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Chris A

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  1. Door #3 whenever I can do it is the ideal solution, especially with horns. I think that was PWK's ideal solution, too. Door #4: make the resonant frequency higher via a stiffer structure, then damp the mouth of the horn until there is no measurable resonance via clamping it to a piece of wood or MDF, etc. I like door #4 as an engineering solution to the problem, and I believe that's what PWK actually did with his loudspeakers and horns. That's also the way that I intend to do it with the future MEH using the thicker/stiffer horn assembly, and clamping the horn mouth to the box. But that doesn't relieve the designer of having to make the horn walls thick enough not to deflect measurably (especially acoustically significant deflections). That means that the walls of the horn can't go into "flat plate breakup modes", instead of the "bell ringing" modes that are typical of metal horns--like the early K-400-series horns. I think that a lot of people were stuck with multicell (sectoral) horns from Altec, et al. that couldn't be clamped to a big piece of wood, etc. so they started to put mastic on the back of the horn in the hopes of achieving door #3 solution. That approach on curved mouth horns doesn't usually work very well--in my experience. Chris
  2. Thanks guys. It's not always clear to me that threads like this one are effective in conveying the ideas of (in this particular case) new loudspeaker design approaches. Your comments let me know that there's an audience that's reading and has interest. I should take just a little break to explain why Klipsch doesn't presently produce MEHs: it's all about the legal uncertainties of the basic ideas, and perhaps a little having to do with "not invented here" (NIH): When you look at the development of of the first electrically-driven loudspeakers around the time of the "talkies"--motion pictures with sound which occurred ~1930-1931, there were similar issues of patent protections and "NIH"--not invented here. Western Electric--the engineering arm of the Bell Telephone system at that time--first produced the now-famous WE horns having very large dimensions, very high fabrication costs (welded steel horns of folded and/or curved horn construction), extremely high weight, very limited polar (audience) coverage and similarly limited low and high frequency response capabilities. Later on, there were innovations by MGM led by two men: Shearer and Hilliard (of Shearer Horn fame)... ...which directly led to the Altec Voice of the Theater (VOTT) series of loudspeakers with greatly improved cinema coverage and frequency response performance, much lower fabrication costs and weight, and lower demands on the driving electronics (amplifiers, preamps, etc.). This led to the "golden age of cinema": While all this doesn't appear to apply to home audio, I think that many horn aficionados today see the connection to real hi-fi beginnings. For many years, the home hi-fi scene has been dominated by Ed Villchur's small "acoustic suspension" (closed box, infinite baffle) direct radiator systems (Acoustic Research Corporation and others) as the costs of power electronics began to fall, and later on, the rise of ported-box bass bin loudspeakers. These loudspeakers made home hi-fi affordable and economically realizable for the small room sizes that were typical of the new "suburban" homes and small high-rise city apartments of that time. Since that time (late 1940s-early 1950s), there has been the realization by some consumers that direct radiating boxes didn't have the same feel and experience of "being there" that fully horn-loaded loudspeakers provided in commercial cinemas. This was the cue for Paul Klipsch's corner horn to arrive on the scene: the Klipschorn. During those days of mono, a single Klipschorn in one corner of the room brought that "excitement" potential to those hi-fi enthusiasts that could tell the difference between direct-radiating small loudspeaker sound and fully horn loaded "live sound": Ever since that time (late 1940s-1960) there has been a tension between the "small loudspeaker" buying public and the horn-loaded enthusiasts. The year 1960 brought with it the innovation of stereo records and stereo loudspeakers. Instead of one loudspeaker, you buy two and set them up in identical twin pairs. The increase in realism of the performance was undeniable. Thus, the real birth of home hi-fi that rivaled commercial cinema sound systems began in earnest: One of the most important advantages of the MEH design discussed here is that it brings the fully horn loaded sound back to the technical/subjective listening forefront of high fidelity--and in an even smaller package than corner horns. The rest of the story about the "why" of MEHs is now put into better perspective, above. Chris
  3. No issues with measurements or simulations. Just engineering knowledge of how stiff the horn should be...and the stiffer, the better. There are even some people that have used concrete as a horn material (but I don't recommend that, however, since it makes the horns into boat anchors--and they cannot be made in thinner sections). No resonances that I can identify, but until I get a horn made and test it A-B fashion... The clamping of the horn's flange on the mouth end to the box gets about 95% of the issues. The rest of the story on thicker horn material is just good engineering practice. Chris
  4. So the bottom line on MEHs designed and built for horn hi-fi duty is that they will: look different than Danley designs (i.e., they will probably incorporate nicer finishes and perhaps incorporate horn veneer, be less deep, and have a wider mouth) have significantly more operating flexibility with their inherent room EQ/boundary gain (even including excellent corner-horn performance if that's desired), cost significantly less than Danley products, have more hi-fi sound, perhaps that rivals any loudspeaker that the user/owner has ever heard, be significantly smaller than a Khorn or Jubilee--at about La Scala II size or slightly wider/taller, but form-factored to fit into room corners much better also perform very well as studio monitors due to their outstanding impulse response, frequency response flatness, almost perfect step response, controlled coverage (horizontally, vertically), and much wider frequency response, especially low frequency response down to sub-30 Hz. Chris
