Chris A

Generally, the comments below can be found in the existing REW tutorial in pdf format. Here are my specific comments based on your questions and process steps, above: _________________________________________ "1.3)...This measurement should be taken with the SPL meter's 'Z' weight??" (Yes.) You can take a sweep at higher SPL than 75 dB. The 75 dB number is actually a minimum level, and in my experience, it needs to be taken at a higher level. I actually recommend taking initial sweeps at 90 dB @1m on-axis. The 75 dB number comes from those people using direct radiating loudspeakers that experience significant amounts of compression and modulation distortion. Sometimes you won't see problems in the measurement or amplification circuits until they are turned up a bit to let them "breathe" a little. _________________________________________ "1.4)...Each sweep should be 20Hz - 20000Hz? (In general, yes.) 512k length. No timing reference needed?" (Yes, not at this point.) Should mic channel be the same as the speaker?? I don't know what you're asking here. _________________________________________ "2.1)...Choose the correct EQ, or generic, from the first menu." I believe you are saying that you should select the correct DSP crossover model from the first drop-down menu within the REW EQ facility. If using a Xilica, select the "XP2040" button, if using a miniDSP 2x4 HD, select the miniDSP 2x4 HD button, etc. _________________________________________ "2.2) Set speaker type to 'Full Range' for each driver in 'Target Settings'. Leave all other settings alone?" The default values for the EQ facility need to be set within the "Preferences:Equaliser" menu from the main menu. "Full Range" should be chosen, the Crossover HP and LP type should be set to "None", and the room curve set to a starting value of 200 Hz and a slope of 0.4 dB/octave...or zero if you're not using room curves, which is a good starting point. The "target value" setting is extremely important--it sets the default overall SPL level that REW will try to EQ to tell the application where to stop EQing. ******[Before starting REW EQ optimization, you need to set the smoothing of the measured SPL response to something like "Psychoacoustic" or "1/6th octave" smoothing in the main window, under the "Graph" menu. I'd set Psychoacoustic or 1/6th octave smoothing as the default value in the Preferences menu. Otherwise, REW will try to correct all the little peaks and dips of the measured SPL response, to no avail. You really have to smooth the curve first before trying to find the PEQ filters to flatten the response.]******* The "Target Level (dB):" control moves the"room curve up or down to lie on top of the measured SPL response. It's best to set this to match within 0.5 dB of a "happy medium" of SPL values. By "happy medium", I mean that the target curve is matching or somewhat lower than most of the peaks in response is applying attenuating filters to the response--instead of boosting filters (more on the "why" of this later). Otherwise, you're going to be using your DSP crossover PEQ filters to boost overall response, and that will generally run you out of overall PEQ boost range before the job is done--which is something to avoid. In general, set this value to maximize the level of overlap of the room curve to the measured SPL response of the driver or drivers taken beforehand. You'll get a feel for this after using it a little. If REW generates too many PEQs to correct the SPL response, you run out of PEQs before the job is done. The ear can't hear those small perturbations of the raw measurement anyway. _________________________________________ "2.3) I have no idea what to do in 'Filter Tasks' ... so I use 20Hz -20000Hz in 'Match Range' and leave the other settings alone." (Here is a screen shot of the "Filter Tasks" dialog: The "Match Range" settings determine the highest and lowest frequencies that REW will try to correct to the target curve level (the "room curve", above). If flattening the response of a woofer, then I'd set these two values to something like 10 Hz and 500 Hz--the upper limit being a little above the low pass crossover point of the woofer to the midrange/high frequency drivers. For midrange, I'd set the two values to be just lower than the high pass frequency and just above the low pass frequency of the midrange crossover points, something like 300 Hz and 8000 Hz, etc. For tweeter, set the first value to the high pass value (usually around 1-8 kHz), and leave the upper value at 20 kHz. The default values of "Individual Max Boost" and "Overall Max Boost" are good--you can leave them where they're at. The next two check boxes are also good to leave at their default values ("Allow narrow filters..." and "Vary max Q..."). The "Match Response to Target" is the trigger to start REW to optimizing PEQs. Once you punch this link, REW will optimize the PEQs for the measurement selected, based on the target values and guidelines you set above. If REW tries to create too many PEQs, you should select the "Reset filters for current measurement" to reset all the calculated PEQs, then usually increment the "flatness target" link upwards by one dB, then try again. If you've forgotten to smooth the measurements to 1/6th octave or Psychoacoustic smoothing, you'll find out at this point that you forgot to do that. Reset, smooth, and try incrementing the flatness target upwards to reduce the number of PEQ filters generated to a reasonable number (usually between 1-->6 filters total for each driver/amplifier channel). _________________________________________ "2.4) Load the files that were created in step 3 into active crossover. Mine happens to be a miniDSP 4x10HD..." When you've achieved a reasonable number of PEQs generated, and the optimized response simulated by REW shows a reasonable flat resulting response, then you move on to exporting these into your DSP crossover--which is a semi-automatic operation for miniDSP crossovers, and manual for other brands of DSP crossovers--worry not, it only takes a few moments to transfer the REW-optimized PEQs into your crossover's application to program it for the channel/measurement you're working on. _________________________________________ "3.2) The mic should still be placed per step 1.2 at this point??" Yes. Don't move the microphone between successive iterations, unless you wish to multiply the work you have to do. In general, once you find a good place to locate the microphone, put a little piece of tape down on the floor on centerline with the center pole of the microphone stand to quickly be able to locate the correct position of the microphone--in case you inadvertently move the microphone between process steps. _________________________________________ "4.1) This is all I have so far..." This is where you combine the output of all drivers/horns together to fine tune the individual channel gains for flat response, then you can begin to time align the drivers. I don't use loop-back within REW to set the channel delays. Instead I use all drivers playing actively during the measurement sweep, then adjust the output based on the resulting spectrograms and the group delay (actually "excess group delay") plot. More on that subject can be found here: The subject of setting the crossover filters to achieve zero phase growth through the crossover interference bands can be found here: https://community.klipsch.com/index.php?/topic/182419-subconscious-auditory-effects-of-quasi-linear-phase-loudspeakers/page/4/&tab=comments#comment-2388972 Chris

