, I manually adjusted the GEQ below 100Hz to get something fairly flat. (My speakers roll off below 40Hz, so naturally I didn't try to fix that range.) Note that again, you have to use the GEQ on both channels.
Finally, change the input back into your music input, and do some listening. Season to taste; I generally like a little more bass than you get out of a perfectly flat response.
Originally Posted by Mike Anderson After you've run the AutoEQ, go back to the I/O page and change the input to Pink Noise. Noise should be coming out of BOTH speakers now. Then go back to the RTA, and look at the spectrum. You may notice -- as I did -- that it is no longer flat, now that you have both speakers going. I had a dip right around 12khz. I went back into GEQ, and adjusted the EQ (for each channel -- so you have to switch between Left and Right, making the same changes in each), then toggled back to the RTA, back and forth, until it was fairly flat. 
At this point you should be running the RTA in "Average" mode, which averages out the peaks over time. You should reset it each time you change the GEQ, cycling through from fast to mid to slow and back to average to get a new reading.
Also, one other thing that's very important: After you've set the EQ, you need to make sure your output signal isn't clipping, because you may be adding a substantial amount of gain to the signal.
So go to the Meter page, and switch Source to Output. While you're playing music that you know has a very loud signal, check the outside meters to see if they're clipping at all. (You probably want to leave this on for a long period of time, so that you end up capturing the hottest peaks you can.)
If you're clipping, go to the Utility menu, and set Gain offset to -5db or thereabouts. Then go back to the Meter page and monitor it again for a while. Go back and forth until you are fairly sure you're not clipping, while not taking too much off your signal. 
For my EQ, I find -5db to be about perfect. I can play several hours of hot signal, and get peaks at around -0.1db below clipping. Ideally, this is where you want it, assuming you don't have to lower it even more to accommodate your preamp inputs.
I've got my setup sounding really, really good now.
Of course it's a bit of a pain, since I'm still working with speaker arrangement, and you have to reset the thing every time you move the speakers, requiring a good 10 min blast of loud pink noise -- but I'm definitely getting it dialed in.
It's true that it doesn't sound perfect everywhere in the room, but right in the listening spot, it's totally sweet!
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DEQ2496 PRIMER 
First some basics. The DEQ2496 can be connected to any Pod (most Line 6 products with direct out) to further shape the tone beyond what you would be limited to with the stock unit. After you have "tweaked" out the Pod to it's best sound, you can go beyond that limit to custom shape a tone to your liking. A number of "matching" experiments have shown that given some good base (stock) tones you can reach a professional level of sound that rivals some of the best CD recordings. It's a process that requires skill, but the capability is there if you want to reach for it. The DEQ2496 is a very dynamic and versatile device with many options (see product description on Behringer's web site). Bottom line? Whether doing just minor touchups or all out tone bending, the DEQ can get you tone you could never get without it. However, tone is only one aspect of total guitar sound and you have to be able to tweak L6 patches for the others. 
DISCLAIMER: Although I have provided files for the Pod and I am now providing files for the XT series, you will have to determine if this is sound you think is worth spending extra money for. Of course you can create your own custom tones, but the process may be a bit much for the beginner. Of course "jumping in" and getting your hands dirty is fun and a good way to learn (gaining experience is a good thing!). Proceed with your own will. 
The DEQ2496 has many connection options. (Look at the product pictures to see the back panel.) For inputs it has balanced analog L & R XLR IN, digital AES/EBU IN, and optical S/PDIF IN. It also has a single (mono) balanced XLR RTA/MIC IN for use with a measurement mic like the ECM8000 to check speaker accuracy across the spectrum (and a AUTO EQ function to automatically flatten a room). Of course it has MIDI IN to control the preset changes or load preset files into the unit. 
The output options are about the same with the addition of an unbalanced L & R AUX LINE OUT with 1/4" phono jacks. These outputs along with either digital output has an additional option of delay (LEFT, RIGHT, or BOTH) of up to 300 ms. I use this delay on the LEFT channel to get that double tracking effect on some of my patches. The analog balanced XLR OUTS do not have this delay option. If you were to want this effect you would have to work with the other output options. 
(NOTE: MIDI out can be used to save single preset files or the whole unit's memory to your computer via a program like Send SX. You will need a MIDI interface to your computer to load or save files. More on that later.) 
