Chris A

Sort of. Private schools only serve to administer compulsory education to those under the age of being an adult and making their own decisions. If this case had been about private schools, it might have been a different set of rules that applied ("might have" are the key words here). Ask yourself if a private school can set rules like public schools (in my experience, they often do this, and sometimes go far beyond what public schools require for continued attendance). It's a contract that the public schools don't get to make. While this might be a consideration for the case discussed above, it is only cursory to the decision made--which is "smooth functioning of the schools". If you think about it, the real problem is between the parents and their school-aged children. If the children were bought up to respect the institutions and the rationale for that respect were drilled into the offspring, my guess is that there would be far fewer issues. Using the public schools themselves as platforms for political dissent (as was the case in Tinker) to me signifies that the parents care less about the smooth functioning of public schools than they do "individual First Amendment rights" of their children. There's always a tradeoff, and "nothing's impossible for those that don't have to do it"...bystanders and perhaps parents vs. the teachers and school administration--and the students that want a good education. ___________________________________________________________________ One of the problems that I see in this line of discussion is that it is actually drifting from the SCOTUS case as highlighted above. I think that's not a good direction to take this conversation. Opinions on "public education in general" tend to become political on forums like this. If the intent is to get this thread locked and/or hidden, that's an effective way to do it. Myself, I like the ability to discuss these cases with people that understand the law--on this forum. Chris

The way I see your comment: If the arguments are wrapped up in political party platforms as "planks" of policy and that are inherently divisive in nature, it's political. If however it's on legalities of the court proceedings, and not really a central political plank or theme--then it's probably "not political"--as is emphasized in this particular thread. All of the above is based on legal arguments based on the First Amendment to the US Constitution--freedom of speech vs. abridgement of those rights for ease of smoother functioning of public school systems...who must also teach free speech to those same students, too. We live in a society based on rule of law (in the US). So it's of general interest--a subject which is in-bounds for the Lounge. If that is now a political issue, I was not aware of it, and I will apologize for tacking this discussion onto the bottom of this thread. I do use the voices and expertise of this forum to better gauge the general mood and biases of members--who represent a small cross section of the general public--to stay informed. I think this discussion is actually a lot more useful than talking about most of other subjects raised in the Lounge subforum... And the discussions here are civil and apparently informed (I hope). Travis is an attorney. So are a few others on this forum (which itself is an interesting subject: why do lawyers like hi-fi?). Making use of their expertise to gauge public opinion is a good pastime, at least for my needs. I can't get that on Facebook or in the comments sections of online newspaper articles. (I don't belong to Facebook or Snapchat--and never will.) Chris

To me, Justice Thomas seems to be trying to apply originalism like a straightjacket, a blunt instrument, if what you say is true just above. The focus of arguments in this case seems to be "how to handle First Amendment rights in a digital age". None of the justices seems to think that the ease of transmitting information digitally wasn't the issue--it clearly is. The test--classroom disruption (as per Tinker)--seems to be used "after the fact" to regulate digitally transmitted free speech, not a test that the students themselves can really use before speaking out to help guide digitally transmitted speech--even that speech which will "go away" after a short period of time. This might be a legal strengthening of the test's applicability (Tinker), but in my view, it isn't one that's useful to the students to know where that line really is--until afterwards, then they are rolling the dice to see if it is judged by school officials to create "significant disruption" or not. How about emails? Private BBs set up to limit access to just a few people (as it seems in this case)? Facebook and Snapchat are clearly not considered private conversation now, even if the audience is regulated by the student on an individual case by case basis (which was not singled out by the court), or that the messages would "time bomb" out of existence (the court appears to be considering this, but doesn't appear to use this in its decision). Even in this rather innocuous case (an outburst of emotions on a time-bombed message board that the school itself probably shouldn't be monitoring because it's so far away from school), students still appear to be "on a short leash" by school administrations as compared to the early 1970s when Tinker was first applied widely. If the girl had simply spoken (in person) during lunch time at a fast food restaurant, or even in a school cafeteria to a bunch of friends, she would have had much more latitude to speak her mind, and ostensibly without consequence--since spoken speech in real time isn't usually captured because there are laws that protect individuals from being recorded without consent (just like teachers). And spoken speech is further protected by the laws that require certification of just who is speaking on recordings--which is merely obfuscation of uniquely identifying the source of the speech--before punitive action can be taken by school officials. The net result is that the students still continue to lose more sensible free speech rights than the prior appeals court decision apparently supported--if that speech is digitally transmitted--anywhere. This question--what constitutes free speech in the digital age--continues to be asked, and useful answers seem to keep getting farther and farther away. No one side has won much here for all the effort expended in the string of cases leading up to this decision. In fact, the line actually seems more blurry now than before this latest decision. The message seems to be, "you can think what you like, but if you are efficient at transmitting what you think to others, you're not allowed to do that...". Remind me, what is the concept of "free speech" again? Chris

