Arkytype

In the mid '70s, one of the Klipsch reps drove a VW van with a pair of La Scalas in the rear facing forward! Stereo separation wasn't much but they would sure part your hair. For those attending the Hollis' Goat Roper "shootout", does anyone remember bringing a CD (on the Columbia label) that had a really strong opening bass line? The artist was perhaps Indian and the cover art was an abstract form. The music was New Age/Alternative ? Lee

Bob, et al, Obviously, I was misinformed by someone at the Heritage Gathering about the grand prize. Sorry if I got your hopes up (no pun intended). Lee

Bob, There will be many posts from the attendees in the next few days. Trey, Roy and the other Klipsch employees could not have been more welcoming of our presence and the many questions we had aboput new and current products. My highlight was visiting the old lab and listening room which is now filled with dozens of older Klipsch models as well as other horn loudspeakers from other manufacturers. My last visit there was around 1979 when the lab side was filled with commercial and several home-made testing gear and the listening side had a three channel Klipschorn/ Belle setup. There was a rumor that a factory tour was planned but Trey and the others kept it a secret until the last minute. The place is huge and has to be seen to be believed!! Some of us will post some pix later. Lee Unless you have talked with Daddy Dee today, you may not know you won the grand prize drawing Staurday night---a pair of Heresy IIIs !!!! (We shoulda had a "you must be present to win rule""!!!) Congratulations.

First, we are concerned with shaping an omnidirectional LF polar response to conform with a cardioid polar response without changing its frequency response. With a cardioid subwoofer we are essentially taking a 'ultra low', essentially non-Q device and modifying it into a higher Q device without modifying its frequency response. You also mention shaping the units "frequency response" into a "cardioid frequency response" pattern, a goal that is as spurious as it is non-sensical. We have no interest in affecting a "cardioid EQ", whatever that is! Of course, I should have written "cardioid polar response". Czerwinski's horn has little to do with a cardioid subwoofer. Despite its attempt to reconfigure the Q of the unit, it has little or no application within the niche we are talking about! Who says that the horn he proposes can't be designed for a low frequency cut-off of (say) 20 Hz? Would that meet your definition of a subwoofer? And a low frequency horn is not an omnidirectional device. Says who? If a subwoofer or low frequency horn (fc=20 Hz) is positioned near a planar surface with dimensions approaching the wavelengths of interest, then, by definition, neither will exhibit an omnidirectional polar response pattern. However, if we fly them with rigging above the center of a large hall, then they would exhibit true "omni" directional polar response pattern. We are specifically dealing with mocifying the polar pattern of an omni-directional source to increase it's (sic) Q and to provide a larger measure of well-behaved control over its radiated acoustical energy. I stand by my original response to your statement: Perhaps if you read Mr. Czerwinski's patent more closely you would see that he is proposing exactly the same end result as the DIY Cardioid Sub article: directional control of a loudspeaker's polar response. To quote Czerwinski's patent application, "The output of the (horn) loudspeaker is a monopole and therefore is omnidirectional." "...When the first and second (dipole) transducers are added...their radiation output in front of the loudspeaker is a dipole.....The combined monopole and dipole produces a cardioid-shaped wave." How does the end result Czerwinski proposes differs from that in the Do-It-Yourself Cardioid Sub article? To quote Shirk, "Herein lies the trade-off. The amplifier power required will increase as frequency decreases..... Two octaves below the design frequency, the output capability of the array will be 6 dB lower...." Congratulations, you read well! Unfortunately your understanding and conclusion that this approach requires "10 times" the power is completely absurd! Let's take a typical horn-loaded woofer with a sensitivity of 100 dB SPL/watt . Is it absurd to think that the same woofer mounted in an open-back baffle would have a sensitivity of 90 dB SPL? I was not referring to the example in the DIY article when I wrote the 10X power factor. Wonderful math, except you demonstrate your complete ignorance of what constitutes an octave considering a sub-woofer is not being used to reproduce mid-bass frequencies!! For a sub-woofer, the most common lower two octaves would be 20Hz-80Hz! The example given in the DIY article is a subwoofer with a high frequency cut-off of 140 Hz. Guess the author is demonstrating his complete ignorance of what constitutes subwoofer frequencies. And as anyone would realize if they read for meaning, as the frequency bandwidth increases, the increasing effects of comb filtering renders this technique problematic, both from a frequency prespective and from a polar pattern perspective! Are you stating the obvious for a reason or did you think I might not have grasp that concept by reading it in the article? Providing a 6dB increase in amplification is not prohibitive! So let's carry this through to the logical conclusion! We are dealing with a subwoofer! A subwoofer whose response will be crossed over on a high end at ~100Hz( 2 octaves ranging from 25-100Hz), or more appropriately at ~80Hz. "Two octaves below the design frequency" of say 80Hz is 20 Hz. 6dB gain is not 10x the power required! Once again, I never stated that 10x power would be required for the DIY article example. Keep re-reading my explanation above--maybe it'll sink in. And to say that more power is required for equivalent gain at low frequencies is a given! Unless you know of commonly used woofers that are more efficient then MF or HF drivers! I was only repeating the trade-off of the proposed design in the article--I mentioned no other type of loudspeaker. And since you yourself make referance to a KHorn, one wonders to what mystical LF transducer exhibiting an efficiency greater then a MF or HF driver you are referring!?!?!?n The K-33 woofer???? Right!!! My reference to modifying my Klipschorns was, like the valve stem remark, written tongue-in-cheek. Wonder if the above statement (as well as your petty personal attacks) was written head-up-a...?