The DEQ2496 can interface with either the Pod XT Pro or the Pod XT bean (or other Line 6 products with direct outs). I have the Pro models and like to connect it to the DEQ via DIGITAL AES/EBU. (use a short XLR cable) There are several advantages to doing this. For one, you stay in digital keeping the number of A/D D/A conversions to a minimum. The signal remains pure until it is fully processed. Also, the levels are fixed and match up perfectly with the DEQ. The DEQ METER screens (I like to watch SOURCE OUTPUT on the PEAK/RMS page 1) show you what your peak and RMS levels are and when you start to clip and limit. This makes it easy to normalize the levels. As a general rule, your loudest lead patch should be just under ZERO and not CLIP. A slight CLIP will not be a huge problem as it is limited and not distorted on the DEQ. Rhythm patches should peak about -4 dB or less. Note that the OUTPUT control on the XT Pro does NOT work when connected this way (one less thing to worry about) and where it is set makes no difference. Only use the CHAN VOL on the XT to set the levels and save them to the XT (or Pod...whatever). So far I have not been clipping on the XT Pro when the levels were set like that. It's a perfect level match between both units. Clipping on the XT is distorted and undesirable. 
As mentioned above, the DEQ output options are many, but only the 1/4" unbalanced (and digital outs) have the delay feature available. The XLR OUTS work fine without the delay if it is not desired. IF YOU CAN ONLY USE ONE OUTPUT, USE THE RIGHT OUTPUT. All of my double tracking delays are on the LEFT side. (You would not want to play to a 20+ ms delay.) The reason I use the right side is because I stand on the right side of the stage. If you are on the left side and have a stereo stage monitor setup, you may want to reverse the L & R channels or reprogram (much harder) all those patches that feature the effect (many). Although the DEQ can have different LEFT & RIGHT EQ settings on each preset, I do not exercise that option at this time. Once you get up and running, you can go nuts with all the capability the DEQ has to offer. 
The DEQ can also be connected through the analog EFFECTS LOOP. I had to use a pair of balanced to unbalanced transformers to get the XT PRO SEND to the proper impedance for the DEQ analog XLR inputs. Then I connected the AUX OUT L & R to the EFFECTS RETURN on the XT Pro to complete the signal chain. You must also set the XT Pro to SERIES EFFECTS LOOP. The results yielded low signal. The DEQ really did not have enough signal coming in from the XT Pro. I tried this to see if I could use the "PHONES" jack on the XT Pro. Yes it worked, but this signal loss issue was a problem. I do not recommend this connection option, however it may be the only way to use the XT Pro outputs like they were designed. It's just another option to try. 
Connecting to the Pod or XT bean is also possible. Here the OUTPUT volume does function and you will need to set that so the loudest lead patch will not go into clipping on the DEQ. Only one setting should be needed. Please note that I have set all of my presets on the DEQ to AES/EBU DIGITAL IN. So if you were to load a file of mine, you would have to go into every preset on the I/O PAGE 1 screen and change it to MAIN IN. This selects the balanced XLR inputs. The XT bean has balanced TRS outputs. Cables are ready made at music stores to go from TRS to XLR, however if you need to know...TIP goes to pin 2 on the XLR, RING goes to pin 3, and SLEVE (shield) goes to pin 1 (ground). Use well shielded cable! The output of the DEQ should provide enough options except for the double tracking effect options mentioned above. I realize switching all the presets to "MAIN IN" is a drag. You will have to create a working DEQ load and extract any new presets I have created off of my master files. You can save them individually after recalling them by using "EDIT DUMP" on the DEQ to capture the single preset to the software. Save it and name it with the date in the name (ie...Slodano Lead 041004.syx). Then after you load YOUR master DEQ file back in you can simply send the single preset to what ever memory position you have selected. Do a memory save (STORE PRESET) on that position and your good to go. After that, you may edit them to taste. I will not normally break out single preset files because of the time and upload issues. It would take me forever to get anything done. (sorry, my time is short) 
MIDI connections are easy. Under normal operation you just need a short 5 pin DIN MIDI cable to go from the (Pod) XT MIDI OUT to the DEQ MIDI in. The presets seamlessly track between both units! (yes, the footboard will change both units) Every (Pod) XT patch can have a totally different DEQ setting. It's RAD! To load and save DEQ presets, you need a MIDI to computer interface cable and FREE software to send and receive sysex (syx) files to and from the DEQ. This interface can be a 15 pin gameport (joystick) to 5 pin din IN/OUT or USB to DIN IN/OUT. I know of some people who have had issues with USB drivers, so ask your local music store if you are in question. I do not use USB at this time. All "SEND", "RECEIVE", & "DUMP" functions are fairly intuitive. Basically, to load my master file into the DEQ you will open the file in the SEND window and SEND it to the DEQ. That will overwrite every memory preset in the DEQ with my presets. Make any changes you want to make and save each preset in the DEQ as you do. Then, when done, backup all your work on the sysex program (with the date in the name) by doing "DUMP ALL" from the UTILITY page 2 screen on the DEQ. (Make sure the screen on the program is clear before you transmit a dump. I have noticed doing something as simple as changing patches on the XT has sent data to the computer program. This would hose up the file. Always "Clear" the screen just prior to doing the dump.) This will send it to your computer. Name and save the file. Now you have your own custom file. If you need to extract new presets from my master file, you will have to load it and save each new file that you recalled with "EDIT DUMP" from the utility screen (page 2). Then save them to your computer and reload your custom file to the DEQ. SEND the single presets to the DEQ. All the settings will load and you can dial to a memory location you want to save it in and do a memory save (STORE PRESET) to store it. Repeat the process for additional new presets. Then resave your custom master file with the new date (good backup practice). The files are small.