Welcome!

user @etc6849 reports that it does: Chris

The "media console" is the problem...in my experience.

No...you don't. Turning off the center loudspeaker in my setup results in at least a 50% reduction in speech intelligibility (subjective assessment). Personally, I don't recommend the Academy unless it's a choice of last resort. Bass response and horizontal coverage issues (sitting anywhere off the exact center sweet spot) for the "horizontal centers" is always an issue. Most that acquire them (the Academy) seem to end up selling them, or not using them. Perhaps you should ask those that have owned them on this forum why they don't use them any more. Subs do help a great deal--the bigger the diaphragm area/horn mouth area, the better, especially into a room volume that you have. Small, direct radiating subs can work in smaller rooms, but in a room the size you have, you'll probably end up killing them with too much power into in order to faithfully keep up with the Cornwalls. Chris

@John Warren, As the OP to this thread, I'm asking you to take this information off to another thread--probably in the Technical/Modifications section, because it appears to be diliberately diluting the topic here--which you've already shown that you really don't wish to talk about. I'm familiar with the topic you mention, above, but your topic really doesn't belong in this section of solid state amplifiers, I believe (and the moderators can certainly overrule if it isn't a correct assessment). This thread has a history to it. I don't wish to add to those issues any further. Apologies for the directness of my request in advance if this causes any issues. Chris