Thanks, Mark. That's what I was asking to see. Chris

Mark, could you describe your software/hardware DSP architecture in a diagram, including sources for FIR and IIR algorithms, and what kind of either development environment you use or how you load third-party software/firmware and supporting software/firmare modules? Chris

I'm not sure that you're going to find someone with experience with both AMT drivers, especially with winged/stacked AMT-1s. I can say that the winged/stacked AMT-1s have a dipole airiness about them that is due to their "two-sided horn" construction. If you are comparing to a Beyma TPL-150/H, you get a very nice sounding dipole effect out of the AMT-1s due to their controlled directivity in both front and rear directions relative to the Beyma AMT. The trough-type horn of the stock single-high AMT-1 is good down to ~1800 Hz, at which point it begins to lose horizontal directivity control. Therefore its on-axis SPL begins to degrade and its harmonic distortion begins to show up at higher SPL (i.e., above 95-100 dB at 1 m). Below 670 Hz, the stock AMT-1 loses both directivity control and the 1/4 wavelength starts to exceed the ribbon length, at which point the SPL takes a nosedive. If you provide the double stack AMT-1 and increase the length of the trough-type wings from the plastic case steel laminations, the low frequency cutoff depresses to about 570 Hz--or slightly lower. The effective sensitivity of the double-stack AMT-1s also increases to about 98-100 dB at 1m on-axis. This is low enough to use the AMT-1s on top of Belle and La Scala bass bins (i.e., horn loaded). So you've in effect got a fully horn-loaded loudspeaker without the attendant problems of a separate horn-loaded tweeter and K-55 midrange--including the time misalignment and non-flat acoustic phase and electrical impedance curves. If you bi-amp the AMT-1 with a Belle or La Scala bass bin and use a DSP crossover to provide EQ, you've got a killer deal. This configuration is what I use for my current surround loudspeakers: Belle bass bins with AMT-1s on top. They sound really good...good enough to be front loudspeakers in a conventional stereo setup. I'll stop there...Does any of this begin to answer your questions? Chris

Ruling of the case finds for the First Amendment rights of the student: https://www.washingtonpost.com/politics/courts_law/supreme-court-cheerleader-snapchat-free-speech/2021/06/23/09b905ba-d42a-11eb-a53a-3b5450fdca7a_story.html The outcome of the trial was very different than the Q&A session--a surprise, IMO. It seems that SCOTUS took the case to merely narrow the scope of Tinker applied by the Appeals court, but did not reverse the Appeals Court decision. Chris

See: https://www3.mbari.org/data/mbsystem/sonarfunction/SeaBeamMultibeamTheoryOperation.pdf, page 43 of 107 To be honest, I'm not really interested in making the horn smaller for this thread's discussion--quite the opposite. I would recommend the following thread for further discussions of smaller horn sizes, use of midrange drivers in MEHs, or across-the-center off-axis MEH ports: Chris

