Z-systems RDP-1 and similar models.
Restock, I lost the last part of your message. Maybe what I want to do is possible with this unit after all. Ideally, I'd like to put the eq between my CD transport and DAC. I've modified a CD player using a reclocking circuit to provide a VERY low jitter transport and I've hotrodded an MSB dac to be very good as well. But they are both coaxial. Are you saying I could use the Behringer after all?
Sorry about the lost message ...here's another try:
In addition to the optical input the Behringer has AES/EBU digital inputs/outputs. These should work with an XLR to RCA adapator for coaxial in/out. Yes, I think you should be able to use it. Note however that the Behringer will do some sample rate conversion (SRC). Overall I am not sure whether more jitter will be introduced by the Behringer (even probable).
As for the DAC after the Behringer: You can use the digital out (optical or XLR) to connect to a DAC and do the DA conversion in a higher quality DAC. As mentioned above, I am not sure whether there will be some SRC step in addition to the digital DEQ that might have negative effects.
As I mainly listen to analog, I hardly use the DEQ, but I was surprised how well it worked when running it into a Lavry DAC. Also I keep it around to do room measurements, integrate the subwoofer better, etc...
Overall a good component at a very reasonable price.
Hope this helps.
For $350 (including mic) get the Behringer DEQ2496. Only you can decide if its audio quality is "audiophile". IMHO the outboard DAC would be an unnecessary complication. If you decide not to keep it in the signal path (and I bet you will) it is still well worth the price just as a spectrum analyser.
What mid bass frequencies are you tryind to tame?
As far as I am concerned, notch filtering is only effective below 100 HZ. Above 100 Hz the problem of room size comes into play. 100 Hz = roughly 10 feet of wavelength, which means the distance between peak and trough is only 5 feet. At 200 Hz the distance between peak and trough is a mere 2.5 feet. This means you only have to move the microphone a couple of feet to get a totally different response from room modes.
Therefore if you try to PEQ "room mode" notch filter above 200 Hz then you are basically chasing your tail...move the mike a foot and you get a different plot. So don't go there unless you are correcting a deficiency in the system or severe seating position problem (like up against a wall). I may be a bit weird but I expect my system to sound good from all around the room...only three feet and less from walls do I normally expect a poor frequency response.
Also remember that notch filters are quite sharp....so they introduce phase distortion. Phase distortion is unimportant at less than 100Hz as you can't even tell which direction these low frequency sounds are coming from. This is not the case above 100 Hz...so a sharp filter above 100 Hz may cause some audible degradation...it may change the timbre of sounds.
I run my Behringer PEQ ONLY to the subwoofer signal for these very reasons.
Shadorne, I'm not absolutely sure what the frequencies are. I think there's a kind of large peak around 200 Hz. I've built bass traps that make the lower bass quite consistent around the room. My speakers are Infinity Preludes that I've separated the subwoofer from the full range tower to give greater placement flexibility and I've positioned them as best I can for a good presentation of depth and also working around the limitations of this being a home theater as well. Are my peaks the sort of thing that should respond well to repositioning withing a foot or so of where they are now? Are there other acoustic treatments that I could use?
Jlambrick...You say "I'm not absolutely sure what the frequencies are". All the more reason to get that Behringer. Its spectrum analyser (also called Real Time Analyzer, RTA) will show you what your problem is. No matter how you fix the problem, room treatment, equalizer, build a new room, understanding the problem is the first step. And, although equalization has theoretical faults, as Shadorne points out, in practice the benefits sometimes outweigh them.
So I say... get the Behringer as a spectrum analyser. Then see what equalization does for you. Trust your ears.
Yes if you love music that runs through one it is an audiophile's parametric equalizer. There simply aren't any rules in this hobby. No standards commision demanding that if something is to be granted the name audiophile it must meet the following criteria. For that matter anything that makes music that the listener loves is by definition audiophile. I don't agree with them much of time but its their party. I am not the sole arbiter of good vs bad. Is anybody's opinion better than someone else's who listens to music?
The best wine in the world is the wine you like best.
I've read very good reviews about the Behringer DEQ2496 and decided to buy one. When I attempted to buy the DEQ2496 from a Pro Audio Shop retailer (Sydney, Australia) he advised me against it as he has had many returned for repair. Beccause of the DEQ2496's unreliability he has decided not to stock them any more. This wasn't a ploy by the retailer for me to buy another model, as he had nothing else to offer in one box that the DEQ2496 provides. The situation may be different on your side of world.
