Chris A

I'm an engineer by upbringing, education and career. My experience has been that it's the choices made during design that predominantly determine the success of the produced products in the marketplace. By way of a now-famous example of this point: in the 1960s-1980s, the "big three" Detroit automakers generally engineered autos with not-so-good quality/reliability standards (by comparative standards to their across-the-Pacific competitors), and the result of what was to become a defining management cultural clash which became "The Machine that Changed the World". It wasn't the assembly workers, but the engineers that designed the finished products (the autos) and the system that produced the product (the manufacturing and supply systems). Ford largely led the way in the US to their own recovery after they were basically being handed their heads by Toyota's and Honda's then-comparatively superior auto design and manufacturing systems in the 1970s-1980s. So yes, largely, it's in the hands of those doing the design (and that especially includes upper management of the organizations that control what the engineers do). We since have gone through "lean", "six sigma", and other "revolutionary" practices (which really disguised how bad their system designs really were beforehand, in terms of what the buying customers got--but not the profits reaped by those holding stock options). That also applied to electronics--and Motorola led the way to their own recovery vs. their competitors in the 1980s. See "The Parable of the Boiled Frog" for the story of those companies that didn't make that transition. So in a long-winded fashion (...I must be suffering from K-forum withdrawal, or at least subjectively realizing the importance of this forum what its loss might mean...), I think I answered your rhetorical question, above. Apparently Class D is much more difficult to do well than it appears, and probably still has a distance to go before it really does overcome class A and AB design performance, and the hunt for, I believe, is still on for the source of the sonic differences between the two approaches. This viewpoint was one I adopted from reading Nelson Pass' views on this subject. I trust that guy in this particular domain. I believe he said that class D will probably ultimately triumph--but just not that he's actually seen/heard yet. Chris

I use the natural acoustic roll-off of the acoustic drivers when I can. Sometimes a large overlap of driver frequency/phase responses between "ways" makes the job more difficult with a DSP crossover since neither driver on each side of the selected crossover frequency is really attenuating by itself at the chosen crossover point. So in those particular cases, more PEQs usually are needed to attenuate the response of both drivers at the ends of their frequency response. But generally, one or the other driver is close to its own steep roll-off in response for many/most applications., and usually combine with the applied DSP crossover attenuation PEQs to give you 10 dB/octave or more slopes on either side of the center crossover frequency that's selected. In the case of the prototype K-402-MEH, the natural roll-off of both the woofer (low pass) and BMS 4592ND (high pass) is ~10 dB/octave at the nominally 550 Hz crossover point: a so-called "fractional order" acoustic crossover, neither first order nor second order. The same is true for the Jubilee bass bins and the TAD TD-4002 drivers: I get about 10 dB/octave effective slopes Chris

Well...the saving grace is that, unless there are strong nonlinearities in the system somewhere, subharmonics are very difficult to generate, much less hear, and the bandpass nature of acoustic drivers act as a strong "crossover" to reject those frequencies. The tests of human hearing above 20 kHz in the 1980s were apparently corrupted by aliasing, which is very audible, so for years, some people thought humans can hear above 20-21 kHz. Alas, such is not the case, and the frequency analyzer in the human ear (the cochlea and its associated structures) strongly rejects frequencies above 20-21 kHz--by apparent design. But as you are probably about to say, one never knows the effects of that added ultrasonic noise--that is impossible to completely filter out in the amplification loop during playback. _____________________________________________________________________________ In the case of SACDs (i.e., DSD music files), they also have huge amounts of ultrasonic noise or aliasing effects that must be filtered out during recording and playback. To my ears, the anti-aliasing filters used during recording do a good job of filtering out recorded frequencies above 20 kHz to prevent aliasing problems. SACDs, as provided by the record companies, seem to have the least amount of editing of the original downmix tracks, which of course preserves the their SPL and phase (transfer function) fidelity, and is probably what I hear as "higher quality" and "much more engaging" subjective listening effects. They are, in fact, extremely enjoyable to listen to in my experience, especially multichannel (5.1) recordings of higher overall recording and production quality than the norm for PCM 44.1 CD files. As most good engineers know, there are always tradeoffs in design--always. I've had a bad experience with class D amplification with DSP-front-end, but extremely good luck with many DSD recordings. Apparently the design tradeoffs selected in the prior case were not as successful as in the succeeding (completely different) case. Chris

