After reading the raves about this product, I finally bought one along with the matching microphone tonite. Put in my system, eager to try room correction. The first 2 attmepts produced some curves that I wasn't crazy about, but seemed plausioble. Now, all it does is push all the bands above 125 all the way to maximum boost, and all the bands below 125 to maximum cut. When displaying the RTA of the pink noise, there is nop more htan a 15 dB range between the highest and lowest levels on the curve (as if that were small!)Also, one of the primary reasons I bought it was for equalizing low frequency room problems, yet it suggests htat anyuthing below 100Hz not be included in the auto EQ. Does anyone know why it is coming up with such odd equalization curves, even though it is reading the data, which doesn't look so bad? Also, how bad is the product at low frequencies?
Don't mean to crash the party, but I just wanted to add that I've put a DSP8024 between my preamp and the power amp bass channels in a biwired & biamped system with ML Aerius i's, and it works a treat for correcting my room's 80Hz resonance with -6db correction. Since my preamp has two pairs of outputs and the DSP8024 doesn't touch anything over 400Hz, I'm not worried about signal degradation - midrange & treble remain pristine.
The auto mic gain sets to +80db for Auto EQ even when the pink noise level is set to -32db, which is not loud but well above ambient noise. Not sure if this is relevant to the discussion about the DEQ2496. There is no auto level for pink noise on the DSP8024, and there is no indication in the manual or on any display regarding the appropriate pink noise or mic gain level. Still, it seems to work well.
Tvad...The clipping when digital data from a 16 bit CD is input to the 24 bit DEQ2496 is certainly unexpected. Perhaps Behringer can tell us how they load 16 bit data into their 24 bit machine. If it is truly the input that clips I suspect that the CD itself is the culprit.
In the making of prerecorded mag tapes recording engineers were prone to cranking up the level until a "little" clipping occured, so as to mask tape noise. Perhaps they are still thinking that a little clipping is OK.
"One has to have a background in sound processing, or have a good tutorial to properly take advantage of the Behringer, IMO."
The longer you have it and the more you use it, you will eventually become very comfortable with what it offers. I was also a bit initimated when I got it, but now feel fairly proficient in its use, and have been able to coax it to due just what I need for it to do.
Tvad...I still suspect that your clipping is internal to the DEQ2496 (clip limiter) as a result of equalization. A 16 bit digital word input really can't saturate a 24 bit D/A unless your processing has led to an output of more than 16 bits. If this is the case output attenuation should resolve the problem.
I have not used the digital input because my multichannel discs don't provide anything but analog.
I am having the problem of the input being to high as well. It is optical out of a Philips DVD player. I had to turn the Gain Offset in the Utility page down to -10 before the clip was not going off all the time. Is there a way to adjust the input, or is this what you need to do?
Zapper, turn up the mic sensitivity in the utilities menu. Tvad is right, you don't need a blasting signal, but if your mic is set too low, it won't "hear" the pink noise.
"While I appreciate the EQ features of the DEQ2496, thus far the DAC sounds mechanical, sterile, and unmusical in my opinion."
I also think the DACS are not very good in the unit. IMO the unit is very transparent if kept in the digital domain and output to a good external DAC (I use a Muse 296), but just doesn't really cut it using its internal DACS. When I got the Muse, I was able to quicly A/B both DACS, and there was a very(!) large difference.
I am having the "Noise to Ambient Ratio Too Low, Turn up Pink Noise Level" message when running the auto-eq. I have the level between -20 and -15, which is really loud when the pink noise is being analyzed. When I turn it up past the -10 level, it seems to work but is very loud, is this the way, or is something else wrong? My roo is dead quite, and the house is in the country, no outside noise.
As I understand it, you are lighting up the red LED while using the digital input. The digital data is as it comes straight off the disc, so there is no way to attenuate it. I assume that the MSB of the 16 bit CD word is treated as the MSB of the 24 bit word in the DEQ2496. True clipping will only occur if all the lower order bits are set as well as this MSB. If the digital input clips it means that the CD data was already clipped (but you never has an indicator to tell you this).
The red LED also lights if the CLIP Limiter inside the DEQ2496 activates, and that might happen due to your equalization when the input is near, but not quite exceeding the maximum. The CLIP Limiter threshold is permanently set at 0dB, but there are some parameters of the limiter that you could play with. Check the manual.