I guess it is worthwhile to comment on what I saw in this video that perhaps others that are not as sensitive to what PWK was saying might understand. What appears to be an "old man" trying to answer the questions from the interviewer (who clearly wasn't Jim Hunter--unless he had his voice fixed an octave higher or so... 👧) was actually diplomacy...in my view. The question was three times asked, "what makes the Klipsch loudspeaker sound so much better--in layman's terms?". Three times he answered: "the absence of distortion." I think he became a little irritated the third time this question was asked, and he had answered it twice before (the reason for his pause in answering the third time), so he stopped to think about his answer so that he wouldn't say something that he would later regret. (At least, that's what I see in this recording.) If I get to that age, I'm pretty sure that I'll probably not be nearly as diplomatic if the same sort of event occurred. I believe the answer was startling--and not at all what the interviewer was expecting to hear (the reason why she asked two more times--because she was convinced that he didn't answer the question). So the Klipschorn was developed to avoid the kinds of distortion that were present in almost all other loudspeakers of the day (and since that time). This is startling and surprising, I believe, and the interviewer wasn't expecting the answer. "The absence of distortion" answer that was not accepted initially (or the second time around) implies many put up with so much distortion from loudspeakers...that they've grown accustomed or acclimatized to that distortion over the years. They no longer think of it as distortion because that's what they've listened to during their lives. I believe that most people have become much more accustomed to hearing that distortion than the "real thing"--i.e., natural or acoustic recordings of acoustic instruments and voices. I think the reason is due to not listening to the "real thing--unamplified". PWK always said that his customers should go to real concerts to re-calibrate their ears. He wasn't talking about going to a concert where the musicians were using amplified instruments or voices--rather just the opposite. Here's the dilemma: if all you've heard is distorted music from loudspeakers--how do you prefer non-distorted music? It's also of interest that many professional musicians (the type of musicians that play non-amplified instruments, that is) prefer the sound of the Klipschorn over other loudspeakers. This has been noted over the years, but the "talking heads" that review for magazines and produce the type of loudspeakers that create the types audible distortion that PWK was talking about--don't want to talk about the type of distortion that PWK was talking about--because their favorite loudspeakers would be revealed to have truly massive amounts of these types of distortion (as compared to well designed horn loaded loudspeakers). What kind of distortion was PWK talking about? Well, most people automatically assume that it's harmonic distortion, but the really objectionable type is modulation distortion, because it isn't harmonic in nature. Well-designed horn loudspeakers basically do not exhibit modulation distortion, and all direct-radiating loudspeakers do--audible modulation distortion, that is. This is because the horn loading of the diaphragms increases the impedance match to air enough that the efficiency of the driver/horn only has to move about 1/5th the distance for the same on-axis SPL as using the same driver in direct radiating mode. This is the real difference between direct radiating loudspeakers and well-designed horn loaded ones: modulation distortion. This is the type of distortion that PWK was mainly talking about. Compression distortion is also a factor, but typically not to the same degree as modulation distortion, unless the direct radiating loudspeaker is quite small in its lower frequency driver diaphragm areas. _______________________________________ As far as the center loudspeaker question that the interviewer asked, I think that PWK saw that the conversation was probably going to be too difficult to productively discuss if the preceding answer was so difficult for the interviewer to accept. I think he punted when the more complex question of center loudspeakers came up. I would have punted, too, I think. Chris

That was an enlightening experience. The difference between the K--402 and the K-510 horns was pronounced during a visit in 2009 to the chamber. I also got to hear the degree of sound diffraction in and around the rear and sides from the cabinet tops and the HF horn stand itself, and how intense those reflections actually were, and how complex their polar patterns were while walking around the assembly. Large horn mouths can control their polars to a much lower frequency and hearing this both in the chamber and outside the chamber told me a lot. Roy also had to spend a lot more time with his "salt and pepper EQ" to get the K-510s to sound about right in the listening room. It was a snap (relatively speaking) for the K-402. That told me a lot more. We learned a great deal in the two days we were there, and the difference between the in-chamber and in-room (the listening room just next to the chamber) was something that I probably won't ever forget. It took a while to assimilate all that Roy demonstrated over those two days, but I believe it was the real beginning of my audio education. Thanks again, Roy. Priceless wisdom was shared in those sessions. PWK's investment in the early 80s was a good one (and necessary, I believe). Chris

For reference (and before questions might arise), here is a fairly well-known diagram of the various acoustic straight horn profiles used in loudspeakers (i.e., not folded or curved): The tractrix and the exponential horn profiles actually share the exact same geometries at their throats, but differ as the horn expands toward its mouth, with the tractrix always flaring out to 90 degrees from its central axis. Indeed, the tractrix formula starts at its mouth instead of its throat, and always starts with the mouth perpendicular to the central horn axis. Typically, tractrix contours are actually about 85-90% as long as exponential contours for the same throat size in practice. Chris