I think that I've run into that myself. 😉 One of the things that Dirac does is apparently to flatten phase response, even bass phase response, a little. This of course greatly increases the perception of bass, something that Rob has mentioned here. In the present case the tradeoff is computing horsepower and added delay if trying to synchronize with a screen, as I do. I'm sure the bass response with FIR flattening below 100 Hz knocks your socks off. Once you get the balance of the Jubilee response flattened in terms of both SPL and phase response I recommend playing some higher quality digital recordings of orchestral or acoustic instrumentation only. These are recordings that don't use a mixing board to stack the music together, i.e., a "natural recording" with all musicians playing/being recorded at the same time. What I've found is that the music will now draw you into the performance like you've not experienced except at a live acoustic concert. If using live recordings, make sure that no amplified instruments are prevalent so that the original phase relationships of the instruments together are largely preserved. If you also place a little absorption on the floor (out to about 3 feet around the base of the Jubilees) to catch the strongest floor bounce, and some absorption on the side walls just where the K-402s and Jubilee bass bins are closest to the side walls, the center imaging and soundstage image will greatly increase their clarity and phantom center strength. The tradeoff here is a slightly less "full" sound quality. You can also repeat this small addition of absorption on the front wall just next to the K-402s/Jub bass bins--just the first two-to-three feet radially (i.e., the first 3-4 milliseconds of early reflections just around the horn mouths). When you get the acoustic treatments right, you can then sit back and listen to something that you've probably never heard before--except by the real thing: musicians playing together in real time, adjusting their performances to blend together. It can take your breath away with realism. Chris

I didn't have that menu item ("edit") for my posts, too. I suppose that something changed in the mean time...? Chris

Thanks Chad...I've got an "edit" function now... Chris

Sorry...I missed it. Implied humor tends to be a reason for blowups on the forum--so I tend to read literally. I still have a sense of humor......typically at a higher threshold. Chris

These are my choices: Chris

@Chad So, one of the changes that I see that's occurred is that the "edit" button has either been removed from the posts--or hidden in a place that I don't see. Is this a temporary thing, or permanent? Chris

Generally, once you get to 30-40 watts/channel, you're at the 99% level in terms of listening requirements (including peak requirements--depending on Klipsch loudspeaker model), not knowing anything about how loud you listen. I've run into one or two guys that apparently listen at 105-110 dB (average) and it's anyone's guess what the peak levels might be. Those individuals are using PA driving levels inside their homes--which I don't recommend if you're trying to retain your hearing acuity. (Since I'm not 18 years old any more, retaining hearing acuity is a priority requirement in my experience.) Having more amplifier power available is only an issue in the following cases: The quiescent noise levels (no source music is playing) into the loudspeakers creates an annoying background noise floor (this is usually the major issue, typically, of using amplifiers having too much available power). Lower power amplifiers typically have correspondingly lower noise floors. That's why most Klipsch owners tend to gravitate toward lower power amplifiers. The owner finds it difficult to keep from turning up the amplifier gain--until either his ears are taking a beating (a consequence of loudspeakers not having lots of modulation distortion--like direct radiating loudspeakers all have), or he manages to destroy the voice coils or driver diaphragms of the drivers in his particular model of Klipsch loudspeakers. This is something that is typically avoidable, but using lower power amplifiers will typically avoid the driver diaphragm destruction issue (but not the hearing acuity issue (depending on Klipsch model), since it only takes a watt or so to have very loud playback levels that shouldn't be endured for very long. Chris