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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 - 

Quote: 
Originally Posted by cporton ...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
Quote: 
Originally Posted by cporton 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. 
Quote: 
Originally Posted by cporton 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. 
Quote: 
Originally Posted by cporton 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 **** around) to confirm this if you are still having difficulty. 

Quote: 
Originally Posted by cporton 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. 
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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.

http://www.pinkfishmedia.net/forum/showthread.php?t=16710
http://forums.slimdevices.com/showthread.php?t=21643&page=3
http://66.102.7.104/search?q=cache:...nalog+modifications&hl=en&gl=us&ct=clnk&cd=35
I found the manual to be pretty sparse on info for the RTA section. Here's a little better guide to using the RTA function. This assumes that the output from the unit is going to an amp and speakers. Use the dual mono mode to analyze each speaker seperately for best results. You may also want to analyze without EQ in several spots in the room to look for hot spots, but generally placing the mic pointing straight up at the seating position at approximate head level will give the best results.
1. Use an XLR mic cable to connect the ECM8000 reference mic into the RTA/MIC input on the back of the unit.
2. Press the UTIL key to reach page 1 (GENERAL SETUP), use the upper small jog wheel to select 'CHANNEL MODE', then use the large jog wheel to set the MODE to 'DUAL MONO'. Press the B key to 'ACCEPT MODE'.
3. Press the I/O key to access page 1 of the I/O configuration.
4. Use the big jog wheel to highlight the 'MAIN IN' box and press the B key to accept the source.
5. Press the I/O or PAGE key to enter the I/O page 2 and highlight the 'WIDTH' box.
6. Press the I/O or PAGE key to enter the I/O page 3, use the large jog wheel to highlight 'RTA/MIC', and press the B key to highlight the +15v phantom power box (if it isn't already).
7. Press the RTA key to go to RTA page 1.
8. Press the B key (AUTO EQ), and you will enter AEQ page 1. 
9. The DEQ2496 will now request that you edit the target curve. Use the A key to select the 'LEFT' channel. With the large and small jog wheels you can adjust the EQ bands to reflect the curve you want to end up with (it can be 'flat' - without any adjustment). Press and hold the B key to reset if you feel it's necessary at some point to restart the curve adjustment.
10. Press the PAGE key until page 2 (AEQ) is accessed. Pink noise will begin, but the EQ process doesn't start yet. Highlight ROOM CORRECTION if desired. 
11. Adjust NOISE GAIN with the large jog wheel as desired.
12. Adjust AUTO EQ to 'slow', 'mid', or 'fast' with the lower small jog wheel for the Auto EQ calculation speed.
13. Press the PAGE key until page 3 of the AEQ menu is displayed. The GEQ is shown with outlined virtual faders.
14. Adjust 'MAX' with the upper small jog wheel for the desired maximum difference (in gain) between adjacent frequency bands. A suggested starting point would be a value of '3'.
15. Press the A key to START AUTO EQ.
16. The pink noise will end, the DEQ2496 will measure the ambient (or room) noise for about 15 seconds.
17. The DEQ2496 will produce pink noise again and will analyze system noise for about 15 seconds. If you get a message that the 'Pink noise level isn't high enough', turn up the pink noise level and restart the Auto EA process or wait until a message appears that it is 'Starting Auto EQ Process'. 
18. Once the Auto EQ process has begun you can view the 'progress' from page 2 (RTA response) or page 3 (Graphic EQ changes). If necessary, the 'MAX SPAN' and 'MAX' values mentioned above can be changed in real time, so that you can see the Graphic EQ changes.
19. Let this process run until it appears obvious that the adjustments begin to 'settle' (or after about 90 seconds), then press the PAGE key until page 3 appears, and press the B key (DONE) to end the Auto EQ process. The unit will default to RTA pages 1-3.
20. The DEQ2496 will have created the corrective adjustments to achieve the target curve that you specified earlier in this process. To view the Auto EQ changes, press the GEQ key. The settings will be visible and available for further editing.