No, it's a modified exponential horn (just like the Khorn bass bin) with a "rubber throat", meaning the expansion is greater than exponential near the throat, but it continues to a taper rate of 70 Hz (from PWK's 1965 JAES preprint #372, pg. 2). The folded horn construction is made up of a series of expanding straight-sided segments which approximate an exponential expansion. If the bass horn were merely "straight-sided" (a.k.a., "conical" expansion), the SPL response would look like the following impedance plot for a conical horn: Exponential horn expansion SPL looks like the plot that you showed, above. Here is a Hornresp area expansion profile put together by Greg B (Edgar) on the La Scala bass bin, showing its modified exponential expansion profile: A conical or straight-sided expansion would appear as a straight line on this type of plot. For those not familiar with basic shape of an exponential horn profile (cross section), here is a little snippet from the exponential impedance plot (above) showing that basic shape: You can see the difference in the throat (small end) areas of the exponential profile vs. the actual La Scala profile with its "rubber throat" expansion. Chris

I use an F3. As far as I'm concerned, any progress toward using current source amplifiers is welcome in this thread. I may not be contributing strongly due to the very apparent "growing pains" of putting out this subject for public consumption (and the battle scars that apparently go with that), but anyone that is actually trying something on this subject has highest priority to post here--in order to make this thread more useful. Chris

No, it's really related to the dimensions of the horn itself. Here's a figure from Beranek's Acoustics that illustrates the effect of a smaller horn with a finite-sized (too small) mouth that doesn't fully expand the emerging sound waves (the solid black line corresponds to SPL output from the horn mouths, and the Belle and La Scala bass bins are both exponential expansion horns): More discussion on this effect here: https://community.klipsch.com/index.php?/topic/161404-a-k-402-based-full-range-multiple-entry-horn/page/30/&tab=comments#comment-2263347 This is why horn-loaded loudspeakers benefit from in-room measurement and EQ compensation. Chris

Klipsch discontinued the Belle in favor of the La Scala reportedly in the mid-1990s when the company was going through changes in their lineup. The Belle was apparently only a special order loudspeaker costing much more from the factory than a La Scala, and was meant to be a style upgrade to a La Scala as a center channel between two Klipschorns (as a special order loudspeaker). As such, there isn't a great deal of difference between the two except form factor and cost. They apparently used the same drivers (at least when the Belle was introduced in the 1970s). Honestly, the Belle, being a flatter and wider loudspeaker seems to be more easily accommodated in many listening rooms. I've found that the depth of the Belle being about 19 inches deep instead of about 24 inches deep of the La Scala (widths are ~30 inches vs. ~24 inches, respectively) is significantly easier to handle in my listening room. So from a listening perspective, what's the difference? If you do something to EQ the SPL response to be much flatter around the 70-300 Hz band, each loudspeaker sounds pretty much the same (they both had to timbre match vis-à-vis Klipschorns on either side). So it's really your choice assuming that you use something in either case to EQ their SPL response to be flatter after placement in-room, in both cases due to their undersized horn mouths from an acoustics theory standpoint. The original La Scala doesn't require bass bin stiffeners, but their performance is slightly improved using in-horn bass bin stiffeners (DIY) to further stiffen the side walls of the bass bin. (The La Scala II uses thicker side walls instead.) The Belle doesn't need any bass bin stiffeners, and its shorter midrange horn and the space between the two bass bin mouths is not really an issue if crossing the midrange to the bass bin at the factory setting (~500 Hz). You will likely pay more for a pair of good condition Belles than La Scala I's. Chris