"This would be accomplished by filling the enclosure with a gas whose density is less than that of air. Gene, would this speaker come with a valve stem for periodic replenishment? " Since you didn't bother to read it, I'll help you out here. The gas should be heavier than air, so it won't leak out. Dayton Wright used SF6 in some of their speakers, although for a different reason. djk---Don't presume to know what I have or have not read. I did read Czerwinski's patent several times. Since I didn't have access to gas density info and the formulas involved at the time, I relied on Czerwinski's patent wording and erroneously concluded the gas density needed would be less than that of air. "This patent (like a couple of his other ones) is an apparent cure for no known disease. " Lack of low frequency directivity is a real problem in PA, as dragonfyr mentioned. I'll concede that lack of low-frequency directivity can be an issue for venues larger than a home listening environment. Actually, the C-V patent is more of a dipole than a cardioid. Dipoles can exhibit 6dB of gain compared with a simple sealed box. Too bad you didn't understand this either. Apparently, it is you who have not read or understood Czerwinski's patent. In it, he states (no less than six times) that, "The combined monopole and dipole produces a cardioid-shaped wave.". Now, I may be an Arkansas Hillbilly educated beyond his intelligence, but Czerwinski's patent description doesn't sound like the resultant polar pattern is a dipole to me. The dipole driver also fires into the horn, the parameters of the driver determine the load match for the distance down from the throat. See the Unity patent from Tom Danley if you don't understand how this works. The Danley patent you cite (U. S. Patent 6,411,718) has nothing to do with dipole drivers or modifying the the polar response of a loudspeaker. The rear radiation from each driver is sealed.

Huh? What in hell does this have to do with a cardioid sub? This has quickly become very confused! Or, more correctly, the replies are rather confused! I think djk was simply responding to your post with info on another approach to achieving a cardioid frequency response. Your insights, I mean observations, or, well, simply assertions, have NOTHING whatsoever to do with what I posted nor the topic of this thread!! And that would be in violation of Forum rule number.......? "Hmmm. Did (YOU) account for the fact that" Czerwinski's horn has NOTHING to do with a cardioid subwoofer. Perhaps if you read Mr. Czerwinski's patent more closely you would see that he is proposing exactly the same end result as the DIY Cardioid Sub article: directional control of a loudspeaker's frequency response. And "In the patent dragonfyr refers to", Dragonfyr NEVER referred to any such patent! In fact, I have never even heard of the $*#&% thing! My mistake, I should have written, "In the patent djk refers to..." May I suggest that you go back and try reading the original post! And with the wealth of well behaved constant directivity and other associated horn formats offering various well-controlled and behaved Q's, I am not sure where it even fits. Nor do I have any idea what the cardioid horn has to do with the cardioid sub that uses the principle of superposition to achieve its resultant polar pattern. And superposition has little to do with excessive amplifier power! In fact is is almost independent of it, provided that the various sources are closely matched. To quote Shirk, "Herein lies the trade-off. The amplifier power required will increase as frequency decreases..... Two octaves below the design frequency, the output capability of the array will be 6 dB lower...." That means (using his example) at 35 Hz you'd need four times the amplifier power than at 140 Hz for the same acoustic output. If you are averaging 250 watts at 140 Hz, you'd need 1,000 watts at 35 Hz. Cardioid subs are rather common and are a proven commodity. This was not a proposed concept paper! And where does this "10X" power requirement come from???? It's probably a fair assumption that (in Czerwinski's patent) a horn-loaded woofer would require a tenth of the power of the non-horn loaded woofers shown in the drawings. And cut holes in the back of your KHorns?[] My first reaction would be to fill in some of the holes located elsewhere! What does any of this nonsense have to do with a cardioid sub!?!? I am very confused regarding the additional non sequitur posts!!! But it appears that I am NEITHER the only,nor the most confused person here ! It appears I'm in a battle of wits with a half-armed opponent. []