For reasons I can explain--these assumptions you state just above are not right--if the MEHs are placed within boundary gain in-room. No larger in cross sectional area than an equivalent ported subwoofer needs--but MEHs typically use two ports per woofer (except the SH-96, which uses one long port per woofer). The other thing is, MEH ports are very short--not like subwoofer ports, which typically require some length--like a tube. MEH woofer ports need to be as short as structurally possible. I currently own K-402s on top of KPT-KHJ-LF Jubilee bass bins, just in front of 14 Hz TH subwoofers in the room corners, crossed at 40 Hz. What I found is that the center K-402-MEH, EQed to a -3 dB point of 18 Hz in its elevated center position adds about 2x to the infrasonic bass experience, probably due to being able to "fill up" the anti-node midwall positions of the room with deep bass response. Having 5 K-402-MEHs, each with ~18 Hz (EQed) response would likely be even better, and is still what I'm aiming to do in my listening room. That point is about 18 Hz in my listening room, with the K-402-MEH in the elevated above floor center loudspeaker position. It does even better than my TH subwoofers in terms of freedom from harmonic distortion and phase shift. Having them in room corners would add significantly to the low bass response and probably deepen the -3 dB frequency. Chris

Not true in my listening room. 😎 But there are acoustics differences that loudspeakers having "full range" directivity in-room have over what the guy that you mention doesn't listen to. Three properties are needed to hear phase in-room: 1) loudspeakers having excellent directivity control down to ~100-200 Hz, 2) loudspeakers having essentially flat "excess phase" response, 3) room treatments (mostly absorption) just around the loudspeakers--within the first 3-6 feet (1-2 metres) that significantly reduce early reflections within the first 4-8 milliseconds from the direct arrivals from the loudspeakers, and no acoustic reflectors around the listening positions. ...then you can "hear phase" in my experience. More on that subject here: Chris

Well, usually the trigger for me is when the discussion turns to midrange drivers on MEHs. That's squarely in the other thread, I think. Otherwise, this thread is going to continue to grow past its already sizeable portions. I've found that there really is never a reason to use midranges in MEHs--just substitute a 1.4-->2" compression driver instead and avoid the midranges altogether. The SH-50 almost doesn't need a midrange--and if Tom D. had used a 1.4" compression driver instead, he wouldn't have needed the midrange drivers (except perhaps for extreme SPL output--which is what he was exclusively designing to). I plan to talk about a Cornwall-sized MEH (the entire front baffle and horns are replaced by a single MEH with probably 1.4-->2 inch compression driver, thus eliminating the need for midrange drivers. I'll probably put that into it's own new thread--when I can finally get enough "round tuits". Lately, life hasn't really allowed for any MEH design or building, but I hope that will change in the next month or so. If you read carefully on Tom D.'s comments over on that "other forum" (the one with "science" in the name), you'll see that he started with a SM-60 as the top portion of his new design (basically unchanged) as a home project, and that's the basis of his new "Hyperion". So I don't believe that he started from scratch with an idea to control directivity from (nominally) 100 Hz, all the way up to 18-20 kHz. That design started on "reuse". He acknowledges that it loses directivity control below ~500 Hz. I calculated it might lose directivity at ~400 Hz, but I guess that's in the "splitting hairs" category. That 400 Hz loss of directivity control is still at too high a frequency for my needs (and those that listen to them, I would bet). Here's the beam width EASE data (-6 dB) for the SM-60M: Here's the same horizontal beam width data for the SH-96 (-3, -6, and -9 dB curves): In this thread, the K-402-MEH design started on reuse of the K-402, and leveraged the idea of a larger MEH based on SH-96 proportions, but definitely without using 11 drivers--rather using 3 drivers (two 15" woofers and a 2" compression driver). The reasons should be clear by now why I took that course. However, if I were to start with a "clean sheet of paper" design, I'd probably try both a slightly smaller MEH--the size of the Cornwall (discussed above)...and one larger size that's a little larger than the K-402 in order to eliminate the need for extra subwoofers+woofers (i.e., horns can be designed to get about 3.3 octaves of passband, and for woofers, that could be sub-20 Hz to 200 Hz), and use the amazing capabilities of the Celestion Axi2050 full-range driver--at least down to 200 Hz to cover the rest. That MEH design would get me out of bed in the morning to design and build. But I get ahead of myself... Chris

