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New Loudspeaker Technologies Applicable to KGI Products


Chris A

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I'll start by copying my thread from another that started this discussion:

 

1 hour ago, Chris A said:

 

Since I was inducted into this thread involuntarily (i.e., post #6 of the previous thread that no longer exists), I think that looking forward instead of backward and discussing some of these new technologies is a lot more constructive, interesting, and friendly, unlike arguments about branding, which frankly mean extremely little to me personally---since my buying identity is not defined by brand alignment but rather by technology alignment (i.e., horn-loaded loudspeakers, etc.).

 

It may be that Roy can't talk about what he's looking forward to doing, so perhaps the members of this forum that don't work for, or are beholden to (by whatever services that have been provided or are hoped to provide to) KGI, can perhaps fill in the blanks.  Who knows, perhaps there might be something that will help KGI's marketing arm open up their current views on market--technology subjects.

  1. One area that is already with us is the use of multi-amping using DSP crossovers instead of mono-amping using passive crossovers, which now very much represent the "lower performance/lowest priced" end of the consumer loudspeaker market.  The evolution to DSP crossovers has already been made in the professional (commercial cinema, studio monitors, PA, etc.).  KGI has made limited use of the technology, but the economics of it and the potential for performance gains is so great that it really cannot be ignored or continue to be assigned to secondary (table radio, specialty, little loudspeaker) markets. 

    The New Jubilee is apparently coming out with DSP crossover as standard issue.  I see this as a heralding event that can easily be applied to basically all other Heritage products as add-on kits, even for very old Klipsch models.  Any upgrades to drivers can be made within these "kits" along with user APIs or third party "room correction software" to dial these in after purchase.  This is a pretty big deal, and I think is already beginning to affect the consumer marketplace.  This is something that will see more and more penetration into the consumer marketplace as younger buyers, no longer stuck on yesterday's technologies, now represent the bulk of the buying marketplace.
     
  2. Multiple-entry horns (MEHs).  This technology is now being used by KGI's competitors in the commercial marketplace, and is no longer an item that can be ignored as "not invented here" (NIH) technology.  PWK himself built his first loudspeaker based on new technologies of his time, which became the original Klipschorn.  This also continued with the Shorthorn, Rebel, Heresy, La Scala, Cornwall, and then Belle, which are all applications of existing technology of their time. 

    Why MEHs?  Coaxial performance in a smaller package without violating Hofman's Iron Law.  DSP crossovers make this an easy and low risk product development path. The performance gains available using this technology are now well known, and the time has come for introduction into the consumer market.  It's been my experience that these loudspeakers generally blow away anything else that I've heard--assuming that linear phase response is retained in their design (i.e., the difference between the newer Synergy and older Unity horns).  MEHs are also less expensive in terms of bought parts--only one horn is needed, and plywood works extremely well.
     
  3. Use of better and potentially lower cost high frequency drivers not requiring such high sensitivity (I.e., coupling DSP crossover/multi-amping with their employment).  They can be used in two-way designs, thus eliminating a horn and extra driver used in the older three-way passive crossover designs.

    It's no longer a secret that by using DSP crossovers, a whole new universe of loudspeaker configurations of extremely high performance levels are now possible that were precluded by the use of mono-amping/passive crossovers.  I'm not sure why this subject is so obscure, but my experience is that the use of very low moving mass drivers (AMT-type) can now be used in fully horn-loaded loudspeakers.  If you've never heard these type of drivers, you're in for a treat, because they present flat-phase response even better than "full range" two inch compression drivers costing many times more.  The AMTs can now be used with DSP crossovers in horn-loaded applications, and there are many manufacturers that make these drivers nowadays.

 

and some follow-up posts:

  

1 hour ago, Edgar said:

Once again, @Chris A, I find myself adding to one of your insightful posts.

 

First, I think that the concept of an API for EQ, room correction, etc., is genius. Those who don't want it don't have to use it. But for those of us who have the background and knowledge, it would be a delight.

 

Second, I'll add that high-quality, high-efficiency amplification is now available for hundreds or even tens of US dollars, instead of thousands. This has already made its mark in Pro Audio in the form of amplified loudspeakers. There's no reason that it couldn't do the same in the consumer market. To have complete control over EQ, power and overexcursion protection, thermal compensation, etc., is the objective. 

 

Both of these also contribute to what I consider the ultimate goal: an all-digital system from source to amplification.