It seems from my research the Tact and Rives provide a very good but more expensive solution.
I question the chorus of suspiciouly loud criticisms of Behringer, particularly with respect to reliability.
Just because a dealer carries Behringer, doesnt mean he might not be motivated to upsell customers to more expensive and/or higher margin products.
I am not about to get rid of my Audio Research and Levinson gear for a full Behringer system, but I use two Behringer amplifiers and a DEQ2496 with my PC and televisions.
They have all worked perfectly right out of the box, I have never turned them off, never had a single problem, they sound great and are dirt cheap.
If I was a manufacturer, distributor or retailer trying to buy groceries by selling mediocre, overpriced, high end voodoo gear, I might be tempted to bash Behringer too.
The rest of us should definitely check out the EQ and mic and see how it can improve your system.
I believe analog is the way to go for a variety of reasons. One is that the phase shift in the analog eq is the oposite of the phase shift created by a room mode--thus you are automatically correcting amplitude and phase. It's also extremely transparent and leaves the upper frequencies alone. As Shadome pointed out trying to EQ above 200 Hz (I might go as far as 250) is not going to work terribly well. Our EQ functions 350 Hz and below, but we do recommend that you only use it up to 250 Hz.
[http://www.rivesaudio.com/PARC/PARCframe.html]Rives Audio PARC[/url]
We are the manufacturer of this product, but we built this product for this exact purpose and built it to be as transparent as possible. The circuitry used in the parametric is the exact same in API mixing consoles, which are undoubtedly the finest analog mixing consoles made.
Thanks for all the responses. It might indeed be worthwhile to buy the Behringer for the sole purpose of being able to see what I'm up against.
The PARC looks like an amazing piece of equipment. Still beyond my budget at this point. I have some connections in the pro audio arena and maybe I could find a poor man's PARC at least until I can afford the genuine article.
Thanks for the thumbs up! You undoubtedly know a lot more about this than I do, since you build the well known and highly praised PARC.
FWIW, I haven't encountered a problem with the Behringer PEQ (I actually have the Feedback Destroyer Pro), however, this is definitely a really cheap piece of kit - so partly for fear, I only use it in the 0.1 channel below 80 Hz. If Eldartford says it is good then it reinforces my experience. I use a Ratshack meter to check settings and I walk around so as not to adjust for one single spot - a very slow and cumbersome process which in the end the "ear" adjudicates. In my defence of such an approximate process, room mode "bumps" are to me fairly obvious. Furthermore, I don't seek to squash everything flat ....I just take the "edge" of the nastiest bumps. My approach is minimalist, I guess. I am also slightly distrustful of automated software algorthms - I like to know what is going on and how much adjustments are being made - rather than let the software take over.
But be warned, I also own a cheap analog mixer by Behringer and it was totally unusable as it degraded the sound. So in general, as the old saying goes, you get what you pay for (I was lucky so far with PEQ and not so lucky with the mixer). Like all those cheap toys with lead paint being returned by Mattel to China, my constant fear with an amazing "bargain" is quality, will it work properly and will it last. Another issue is that you practically need an engineering degree to work the darned Behringer Feedback Destroyer, although I understand the PEQ 2496 is a little easier!
My comments are based on experience and a little back of the envelope physics. My comments also apply only to correction for room modes in the LF. If you want to "tailor" the sound (rather than correct for room modes) then an EQ can do the job well over the full range, the basic rule is to always use it sparingly. As you may now realize, I may be overly fearful of using such a cheap item over the entire frequency range.
Rives, interesting comment about the analog equalization - I usually read about the negative impact of the phaseshifts in analog EQ. I didn't know the phaseshift would be the same as the one imparted by the room. Do you have a good reference (even a very technical one) that explains why?
Also, I am curious to hear your opinion on Room equaliztion. The effect of modes on the spectrum of course depends largely on the speaker position and as well as the listening position. While I have not too much problem compensating for modes excited by speaker position, I am not so sure about compensating problems due to the listening position. I do listen not always in the sweetspot or listen with my wife. Wouldn't it be better to average over several microphone positions, rather than choose a single place to correct for?
My three Behringer DEQ2496 have functioned flawlessly for several years, and if one should fail I could replace it for less money than repair of the typical "audiophile" equipment. No doubt Behringer makes some items that are not designed to the audiophile market, but the DEQ2496 is not in that category. Again I say...get your hands on one and form your own opinion.
No doubt the PARC is great, although it has no RTA and lacks other features of the Behringer. Lexus cars are great too, but I drive a Honda.
