Equalizer update - Page 9

More good links to good information about crossfeed:
Fixing the Blobs in your Head
How we hear
The problem with headphones
Fixing Headphones using electronics
Using Computers to fix Headphones

Digital crossover and DIY speaker system

Well, my DQ-10s have given up the ghost. While in the process of equalization, I heard distortion from the left tweeter. Since I "just happen" to have a replacement DQ-10 tweeter in my audio garage, I popped the left grill off to effect the repair.

Upon inspection, I noticed small cracks in the woofer and midrange surrounds. An inspection of the right speaker shows similar degradation. All five transducers in each speaker, except the supertweeter, should probably be replaced or reworked. But, the cost of obtaining replacements for all four drivers in each speaker is more than an new set of more-modern speakers.

So, I'm very intrigued with the notion of using this digital crossover to construct a new speaker system.

Any recommendations on an attractive, 3-way enclosure plus three well-matched transducers to be directly-driven by the six 100W channel amps in a Yamaha receiver? Should I buy a second digital crossover in order to drive a powered sub, or should I scale back to a two-way speaker design and use one or two subs as the third frequency band on a single digital crossover?

I see many potential benefits to directly driving each transducer in each speaker, ala a studio monitor. And, this project sounds like a lot of fun. But is this approach really likely to sound equal to or better than one of the many high-quality speaker systems out there at the same cost point (~2000)?

Greetings!

I thought the following might proove useful for recent owners of the DEQ2496:

1) Input clipping? (With regards to digital optical input into EQ)

The input level LEDs may occasionally flash red to signify 'clipping'. I thought this to be highly unusual as my input was pure digital from my DVD player's optical out. I used the below procedure and came up with the following findings:

First, activate the 'METER' button to display METER menu page 1.
With the CD playing, and with 'Input' as the source reading, even the loudest tracks should only produce a peak reading of '-0.0 dB'.
This indicates that the peak digital input capicity of the EQ was reached but never exceeded. Only when the peak reading 'Clip' is produced do you have a problem with the unit. Remember the above is only applicable to my experimentation with optical digital input.

My Bon Jovi 'Keep The Faith' album consistantly produced peak readings of '-0.0 dB' over and over again. On occasion, one or two Def Leppard tracks from 'Vault' woud exceeded '-0.2 dB' peak. While all Dave Brubeck 'Time Out' songs never exceeded '-0.9 dB'.

The same is also true for peak meter readings for output (as long as there have been ABSOLUTELY NO MODIFICATIONS to any settings like the GEQ, PEQ, Utility Menu etc...) However, once any of the above audio-related-settings are changed from the default, things get a bit impractical.

2) Inconvenience: output clipping

I realised that the red LED flashes I saw in the above situation were actually to do with output clipping. Of course this was not due to incompability with external equipment like my amp but rather, the EQ's ability to produce a higher output to meet my equilisation demands. (The LED always measures input signals but when clipping occurs, regardless of wheter it is output or input clipping, the red 'cliping' LED will flash.)

I discovered that this was due to the changes I had done to the GEQ. I had boosted the bass by a few dB and reduced treble by a max of 8.0 dB. So if I remove the bass boost, leaving only the treble cut, then clipping should no longer occur right? Wrong!

Output clipping continued to occur despite the GEQ being almost flat except for the drop in treble. I increased the treble again to completely flatten the GEQ and sure enough, clipping stopped. This didn't make sense. It seemed that any change of more than a few dB, wheter be it a boost or cut, would result in output clipping.

I finally solved the problem by changing the 'GEQ-MODE' (in Utility Menu) from the default 'TRUE RESPONSE' to 'UNCORRECTED'. Clipping stopped and only returned when there was a gain in any sector of the GEQ settings.

As stated in the instruction menu, the true response modifications help to smooth out the response curve. However, what it probably did not mention was that in order to do so, certain frequencies might actually exceed those seen in the GEQ settings. Either that, or the modified signal output has an accuracy of +/- 0.7 dB (for the case of my music and my relatively mild GEQ settings.)

So how can I get a true response from my desired equilizer settings without clipping? The solution was to drop the 'gain offset' (again in Utility Menu) to about -1.0 dB. (For my case -1.0 dB is fine. People requiring more extreme GEQ curves might have to go much lower than this. Perhaps past -3.0 dB mark and increase amplifier volume to compensate.)

Other modifications which resulted in clipping were adjustment made to stereo width (expansion; values larger than 1.0) and the PEQ (I've not experimented with DEQ/DYN settings just yet).

My impressions of the unit so far:

The above is my complaint with regards to the DEQ2496 at this point of time. Owners running a weaker analog in signal will probably be less restricted than users running unmodified digital optical inputs into the DEQ2496. Statistically, at least half of all changes to PEQ/GEQ settings will require owners to drop the unit's gain offset from the original factory default level of 0.0 dB. I'm fortunate that a drop of only -1.0 dB is sufficient for my purposes so far. Even if I keep the 0.0 dB offset, the maximum clipping I'll get is a cut of 0.7dB with my loudest songs, which is hardly noticable.

