Both inputs can be hooked up. You select which to use via the I/O screen. |
Streetdaddy...The straightforward explanation is that the input level is too high. What signal are you puting into the DEQ2496? |
Ecruz...The Behringer has a "hard bypass"...you can turn its power off, and the signal will pass through it without any analog or digital processing. Try leaving all the wires connected and shutting the Behringer power off and see if the problem goes away. |
Streetdaddy...Try analog. |
Ecruz...What if the Squeezebox is turned on and in the room but not connected to the audio system? It apparently radiates a signal, and maybe that interferes with the Behringer.
Why don't you talk to the Squeezebox service people. |
I find that the auto EQ works fine for all frequencies, if "fine" means perfectly flat response 2Hz -20KHz (in my system). I have a multichannel system, and two DEQ2496, and I do the front left and right and the surrounds each channel separately. Because my room is not symetrical, and one speaker system (MG1.6 and a subwoofer) is in a corner the EQ curves are quite different. I have a single channel parametric for the center front, and, by comparison it is a big pain you know where. I do run the white noise up to 85 dB or so when I do the auto EQ. After the auto EQ is done, and while the noise is still on you can turn down the volume and see if the frequency response changes. Mine does not. |
Ecruz...
1. Auto EQ works fine below 80 Hz (regardless of what the manual says).
2. Don't try to get flat response any lower than what your speaker is capable of. If you do you will mostly screw up higher frequencies.
3. I use the "FAST" setting, and after things have pretty much settled out I go to MID or SLOW. However, I find that FAST doesn't actually bounce around any more than MED or SLOW. If you were trying to do auto EQ while various noisy activities were going on (perhaps during setup for a live show) the SLOW setting would help.
4. The Delta MAX, and the MAX SPAN parameters allow you to constrain how much EQ is applied. I leave these wide open so that the DEQ2496 can do the best it can.
5. I adjust the gain so that the sound (as picked up by the mic, is about 85 dB. Where this puts your system's controls depends on your system. |
Ecruz...In my experience 5 minutes is way overkill. 30 seconds in FAST mode just about nails it. Then I watch in SLOW mode, probably for another 30 seconds just to be on the safe side, but nothing much changes after the FAST phase. |
Warnerwh...I read it. Will there be a test? |
Ecruz...If you move it between the CDP and the preamp you can't use it for any other source. If you do move it up you should try feeding it digital from your CDP. Your output from the Behringer will be through its D/A anyway, so why run the signal through an extra D/A in the Sony and an A/D in the Behringer? |
Ecruz...What is a Squeezebox? |
I have not used the I/O selection capability, but from what I read in the manual (page 13) you could select either the digital or analog input, so you could accomodate one additional source. Try it. |
Ecruz...What does the error message say? I never saw one.
I would try removing the line conditioner and see if the problem still occurs. |
Ecruz...Seems to me that a "quality issue" would show up all the time. I rather suspect there is some kind of power or ground disturbance that the Behringer is picking up. Sometimes an input wire from a turned-off source can be noisy...noise that goes away when the source is powered up. In my experience this noise has been Hum, but the Behringer A/D could get all confused trying to interpret it as digital data. |
Tvad...How do you measure the imbalance? Did you create it by your EQ curve? 3 dB would be a fault condition for the electronics. |
I read somwhere that Behringer service was poor. I guess that advice was all wet. |
Streetdaddy...I realize that the white noise SPL is not displayed on the autoeq screen. The 85 dB (approx) that I suggest is based on measurement using my Radio Shack meter. I don't think that the exact level is critical, but to minimize the effect of ambient noise it should be loud. But not too loud, and if the bar level indicators on the left of the screen are hitting the top this may be the case.
Are you sure that uou have selected the mic input to be equalized?
Keep fooling around and you will get it figured out. |
Ecruz...Possible explanation ...
1.. The analog circuitry is just input and output unity gain buffers. These are easy to implement with near zero distortion.
2.. The D/A and A/D modules are about as good as they come.
3.. The heart of the unit is its digital processing which can't be tweeked by hardware mods. |
Streetdaddy...Glad to hear that you are making progress. My room is also very asymetrical and every speaker of the multichannel system is different. If +15 dB caused clipping your signal level must be on the high side.
What is the 20Hz EQ for the other speaker? I would try swapping the speakers...perhaps the problem (need for +15dB) is in the speaker - not the speaker location. |
RTA page 1 MAX and RANGE are just so you get a good display and will not have any effect on the EQ. You want to select MAX so that the display never hits the top, and you want to set the RANGE so that there will be something to see even during quiet parts of the music. Your settings will depend on the gain of the amps and sensitivity of the speakers, but on my rig MAX = -20, and RANGE = 60 works.
(MAX is negative, because it is dB down from the reference level. In other words, -10 is louder than -20).
If you want to roll off highs at 1 dB/oct hit the ROOM CORR button. Most people like this, but you have your choice. |
Streetdaddy...I have been down the RS meter/Rives CD, and with it I could never get the same results two times in a row. Perhaps it is a matter of exactly where you locate the mic or the meter. On the average, the RS meter results agreed with the DEQ2496, but were much more variable. Of course you should not expect agreement down to the last dB. The world is not that exact. I assume that you are using the tracks on the Rives CD which are supposedly corrected for the considerable errors of the RS meter. Also, I have heard that there have been several lots of RS meters, and all may not have the same errors. |
Streetdaddy...FYI I had to Google "Airport Express" to find out what it is. Yes, extra A/D and D/A are theoretically best to avoid. But a digital chain such as you are using needs to be consistent with respect to the format of the digital data. I think that the airport express has an analog output. Try it and see what happens. |
And to add to Ecruz's observation, if your speaker can't do 20 Hz (and few can) set that frequency to -15 dB so as not to waste amplifier power and to avoid futile effort by the speaker. |
Kck...When measuring ambient noise (in the home environment) it should be dead quiet. I think that the ambient noise measurement may be important for pro sound applications where things may be going on while the sound guy is trying to get set up.
You might try using the analog output of the DEQ2496 into your amp. The D/A in the Behringer is probably as good as the one in the CDP, and one less item in the chain might help to isolate your problem. |
Analog in and out is very high quality. The advantage of the digital interface is theoretical...maybe real but I can't hear it. I also have the DEQ2496 between my preamp and the electronic crossover going to the power amp. In general it would be better to put it in the TAPE loop (I don't have one) because the signal level would not be affected by the volume control. I avoid low signal level through the DEQ2496 by turning down the gain of the electronic crossover so that the preamp output signal is higher. |
Kck...Below 100 Hz is probably a range where EQ is badly needed. The DEQ2496 suggests that the automatic equalization may not work well below 80 Hz, but for me it works fine.
Here are a couple of suggestions.
1...If you want more bass, turn up the subwoofer volume so that the DEQ2496 is cutting it down where needed instead of boosting.
2...Similarly, when you set up the "target" response curve, instead of boosting the low end try cutting the highs. The "ROOM CORR" function will do some of this for you. |
Tvad...Unmusical? Check out RAM mods for the DEQ2496. |
Tvad...You don't need to select the noise as input. Doing the EQ thing automatically turns it on. The input for EQ is the mic (obviously). |
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. |
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. |
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.
What levels are you seeing on the green LEDs?
Are you having fun yet? |
Shadorne...I have noted that while it may take a lot of effort to get it figured out, once you do that it is easy to use. |
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. |
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. |
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. |