  5. SH-50 specifications: Coverage Pattern ....................................... 50° horizontal x 50° vertical Operating Frequency Range .................... 50 Hz - 18 kHz +/- 3 dB .................................................................... 37 Hz – 24 kHz -10 dB Sensitivity @ 1M ......................................100 dBSPL ................................................................... (Measured as 2.83V input, 1M whole space) Maximum Output ....................................127 dBSPL Cont.,133 dBSPL Peak Input Power Ratings ......................... 1000W continuous, 4000W Peak Nominal Impedance .................................. 4 ohms Minimum Impedance ................................ 3.5 ohms @ 1 kHz Recommended Processing ................... 40 Hz HP @ 24 dB/Butterworth Drivers ..................................................... LF 2 x 12”, MF 4 x 5” HF 1 x 1” Dimensions (in.)......................................28 x 28 x 25.5 Weight...................................................133 lbs SH-46 specifications: Coverage Pattern ................................... 40° horizontal x 60° vertical Operating Frequency Range .................. 71 Hz – 14.5 kHz +/- 3 dB ................................................................. 58 Hz – 16 kHz -10 dB Sensitivity @ 1M .................................. 106 dBSPL (Measured as 2.83V input, 1M whole space) Maximum Output ..................................135 dBSPL Cont., 141 dBSPL Peak Input Power Ratings ........................... 1400W continuous, 5600W Peak Nominal Impedance .................................4 ohms Minimum Impedance ........................... 3.4 ohms @ 810 Hz Recommended Processing ................... 60 Hz HP @ 24 dB/Butterworth Drivers .................................................... LF 2 x 12”, MF 4 x 4”, HF 1 x 1.4” Dimensions (in.)...................................... 29 x 22.3 x 22.7 Weight..................................................... 118 lbs As you can see, the output levels of the two MEHs by Danley are very high (127 dB, 135 dB music power), and as a consequence are three-way designs using seven separate drivers. They are both designed for "arrayability", i.e., stacking two or more units together to cover more angular arcs. In the case of the SH-50--it's 50 degrees each axes (hor, ver), and the SH-46 is 40 degrees horizontally and 60 degrees vertically. These are probably not the requirements that I would design to for home hi-fi use (but I'm sure that the two models sound very good nevertheless). _____________________________________________________________________________________ Notional Design Requirements for Home Hi-Fi MEHs In my view, the design requirements should include wide horizontal coverage of at least 80-90 degrees, and at least 60 degrees vertically (in order to avoid pattern flip of the polars below some operating frequency). FYI, the K-400 series midrange horns (including the K-500. 600, etc.) from the Klipsch Heritage series begin to experience pattern flip below about 2-->2.2 kHz due to limited vertical mouth dimensions. Additionally, I'd recommend minimizing the number of crossovers and drivers--to two woofers and one 2" compression driver in a two-way operation. This will simplify the crossover by 50% or more, and offer even better smoothness and inherent time alignment than a three-way design. If necessary to use three-way, using a dual-diaphragm compression driver (like the BMS 4592ND) bi-amped is definitely the way to avoid breaking up the MEH into two or more horn apertures (i.e., separate tweeters are not desirable), or employing midrange drivers to bridge any woofer-tweeter frequency response gap, which also adds significant physical complexity in terms of mechanical clearances, disruptions in polar coverage at 1-2 kHz, and having to add more PEQs to compensate for the 1-2 kHz crossover frequency on the multiple-entry horn. The two-way requirement (or using dual-diaphragm 2" compression driver) dovetails nicely with the maximum SPL output requirement, discussed next. The output levels required by home high-fidelity loudspeakers is more like 120 dB operating (for Klipsch horn-loaded users) and probably 110-115 dB by others that have imprinted on direct-radiating two channel operation. I think 120 dB is a good design point. There is no requirement to array MEHs for home hi-fi operation. This releases the requirement for arrayability, which affects the design parameters for the horn itself, and the phase/frequency curve characteristic, and an implied requirement for passive crossovers for outdoor PA use in remote installation locations. [The DSL Synergies have been designed for arrayability (and SPL) as perhaps the highest priority/precedence requirements.] Realizing that arrayability is not needed opens up the MEH design approach to other less intensive design and fabrication approaches for the crossover networks, and really opens the door to using DSP crossovers instead of passives. This increases the MEH design space fairly dramatically, allowing for optimization of polar coverage and minimizing impedance bounces in the horn/driver assembly for home hi-fi use. It also reduces to the design complexity by probably an order of magnitude, and permits even flatter phase and group delay plots, and almost perfect step response (using FIR filters available in something like a single miniDSP 2x4HD crossover [$200 for two stereo loudspeakers] which support FIR filtering to completely flatten the phase and group delay plots above 200-300 Hz...or perhaps lower frequencies). Chris
  6. Chris A