By the way, one saving grace that I see from the bass reflex in a horn (adding bass reflex ports well inside the horn mouth aperture--like in Roy's now-expired patent) is that the loading of the woofer diaphragm by the horn itself will serve the same function as a high pass filter in a direct-radiating bass reflex box/woofer to protect the woofer without effectively increasing the low frequency cutoff that a high pass filter induces: the increased impedance of the horn loading can be made to support the woofer diaphragm from greater than Xmax excursions that would unload or otherwise damage the woofer because of the lack of acoustic suspension loading of the woofer diaphragm. However, Danley "strongly recommends" a 40 Hz Butterworth 24 db/octave high pass in their bi-amping SH-96HO, SH46, and SM80M owners manuals. Clearly, the degree of horn loading of the woofer diaphragms and the expected output SPL of the loudspeakers in service (PA is usually extremely high SPL relative to home hi-fi) play a role in this advice. Chris

You can argue with the author...(i.e., not me): My view is to try it to see if it works like he says. I think that the application is perfect (TAD TD-4002 on a K-402 horn). Chris

You can only do that for one point in space (in your listening room, etc.) by measuring and correcting SPL response in-room. Everywhere else in the room, it's not flat SPL and/or phase response. When you design for anechoically flat power response over a certain sector of a horn's output, you've done a whole lot more than what you can do using DSP to flatten DSP at one place in the room. In fact, you can't correct for power response issues (a.k.a., polar coverage) with DSP. Additionally, the reason for using transconductance amplifiers (as least in my view) is not flatten power response in-room, but rather reduce distortion due to room/horn/driver effects on back-EMF from the drivers. Transconductance amplifiers ignore the back-EMF noise from the drivers in order to produce a cleaner output that more closely matches the input signal to the preamp/amplifiers. Chris

As I might have indicated earlier, really nothing has been released on the "new Jubilee" ("new Jub"), but it was revealed that Roy's now expired patent is one of the contributing technologies to the new Jub bass bins (complete with "airflow animations" of the bass bin with a single 12" woofer and bass reflex ports embedded within the horn itself). So anything we talk about here is mostly conjecture, and I've learned over time that Roy likes to do "drive by's" on those sort of comments and speculation. I really don't blame him for picking at the speculation before the product specifications have been released. However, in the past, Klipsch doesn't release a lot of detailed information on most of its loudspeaker models--much like their competitors. Fortunately, Roy has, over the years, shared some aspects of the current Jubilee performance (the two-way version using the KPT-KHJ-LF bass bin--apparently now discontinued according to Roy himself). So I really can't comment on the "new Jub", and if pressed, I would have a difficult time recreating the exact requirements that Roy, et al. used during the creation of the K-402 horn. The KPT-KHJ-LF bass bin has a JAES article on that effort that outlines the requirements quite clearly. Unfortunately, that's the part of the Jubilee that Roy has chosen to change in terms of its design details. I assume that the design requirements have also changed (but I expect he'll let us all know if that's true or not). I suppose everyone can say that they have their "druthers" in terms of loudspeaker design. My preferences include the ability to minimize phase and group delay growth, thus improving the impulse (a.k.a., transient) performance to as low levels as is possible--along with all the other things that the Jubilee does well. I've also found that putting the woofers on the same aperture as the HF driver on the K-402 Horn in the MEH design that I've documented (K-402-MEH thread) actually improves performance in some areas, and trading away some areas of performance that I personally value less (but as it turns out--this isn't very much--the trade is a very good one--in my estimation). I try to highlight the requirements that I design to, and their rationale. Oftentimes, this information is largely omitted from other DIY efforts, and particularly manufacturer's information about their loudspeaker designs. Roy has done a good job in communicating his design requirements (in general) for the current Jubilee over time--perhaps not all requirements, but many of the major ones. So, over time, I also try to reveal my requirements for the K-402-MEH and all other related MEHs that I design. In general, here are the requirements categories that I designed to (with discussion of some requirements thresholds in some areas): I think it's worthwhile to point out that there are actually many requirements, and each has its own threshold value related to the exact loudspeaker design that's chosen. So it's difficult to relate all requirements without handing you a full specification (which I could do if I spent many months developing--which I'm not really interested in doing currently). However, if someone did start to document full specifications (like the military requires for its developing major products) virtually everyone would benefit. Chris