 

1 hour ago, Chris A said:

One thing that I did miss that I should probably add to my last post, above:  PWK was a very early adopter of stereo/multichannel technology that came out of Bell Labs studies.  He basically led the industry in promoting his three-channel systems (i.e., two corner-loaded loudspeakers and a very good center).  This is also highlighted in Greg's ("Edgar's") post just above in his signature line.  Look at the date of that quote, and look at the date of the stereo record and tape introduction into the consumer marketplace.

 

PWK wasn't "behind the times", rather he led, but in another direction than the Villchur "small loudspeaker" marketplace.  His company survives intact today, unlike all of those other consumer-based loudspeaker companies.

 

Looking forward (i.e., not really backward), some of these technologies have already been introduced into the consumer marketplace, as mentioned above.

 

Chris

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I've been thinking more about the high-efficiency amplification topic. 

 

Imagine 90%+ efficient amplifiers in 100dB+ loudspeakers. Both are readily available right now. Now let's say that the average power delivered to each loudspeaker is 10W. It's probably less than 1/10 that, but let's be generous. A 90% efficient amplifier needs 11W input to provide 10W output.

 

Let's say that we're powering the amplifiers with a string of 12V, 12AH motorcycle batteries. I know, for example, that the TPA3255 Class-D amplifier can work between 18VDC and 50VDC, and since we don't need a huge amount of power let's put two batteries in series for 24VDC.

 

11W at 24V is 458mA. So the 12AH batteries would last 26 hours before needing recharge.

 

Imagine an entire system in which the only connection between the source and each loudspeaker is a single fiber optic cable. Or perhaps no physical connection at all; it's all done via Bluetooth or other RF technology. No more exotic interconnects or speaker wires. 

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Wow.  Burning Man festivals could crop up everywhere.  Kind of makes you rethink Woodstock (1969) in totally different terms.  A solar panel in full sunlight could easily get close to that kind of power output to keep them recharged during the day. 

 

Of course, there is a lot more to this than festivals and outside concerts, but the thought did initially come to mind. 😉

 

There are more technologies that haven't been discussed, one of them that fires the imagination:  using time-domain FIR filtering to improve upon the transient response of the loudspeakers (a.k.a., Fulcrum Acoustic "Temporal Equalization").   Here's a really good video overview of that from David Gunness himself:

 

 

Here's the follow-up video that Mr. Gunness mentioned in the above video:

 

 

Chris

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1 hour ago, Chris A said:

There are more technologies that haven't been discussed, one of them that fires the imagination:  using time-domain FIR filtering to improve upon the transient response of the loudspeakers (a.k.a., Fulcrum Acoustic "Temporal Equalization").  

 

Everything that he said was true; I only wish that it was so simple in the real world.

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on the cheap side a TPA3116 mono (or two channel) amp can be carried in one pants pocket and a 20v tool battery with dock in the other.  Limiting low end to ~100Hz would save on speaker size.  I've noticed my kid thinking a little Karlson box had good bass (jazz trio - piano drums - upright bass) - it has a good amount of impact from the cavity and Beta 8CX vs his Rokit8's which go much lower on the scale.

 

A little basshorn made of 12mm plywood - say a scaled Cerwin AB36 using  Delta 10 and with hinged wings could make a lot of noise outdoors on 20-40 watts.  It would be about the size and shape of a full tower ATX case.  For "non" critical sound I would perch one of those Dayton round re-entrant horns on top plus whatever bullet tweeter that's cheap, smooth and with replaceable diaphragm.

 

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No offense, but I have to be honest: there is nothing about Karlson bass enclosures that interest me.  Or split K-tubes.  These actually have terrible acoustic performance--especially polar performance.  These were inventions of the 1950s that were initially made without access to proper measurement gear that we have today.  PWK tested them and handily rejected them as has been discussed on this forum many times.  I agree with his assessment. 

 

I'm also not into "cheap" or compromising bass frequency response in order to make the enclosure "small".  Candidly, lower cost amplifiers can be had, but, for me, life is just too short to spend it on trying to make equipment that's too cheap produce the kind of sound that I want to listen to.

 

Generally, to help others understand the rationale for this thread: it was created to be about reproducing extremely high fidelity sound using emerging and modern technologies, not about rehashing yesterday's technologies because they are cheap, available, or small. 

 

Chris

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6 minutes ago, Marvel said:

A serious question for Chris and Edgar (Greg?).

 

The quality of class D amps for high fidelity. Are they good enough? Some people swear by them, some don't, or only use for bass only.

 

My personal opinion only:

 

I have a Rotel Class-D amp that uses ICE modules. I don't like its sound in any portion of the spectrum.