You can reduce the gain in the Utility Menu using the Gain Offset parameter. I have mine set to -2db. When I spoke to the Beheringer folks about the high input levels, they mentioned that it's Ok if the level goes slightly into the red zone as long as it's very brief. I would agree with that in terms of my listening experience, as I don't hear clipping artifacts unless the levels are significantly into the red.
I don't know the DEQ2496 but the Feedback Destroyer Pro took me a while to figure out. The PEQ gain and frequncy settings were obvious but the Q filter widths....well I had to use a calculator to work out what it all meant in terms of Hz so that I could plot out my corrections on a log DB vs linear frequency plot...
Pilot error may be linked to the fact that it does not indicate what settings have been programmed in ....you have to run through each filter to see what has been set.
but for the price....I'm not complaining! This machine is great!
Drubin...I agree. I think that the problem is partly due to the manual being a translation from German, but mostly because it is written for pro sound people who are pretty savy, and explanations which would be helpful for the average audiophile are omitted. Also, the darned thing does so many things, and with so many options that writing a good manual would be a real challenge.
Tvad..."betweeen 3db and 1 db depending on the recording". If you are looking at a stereo recording it is normal for the two channels to be different. To determine if the Behringer channel gains differ you need a mono CD (or a test CD like the one Dennon makes).
When I said that 3dB would be faulty electronics what I am suggesting is that some setup or EQ condition must be causing the inequality because circuitry is never that bad.
When you say that the inequality is measured at the input this suggests that it isn't in the Behringer.
When in DEQ mode, if you push the top small dial you go into a full spectrum gain mode. Tvad, is it possible you accidently raised the left gain when you were in dual mono mode? I think it would carry the imbalance on through when you switched back to stereo link mode in the utilities.
This might interest me given that most CD's are intentionally mastered with 1 to 3 db of added compression compared to the studio mix. Unfortunately a lot more compression is often used (especially on rock) to achieve a "loud" CD, where the artist simply wants it to sound louder than other CD's.
That link you gave is very good advice. Could not agree more with the authors criticism of many speaker designs.
I would only add that even if serious room acoustic treatments can be disappointingly ineffective for very low frequencies.....Soffit mounting of speakers is well known to be effective (by professionals).
There is a good reason for soffit mounting....it eliminates the problem of omni-directional bass frequencies which radiate towards the wall behind the speakers creating havoc with quarter wavelength cancellations. (The wall behind the speakers has the single biggest effect on the bass sound in a room as it creates the biggest coherent reflection....the second biggest issue is the wall behind your listening position...third are side wall reflections in mid and upper frequency ranges)
BTW....why do cheap car stereos systems often sound pretty amazing.....it is actually very simple, two words....Soffit Mounting.
BTW...why all the recent interest in bass traps....more speakers on the market that are nearly flat anechoically to 20 Hz, which radiate in all directions in the bass and play havoc with room modes...
For those who think it was as simple as buying a speaker flat 20 Hz to 20 Khz...do some research!
Tvad...I never tried the stereo autoEQ. My impression is that it just transfers results measured for one channel to the other one. That doesn't sound right to me. Perhaps it drives both channels for the measurement. Is that so? (Now it's your turn to be the Behringer expert!)
Smooth, rather than flat is the most important thing about bass response, and the flat autoEQ will give you this, so it is a good starting point. Your tweeking to achieve a frequency response that sound good to you can be done when you set up the target curve for the autoEQ. Many people find that they like different curves for different types of music, and the Behringer memory feature accomodates this.
The one Behringer gives as a default is not very good imo. Also you have to learn how to adjust the bass. The bass is critical and will affect the entire presentation.
Most important though is to learn what affects what. It takes some time to do this well. Fortunately you have plenty of latitude on adjustments.
Funny you say that Tvad, the first time I AEQ'd I thought it sounded like crap too. I really don't know what I did, but the last time I tried it, it sounded MUCH better. Anyway, I also ended up tweaking.
Tvad...I do each of my five channels separately (I have three DEQ2496). The EQ curves for the five speakers are quite different, although they are identical models.
There is a noise gain on the smaller dials. There is also a RTA mic input level in the utilities menu, mine is set to +15 volts. Be sure to place your mic as closely as possible to where your ears would be when you are in your listening seat. I can't wait to hear what you think of the room correction.
You will have to choose your input path. I believe you will output analog and digital simultaneously no matter which input you choose. By the way what do you think of the DACs?
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