21. Repeat the process for the right channel on the unit.
In the beginning of the Auto EQ process, the DEQ2496 can be set to 'STEREO LINK' mode (in the UTILITY menu) if you want, so that the Auto EQ settings from one channel will be copied to the other channel.


I just got this from Berhinger. My misunderstanding.
The centre frequencies on the graphic EQ mode ('GEQ') are preset, but on the parametric EQ mode ('PEQ') you can define the centre frequency.
When you are adjusting the centre frequency ('FREQ') of one of the bands in the parametric EQ mode, the adjustment can be made on a 'coarse' or 'fine' scale - to switch between the coarse/faster-scrolling and fine/slower-scrolling adjustment of the centre ('FREQ') frequency, you press the large data wheel like a button. This switches between adjusting the frequency parameter in 1/6-octave steps, or adjusting it by 1/60th-octave increments.
We hope that we have been able to help you with this information.
Best regards, 
Chris Roy
Your BEHRINGER Customer Support Team


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## cixelsid

Part 2

Suggested target curves and setup techniques for the Behringer Ultracurve digtal	
Scope and Objective The following are some notes on how to use the UC and how to combine it with other measures to achieve the best results. It will only cover the EQ settings, room acoustics, suggested target curves etc. It is assumed that correct level setting, connection etc. have been sorted. Whoever needs more info on this may be directed to my review at enjoythemusic.com . Now, the Beringer, just like the Derringer it is some punwise compared to, is a specific tool for a specific job. Correctly applied the effect of either is mindblowing. Used wrongly it can make a bad situation worse. It is ESSENTIAL to understand a little about speakers in general and room acoustics, as well as understanding the specifics of your speakers. As a result this whole "little" article has to become quite long. Lets start at the beginning - the room..... The room of doom - why listening to a stereo indoors is bad idea For a room in which a speaker plays we find three predominant frequency ranges through which the room behaves dramatically different. For frequenies in the midrange and treble, where the room dimensions are large, compared to the wavelength of the frequency reproduced the room behaves in general diffuse but consistent. We shall call this range the reverbrant range, as this describes it's behaviour well. In this range we can think of sound to be similar to a ray of light and being reflected, diffused or absorbed by room surfaces. Depending upon the directionality of the speaker and the reflectivity of surfaces in the room that sound is radiated towards the room will reflect (or not) and generate a reverbrant field that merges with the direct sound of the speaker. If two or more speaker are used (stereo, multichannel) it is essential for correct reproduction that the acoustic envoironment around each speaker is as symmetrical and identical as possible. Gross asymmetry (like heavy drapes on the left wall and a naked wall on the right side) MUST be avoided at all cost. Please also note that few absorbing materials are consistently absorbing over a wide frequency range. I personally prefer the use of diffusion as much as possible with some absorbtion behind and around the listening position to excessive damping, as the latter often leads to a "dead" and unnatural, lifeless and amusical sound. Contrary to many claims made by their manufacturers, there is NO WAY to effectively deal with the reflections in the reverbrant range and the only real way to achieve a resolution of problems is physical, be in using controlled directivity speakers or absortive/diffusive room treatment. In most modern living quarters the reverbrant range starts somewhere in the 200 - 300Hz range. The larger the room the lower extends the reverbrant range, though usually the ceiling/floor distance is the limiting factor. The common 2.4m high ceilings will lead to a main mode at 143Hz and a strong second mode at 286Hz, demarking basically the reverbrant frequency range. Below the reverbrant range the room transit into the modal range. Here the behaviour of sound in the room is dominated by resonances, in effect the room becomes a resonant system where both the position of sound source and sound receiver (ear) with respect to room boundaries becomes a dermining factor in the resulting frequency response. In the modal range deviations from a flat response by more than 20db are common. Very few effective ways exist to deal with the problem in a mechanical fashion (room treatment) and even the various "Bass Trap" products are contrary to their marketing NOT effective in combating these problems. In the modal range we will encounter both pressure maxima (resonant reinforcement) and pressure minima (resonant cancelation). Certain effective speaker setup systems like Wilson Audio's WASP rely on optimising the speaker and listener position such that the number of frequencies at which the listener receives either a boosted or cancelled signal are minimised. However, again effectiveness is limited. As a rule, pressure maxima exist near all room boundaries and pressure minima are scattered like "black holes" across the room. In a pressure minima there is simply no sound pressure, thus no sound and deep notches occour. In a maxima there is a boost and thus at the room mode a strong peak. While there is no way "to fill in" a "black hole" a number of solutions exist to allow peaks to be dealt with. As you are reading this in the context of using an equaliser that you will chose to equalise these peaks out. This usally requires parametric equalisers or dedicated room correction systems (like TACT), simple graphic equalisers are not able to help, as they offer to coarse a view of the spectrum. Below