Finally, I would advise owners of the DEQ2496 (especially gerG who already has a particle stuck on the inner surface of the display) to minimise the possibilty of dust and insects getting near to the equilizer! An ant or two was seen walking around inside the display of the behringer! I saw one crawling out of the buttons of the front panel. Behringer could have done better produce higher quality buttons.

Other than the above two complaints, this equilizer has turned out to be my most significant audio purchase I'm ever likely to make from now onwards. Even upgrading from a buffered Cmoy to a Rega Ear or even a Talisman wouldn't be as significant.

Best sacrifice my wallet made for this hobby.

Hey jhenderson, if it's just the foam surrounds that have fallen apart, check out this site: http://speakerrepair.com/

They're in orange county so you could even drive down there on a weekend if you really wanted to.

getG -

The Yamaha has a special 6.1 input mode, which relegates it to a 6 channel amplifier, devoid of any pre-amp signal processing or routing (except for volume control). I was planning on running the six Behringer outputs into these line-level inputs, then adjusting the digital crossover to compensate for the amplitude differences and frequencies required for each individual driver. Once done, I think the Yamaha volume control will act as a pre-amp for all six channels. Think this will work?

I don't follow your comment about the "3.5 channel" operation of the digital crossover. If I drive each transducer in a three-way satellite with each of the three digital crossover outputs (for each channel), the woofer channel would be a lowpass, the midrange would be a bandpass and the tweeter would be a highpass, correct? How could I extract only LF signals for the sub without another crossover channel?

Wait.. I think I see. You're saying that I route the lowpass line level digital crossover output to the Yamaha amp line level input in parallel with the sub's crossover circuitry, so that they both see woofer LF signal, then the sub crossover rejects above ~90 Hz? Seems like this simply shifts the problem. Don't I still need an active, line-level crossover for use by the sub near the electronics rack? I wouldn't want to run a line-level signal across the room to a powered sub...

gerG -

What's your position on scaling back to a two-way satellite, plus one or two subs versus the three-way satellites + 1 sub discussed earlier? I am attracted by the simplicity of this latter approach: Each transducer gets an amp; The digital crossover has enough channels; I could procure higher-grade transducers for the same money, since I'd need only a mid-bass plus a tweeter. I'm thinking of something like a ScanSpeak 7" mid-bass + 1" tweeter for the satellites. Don't know what to use for subs yet. Perhaps I could just use my single, DQ-1 subwoofer initially.

I have a stereo Dyna 200 W amp w/ext capacitor for powering the sub(s). So, I am not short on amplifier channels, just crossover channels.

I've been using a DCX for an active crossover for a while now, and a DEQ for over a year, but I just found this forum. Most audio forums are quick to dismiss these items as "pro gear" and therefor not worthy of fooling with.

Glad to see you guys like to experiment!

For Flea, regarding the clipping issue, are you actually hearing the clipping? Unless you're hearing audible distortion, this is a relatively common indication for most digital devices. Though the CLIP LED's will illuminate, a correctly mastered CD will never actually exceed the 0dBFS ("decibel - full scale") limit, even when it is indicated as such. The actual signal level on modern mastered CD's can be literally -.01 dB short of full scale, but the resolution of the meter circuit under real-time conditions may not discern between this level and 0dBFS (max/clip). It is quite common for manufacturer's to implement a circuit that illuminates the CLIP indication PRIOR to actual clipping, so that the engineer can correct any issues before an audible result is heard.

You will know immediately if the input is ACTUALLY clipping, as this form of digital distortion is particularly uncomfortable to listen to. Also, the ULTRA-CURVE has a limiter built-in to it that should alleviate any possibilities of clipping within reasonable limits.

Az

When I first equalized with the 2496, I failed to constrain the permissible step between adjacent frequencies. I tried auto-eq-ing to provide a quick starting point. Well, the auto-derived response changes were fairly severe and when I tried playing some music using this curve, the equalizer clipped very badly. The sound was extremely distorted - very noticable. But, until I read Flea's report, I didn't understand the problem or how to fix it. So, under the right conditions, the 2496 can definitely produce audible clipping.

It's great to talk with someone actually using the DCX the way I intend to. I am just now assembling my satellites and subwoofer in preparation for tri-amping using the DCX. I have about five-hundred questions rolling around in my head about this configuration.

To start, I am concerned about protecting the tweeter. Though I intend to be careful, during initial configuration, I could steer an inappropriate frequency band to the tweeter. Is there an effective way to mitigate this risk? Will a fast-blow fuse do any good?