Eugene Czerwinski is better known as the founder of the loudspeaker company Cerwin Vega. This patent (like a couple of his other ones) is an apparent cure for no known disease. Perhaps his most noteable patent is # 4,101,736 which offers a " ...Device to effectively enlarge the volume of the speaker enclosure...". This would be accomplished by filling the enclosure with a gas whose density is less than that of air. Gene, would this speaker come with a valve stem for periodic replenishment? It would seem it is easier to get a patent for an "improvement" of an existing invention than for the invention itself. In the patent dragonfyr refers to, Czerwinski cites several well-known existing patents and then offers (without proof) that adding woofers to the outside of a bass horn will create a "cardioid-shaped wave." Hmmm. Did he account for the fact that the external woofer would probably require ten times more power than the horn-loaded driver to have the desired effect of acoustically combining to form this long-sought-after Holy Grail of directionality? Can't wait to cut holes in the rear of my Klipschorns to try this.

The Malco Theater in Rogers, Arkansas (exit 85 off I-540)is Klipsch powered. The owner of the Malco theater chain (which started in Memphis) lives in the area and a Klipsch engineer who also lives in the area works with him. It may be that all 35 of the Malco theaters in the mid-south use Klipsch, but I can't confirm that. Rumor is the Malco owner is building a multiplex theater and an Imax in the NW Arkansas area. They too will be all Klipsch powered. The major difference (improvement) I hear is the dialog is very well defined. The siblants aren't spitty or sound compressed as in other venues. Lee

Al, Thanks for the kind remarks. The box was constructed from spare parts needing a good application. The simplest version to build is using a 4PDT relay which you control from your listening position with a pushbutton. If you have a Tripathi-based amplifier, you cannot use a common ground as the outputs are floating. The 4PDT relay allows for switching of the individual left and right channel grounds. In addition to a simple A-B comparison, the other circuit in my box allows a polarity inverting function with the ability to select a combination of which relays are slaved together. This allows independent polarity swapping of both squawkers or both tweeters. If there's enough interest, I can post a schematic for the DIYer. While this is not an A-B-X box as described in my earlier post, it will help you make informed comparisons. Lee