(Please remember that the subject of this thread is the K-402-MEH--and not all MEHs or all design trades. There is another thread for those discussions.) I saw the effects of the woofer ports in the K-402-MEH, crossing between 400-500 Hz to the 2" compression driver: Of course, this was before I knew how change my approach to crossover filters between "ways" on an MEH, so the effect you see above is not what I'm actually seeing in the K-402-MEH nowadays. This is apparently a "worst case" condition--minimal frequency overlap between drivers via higher order filters. I now use much more frequency overlap and lower order (actually "zeroth order" without phase growth through the crossover region--then I add back 90 degrees of delay to the woofer channel)--because I can do this without the negative effects of overlapping crossover interference bands in the MEH horn aperture like you get with non-MEH horn designs (i.e., using multiple horn apertures--one per driver--in the traditional sense). But it is instructive to see that the polar coverage effects of off-axis port is scalable with frequency. Just because they're woofers doesn't mean that the effect is any less important--in fact, it may be more important at the 400-500 crossover region because this is where the ear is apparently most discerning of discontinuities in polar coverage---and one of the reasons why I don't understand why virtually no one pay attention to directivity below ~1 kHz--even Danley seems to be paying only lip service to "full-range" directivity control. In my listening and (and others at Klipsch in Hope, etc.) it makes a huge difference in sound quality (i.e., much better sound quality) to be able to control directivity below 500 Hz. I have to say that I'm not really that interested in investigating port placement across horn mid-wall areas. Mid-wall is where most of the acoustic wave energy transmission is occurring, while the corners of the horn are somewhat shielded from the effects of acoustic waves. Also, the fact that Danley only uses ports in the horn creases (over a period of 20+ years) tells me something, too. ________________________________________________ Personally, I would rather see a trade on the size of the ports and the acoustic efficiency/flatness of the lower frequency driver response. In my measurements of the SH-50 drivers (with and without crossover networks), I saw interesting behavior in terms of the SPL response (vs. frequency) that I compared to the K-402-MEH, and found that the K-402-MEH actually exhibits flatter SPL and phase response than the SH-50 drivers do. This is four and a half inches for the 15" woofers in the case of this thread's subject--not six. I don't really see the point you're trying to make. It feels like you're going in a direction that I wouldn't choose to go with full-range MEHs ("full range" as defined by myself in this thread). ________________________________________________ With the Celestion Axi2050 on a horn the size of a K-402, the crossover could easily be an octave lower than the ~500-550 Hz it is in the case of the prototype K-402-MEH (again to the point--the actual subject of this particular thread...not all MEHs in general). This means that the woofer ("off axis") ports can be placed 8-10 inches away from the horn throat--which not only pushes down the effects of polar coverage perturbations an additional octave into a much less critical listening band (in terms consistent polar coverage), but it also results in better loading of the woofers within the horn, per this type of effect that Tom D. pointed out in his white paper on Synergy™ horns and tapped horns where placing the woofer and midrange ports a bit further away loads them much more effectively due to the slower taper rate found in the straight-sided horn profile: This is a win-win design trade in my experience. One apparently wins in two different ways by crossing over an octave lower. The only tradeoffs I see are slightly increased FM distortion of the compression driver (which is still vanishingly low--apparently below audibility) and the reduced ultimate SPL from the loudspeaker due to the increased travel requirements of the diaphragm to reach down to 200 Hz. The plane wave test results of the Axi2050 leads me to believe that there is still unrealized low frequency performance that's available there (note the overall SPL in this plot from Celestion): In fact, it may the reason to explore horns with greater depth than the 17" of the K-402, i.e., scaling up the size of the K-402 horn very slightly to see the effects of better woofer loading and pushing down the effects of the crossover frequency to an even lower listening band, perhaps to ~200 Hz . JMTC. Chris