[I don't see an "edit" button (...it apparently disappeared since yesterday's forum Russian DDOS assault...), so I'll have to add another post]: One of the issues that I've got with taking measurements at the listening position is that you're baking in those nearfield reflections in the FIR filters, which is why I asked about microphone location. I've found that the effects of using Dirac Live (Full) to be problematic in that it doesn't allow the user to take "minimum phase measurements" in-room. Instead, it forces the user to move the microphone back too far (at the LPs), then doesn't do a very good job in excluding the nearfield reflections, particularly in the 100-300 Hz band with the Jubs (which don't have directivity issues at this band--like almost all other loudspeaker do). I eventually had to find a way to shut off Dirac entirely so that it didn't try to boost the room nulls and attenuate the 100-200 band--fairly significantly. Everything sounded very "thin" with Dirac on. Once I turned it off and went back to REW measurements using the Xilica to correct, the "thinness" issue disappeared. YMMV. Chris

Thanks, Rob, for your responses. I'm glad that you are finally getting the sound quality that you were after, starting about 2 years ago. Did you ever get any further DSP support from Roy? The comment quoted just above about having the Jubs in the corners, I believe, is probably not very useful since the combination of the Jub KPT-KHJ-LF bass bins in parallel with the KPT-1802-LF subwoofer makes the bass response in-room not an issue with the Jubs out of tight corners. [mikebse2a3 is apparently doing a similar thing, apparently to avoid using absorption panels just at the exit to the horn mouths. He inserts an air gap instead. YMMV.] If using TAD TD-4002s, that "big boost" above 10 kHz isn't really an issue. It isn't really a "big boost" since the raw response of the TADs on K-402s actually rises above 13 kHz, without chattering, so that roll off shown is actually "baked in" via EQ (assuming you're using TADs). The TADs don't chatter. I would bet that Mitch probably hasn't heard them--since I believe he's been using JBL 2" compression drivers, and all of them do chatter above 10-13 kHz, so perhaps he's designed in something that actually isn't desirable. I'd recommend talking to him about putting that response back (unless you're no longer using TADs). If those compression drivers are Celestion Axi2050s (as the throat of the K-402s appear to be from your small picture you posted above), then I'd say that allowing the axial response above 10 kHz of them to roll off takes away significant amounts of "sparkle". At least, that was my experience with the Axi2050--which literally comes alive when that top octave is EQed flat to about 18-19 kHz on-axis. I recommend revisiting that. The effect of flattening the phase of the Jubilees as much as possible (in your case, using pretty significant FIR filters with a lot of delay) will not only make the bass so much better, it will also change the "listenability" of the loudspeaker quite significantly. See the following thread for a discussion of that phenomenon: Not sure of the context of this. On a side note: have you provided direct DSP dial-in support to his particular room/setup that I'm not aware of? I did spend time over the past two years helping Rob (always free of charge). I'm glad that he's finally getting to the sound quality that he's been after, albeit at a much higher investment level, unfortunately. Mitch has always been friendly, polite, and helpful. I'm glad that the FIR filtering seems to be working out for you, Rob. Chris

Try this: https://community.klipsch.com/index.php?/topic/155096-the-missing-octaves-audacity-remastering-to-restore-tracks/ The sibilence is likely in the recording, perhaps emphasized by the SPL response of the Cornwall IVs in-room. In the case of the Nora Jones Black Hole Sun recording (live at Detroit), it's around 2-3 kHz, which is lower frequency sibilance (it's usually at higher frequency--around 5-7 kHz). You can see it in the recording itself. You can better employ "room correction software" (i.e., flatten the in-room SPL and phase response). I don't recommend using amplifiers with high output impedance (i.e., SETs without global feedback) unless you re-EQ your loudspeakers to flat response. Chris

@Chad It seems that the forum is extremely sluggish this AM (16 Aug. 2021, from 04:00 local time to the present). Looking at the "Online Users" pages, there are sometimes over 100 pages of member activity listed, and all the users have visited within the last 30 minutes. Is there a new kind of web crawler that's vacuuming up all the threads, or is this a signal that the time stamping of the member activity is somehow compromised? Anyway, since you typically don't spend much time on the forum, I thought it might be news to you that the forum is experiencing big delays. Chris