the modal range our room behaves as more or less ideal pressure chamber. Room modes stop to exist at a frequency where 1/2 of Wavelength of the tone is longer than the longest room dimension. In my listening room this dimension is depending how you look at it 5.5m or very long (through the open door into the hall). For a 5.5m dimension the room becomes a pressure chamber (in my case a leaky one) below around 30Hz and here a notable bass boost is present helping to extend the bass of large speakers down as far as one wishes, assuming there is still SOME usable output. In this region the positioning of listener and speaker becomes competely irelevant. It should be noted that the demarcation between all three of these ranges is somewhat fluid so it is a question more of abstract judgement where one starts and the other ends. However once we are in the modal region all sorts of proverbial hits the fan and live becomes VERY interresting. In the reverbrant region things are interresting too, but primarily due to the stupidity of modern speaker designers. Only in the pressure region at very low frequencies is life easy and predictable. But usually not only do we have a lot of problems with the room, we often have as bad or worse problems with the speaker.... Friend or foe - the Loudspeaker designer and his products Considering the extensive body of knowledge that exists on room acoustics, the "average room" and so on it comes again and again as a surprise to me how ill considered and conceived most socalled "High Fidelity" loudspeakers are, especially if one compares the sittuation to the sound reinforcement sector and serious studio monitors (meaning NOT the Yamaha NS-10 or BBC LS 3/5). Speakers interact with the room in various ways, both due their own behaviour and due to the different regions of the room discussed above. The key to much of percieved tonality is the behaviour of room and speaker though the reverbrant range, because in this range we find the fundamentals of most instruments and the human voice as well as all the harmonics. Above I commented that in reverbrant range the speaker behaves a lot like a light source. For a long time two specific types of speakers have been sold to the unsuspecting public that have by design severe behavioural problems in this range. One type is the Full Range Dipole and the full range omnipolar speaker. Neither type is found in serious Studio or sound reinforcement applications for exactly this reason. Dipoles and Omnipolar Speakers create way more problems than they solve, by radiating energy over a very WIDE frequency range and also a very wide area. Think simply of a naked lightbulb in a room. THis is your omnipolar speaker and to a lesser degree your dipole. this makes the frequency response in room within the reverbrant range maximally dependant on the room symetry, absorbtion etc. and thus will require major efforts on room treatment to correct for the problem. Of course, a notable minority of listeners actually likes the presentation of omni's and full range dipoles and they by all means are welcome to their preference. Another major problem source are modern "conventional" HiFi Speakers. They missuse a semi omni directional HF unit (the dome tweeter was designed by Stu Hegeman for applications radiating upards and giving omni directional hemispherical response) in combination with a cone unit that at higher frequencies coming up to the crossover frequency is actually quite directional and as a result the directionality changes strongly with frequency. The use of non coincident sound sources also adds to the problem as now in at least one plane (lateral or vertical depending upon orientation) the radiation is turned into several "beams" going off on sides of the main (forward) one. All in one unholy mess and one that has done much to cause the rather unnatural, upper midrange emphasised sound of modern HiFi systems, by placing a depression in the upper fundamentals and a strong emphassis onto the lower harmonics for the response off axis, assuming a flat on axis response. Here you cannot really do anything with equalisation, as "fixing" the problem in the overall response will severely unbalance the on axis response and thus the first arrival of sound. THAT SAID, if no other measures can be taken to address the problem, equalising the in room reverbrant response flat will sound subjectively better. However, a better solution is to either switch to controlled dispersion loudspeakers or if a change of speakers is not possible at least all reasonable means should be employed to correct especially the excessively wide dispersion in the upper midrange of common dome tweeters. One of best solution are the type of diffraction control felt/foam rings embodied by the AIG "Imagers". http://www.audio-ideas.com/tweaks.html Room treatment of various sorts can be used effectively in this range to eliminate first reflection points (primary ceiling and floor) and an arrangement that places the speakers along the long room wall wide apart and strongly toed in can further help to reduce problems from non too ideal directivity of the speakers. I generally advise against removing sidewall reflections with damping material, as the sidewall reflections somewhat match the behaviour of a concert hall and thus are more less similar to the natural behaviour of music and can in modest amounts enhance the naturalness of presentation, diffusion is preferable. If fullrange dipole speakers are used (Maneplanars, ESL's etc...) it is essential that the rear wave is in some way diffused and damped, to avoid strong reflections. My preferred suggestion would be to place diffusors fairly close behind the Speaker, like for example variations on the Argent Room-Lense theme. Unlike with dynamic speakers, where a location close to rear walls is entierly possible and indeed even desirable (more