This isn't really as difficult as it appears--in my experience. If you look for 96/24 (internal bit rate and bit depth), you'll see that this will cut way down on the number of available units. In particular, those DSP units that were designed before ca. 2008-2010 will not pass this bar. In my experience, there are at least two good choices based on value-for-money: Xilica XP series and miniDSP HD series. Other companies make 96/24 units, but for various reasons, they are usually screened out of the decision process. And there are many 48 kHz units available, but are not advertised as such unless you dig into the specification sheets a bit. For instance, the Behringer DCX2496 (its name gives away the internal bit depth and rate per channel), the issue is fidelity--but not the digital side, but the analog sides (front end and back end). The dbx DriveRack PA2 doesn't make the 96 kHz cut, nor does the Ashly Protea series, as well as the EV Dx38 or DC-One. This also applies to the Dayton Audio DSP-408 (a.k.a., very inexpensive automotive-focused DSP units). The Yamaha SP2060 passes the 96 kHz bar, but is more noisy than the Xilica XP series. It can work (just like the miniDSP HD series), but the issue is usually cost/performance. The Yamaha SP2060 is usually expensive--even used. The Lake processors and even the DEQX (I suspect since they don't list the internal sampling rate) also do not come into solution due to either their internal sampling rates or their prices. So the question is: if you are focused on internal sampling rate, is this because you're worried about fidelity? In my experience, there is a very, very slight difference with the 96 kHz units that fall well into the "subjective" category. I believe that I hear that higher fidelity with the Xilica and to a similar degree, with the miniDSP 2x4 HD, although all the miniDSPs are slightly more noisy than the Xilica XP series in practice--usually due to the quality of the connectors (XLR or Euro/Phoenix balanced connections vs. RCA unbalanced). The miniDSP 2x4 HD uses only unbalanced RCA connections--the reason for its higher noise, I believe--but the miniDSP 4x10 HD can use both XLR or RCA, which is a way to avoid having to use in-line filters like the Jensen ISO-MAX series, which are fairly expensive and will usually push the overall price up to Xilica levels if they are needed to control noise. So if you're dealing with loudspeaker sensitivities and overall fidelity at the Klipsch Jubilee (i.e., K-402 and a good 2" compression driver) level, I recommend the Xilica. Anything else with slightly lower sensitivity and fidelity, the miniDSP HD series will typically be able to be integrated, albeit at a slightly higher noise level. Chris

If you can find the larger SEOS 30 or (much better) K-402 horns--after dialing in the EQ to compensate for the fact that these are controlled directivity horns--I think you'll be so impressed that you'll never be able to go back to anything smaller. The Axi2050 on a K-402 easily crosses at 225 Hz without boosting EQ (except above 15 kHz), and pretty much matches my TAD TD-4002s that I've listened to for 12 years now (on K-402 horns). We have a fairly large contingent of folks here that have heard or own K-402s, and have mostly imprinted on them as their acoustic reference baseline. You should hear them. They pretty much blow away the smaller K-510s that pioneered the originating horn profile. The Axi2050 on a K-402 horn with a newly designed folded corner horn bass bin is the basis of the new and upcoming Klipsch Jubilee, the flagship model for KGI: Chris

Welcome!

Some words from Mr. Danley in another forum: This is the conformation that I was expecting to report on several years ago when I designed the K-402-MEH that you see in this thread. All the Danley Unity™ and Synergy™ horns have the port locations in the corners of the horn, and as I had suspected--there was a reason for this placement--consistently seen in their designs. Chris