Restock: I do not have a reference, though I imagine that somewhere in the Harmon reference white papers it's there--just a matter of getting through them all. It's pretty common knowledge amoung acousticians regarding the phase shift. As to the averaging and positioning, you are correct. We always recommend you work with speaker position and listening position before calibrating the EQ. You will find that once calibrated it's better and just about all locations, even if you calibrated for only one spot. We've done tests with averaging and usually get worse results because it's masks a portion of the problem. This is not intuitive and not what we originally predicted. It was literally hundreds of calibrations and testing that led us to this conclusion.
Restock--yes the physics are surprising, and I'm a physicist, but if you look at the results individually of what's then averaged it makes sense. The averaging becomes a mask of some of the biggest problems. Because the biggest model problems also yeild the biggest nulls at a different location. When you reduce the peak, oddly enough the null is not as larger an aberation relative to the rest of the field. However, if you average those nulls in then you never deal with the biggest problem axial modes fully. It really does go against common sense until you examine it fully.
I am with Rives. You should not average listening points to get an "average" signal as this totally defeats the purpose of correcting for room modes by smearing/smoothing out the problems.
However, in contrast, you should not seek to iron out every bump/null at every 1 Hz data point and get a ruler flat response at one spot....who is to say that one spot is the most representative of the signal quality within your sweetpot area (and after all your ears are 6 inches apart and do some averaging themselves). Who is to say that your zealous corrections are not making it worse 2 feet to the side or causing phase irregularities from sharp filters, for example?
Go for the big broad bumps at lowish frequencies and forget chasing 2 Hz nulls. Once you are satisifed take some measurements around the sweetspot and see how wide an area you can achieve. The biggest sweetspot comes from having a largish room and from treating all the corners and as much of the room as you can with absorption (includes furniture) that works down to say 50 Hz (big thick absorbers). Another factor is leakage....a room that leaks (wood frame) is better than a basement with five surfaces that are concrete. (although a leaky room may disturb others in your home it is usually better sounding)
Perhaps Rives can add whether it is best to stick with 1/6 octave or to do each frequency Hertz by Hertz up to 200 Hertz or so?
Remember also that many speakers produce 10% or more harmonic distortion at realistic levels when driven at low frequencies below 80 Hz (especially small ported designs) In this case your response plot may become misleading, as what you think is a mid bass problem may be caused by excessive harmonics from low frequencies causing mid bass bloom.
A bass guitar with fundamental notes between 50 and 100 Hz will produce 2nd harmonic between 100 and 200 Hz the second harmonic and higher will be part of the signature sound of the bass guitar anyway. However, your ears are roughly 10 decibels more sensitive to 150 Hz than 75Hz and therefore a second harmonic that is only 30% as loud as the fundamental will sound equally as loud as the fundamental note. If you add typical speaker or harmonic amplifier distortion to this then you can easily see how mid bass can get over boosted and the guitar much stronger than the musican/recording engineer intended...all without even considering any room mode problems.
Rives, your last explanation does make sense. Of course, I don't have nearly the
experience you do with respect to measuring different rooms and corrections ;).
Shadorne, thanks for your response - good points as well. So far I have not
really gone the equalization route (except for smoothing out the subwoofer
response). Mostly very careful room placement and a little treatment.
Shadome: We measure with a continuous sweep, so it's not averaged at a specific octave. However, we do have a psycho-acoustical response filter that integrates much like the human ear. Lower octaves are integrated with larger intervals than higher octaves. I also agree that corrections should be for significant issues--not 1 Hz issues, but the psycho-acoustical response filter takes care of that automatically for us.
Bflowers...Back in the 1950s when I became interested in audio, equalizers and tone control circuits were lousy, and audiophiles were right to avoid them. But the circuits that have been used for the last couple of decades do not have significant problems. However, it takes a long time to live down a bad reputation, and avoiding equalizers is an article of the audiophile religion. (You used the right word..."blasphemy").
I agree with the notion that no eq is the best eq. Even though we sell the PARC I'ld rather have a room that didn't need it, but that's not always possible If you need an eq, it's far better to add that into the circuit than let the room dictate what your sound is going to be like (particularly in the low end). The topology we use is straight out of API mixing boards. These guys make 1/2 million dollar all analog boards--considered the best in the recording industry. Chances are many of your recordings have this circuitry and much more on the recording end of things. So why not use it on playback if you need it? It absolutely beats the alternative.