later), dipoles require "room to breeath". The area behidn a dipole speaker should in addition be quite absorbtive, after all we need to kill most of the rear output. The same of course applies also to Omnis, but things are much worse there, because while dipoles have at least a "null" (an area with no sound radiation) on their sides omni's radiate to everywhere, while really radiation is only desirable into the direction of the listener. All the extra radiation needs to be diffused and damped. From the above it should be clear that before you attempt to equalise your speaker flat throughout the reverbrant range (or before you use other digital room correction products) you need to address as much of the directionality problems of your speakers as possible. Of course, some of us do not have such problems, this includes people with 12" - 15" Coaxial speakers, larger fullrange speakers (especially if additionally loaded into a front horn). I'll come back to this in the section on the modal region - but the old Hartley Concert Master Speaker (22" Dipole Woofer and 10" sealed box fullrange speaker) is surprisingly close to an ideal domestic speaker. Forever onwards Behringers Now in to the meatier chunks. Above I said that in the modal range the room becomes resonant. One option is of course to locate speakers and listeners in pressure maxima, that is near walls. So if you move your listening position as close to one of the long walls, centered and place the speakers along the other long wall, strongly toed in and (if neccesary) Imagered, you will have sorted many of the problems for the reverant range and you will have coupled yourself (listener) and the speakers maximally to room modes. Of course, the bass is gonna sound hugely boomy, but we have an equaliser for that. The one problem we will have is the floor/ceiling mode where we, due to a seated hight of around 90cm - 1m are precariously close to at least one pressure minimae. This usually falls around the 100 - 150Hz range and not much can be done about this, it will fall whichever way it will. in the context of the other main modes hoever both speakers and listener are well placed. As a further freebie placing speakers and listener close to the walls will both extend and increase their bass output, which after equalising the resulting system will have less actual power input at low frequencies and thus less distortion, so in the bass the system will player lower and louder without strain. Of course, such an approach only makes sense if a suitable equaliser or room compensator is in the system, it effectivley becomes an absolute requirement. There is of course another trick here which we have so far missed. While all normal enclosed dynamic speakers operate in the modal region effectively as omnidirectional radiator (regardless of being a domestic LF Horn, Reflex, sealed), using a DIPOLE to cover the modal range will actually result in a rather well behaved LF behaviour. The way the dipole interacts with room modes will result in a much more even LF response. What follows from this is of course that 99% of ALL HiFi speaker desgners got it all *** forward. They make hybrid Electrostatic or ribbon Dipole speakers that are dipoles where they should directional (midrange/treble) and that omni directional where they SHOULD be dipolar (bass). Normal Box speakers really can only take refuge to special room setups or equalisation. It seems in the last 70 or so years of sound reproduction only three or four companies ever "GOT IT" (or at least got it to some degree), namely the earlier mentioned Hartley, Celestion with their SL-600 & dipole Subwoofer combo, Gradient in Finland and the now defunked Audio Artistry headed up by Sigfried Linkwitz. Finally, below the modal range our best choice (should we wish to extend our response that low) is a simple monopole subwoofer, either sealed, passive radiator or vented with a suitably low cutoff. I hope this somewhat exhaustive coverage of room and speaker acoustics has helped a little to understand what can be done and what not. Nor onwards, onwards Behringers - we are going to put some of the above to use. How flat are you - equalising with (un)common and muscial sense Now we can put our new gizmo to work. Hopefully we have re-arranged the speakers and listener for maximal coupling to room boundaries or have changed our speakers to a pair of Oris 200 horns with a major bad dude style dipole woofer system, or if we insist on using badly designed speakers (strictly in the sense of working well within an acoustically small room) we have treated the room suitably and so on. Our first task is to understand the limitations of our speakers. Let's for arguments sake take an excellent canidate for equalisation, namely a Lowther Fidelio or Acousta. While offering fairly decent directivity, the response of the actual speaker is extremely uneven. If we do not employ an added active subwoofer (which can be integrated much better if an EQ is employed BTW) the LF response is realistically no lower than around 80Hz. Below this there is not much happening and while we could use the full range of the Behringers various EQ sections to make the System flat down to 20Hz, power handling would suffer dramatically. Thus first be aware of the limitations of the speaker. Then consider the classic "rule of the 400000". This states that if the speaker is 3db down at a given lower frequency it should be 3db down at an HF point that will multiplied with the LF limit give 400000. So if your LF cutoff is 80Hz and we will equalise the system to be flat at 80Hz (or 3db down) then we should retain a flat response up to 5KHz and then roll off at an equal rate as the LF rolloff. Thus before running ANYTHING like Auto Q please set your target curve in our case to flat from 80Hz - 5KHz and then with a gentle rolloff (say 6db) towards 40Hz and 10KHz with a steeper