Dean, Bob, Al, et al: (sorry, couldn't resist) Guys, Thanks for the enlightenment & confusion :>) about an interesting topic--the "sound" of passive components. The subject of measurement vs. audibility is familiar to many of us whether it involves cables, capacitors, amplifiers or other devices in the signal path. IMHO, to properly assess the audibility of Component A vs. Component B, a double-blind test using an AB-X switch box is the only "fair" way to eliminate ones prejudices. Not knowing if you are listening to Component A or B (or no change) when you press the button levels the playing field and usually quiets the BSers. Recently, when I received my Tripathi-based TEAC A-L700P amp a few weeks ago, I was convinced it had deeper bass than my 50 WPC McIntosh amplifier. After manually swapping amps several times it became clear a switch box was needed to make a faster A-B comparison. I built an A-B box that would allow me to not only switch between two amplifiers, but also had inverting polarity terminals for testing the audibility of absolute polarity. After using my SPL meter to carefully match the gain of both amps, I proceeded to A-B the TEAC and McIntosh. I was surprised to hear NO audible difference between the two. At first, I thought the box wasn't switching properly. No matter the program source (pink noise to Pink Floyd), both amps (to my ears) were identical. The deeper bass initial impression?---obviously a subjective evaluation on my part. Could it be I wanted to hear deeper bass? Having satisfied (for now) the issue of amplifier differences, I tried "A-Bing" for the audibility of polarity. The tweeter outputs from my ALK ES-400 networks were fed to the A-B box and the outputs were connected to the Beyma CP-25 tweeters. In position A, the polarity of the signal to the tweeter was normal, and in position B it was inverted. BTW, this test was done using each Klipschorn and then both. Again, pink noise to Pink Floyd and no audible difference. With the Ivie IE-30 SPL at the listening position and listening to pink noise, I could see (and hear) the comb filtering (from the acoustical combining of the squawkers and tweeters) in the 1/3-octave display. Changing to the opposite polarity only slightly shifted the display. Moving the B&K 4133 mike left and right a foot or so easily duplicated the polarity change seen switching from A to B or B to A. The SPL meter detected a change, but with the short wavelengths involved, there was probably an infinite family of destructive and constructive interference going on between the two squawkers and tweeters. The next test was more revealing. Swapping the polarity of the squawkers was plainly audible. Finally, a comparison that could be heard!! Simple physics would demonstrate the wavelength emanating from the squawkers vary from about three feet to three inches. Our ears are far enough apart to make polarity inverting audible. Everyone has done the next A-B comparison. Swap the polarity of one bass driver while playing source material with deep bass. You should hear a pronounced difference when both drivers are properly connected. This is usually referred to (incorrectly) as "phasing" your speakers. Getting back to the audibility of Component A vs. Component B: If one accepts the premise that we don't have an "audible" memory, stopping a test to change out a capacitor, inductor or other device would seem to negate the purpose of the test. If (for example) Capacitor A caused a subtle high frequency roll off and Capacitor B didn't, an A-B test (or A-B-X test) would (or should) readily identify Capacitor A. If, on the other hand, it takes a few minutes to manually swap components, could one hear the same roll off? If our quest is to realistically re-create in our listening room a live performance, then the efforts of Al, Dean, Bob and many other Forum members will continue to advance the art. I'm only suggesting that we might need to re-think how we arrive at the perceived audible differences between two components (active or passive) by changing our testing methodology from a subjective/opinionated-based one to a more objective/scientifically-based one. Lee P. S. I'm aware my swich box is A-B only and my testing was not douoble-blind. My purpose was to hear differences, not which sounded "better". I'm working on an A-B-X box design.

Hey, Seadog, et al, I'm still offering to host a mini-Pilgrimage with or without Trey's blessing. If we can't arrange for a tour of the plant in Hope, there is a Klipsch engineer based in Rogers, Arkansas who is working with the owner of the Malco theater chain to develop new pro audio loudspeakers. All the Malco theaters are equipped with Klipsch loudspeakers. We could possibly get a tour behind the screen and watch a movie. I saw "Ocean's 12" there a few weeks ago and while there weren't a lot of surround effects, the dialog was crystal clear. Popcorn was good, too. As an alternative, there's an ElectroVoice pro loudspeaker manufacturing plant not far from Conway. It has an amazing CNC router that cuts loudspeaker cabinet parts in just a few seconds. They make almost all the components there from winding the voice coils to charging the magnets. We could probably get a tour to see the "enemy's" secrets. Bob Crite's (BEC) place is just a little further down the road from EV and we could see his place, too. After that, it's lunch at Whataburger. Trey and/or Amy is certainly invited if the event happens. May would be the best month for tolerable weather. In the summer, the temperature and humidity are about equal from June to August. While a trip to Indy would be informative and entertaining, it probably wouldn't allow attendees to bring their single-ended tube amps or loudspeaker mods. The 2004 Klipsch Gathering was a well-attended and received informal listening & BS session. That's the atmosphere I'd like to contnue with Klipsch Gathering 2005. BTW, my belated vote is for Hope. Lee

Be aware that most variable autotransformers will deliver up to 140-160 volts. You might want to measure the voltage and put a mark on the dial at 120 VAC. Over voltage usually causes more problems than under voltage. BTW, General Radio patented the Variac in 1934. The term Variac, like Coke and Kleenex has become a generic word for a trade name registered by General Radio way back when. Lee