I'm generally aimed at just eliminating the non-minimum-phase portions of the loudspeaker response (i.e., all pass), but note that bass reflex might be considered by others as "minimum phase". I don't accept that...kind of phase growth. Chris

I agree and actually see them as more than just "interrelated", but actually different aspects of the same phenomenon, so I'm also just as concerned (actually more concerned) with transient response. That's where the phase growth is actually heard--during transients (per Griesinger's article on the subject where the phase shift results in a time domain distortion of the peaks of transient response, [see slide 17 for the punch line] which is what the ear is actually listening for). Chris

This is fair, but note that full front-loaded horn subwoofers typically are typically more efficient than direct radiating subwoofers, and are much less affected by compression distortion (including voice coil heating) than direct radiators. I often have a trouble keeping separate those that prefer more amplifier/required driving power from those that seem to approach from the other direction to the argument, especially when we are not talking about "subwoofers" (which I wasn't talking about--at least those subwoofers not on the same aperture of the MEH, not using a separate box for the subwoofer). Those that prefer more power are typically not nearly as verbose as those that argue for very low power amplifiers everywhere with horn-loaded loudspeakers/subwoofers (like tubes...especially SETs). For woofers reproducing bass (i.e., above subwoofer frequencies, perhaps defined as above 30-40 Hz), power isn't an issue unless using something like very low power tube amplifiers (and that particularly includes SETs, which typically don't use global feedback because of their lack of forward loop gain). So I'm really talking about full-range MEHs that include bass reflex ports on their apertures (i.e., again, we're back to the definition of "full range MEHs" vis-à-vis just "MEHs" which you apparent assume to mean a separate bass bin. That's really not the subject of this thread. I set this one up to talk about bass reflex ports on the MEH horn aperture. This is actually not true, either the last thread (K-402-MEH) or in this one. I was actually referring to woofers on MEHs, and why I don't support the idea of bass reflex ports on the MEH aperture for their woofers, for the reasons I stated. ______________________________________________________________ If you want to argue bass reflex vs. sealed (not including MEHs), then that's probably another thread that can be opened up to the larger K-forum community that doesn't deal with MEHs--because that topic will probably eventually degrade into a moderator-locked thread (...if I read my tea leaves correctly...). I restricted the applicability of bass reflex here to MEHs (on the same aperture) to keep that from occurring. It's very similar situation to arguing "objectivity vs. subjectivity in audio" in my experience, and some people apparently just want to argue instead of just understanding one another and agreeing to disagree. I prefer not to be the OP on that thread, by the way. Someone else can open up that thread, if you don't mind. ______________________________________________________________ Just so you know, the above pictures of separate HF horn from bass bins/subwoofers really isn't the subject of this thread (for clarification). _______________________________________________________________________ I actually consider it to be discourteous to say what you did here (quoted just above), but I guess it's the norm on other audio forums to say this. It says that you don't care what anyone else (or everyone else) is saying, and I don't believe that's really the case in this instance. Nowadays, I really do try to keep the threads that I create (i.e., as "OP") to be shorter such that these kind of statements really can't be justified. That's why I tend to move certain subjects off of some threads (as the "OP") when it appears it's going to devolve into a long discussion that is really departing from the central theme of the thread. Chris

You can use what you want for amplification. Generally, it's the high frequency channel that requires a higher quality low-noise amplifier. I do not use the same amplifier model for the bass channel (a Crown D-75A)...as the high frequency channel (First Watt F3) with the TAD TD-4002 drivers that I use. This is the configuration that I've used for well over a decade. It works extremely well. The only issue that I see is when trying to use amplifiers having high output impedance (like SETs: single ended triode tube-type amplifiers). These actually change the SPL response of the loudspeakers, so in general, they will need in-room acoustic measurement to compensate for the changes in SPL response using the Xilica due to the higher amplifier output impedance alone. In all cases, you'll likely need to measure the SPL response in-room using a calibrated microphone and a computer-based acoustic measurement app (like Room EQ Wizard or any of a number of bought apps). Using the Xilica to dial-in at least the SPL response of the Jubilees is I believe required with Jubilees to get their performance up to their best. Having someone come in to do that dial-in that knows how to do it is a good plan. Note that the acoustic drivers do need a little break-in time (at least an hour using significant lower frequency drive signals to exercise the diaphragm surrounds until they break in) until their steady-state performance levels are largely reached. Chris