I've got a few questions, since this is the first time that someone from this forum has used Mitch's service and his recommended hardware/software: Can you tell me where the microphone is for these measurements, i.e., how far from the front baffle? If you're taking measurements at the listening position(s)--more than a metre away--can you post the phase response plot and a spectrogram plot (i.e., before FIR filtering)? What order of crossover filters are you using? Second order, fourth order, etc. Are you still using REW to take measurements, or something else? JRIver (running on a PC) typically requires a pretty hefty PC (in terms of its computing horsepower) to apply the FIR filters--can you identify what PC you're using? JRiver also requires that your PC is running all the time and is your digital preamplifier. How do you handle analog sources, such as a turntable? Do you ever plan to run more than just stereo with a subwoofer, i.e., how would you convert to a 5.1? How would you expand to 6 channels using JRiver? Can you tell me how many FIR taps are being used in each channel (bass bin, K-402/TAD)? The Lynx Hilo apparently goes for $2300 (USD), the Earthworks microphone goes for $600, and the microphone preamp goes for ~$800 (i.e., the total microphone cost is $1400). Audiolense XO costs ~$460. Is that the product you're using from Audiolense? Any other add-on software that's required? Can you give us a ballpark on the service costs using Mitch's service? Chris

😊 Perhaps the dust has settled just enough to comment now. Note that my experience was with the Hypex FusionAmps (which were designed after the NC400)...was a fairly terrible experience...one that I couldn't believe was occurring when I first heard them after dialing everything in carefully. This leads me to believe that the designers of the Hypex FusionAmps either didn't catch the problems using their ears (and/or loudspeakers and listening rooms), or they just didn't care. Neither case is very attractive to talk about, and it has shaken my trust in this company's products (i.e., all of their products). _______________________________________________________________________________ I think that this is a very good point of view, and one that should elicit a little self-reflection on this pastime/hobby (i.e., not a profession to make money for living). I also think that a lot of folks like to obfuscate this point because they haven't taken the time/effort to learn how to do acoustic and electrical/electronic measurements--and interpretation of the results. The onus is still on them, I think to negate this balanced viewpoint enunciated just above. It includes both listening and measurements-together. Perhaps...or perhaps you might... I'd re-read what Langston said: It's been my experience that the subjective aspects of listening can sometimes reveal great truths about our pastime (listening to great music being reproduced), if we allow ourselves the opportunity. I found a great truth a couple of years ago that was astounding to actually hear, and one that basically no one apparently talks about. I attribute this situation to the problems that most people experience in terms of the capabilities of their setup: 1) not having full-range loudspeaker directivity, 2) not having full-range horn loading (to achieve extremely low modulation distortion), and 3) not controlling nearfield (early) reflections. Once you control for these three deficiencies (all simultaneously), then phase response, the other half of the transfer function, becomes an audible variable that differentiates "so-so" subjective sound quality from "outstanding", and suddenly becomes a key performance criterion. Apparently, few people have taken the required steps to hear this effect (especially researchers who write technical articles). Such is the one-sided nature of a world that is overwhelmingly populated with direct-radiating loudspeakers that have no real directivity control below 1-2 kHz, lots of limitations on the dynamic range of their loudspeakers before modulation distortion dishes out so much "mud" that they can't hear the music details (including the same issues in the recording that they select to do their subjective listening studies--as Greenfield and Hawksford apparently found), and in-room early reflections that destroy the subjective effects of flat phase response loudspeakers in-room. Apparently, this same type of experience exists in amplifier performance, but perhaps not as well understood as in loudspeakers, and the measures used to date that apparently don't identify the objective measurable differences between amplifiers that lead to the differences in these sort of subjective listening experiences. Your mileage may vary (...but not by very much, in my experience). Chris

This is what I use on top of my Belle bass bins for surround loudspeakers in my 5.2 array, except I'm using a single AMT-1 without wings presently: The advantage of the AMT-1 is that it has outstanding cleanness of response and the SPL/phase response (i.e., its transfer function) is extremely flat over the 600-20000 Hz bandwidth, with only some flattening EQ required to achieve really outstanding SPL and phase response...which can be performed at the preamp: In terms of bang-for-buck, I know of nothing else that comes close, especially considering that the AMT-1 replaces both the midrange and tweeter drivers/horns, avoiding the polar coverage/lobing problems and all the other issues of using two horns to do what one horn and easily do (like the Jubilee does). The only downside in this case is fairly narrow vertical polar coverage, but on the flip side, ceiling and floor bounce are no longer issues, so low ceilings and bare floors are much more easily handled than the stock Heritage midrange/tweeter configuration with its pattern-flipping midrange horn issues. Chris