rolloff below 40Hz and above 10KHz. This will ensure that the resulting speaker will play musically coherent and "sounds right" plus, it will not attempt to make speaker do something it is not capable of, with potentially fatal consequences for the driver. Being lucky enough to own a system that is capable of a flat response from 20Hz - 20KHz (equalised) I chose to make my system for starters flat from 20Hz - 20KHz. Due to the sophisticated memory management of the Behringer EQ it is possible to overlay any "psycoacoustic" and personal curvers later, so may as well start with clean sheet. With some other systems (especially TACT but also the Accuphase DG- 28) it is neccesary to overlay any such psychoacoustic curves on top of the target curve, as a later addition of this is a little trickey. I come back to psychoacoustic and "generic" recording compensation is a bit, now for dealing with room modes on the Behringer. The TACT does that job for you and well, the Accuphase does it not at all, so the section now is "Behringer only" while all else said above holds a more generic applicability. While discussing the room modes I suggested that only a prametric EQ can actually effectively deal with them. Luckily the Behringer has three per channel on board. They have to be set manually and this takes time, plus a modified radioshack SPL Meter, but the results are very convincing. To access the parametric EQ's choose the Feedback destroyer section, and set all filter to manual. Then enter the analyser mode, chosing the front panel bypass allows you to do the setup without all noise blasting you away. Go into the Analyser setup and select tha analyser output to be a sinewave, suitably low in level. You can then use that sinewave together with your SPL meter to sinewave sweeep the 20Hz- 300Hz range, one channal at a time. If you use speakers with limited LF power handling PLEASE go easy on the volume. You need to make a first quick run to establish all major peaks. In my case these where around 45Hz (related to the mode of room and hall) and 60Hz (related to the 5.5m room length). Just note how much these peaks are above the rest of the spectrum in this frequecy range and the "peak" frequency. Program the attenuation and frequency into the parametric EQ, normally the three filters shouldsuffice, if you have more major peaks, go for the highest level ones. the bandwidth is a bit trickey, you could work that out from the measurements, but for most folks a simple interactive methode will work better. Start with a setting of 20/60 Octaves for the bandwidth, that is 1/3rd octave. Repeat the sweep and vary bandwidth and attenuation (and if neccesary center frequency) until you get a response around the former peak that is well integrated with adjecent frequencies. Bsically we don't want any drops or lifts, if the peak is not entierly symmetric it is better to allow a slightly depressed area than a peak. This process will take a while - once done write down the settings and save them to a program slot. Now you can pull the Mic out and connect it up. Place it on mic-stand (real or improvised) so taht the mic is as close to the point where your ears will be when listening. Make sure your analyser is set up correctly (pink noise output) at a sensible level (mine is at -28db), that whatever target curve you require is seleced and that you have selected a program containing the previously established settings for the parametric EQ's to kill the major room modes. Now hit Auto-Q and let her rip. The result should about match your target curve, though some speakers may require more boost at certain frequencies than the UC is happy to apply. If you feel this should be corrected, you can boost the sliders left out a little so that UC picks them up on the next run and repeat. I would recommend to look out for individual sliders that are either extremely boosted or attenuated compared to the rest of the sliders on either side of them. Take those back into normal regions. I would suggest to apply no more than around 6db equalisation, if you need much more than this it's either an extreme resonance mode somewhere or the Target curve was still unrealistic of the speakers actual abilities, so correct. In the end, both channels will be as flat as they can sensibly get. You can listen now but you may find the resultant too bright, as most recordings are mixed and mastered in non-flat envoironments. This brings us to some "default" psychoacoustic corrections that are expedient to apply. I do so by keeping a truely flat 20-20 Programm, one with my "psychoacoustic" curve and a third that combines both. This way I can easily go back and forth. I would recommend the following EQ Applications for "pleasant" sound with most modern (post mid 1960's) recordings or remasters of older recordings: 1) Boost the range below 125Hz uniformly by 2 - 4db (depending upon taste - I use 2db) and take the sliders above 125Hz so the form a falling slope back to 0db, with 0.5db (1 step) per slider or 1.5db per octave. I personally also have 20Hz at -1db compared to 125Hz and 31.5Hz at -0.5db. This is simply to slightly limit the LF Boost 2) Apply a similar slope (0.5db per slider / 1.5db per Octave) from 2KHz upwards, meaning -0.5db @ 2.5KHz and then on to -5db @ 20KHz. You may experiemnt with increasing the point where the olloff beginns somewhat. 3) Put a 4db notch into the response around 2.5 - 3KHz, returning to flat at 1Khz and 6.3KHz. This is the classic "BBC Dip". The resulting curve is what I use on a daily basis and was arrived at based on the study of the various literature and a noting down of the most often applied EQ settings. It offers a good compromise between neutrality, sweetness and pleasant sound. But feel free to vary this basic recipe to taste. I hope the above missive helps all of you Behringer purchasers to get best from your systems and the EQ.	