 

I have several TPA3255 amps from 3E Audio. They are superb in the bass, but only OK elsewhere.

 

I auditioned Bel Canto Class-D amps many years ago. I found them to be very good in the bass, just OK elsewhere. They may have improved since then.

 

The QSC Class-D amp that Roy used at the Bonehead Class sounded very good to me.

 

- Greg

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57 minutes ago, Edgar said:

@Chris A, I have a question about MEH.

 

Given the high throat pressures that are likely to occur in a MEH (higher, at least, than in separate direct radiators), isn't the tweeter diaphragm excessively modulated by the woofer motion? How is that dealt with?

By ignoring it's existence, and calling it a worthwhile compromise, like all things man made.

 

As far as I can recall, PWK would still be pointing out that Intermodulation Distortion in a Loudspeaker at adequate listening levels is orders of magnitude greater than in an Amplifier, yet those numbers NEVER show up in a specification! I'm pretty sure he used Altec 604 measurements to prove his point.

 

The only reason he went from tubes to solid state in the 80's was that BGW and Crown were designed to minimize Otala Distortion and performed better than Valves with regards to Transient Intermodulation Distortion, which was the Achilles Heel of the early SS amps.

 

In support of modern tools and drivers, as noted by the title of the OP, we should be continuously evolving. However, to quote Geoffrey Chaucer (from 14th century England): "The life so short, the Craft so long to learn."

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7 minutes ago, ClaudeJ1 said:

As far as I can recall, PWK would still be pointing out that Intermodulation Distortion in a Loudspeaker at adequate listening levels is orders of magnitude greater than in an Amplifier, yet those numbers NEVER show up in a specification! I'm pretty sure he used Altec 604 measurements to prove his point.

 

Years ago, when I actually worked at Altec Lansing, D.B. "Don" Keele gave me the frequency response of a driver to analyze, but didn't tell me what the driver was. I ran it through some analysis that I developed for my Master's Thesis, and reported back to Don: "There is a reflection at x microseconds, another at y microseconds, another at z microseconds ...". Only then did Don tell me that the driver was a 604. I had found the multiple reflections between the woofer cone and the back of the horn. So a coax driver has more problems than just intermodulation distortion.

 

Everything is a compromise.

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1 hour ago, Edgar said:

...isn't the tweeter diaphragm excessively modulated by the woofer motion?...

This is a very good question, and one that I believe addresses the "fears" of all who shy away from MEH designs. 

 

I'll have to look for the link to the reference, but I read paper that identified this and measured it, and it found that it's inaudible. That's a pretty big result/find.  I can think of a couple of reasons for this:

 

1) The Huygens-Fresnel principle works one direction, but if you look at the reverse direction (reflections and other acoustic waves coming back into the slit/throat aperture, you'll find that the "rejection ratio" from front-to-back is extremely great.  If you also look at electro-magnetic multiple-entry horns, this is also a principle that's been observed and used to advantage. (Other E/M issues exist in multiple entry horns that make them more challenging to design well.) 

What happens on the back side of an acoustic horn (i.e., the higher order dynamics) tend to stay there, and the same is true for reflections and other energy coming back from the front side of the horn.  I believe that this is the principle that allows horns to work so well without higher order effects completely blanketing their performance.

 

2) There is a frequency separation of the pass bands between the throat-mounted driver and the lower frequency drivers.  This isn't a trivial factor.  Higher order harmonics can be excited in the horn--perhaps on the lower frequency drivers, but subharmonic frequencies getting into the throat-mounted high frequency driver are strongly rejected by the aperture of the horn at the throat and design of the higher frequency driver itself.  The frequency spectrum inside the horn is also "pink", their amplitudes are being reduced by 1/f factors as the frequency rises (which is the same the principle that I use to demaster recordings).

 

3) Danley makes a big deal in the Synergy patent (USPTO 8284976) over the original Unity patent (USPTO 6411718) that he is crossing over below the first notch frequency of the horn (i.e., this is apparently the only difference between the two patents). 

I now see this as a way to keep the lower frequency driver spectrum (from drivers closer to the horn's mouth) from affecting the higher frequencies from the throat of the horn because they are using Synergy horns at 120-130+ dB(SPL) for PA use--well above those SPLs used in home hi-fi loudspeakers. 

 

So if you're worried about this, simply cross the lower frequency drivers below the first notch frequency--but then you're in patent trouble if you do via the Synergy patent.  I don't cross below the first notch frequency, and the effect of the lower frequency drivers on the compression driver is inaudible and for me not visible in electrical impedance plots.