"Operate as it should" includes a wide variability in interpretation. Perhaps if you can briefly discuss what kind of music you tend to listen to (i.e., rock, classical, jazz, dance, etc.), how loud, how soft, and what your expectations are, it will be much easier to answer your question. It's not yet clear what your intent or expectations are. For instance, if you were to identify with David Mancuso (RIP) or Colleen (Cosmo) Murphy's environments at The Loft parties in New York City, then that assumes a certain type of music, an average loudness level, and music genre type/source (i.e., lots of older music only found on vinyl), and a certain musical or entertainment aesthetic. If however, you're into close monitoring of audiophile style recordings of largely digital provenance, then that conveys an entirely different aesthetic. In the first scenario, you might be quite satisfied, just guessing from the picture (assuming dimensions there). In the other scenario, not so much, especially if the room dimensions don't allow you to listen from far enough away from the loudspeakers for them to coalesce well and develop a natural bass response without the entire room being in the "pressure zone" like listening inside an automobile interior. Having a bit more room internal volume and distance allows the Khorns to "breathe" well and give the listener a sense of space. I have found that there are effectively minimum room dimensions that limit using stock Khorns (...or any sort of loudspeakers, for that matter...). That seems to be about 8 feet of height (due to the collapsing polar midrange horns used), and about 13-14 feet for length and width (assuming an enclosed room). At smaller dimensions, the limitations of using Khorns (in particular) begin to become dominant quite rapidly. Some of the source of these issues has to do with not being able to move back a little bit from them and space them a bit wider side-to-side to further increase that listening distance. Anyone walking around the loudspeakers and interacting with that sound field just around the loudspeakers will be immediately apparent, I've found...even if not in a direct line of sight to the loudspeakers from the listener. The openings in the room to other rooms that you show definitely aren't show stoppers--in fact, they will likely help psychoacoustically, but the minimum dimensions of the room itself (length, width, and height) will likely be significant in my experience, even if you go to great lengths to EQ and time align them (i.e., correct their output for extreme nearfield listening). Some minimal discussion of room acoustic treatments in the nearfield is likely going to be fruitful going into that dialogue. Like how to handle the ceiling and floor bounce and keeping the area within about 3-4 feet of the front panels of the Khorns free of acoustically reflective objects (or at least heavily covered with absorptive material)--including electronics and furniture. I'd avoid "entertainment centers" and anything that places electronics near the loudspeakers within 4 feet on centerline height with the midrange horns. Chris

In terms of drivers and horns, that's what the K-402-MEH is within a single horn aperture in order to pick up the added horn gain and directivity below ~500-600 Hz (the crossover frequency) and pick up coaxial capability--which is a big deal in terms of polar coverage and absence of polar lobing, as well as having a much more compact package (~1/3 the height and visual presentation). What you describe is also essentially what the KPT-942-B THX is. Chris