Having fun with an Equaliser and "Bauert Bands"....	
In the 1970's a Major Dude in Germany called Jens Blauert wrote about the effect of boosting/cutting narrow on the spatial localisation of sound sources. This is a piece of work on psycho Acoustics that is still not well known even in Germany, never mind anywhere else (Jens Blauert "Raeumliches Hoeren" S. Hirzel Verlag, Stuttgart, 1974). Anyway, the upshot of Mr. Blauerts work is that five critical bands exists that effect the localisation as front/back and high. In the context of Stereo controlling these narrow bands by a few db will give more "presence"or a more diffuse, distant sound perception. As most if not all modern and even many early recordings are clearly miked way too close and give a way to "present" sound for my liking I choose to investigate my "psychoacoustic" target curve for my digital EQ. A more present perception can be attained by boosting the 270...550Hz range and by boosting the 2.7...5.5KHz range (note the difference of a whole decade and identical bandwidth of these two bands), or indeed, reducing the level in those bands can reduce the "presence". I choose originally to boost the range below 125Hz uniformly by around 2.5db with a gentle rolloff towards flat that reaches 0db at 315Hz, thus effectively reducing the output in the 300...500Hz, so already one layer of de-presencing has been completed. In the upper midrange I choose to put a fairly deep (-3db) notch around 2.5KHz, which I felt left a good vocal intelligibility while making the sound much less "in yer face". Whenever I went higher with the centre frequency I found that the sound lost too much "clarity", very much in line with Mr. Blauerts ascertains. Both these changes where arrived at purely by listening, without knowledge of Mr. Blauerts work. They have been longstanding in my system in addition to a gentle HF rolloff (1db/Octave) above 2.5KHz. Based on Mr Blauerts points about the "height" band 7...9KHz I boosted this by about 3db and indeed got a bigger impression of Soundscape height by further boosting the range up to 20KHz in accordance I got more depth. Equally, a corresponding band is a decade lower between 700Hz and 1700Hz which I also boosted by around 3db. The effect of those additions to my "standard" psychoacoustic correction curve dramatically increased the sense of spaciousness and the backwards reach of the soundscape. I now recommend a target curve for the presetting of room equalisers and room corrections as follows: Look at the capabilities of your speakers in terms of low and high frequency reproduction. Choose a curve that is flat between two points with a simple first order rolloff programmed in the high range and low range that makes sure to be 6db down at the tuning point of the Speaker and with -3db points that if the frequencies are multiplied you get 400000. So if your speaker has a reflex port tuned to 40Hz the -3db point will be at around 60Hz. Divide 40000 by 60 and place a first order HF rolloff at around 6.666KHz from our formula, I'd say put it a little higher at 8KHz. This basic curve will make sure that the equaliser will not try to make your speakers work in ranges where they will be heavily distorted and that the musical balance (rule of the 400000) is retained. Add to the above curve a 1db/octave (on the Behringer EQ this means each slider is moved by the smallest possible step) above 2.5KHz (meaning 2.5KHz is FLAT). Now OVERLAY a secondary curve that boosts the range below 125Hz by 2 - 3db (more boost - warmer sound) with a gentle rolloff towards higher frequencies, reaching back to flat at 315 Hz. Boost the range between 800Hz and 1.6KHz uniformly by around 2db with a 1db boost in the two adjacent bands (630Hz and 2KHz). Place a -3db dip at 2.5KHz with a 3db/octave slope. Boost the range between 8KHz and 16KHz uniformly by 2db with 1db in the adjacent bands (6.3KHz and 20KHz). The addition of the above fairly complex frequency function will give a much more "pleasant" sound while it fits within the TRUE capabilities of the speaker into the tolerance field recommended by the IRT (German Industry body lead by the federal radio stations, similar to the BBC's now defunked research centre) for domestic/studio monitoring. If your speaker system genuinely is capable of covering the 20Hz - 20KHz range your speaker system will still be 100% compliant with the IRT recommendations for Monitors, while having a subjective tonality and spatiality that is much improved over a fully linear response. It should be noted that the above settings assume of course a speaker with controlled dispersion, if you have the classic 2-Way small monitor with it's very uneven off axis response you may have to modify the speaker and suggested curves to make the fit work.	

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


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