What this is basically saying is that the "back chamber phase inverter"--i.e., bass reflex or tapped horn--continuously moves its effective acoustic center away from the listener as the frequency decreases (or lobing increases the effective distance between the direct radiating woofer and/or port and the listener) below a certain break-point frequency where the phase inverter, i.e., port resonance, kicks in. This seems to be the common denominator of these designs, and unfortunately, this isn't a desirable characteristic--in my experience...and the source that Toole cited in his book (referenced here in the K-402-MEH thread). Chris

This is actually an issue that I'm afraid we might run into Roy on, because his (now expired) patent on a bass reflex--horn loaded bass bin is not in principle different from a bass reflex-ported MEH or even a tapped horn (a patent that is still in force that's owned by DSL). These are all different ways of looking at the same thing: a phase inverter on the back chamber of a horn-loaded set of woofers. Roy seems to think that good things happen when you use a phase inverter in a bass horn (which he thus far has not discussed openly--in terms of which parameters get "better" and which get "worse" when you do that). I have an idea of what happens, but no data on his particular new bass bin. Here is the data for an SH-50, showing the effects of the bass reflex ports below 80 Hz: And the KPT-1802-HLS in a home environment (the purple/indigo trace that parallels the tapped horn sub trace): You can see the sealed design K-402-MEH bass bin phase response in nominally half space in the bluish/cyan trace, above, which is much flatter (and therefore better in my estimation) than the internally ported 1802-HLS, the tapped horn, and the Cornwall (a conventional direct-radiating bass reflex woofer). Steeper slopes are worse, flatter are typically better. The current Jubilee bass bin shows the effects of an exponential horn expansion below 1/4 wave resonance (i.e., 40 Hz). So I'm a little hamstrung in answering this question. Roy's initial invention had the "preferred embodiment" using passive radiators, which he shifted to using cardboard tube ports after the initial KPT1802-HLSs (and apparently the few 1502-HLSs that were built and sold) for home hi-fi duty. The difference between cinema duty and home hi-fi duty appears to be that home hi-fi (home theaters) typically use the 10-30 Hz pass band as the operating frequency band of the subwoofer, while the commercial cinema use says "26-240 Hz" for the -3 dB passband. He's now saying that the "new Jubilee" bass bin (internally ported with a single 12" woofer) is "better" than the current KPT-KHJ-LF bass bin with dual 12" woofers and no ports. But he apparently declines to say why--in quantitative terms. What engineering parameters improve, and which degrade? My preference today, if asked, is still to not use an internal reflex (back chamber phase inverting) port. I use tapped horn subs, but the reason is form factor and their relative efficiency and lower compression distortion than direct radiators (especially ported direct radiating subs). If I didn't use tapped horns, and I had the space for them, I'd probably use either a multiplicity of K-402-MEHs in boundary gain (i.e., five surround channels) as a first choice, or alternatively, a conventional front-loaded horn having a very long internal path length for efficiency and avoidance of compression distortion (i.e., more than one of these). Since I don't have floor space for a conventional front-loaded horn, the MEH solution would be my preferred option. Chris

Here is another comment from the other thread that I think can kick things off: Now we're discussing amplifier power required, when not too many pages before this in the K-402-MEH thread (and other related threads), it was noted that amplifier power isn't a factor in today's economic environment. Why the shift towards this topic, but flipping the stance/position in this case? It doesn't seem rational or relevant to MEHs. It only seems argumentative. Chris

This thread was created based on the discussion of the relative merits of bass reflex vs. sealed bass bin construction for multiple-entry horns (MEHs). The genesis of this thread can be found here: https://community.klipsch.com/index.php?/topic/161404-a-k-402-based-full-range-multiple-entry-horn/page/57/&tab=comments#comment-2634408 which is also repeated here for convenience: ...and the ensuing discussion. Chris