Dang, I'll write "There's an S in Klipsch" 500 times. Lee

A few weeks ago, I approached Daddy Dee and Troy about the possibility of another Klipsch Gathering in Arkansas. Last year's was a lot of fun and we got to listen to some cool gear and swap lies on the porch. FWIW, here's an alternative idea for Klipch Pilgrimage 2005. We could schedule a plant tour in Hope on Friday morning say at ten o'clock. That way, those who fly into Little Rock Thursday can caravan to Hope. Those that live closer can drive from Dallas, Shreveport, etc. directly to the plant. After a tour and lunch at PWK's favorite diner (on Trey's tab), we can drive to Conway (where I live) and spend the afternoon and early evening setting up and listening to any gear the attendees might bring. After dinner (on Trey's tab), we can return for more listening. Saturday would be spent listening and swapping lies. Since my listening room is pretty large, perhaps with a little persuasion Trey could ship some of the new Reference products and electronics for all to audition. Other attendees can feel welcome to bring loudspeakers, or electronics for a show, tell and listen. I can accommodate three or four who don't mind a couch or air bed for a couple of nights. I cook up a mean southern-style breakfast, too. Just bring doggie treats for Hemmy and since Faulkner county is dry, your own booze. Lee

90 dB sounds about right. Unfortunately, those of us who mixed rock and roll in the wee hours of the morning in the '70s may have started out at 90 dB. After an hour or so, the amps were clipping and nobody noticed. Must have been the medicinal herbs that were prevalent back then!! Recording engineers wanting "real-world" speakers would buy the single 4-inch driver Auratones and set them on the edge of the meter bridge. If you can hear all the instruments on these, then your mix was done. Lee

Colin, Print the F-M family of curves then invert the page top to bottom. Looking thru the page, 20 Hz should be on the upper left frequency scale and the lowest SPL curves will be at the top. Now you can get a better picture of the human ear's "frequency response". As you can see, human hearing is not "flat" at any SPL. A question for all you budding recording engineer out there: at what SPL level should you mix a multitrack recording to a two track mix and at what level should the consumer listen to the recording in his or her home? Try this site to test your hearing. Haven't tried it, but it looks interesting. http://www.phys.unsw.edu.au/~jw/hearing.html Lee

Gil, According to Roger Russell's web site, the Klipschorn was reviewed in the November 1986 issue of "Audio". Lee

Mike, Are you saying there is a null centered around 77.8 Hz after running the FBD in AUTO? If so, then that's normal and, as you suggested, is the floor to ceiling frequency. Frequency of mode = 565 Ft/Sec//8 feet= 70.6 Hz mas o menos. If you are measuring a null at 77.8 Hz without any filter engaged, try moving the mike up or down or left or right. Peaks are a lot more common than nulls. Colin, The general rule of thumb (or ear) is that you can't fill a hole or notch in the response. Besides, you can't hear them anyway. Keith, Your AUTO EQ settings don't look too bad. You've wisely disabled the lower frequencies in AUTO mode. Try miking about a meter from each loudspeaker and don't forget to mute the one not being EQ'd. "Flat" measured response at the listening position is usually too bright. Using one meter measurement distance will allow the woofer, squawker and tweeter to meld their outputs and will also minimize room contributions. Get the loudspeaker "flat" at one meter, and it'll be 'bout right ten or fifteen feet away. Daddy Dee, That a good price. FWIW, Behringer has updated the firmware for the DEQ2496. Check your manual to see how to determine your unit's version. You can download and update your unit using the MIDI port. Lee

I've started a new topic re the Behringer DEQ 2496 on the 2-Channel Audio thread that might be of interest to the tweakers. The DEQ 2496 menu structure can be confusing. The biggest mistake you can make is to spend a half hour equalizing the left loudspeaker with the GEQ function and then forgetting to switch the EQ to the RIGHT channel. Then you wonder why the EQ knobs don't change anything. Once you get the "sound" you like, be sure to save your settings. Lee