Unfortunately, the comments that the reviewer made (in another "talking head" review) aren't terribly comprehensive, and too subjective. It's like he wanted to find faults--but couldn't, and was surprised. Perhaps he's used to hearing not-too-good direct radiating loudspeakers that "measure well" (well, except for modulation and compression distortion and directivity control response) but perform rather poorly as compared to well implemented fully horn loaded loudspeakers like MEHs. One of the things I've noticed is that he listens only to pop/rock (and its subgenre) recordings: https://open.spotify.com/playlist/0gBVe7rwdZojX41LakrUUy Where I really hear the difference in loudspeakers is not using these types of recordings. YMMV. Some comments by Mr. Danley around the loudspeaker and other designs/experiences on another forum--starting here: https://www.audiosciencereview.com/forum/index.php?threads/upcoming-tom-danley-hifi-speakers.19482/page-5#post-810432

Free air resonance is not the same as horn-loaded F3 (the frequency at which the output of the bass bin is down 3 dB from "nominal"). While higher free air resonance will typically raise the horn-loaded F3 point slightly, it's not nearly as pronounced as your statement. In fact, the real limitation of both the Belle and La Scala Bass bins on their F3 is the designed-in "cutoff frequency" of their exponential horns--and not really the woofer's free air resonance. See the following figure that shows the horn normalized throat resistance (i.e., horn loading) vs. frequency for several horn profiles, showing the effect of the exponential horn volume expansion profile, with its definite "cutoff" frequency due the horn profile itself: Chris

Roy has already stated that the YouTube video is in error due to showing the Jubilees outside of room corners--which are still required. (He said the marketing guys took a few liberties in the video without asking him first.) If you were thinking about buying a pair and pulling them away from the room corners--think again. Adding the horn lens change is just a small concern as compared to that error. Chris

That's not a phase plug--it's actually an acoustic horn lens--mounted at the horn throat entrance. Phase plugs are important to prevent high frequency cancellations, while horn lenses are used to spread out the polars via "refraction". The only issue is that you also get unavoidable diffraction from horn lenses--just like the optical varieties do, too. It's a toss up whether the diffraction overcomes the refraction capabilities of the horn lens. Chris

See https://www.grainger.com/product/GRAINGER-APPROVED-Ribbed-Push-In-Rivet-5RKV7 The K-402 mounting hole diameter is 0.3 inches (7.62 mm) I used the 0.25 inch version--and that works, too. No. The K-402 horn itself is good down to 172 Hz before the axial length begins to approach 1/4 wavelength. (The horn mouth itself is good down to ~40 Hz for directivity control.) At that 172 Hz point, the wavelengths are 78 inches long, and the small mouth flange itself isn't significant. The Celestion Axi2050 can be used down to ~225 Hz. Roy's preliminary data sheet shows that he's crossing at 340 Hz, so you're in good shape without any tapers or "baffle step" considerations. See Chris

My comment...as per your request: That rise in harmonic distortion at 200-300 Hz looks like an issue. I don't know what that is. It could be a measurement issue. If it isn't a measurement issue, then it would be a pretty interesting woofer/box issue that would likely be audible at the 95 dB (on-axis--and I assume the microphone is at one metre from the loudspeaker). The data sheet on the R-41M (https://d2um2qdswy1tb0.cloudfront.net/product-specsheets/R-41M_Spec-Sheet_v01.pdf) mentions that the crossover frequency is 1730 Hz, so that harmonic distortion (HD) is solely in the domain of the woofer and its box. I would say that the R-41Ms are really suited for lower level listening as compared to other/larger Klipsch loudspeakers, based on the above measurement. Having harmonic distortion levels that are only 30 dB down (at 95 dB on axis at 1m) indicates to me that there are also pretty audible modulation distortion issues that would be audible at the 200-300 Hz (about an octave below A440 tuning fork frequencies) and above--up to the crossover frequency of 1730 Hz. Chris