 

Chris

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1 hour ago, Edgar said:

 

Years ago, when I actually worked at Altec Lansing, D.B. "Don" Keele gave me the frequency response of a driver to analyze, but didn't tell me what the driver was. I ran it through some analysis that I developed for my Master's Thesis, and reported back to Don: "There is a reflection at x microseconds, another at y microseconds, another at z microseconds ...". Only then did Don tell me that the driver was a 604. I had found the multiple reflections between the woofer cone and the back of the horn. So a coax driver has more problems than just intermodulation distortion.

 

Everything is a compromise.

Indeed, everything is a Compromise. Yet, there are still people who buy (in low numbers) the Great Plains Audio Upgraded version of an Altec 604. I heard those in my teens on McIntosh tube gear and I thought they sounded pretty good. Little did I know then.

 

That being said, Nelson Pass uses a Tannoy Co-Axial speaker to audibly differentiate the sound between amplifiers. Go figure.

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35 minutes ago, ClaudeJ1 said:

Nelson Pass uses a Tannoy Co-Axial speaker to audibly differentiate the sound between amplifiers. Go figure.

 

Consider the Tannoy coaxial's phase response--combined with its (largely) full-range directivity.  I think that's what they are listening to.  Mr. Pass has talked about phase response as a factor,  I'd guess that the human hearing system is extremely sensitive to amplifier--loudspeaker-direct-arrival phase response--if the phase response isn't already scrambled by one or more component in the system (including the room itself). YMMV. 

 

An order-of-magnitude phase response plot from the 15" Tannoy HPD in an enclosure:

 

image.thumb.png.a1b834b2bb0b34669c980c983cb8f25f.png

 

I find that that a dialed-in Jubilee or an off-the-shelf SH-50 will magnify phase response issues of amplifiers, recordings, and EQ (minimum phase corrections, that is).  You can easily hear that phase response when you also control the early reflections in-room. 

 

Chris

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43 minutes ago, ClaudeJ1 said:

Indeed, everything is a Compromise. Yet, there are still people who buy (in low numbers) the Great Plains Audio Upgraded version of an Altec 604. I heard those in my teens on McIntosh tube gear and I thought they sounded pretty good. Little did I know then.

 

We cannot reliably hear what we can measure. We cannot reliably measure what we can hear.

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1 hour ago, Chris A said:

This is a very good question, and one that I believe addresses the "fears" of all who shy away from MEH designs. 

 

I'll have to look for the link to the reference, but I read paper that identified this and measured it, and it found that it's inaudible. That's a pretty big result/find.  I can think of a couple of reasons for this: <snip>

 

Thank you, @Chris A, but none of this appears to address my actual concern. My question is not prompted by consideration of reflections at all; it is prompted by consideration of the direct bass wavefront propagation. The bass wavefront enters the horn at some point between the vertex of the cone (where the tweeter is located) and the mouth of the horn. Having entered the horn, that wave propagates according to Huygens' Principle (I learned it as just "Huygens". I don't know when the attribution to Fresnel was added.) not only forward toward the mouth, but also backwards toward the throat. As it moves toward the throat, the cross-sectional area is decreasing so the pressure is actually increasing, reaching a peak right at the point where the tweeter diaphragm is located. So the air pressure seen by the tweeter is modulated by the bass waveform. Why isn't this a problem in the real world?

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3 minutes ago, Edgar said:

Thank you, @Chris A, but none of this appears to address my actual concern.

 

Yes, I think you missed my points.  Perhaps a PM/email would be a better place to discuss.  (If you still want to discuss it here, I will of course.)

 

Chris

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1 minute ago, Chris A said:

 

Yes, I think you missed my points.  Perhaps a PM/email would be a better place to discuss.  (If you still want to discuss it here, I will of course.)

 

 

I don't think so. The only one of your points that addressed my concern was #2, in which you state the "what" but not the "how". The "how" is what interests me.

 

Perhaps we should take this offline for the moment, perhaps posting a summary afterward.

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I would be very interested in a summary, thanks in advance.

 

If I recall correctly, DSL uses a second order high-pass-filter on the throat-mounted CD (a capacitor and a coil of low DC-resistance), so that the voice-coil of the CD is short-circuited by the low Ohm coil, to prevent it from being "excessively modulated by the woofer motion". This could also be done by the low output-impedance of an amplifier when the CD is directly coupled to the amplifier. Please correct me if I am wrong.

 

Best regards

 

Steffen

 

 

Edited by Supersteff
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