    Jubilee advice needed.

    The red is first order filters, while the blue is Bessel 3rd order. I've been experimenting with lower order filters in the MEH and the Jubilees over the past couple of months and have found that they produce much flatter phase and group delay plots, and also sound significantly more "together" from a subjective viewpoint. If you want to try first order filters on your Jubs or Jub-Khorns, let me know and I'll assist in any way that you want me to help. Note however that I'll be in and out of pocket for the next 2-6 weeks from today. Chris
  7. Chris A


    Jason, I'm sorry to hear this. I can also understand how difficult it can be to write the words. Thanks for letting us know. I hope you and your family find closure, and over time find a way to accept her passing--while remembering all the good stuff. Chris
  8. Chris A

    Jubilee advice needed.

    You're welcome at my place if you're through D/FW International any time. Yes, that's precisely what I was referring to: "realizable high fidelity". One is suddenly opening up to living in a glass house when making those sort of arguments, I think. They don't really make a lot of sense when you also look at high-priced DACs & turntables, very high priced passive active preamps, tube electronics, and other "audiophile-quality gear". That's the point that I was talking about. Jubs are probably the best decision that you can make for a very small listening room (of which the person that asked the question actually has more than a sufficient-sized one for Jubs in my estimation). [When you feel your knees banging on the Jub bass bins while being seated in your prime listening position, then I think it's time to consider headphones instead--but that's just my opinion on that subject ] The effect that you mention--that they sounded better as they were turned up--is usually due to the ratio of reflected to direct energy flipping over as the SPL goes up. That's very easily controlled using room treatments, but pretty limiting in terms of what you can say about the performance of the loudspeaker if it's not yet accomplished. That's why I commented. That's not a particularly good way to judge loudspeakers (of any sort) and certainly not at the performance level as Jubs, IME. IIRC, the pair that you heard were very newly installed--as mentioned in advance of your visit on this forum that I remember reading--and perhaps without room treatments/sufficient floor coverings and/or dial-in time (i.e., the owner mentioned more than once that he's pretty busy and presently that time is in short supply to tinker with them). I know about that situation as I had to learn about it myself over the years. That owner is also a member of this forum (...you remember, the guy who was kind enough to honor a request to hear them out of the blue). I recently saw the same thing occur on that exact situation happen over on the diyAudio forum recently (...and the owner's response detailed here.) The guy that did it (user:Joshcpct) is now "famous"... Chris
  9. That's for horn-loaded bass bins. For direct radiators with horns on top, the bass bin will usually be nearly in alignment as-is (due to the phase delay of the crossover network itself), or a small amount of delay can be applied to bring the bass bin back into alignment at the crossover frequency. Look at the REW spectrogram views to see if the bass bin is leading, aligned, or lagging the higher frequency horns/drivers. If the bass bin is leading the horn, you can add delay that's read directly from the spectrogram view. Chris
  10. Chris A

    Jubilee advice needed.