Those familiar with the Behringer DEQ 2496 know it is an audio "Swiss Army Knife" with multiple functions. The built-in FeedBack Destroyer (FBD) function may not be the first feature an owner would use in his or her listening room. But it might be one of the most useful. I discovered the FBD's usefulness after deciding to construct some diaphragmatic bass absorbers to tame some of my room's Eigenmodes. Before buying the materials, I decided to try an experiment to accomplish the same thing electronically. A Bruel & Kjaer 4133 mike was placed at the listening position and the output was connected to my Ivie IE-30A RTA. The line level output of the Ivie fed an AUX input of my preamp via a "Y" phono adapter. Since I was only interested in notching out the room modes below 400 Hz or so, I disconnected the squawker and tweeter drivers in my Klipschorns. Starting with the preamp balance control fully counter clockwise, I slowly raised the volume until feedback occured. The Ivie and the DEQ2496 (set to RTA mode) showed a spike at 80 Hz (the floor to ceiling modal frequency). Increasing the volume further brought another mode to life. At this point, I decided to activate five of the ten FBD filters (there are ten per channel) and let the DEQ 2496 automatically do its "search and destroy" thing. In a matter of a few seconds it had found and notched out four offending room modes!! The great thing about the FBD filters is that they are 1/60 of an octave wide which translates to "inaudible" in their action. Of course I wasn't content to just sit down and listen to music. I decided to set the FBD filters for manual mode and spent the next half hour accomplishing about the same thing that had been done automatically in a few seconds!! This is a work in progress. The next step is to place the mike in several locations throughout the room and see if the results change. Since I'm educated beyond my intelligence, doesn't it follow that if the loudspeaker doesn't excite the room modes in the first place, the modes won't be an issue? Having an open mike and loudspeaker in the sme room can be an invitation to blown fuses, ruptured cones or loss of your lease! Be careful. Any thoughts or suggestions are welcome. Lee

Sorry 'bout that---I accidently hit the SEND button. Mike, et al, Here's the method I use for the initial EQ-ng of my system with the Behringer. Place the measurement mike about a meter in front of and centered on the mid/high drivers of one of the loudspeakers. Be sure to mute the other loudspeaker. Initially you are only interested in the 400 Hz to to 20 kHz spectrum display. Adjust the amplifier level until the 1 kHz band lies vertically on a Y-axis reference. This will give you a "0" reference for the subsequent EQ adjustments. Start with the vertical resolution at 5-10 dB per division. As you continue to make finer EQ adjustments, increase the vertical resolution. Now adjust the graphic EQ (GEQ) settings to tame the peaks. If the peaks are isolated, try using the parametric EQ (PEQ) function. You can adjust the filter action from 1/10-octave to a gentle shelving curve. Notice I haven't mentioned dips or notches in the response. Unless there is a gentle rolloff or dip in the response, don't try to fill that notch---the ear doesn't hear it anyway. A slight dip can be corrected using adjacent 1/3-octave filters or just set the PEQ to a 1/2 or 1/1-octave setting. Some of you have successfully used the Auto EQ function of the Behringer. If you are a newbie to equalization, give it a try. The suggestion to have it ignore frequencies below 200 Hz is a good one. Now, you can either spend several hours trying to achieve a "flat" response or spend a half hour or so getting both loudspeakers to have greatly improved imaging. Al K. has suggested setting both channels to the same EQ settings above 1 kHz or so. I agree. At a one meter measurement point, a loudspeaker shouldn't be influenced by the room from 1 kHz and higher. Besides, you'll save time by setting both EQ filters to the same values. Of course, you need to measure the other loudspeaker to make sure there aren't resonse problems caused by incorrect tap settings on the squawker or other crossover network problems. As far as what to look for below 400 Hz, keep in mind the room shape and size will pretty well determine your bass response. While the mike is at the one meter position, just tame the peaks to get the smoothest overall response with respect to the midrange and treble levels. Now, move the measurement mike to the listening position and notice that the response below 400 Hz may have changed dramatically. There may be several peaks and dips in the response that weren't present a meter away from the loudspeaker. The best bass EQ is to apply appropriate room treatments (membrane absorbers or other resonators) optiomized to trap the problematic room modes. There are several web sites offering room mode calculators--just plug in your room dimensions and the tangential and axial modal frequencies will be listed in tabular and/or graphic form. If you want to try EQ-ing the bass peaks electronically, use the PEQ mode. With a 1/10th-octave notch, you'd be surprised how well it "traps" the modal frequencies without affecting the bass instruments. Those familiar with EQ-ing home systems know that a system that measures "flat" at the listening position is too bright. If you try the above "flat at one meter" approach, and the room has adequate absorption, the response curve will be pretty close to ideal. Lee