Well, to be consistent, if you're going to ban discussion of K-402 horns, you'd have to ban any discussion of building your own Khorn, La Scala, Belle, Cornwall, Heresy, or any other Klipsch design on this site also, as well as DIY variants such as CornScalas and other home brew designs, like direct radiating bass bins...if it really is in "bad taste". I think it's a compliment to the basic design that it's even attempted. Besides, the K-402 horn profile is well over 20 years old now--mid-late 1990s--well beyond what any patents would protect...and there never was a patent on the K-402 horn profile, surprisingly. A few people have done DIY on Klipsch products over 70+ years. My father did a single Khorn in the 1950s while teaching EE as an assistant professor at SMU on a near-starvation salary using slave labor (also known as grad students) to test the resulting DIY Khorn in the EE labs. That's a complicated plywood design that required tips and pointers from the original designer himself: PWK. The two men met to discuss the DIY build in the mid-1950s on campus. It certainly wasn't the first DIY instance nor the last that involved PWK's basic loudspeaker designs. I don't think the marketplace is going to be filled with DIYers anytime soon building their own K-402 horns--because they aren't easy to build in DIY fashion. They're large horns--larger than any other 2" throat horns that I'm aware of, save the old WE horns of the 1920s-30s that are curved axis horns. So the visual image is not something that a typical female interior decorator would immediately approve of...although, that's really a conditioned response to "how things should look" which I actually don't share with those that complain. Additionally, some tolerance on symmetry of the finished area expansion and horn wall straightness/angle coverage is warranted acoustically. The K-402s also require EQ to function properly as hi-fi horns, due to their controlled directivity design which requires EQ. Controlled directivity is something that you don't get with the older K-400 horn design which experiences vertical axis polar pattern flip below ~2.2 kHz. Chris

Yes--to the throat, not the mouth. The K-402 employs a different design that was developed since the original K-400 horn was developed. The curved sides near the throat of the exponential expansion K-400 horn create issues with polar coverage vs. frequency, whereas the straight-sided walls of the K-402 near the throat avoid that issue, which also reduce the axial length of the horn. If the Klipsch Professional line of loudspeakers is still being sold, my guess is that you can still buy a KPT-402-HF assembly with stand and K-691 compression driver. Chris

Yes, in terms of the horn's acoustic length for crossing over well above ~170-180 Hz, since the axial length of the horn supports a 1/4 wavelength down to that frequency. Chris

You'll get a horn that's 3x deeper than the K-402 if using a tractrix calculator. The K-402 has a tractrix mouth flare only. The balance of the horn (about 2/3s of it, that is) is straight-sided. That is to say, it's not "conical" but straight-sided. For reference, the conical equation for area expansion, i.e., S = S1x2, will not be correct (where S = area of cross-section, S1 is the area of the horn throat entrance, and x = distance along the central axis of the horn, starting at the horn throat entrance). The K-402 straight-wall section has a more complex area expansion formula. The flanges are 2" wide (4" total on the mouth width), so the horn mouth itself is about 35.5 inches wide and about 21 inches high...and about 17 inches deep. After the mouth is constructed first (as the tractrix equation is usually portrayed--not from the throat), the horn itself is fairly easy to construct graphically: Just cut off the throat at the right diameter. For 2", the K-402 is 17 inches along the centerline. If constructing a horn for 1.4 inches diameter throat, it's about 17 5/8 inches, and for a 1" throat, it's about 18 inches long.

That quote was directly from the source you mention. I'm sure that he doesn't wish to get pulled into this kind of discussion by the person I quoted just above, however. The person I originally quoted has categorically produced products that have been uniformly praised by those who have tried them on their high fidelity fully horn loaded loudspeakers. To others contemplating Hypex products, clearly Claude hasn't heard what I heard with two FA122s on TAD TD-4002/KPT-KHJ-LF Jubilees, carefully dialed-in (SPL and phase response). It doesn't matter what somebody else said about feedback in class D amplifiers, and how it's "different". I cannot recommend any Hypex product or any other class D amplifier based on the same type of technology--based on what we heard here. It was that bad...at least as bad as any amplifier that I've heard on this setup. This is why I am speaking up on what I heard based on that personal experience. The fact that Hypex produced such inferior products (both amplifiers) and put their name on them leads me to believe that all they are doing is linearizing amplifier output via large amounts of feedback so that it measures well with the measures that they have picked. However, listening to the result of Hypex's approach clearly shows that they are not picking the measures that show the problems that are induced by using those methods. Chris

Here is a quote: I heard what I heard. It wasn't good. My wife also heard it and asked what I had done to our sound quality. Here is the link to the entire post in this thread that you only partially quoted, thus ignoring the core information that I had posted: I don't recommend ignoring what is said in that post, or the quote that I added just above. The source of that quote is unimpeachable. And my ears agree with him. Chris