    The smaller the room, the more you need the directivity of the K-402s and KPT-KHJ-LF bass bins. Heresies and other speakers with direct radiating bass bins just can't provide the needed directivity. Having a fair amount of absorption--like the person that asked the question has--is also required, just like that which is used in small monitoring control rooms: The logic that is tied to how loud some loudspeakers can play (or not play) or their small physical size doesn't come into the equation. That sort of thinking is actually not applicable, IME. It's all about full range directivity in small rooms and that's a function of the mouth size of the horns, and reducing the reverberation times of the room to below 0.3 seconds (that is, RT20, RT30 reverberation times). This may seem counter-intuitive, but I can assure you it is the case. The real problem with small listening room acoustics is early reflections. Control those early reflections, and you will experience an immersive sound that's truly spectacular. Jubilees I believe will sound absolutely spectacular in that room. Put the money that many people put on electronics...on the speakers and room treatments instead and the results will be dramatically different. Chris
  11. Chris A

    Qobuz is now available in the US

    One little issue that I ran into: in order to listen to their "free trial", one must put in a credit card number or Paypal account number... ...that's not going to happen anytime soon. Chris
  12. Chris A

    U2's One: bad mastering or Klipsch's weak spot?

    This is true--if you're talking about demastering it at home or you are considering buying a "remastered recording" CD (which means that they crushed the dynamic range further in order to make it even louder than earlier CD versions). Demastering is just the same as using a very high quality EQ unit (sort of like a Cello Palette--but digital EQ inside the digital player), and taming poor mastering EQ, and perhaps taking the time to remove some line noise in the recording (50/60 Hz, 100/120 Hz, HVAC fan noise at 19, 38, etc. Hz...which curtails modulation distortion sidebands at midbass and above frequencies). Except that if using demastering, you do it once and save--instead of doing it every time that you play it back with an EQ unit. Demastering also also shows you the spectrograms and spectrum plots what you've got so that you can zero in on the obvious issue areas, facilitating the update iterations until the corrections sound best. Even if you correct the noise and EQ, the CD tracks can be trash to begin with--especially if someone used way too much compression and clipping (limiting). Once the mastering people use a compressor on the music--that's it. Your ability to recover the dynamic range (the thing that most owning hi-fi rigs actually like to hear) is basically gone forever. One can reconstruct clipped peaks, but clipping permanently degrades the tracks even if the clipped peaks are reconstructed--because you lose all of the higher frequency content of the clipped peaks during the clipping process. De-clipping can only reconstruct the fundamental frequency for each peak--not the higher harmonics that reside inside the peaks. That's permanently removed during mastering clipping (the euphemism of clipping is called is "limiting") . So, bottom line: demastering can make a big difference in listenability--but the quality of the tracks is basically set when you get them on CD. High quality pristine music tracks always sound much better. Applying demastering to phonograph records is possible but more time consuming since you first have to play the record into an ADC to capture it on 44.1/16 or higher bitstream, then separate the individual tracks by hand, name each individually, then remove sub-harmonic artifacts that are on the record (warp, ticks, pops, etc.). Then you can demaster the EQ. Fortunately, the format of phonograph records is just barely able to hold the music as-is, so the record companies cannot do to them what they do to CDs--else the record needle would jump out of the groove. The dynamic range of the records is often 4 or more dB higher than the CD release because of this limitation of vinyl. Chris
  13. Right. I always try to keep the concepts as simple as possible (like A. Einstein recommended) when introducing new concepts in order for those learning to hang onto the facts at first, then deeper learning comes later on. The real world truth is hopelessly complicated if you keep looking closer at the problem. Chris
  14. Basically, you have to add more delay on the HF channel--like what you said above. You use the spectrogram plot to verify that your calculations and assumptions are correct, and you do the fine tuning using the spectrogram. The problem is that you really can't completely compensate for the phase mismatch in and around the crossover frequency band due to the steepness of higher order crossover filter phase changes. So there is a bit of a balancing game that I use to allow some delay through the crossover region. I look at the group delay (GD) plot to make sure that there aren't any spikes just above or below the crossover frequency that is introduced by the time delay that I use. For instance, the negative-going spike at 500 Hz is due to the crossover interference between the AMT-1 and the Cornwall woofer, even using a first order filter. In this case, that group delay spike is due to the vertical separation of the AMT-1 from the woofer, even though the time delay is effectively compensated in this plot. Chris
  15. For a pair of drivers crossed over, i.e., woofer-midrange, midrange-tweeter, etc., it's 90 degrees of additional phase delay on the lower frequency driver for each order of the filter. For a fourth-order set of crossover filters, that's one complete cycle of delay on the low frequency driver that should be compensated via delay of the HF channel. Chris