What procedure do you follow in order to adjust the filters on the DCX to insure optimal flatness at the crossover frequency? Similarly, how about insuring proper phase response for each driver? Is it reasonable to do DCX configuration in the listening room, or must they be performed in a near anechoic environment to achieve any reasonable accuracy?

Is there a text resource that I could read that explains some of this?

What kind of tweeters are they? I use some old EV T-25s and they're pretty much indestructible so that's not a worry. Dome tweets are probably the most delicate. If you can find the specs on the driver, it will give you a good idea how much extension you can have. OF course, the main killer of tweeter is clipping, and one of the beauties of active bi and tri amping is that your tweeter amp will never be stressed by big bass notes and will clip the tweeters a lot less.

For creating the crossover points, I used an RTA. I couldn't find specs on all my drivers, so I measured them while the passive crossovers were still in the cabinets. Unhooking all but one driver at a time gave a very clear xover point and curve on the RTA. This gave me a reference point to start from. If you have the specs for your drivers, you can get a pretty good idea where to start from there.

After that, it's a fair amount of trial and error. I started with a 48db Linkwitz-Riley arrangement at the xover point of the orginal passive crossovers (which were second order Butterworth) Using the RTA, I simply tried several combinations and looked for smoothness, listening to the setups that seemed better as I went.

The end result was a far smoother crossover point than was recorded with the speakers using the passive xovers they came with. I ended up with 4th order L-R crossover type, but that's just what worked best with my setup. Theoretically, a steeper slope will most often be best, but not always. It depends on the cabinet, drivers, etc.

The DCX automatically corrects phase errors, so that's not an issue. (For that matter, L-R slopes above second order have very few phase errors anyway) There are tests for this, but they're long, complicated, and not needed. So unless you really want to know, I won't bother.

I see no reason to try to get any kind of anechoic response out of your speakers since they'll never be used in an anechoic chamber. Set 'em up for the room you'll be listening in. Another great thing about active crossovers is that it's simple to change things if you change drivers, the room, or whatever. You can compensate somewhat for room problems with the xover setup. Try that with passive crossovers!

Az

I am building North Creek Rhythms as the satellites and two Poseidons for subs. The Rhythms normally use a ScanSpeak D2905/9900 for the tweeter, but the owner of NC now recommends using the North D25-06S silk-dome tweeter instead, so that's probably what I'll end up with.

I am really happy to hear you discount the damage susceptabilty. I thought that I'd use an existing set of full-range speakers to rough-in the crossover settings for each amp output, then migrate over to the new speakers.

When you measure the in-situ performance of each of the individual drivers to determine their specific frequency responses, do you position the mic very close to the speaker in order to minimize room effects? Or, should that be done at the normal listening position?

I've purchased both a DCX and a 2496 equalizer for use on the system. But, do you find that both are useful, or does the DCX handle the crossover function as well as room equalization by itself?

What about subwoofer equalization. Though I plan on building some bass traps and improving the room, presently, it's a regular 15 x 15' living room with hardwood floors, rugs drapes and furniture. It has a vaulted ceiling and the rear wall has very large arches that open to the dining room. I am suspicious of all RTA measurements below 100 Hz. How did you configure for optimal bass response?

Wow, those rhythms look awesome! Those are going to sound great actively bi/tri amped.

Yes. Get them flat with close micing, then you can move the mic to the listening position for level matching and room correction.

I also use the DEQ2694 for RTA and SPL meter purposes. Originally I was not planning on inserting into the loop, but since I can use the digital in, and then the digital out to the DCX, and the DCX has limited processor power for EQ, I have it in the loop and it works great.

For the sub, you may find it useful to have some sort of parametric EQ for room modes. Although the 1/3 octave RTA doesn't really have the resolution for some low bass peaks and valleys, you can still use it to find them with a little detective work and a test tone CD. A Behringer Feedback Destroyer is a very inexpensive solution for providing some sub EQ since you're using all six channels of the DCX for your mains.

Az B -

What are you using as a digital volume control? I'd like to drive the CD player TOSLink output into some sort of digital pre-amplifier providing volume control adjustment in the digital domain. The only place I intend to go analog is at the outputs of the DCX, so I think I need a digital volume control somewhere in the TOSLink path.

Originally, I planned on using a Yamaha receiver as both my 6 channel amp and my digital volume control. But now, I am concerned that the Yamaha's 100 W/ch may be inadequate (for the subs, particularly). Worse, there is a technical snag with the Yamaha: It features an input source mode in which the 6 amp channels may be directly driven by six RCA inputs (perfect for the DCX). But, this is mode-selectable. So, if the unit were ever inadvertently switched to any other source (such as the FM tuner), full-range signals would be driven out of all of it's amp channels, damaging my tweeters. A shame, since the only source I intend on using is the CD, but the FM tuner mode can't be locked out.

I could use a laptop PC as my source, via Foobar (my CD collection is already ripped lossless), but I'd prefer to avoid this if possible. It'd be nice not to have to 'boot' my stereo.