Colin, et al--- I've used the DEQ-2496 for several months with my horns. Since it is an electronic version of a "Swiss Army Knife", I'd recommend downloading the owner's manual from the Behringer web site to understand all the neat things it has built in. I purchased the DEQ-2496 after converting the AA networks to AL's Extreme Slope networks and changing the K-400 horn and K-77 to the Altec 811B and Beyma CP-25 respectively. Changing the networks and horns made an audible improvement, to be sure. However, not one to leave well enough alone, I wanted to try smoothing the midrange with a little EQ. The Behringer unit offered the most bang for the buck by offering some powerful EQ modes (parametric, 1/6-octave, shelving) and the ability to store EQ settings. It has a hard-wire bypass so you can A-B the effect of the EQ setting. I'd strongly recommend getting the Behringer measuring mike and using the built-in pink noise generator, RTA and AUTO EQ feature. This will get you in the ball park. Then you can spend the next 10 hours tweaking!! BTW, the 60-band real time analyzer display is driven from the internal fast Fourier tranform circuitry. I've found any spectrum readings below 100 Hz or so to be too high based upon the input source. As someone pointed out earlier, the room is just as important as the loudspeaker. No amount of EQ is going to correct for undamped low frequency room modes or excessive reflections from hard surfaces. Lee

There used to be two things in life that got my dander up: Toilet paper loaded bassackwards and the term "RMS power". After adding a second TP holder for my lovely significant other, that left only the RMS power issue to deal with. There is no such quantity as "RMS power". You won't find it in any textbook or in the National Electrical Code. You will, however, find it in the sales brochures of most amplifier manufacturers. McIntosh Laboratories has used the correct power rating terminology of their amplifiers for many years. It is: "Continuous average sine wave power". While we should measure the voltage across the load with a true RMS voltmeter or the current through the load with a true RMS ammeter, that doesn't mean the resulting power quantity computed using Ohm's law is "RMS watts". Assuming the source is a sine wave, you are measuring the average power delivered by the amplifier. This is equivalent to the DC heating power delivered to a specified load resistor. I suspect the RMS power term came into use as a result of the FTC's misguided attempts to standardize power amplifier meaasurements in 1978. Back then, an honest 30 watt amplifier could be rated at some inflated value using the now-infamous Instantaneous Peak Power rating.

Mike, Chances are those frequency response graphs were generated with an X-Y plotter. PWK had a Hewlett-Packard plotter in the Hope lab in the '70s. The Y-axis was fed via his Logarator (sp?) which was an RMS-to-DC converter. If the integration time of the Logarator was set for "slow", any fast changes in the frequency response would be smoothed over. Also if the sweep time of the 20-20kHz X-axis was set to a fast sweep time and the pen speed was set to "slow", you'd get the same smoothing effect. I doubt PWK had any malice in mind publishing these smoothed curves--he probably wanted the potential customer to see an "average" of how the loudspeaker would perform in a home environment. Lee

Let's say your powered sub woofer is cranking out 100 dB SPL on the peaks and the ceiling construction is rated at a respectable STC rating of 60. This STC rating is measured at 500 Hz and the partition will offer a loss of 60 dB at 500Hz. However, at 63 Hz (well within your sub's frequency range), the transmission loss of your ceiling barrier can be as low as 45 dB and still meet the overall STC 60 rating. That means you will measure around 55 dB SPL in the upstairs bedroom. Probably not acceptable. Starting with an eight-foot ceiling height limits your choices for a noise barrier unless a finished ceiling height of seven feet is appealing to you. Decoupling your HT's ceiling and walls from the rest of the structure would seem to offer the best bang for the buck. For the ceiling, my advice would be to blow cellulose insulation into the overhead floor joists. Using fire-rated 5/8-inch gyp board, screw one layer to the joists and tape the joints. Caulk the perimeter with non-hardening acoustic caulk. Next, install resilient channels (RC) screwing thru the first gyp board layer and into the joists. Install the first layer of gyp board to the RC. Leave a gap no wider than an 1/8-inch or so around the perimeter. Tape all joints and caulk the perimeter. Install the second layer at right angles to the first (or overlap joints) following the same procedure except tape and float the joints. Follow the resilient channel manufacture's instruction for ceiling applications. If your walls and door(s) have a low STC rating, chances are bass energy will still manage to escape your HT room thru flanking. If your budget allows adding RC and a layer of gyp board, go for it.