You can do it that way--no issues. Like I said above, I usually take sweeps at 85-90 dB, but that's entirely up to you. If you've got noisy appliances (or barking dogs), etc., in the room, taking measurements at a higher level will significantly decrease the effects of those type of incident noises. You can reset the "target level" (TL) , up or down, to avoid the issues that ensue from setting the level too high or too low. If REW complains, sometimes it sees a lot of boosting or attenuation about to occur, and it's making sure that you want to do that. If so, just click "ok" to proceed. If you select a frequency range that's too small, you'll end up with portions of the loudspeakers' response that's still not flat. Usually, I find that setting the frequency limits of EQ is fairly intuitive, so I usually just iterate the frequency limits until I achieve what I want. Sometimes, I even add PEQs by hand (manually) to get what I want, especially on the ends of the frequency response spectrum, where REW's EQ facility won't boost the response when I desire that it will boost response. Generally speaking, one of the first things I do is to set the vertical and horizontal scales on the plots so that I can see what I'm doing. The plots you show above have the vertical scale magnitude way too zoomed out. I can't see anything from those plots you posted, to be honest. I try to use a vertical scale so that I can see at least 2 dB increments in the vertical scale bar. You set the vertical scale by moving the cursor into the plot, and then a little set of "+" and "-" icons show up near the top of the vertical scale in the plot area. You click on the "+" icon to zoom in until the vertical scale shows 1 or 2 dB minor increments. Chris

Yes. The microphone doesn't move. The dialing-in process starts with flattening the SPL response of the individual drivers/horns, then progresses to time alignment of the drivers using full-range sweeps of all the drivers together...with the crossover regions between drivers also set to their desired or natural crossover frequencies (determined via REW measurements). Finally, the SPL response across the crossover regions (as well as setting the driver polarities) is the last portion of the dialing-in of each loudspeaker. Once this is done for one loudspeaker, the time delay and crossover filter settings (if any) can be transferred to the other stereo loudspeaker of the exact same driver/horn configuration--to save time. Then the other loudspeaker must be checked via REW measurements with the microphone moved to be 1 m in front of the loudspeaker under test, then the EQ, delays, and individual driver channel gains adjusted as necessary to complete the individual loudspeaker dialing in process. In a multichannel array (5.1, etc.) the settings can be cloned for the other channels if the loudspeakers are of the same design/model (an ideal case that very rarely, if ever, occurs). In practice, since the center and surround loudspeakers are usually of a different configuration, they must be individually dialed in. After all the loudspeakers in the surround array are dialed-in individually, the 5.1/7.1/etc. channel gains and delays are set using something like a preamp/processor...or the DSP crossovers themselves. Then the microphone moves to the listening position(s) to adjust ONLY the L, C, R, SR, SL, sub, (etc.) channel gains and overall loudspeaker delays in order to achieve a time aligned and SPL balanced array of loudspeakers. So this is a continuous process that first uses the microphone in one position per loudspeaker, then if the loudspeaker array is multichannel, the microphone is moved to the listening position to set only the surround loudspeaker channel gains and delays (i.e., 5.1 delays)--NOT to reset the EQ of individual drivers. Chris

That's the input channel for the microphone--left or right channel from the viewpoint of the computer. Yes--output channel PEQs. The only times that I use the input channel PEQs is when: 1) I need to correct the SPL across the crossover interference bands between two drivers/ways of the loudspeaker. 2) when I begin to run out of output channel PEQs, and need additional PEQs to do fine tuning of the SPL response. Note that the input channel PEQs cannot correct single drivers/channels, but rather the summed (total) output of the loudspeaker --like the old preamp tone controls. Never--unless you're trying to adjust subwoofer response at the listening positions (i.e., well into the "modal region" of the frequency response where the longest dimension of the room is shorter than a half wavelength of sound, e.g., anything below ~30-50 Hz for regular-sized home hi-fi listening rooms). The problem with trying to take measurements at the listening position is separating out the effects of in-room reflections (which cannot be EQed--i.e., non-minimum phase acoustic behavior)--which apparently cannot actually be done correctly. Even the more expensive "room correction software" applications such as Dirac, etc. cannot seem figure out what is direct arrival acoustic energy from the loudspeakers (i.e., "minimum phase") vs. in-room reflections (i.e., non-minimum phase), and they all try to boost response cancellations (nulls) when trying to take measurements from the listening position. Chris