Heard a KILLER new Electrostat - JansZen!!!!!!!!!

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Stereo image: Thanks for mentioning this one! Our speakers have none of the same issues as ML's. For one thing, although you can't see through our speakers, the lack of rear radiation has quite a few sonic advantages. Not everyone will agree. Some instead feel that realism is increased by a large area dipole's greater addition of room ambience to the recorded ambience, although probably not by the peaks and dips from comb filtering caused by interference of the rear wave with itself and with the front wave, or the image smearing from widely spaced arrival times for relatively high amplitude rear reflections, or the lack of damping of the membrane motion that comes with using a see-through radiator, and which can tack resonant tails onto transients that obscure their true nature. This note addresses just a few of the aspects that set JansZen speakers apart and let them reproduce sound more naturally and realistically than other speakers.

Count me as one who agrees that a monopole is easier to get right in most rooms. Personally, I prefer the flatter frequency response and better time-domain behavior. Plus, not too many recording/mastering studios use dipoles as reference monitors ;)

Regarding damping of ‘resonant tails’ could you please explain further why an asymmetric dampening of a diaphragm’s output is better than one with equal resistive forces (open air).
I get the acoustical room interaction bit; it’s the membrane motion management I’m interested in.


Restricting myself to ESL's alone: ... Ours are true lines, achieved the old fashioned way, by simply using flat sources whose widths are tailored to create the desired dispersion within their frequency ranges. This has advantages in terms of smoothness of the frequency response, uniformity of the SPL and spectrum across a wide area, and integrity of the membrane shape over time.

Can you please explain how the flat and narrow tweet/mid segments achieve a wide enough dispersion profile. I usually though that there would be a bit of ‘beaming’ unless waveguides of some form are used.
Do you have polar response plots you could share or point me to?
 
Regarding vertically integrated, self-powered active speaker, while it’s the ideal situation, the audio world somehow decided that users need to be able to make these (usually uniformed) decisions.

It’s as if the car industry sold cars without motors, and there was a whole industry supplying Diesel, gasoline, electric, steam, flux-capacitor and other variants of motors so users can mix-and-match.

Makes absolutely no sense, but there it is. It’s the sad reality of the game at this stage.

I’ll always applaud a design house, such as Meridian, that does it right.
 
Regarding damping of ‘resonant tails’ could you please explain further why an asymmetric dampening of a diaphragm’s output is better than one with equal resistive forces (open air).
I get the acoustical room interaction bit; it’s the membrane motion management I’m interested in.

In an ESL, open air provides most of the mass load, but very little damping, i.e., none of what might be called resistive force. A mass load is reactive, rather than resistive, in the sense of tending to stay at rest or in motion. The other reactive element is the elasticity of the membrane, which provides most of the spring load, but depending on the material, again not much damping. As you know, when a mass is on a spring with no damping, like a car without shocks, it bounces on its own after it gets a force or motion input. The best amount of damping, which is enough but not too much, is when an impulse to the system causes it to move without overshoot or ringing, but does not impede its motion like molasses. Properly tuned car shocks do this.

An ESL membrane's fundamental resonance occurs as the combined mass of the membrane and the air bounce on that membrane's spring. Our membrane material, which we believe is unusual in several ways, does provide quite a bit of damping, but not enough to reproduce a tone burst with no overshoot or tail at all audio frequencies. Other than the choice of membrane material, the measures required to provide damping happen to block the view through the transducer. Depending on the design of the transducer and the frequencies being reproduced, features that impart damping can be applied to both sides or, as in our case, just one side. I can see how it might seem like damping from one side would create an asymmetrical effect, but it does not. Membrane motion remains symmetrical, because the damping force is coupled consistently to the membrane regardless of the direction of membrane motion.

Can you please explain how the flat and narrow tweet/mid segments achieve a wide enough dispersion profile. I usually though that there would be a bit of ‘beaming’ unless waveguides of some form are used.

For a given transducer size, the higher the frequency, the tighter the beam. For a given frequency, the larger that transducer, the tighter the beam. There are also diffraction effects that cause lobing, but this does not affect the basic relationship between frequency, transducer dimensions, and beam divergence.

A [vertical] line array such as ours is tall and narrow. This means that the horizontal dispersion will be wide, and the vertical dispersion will be narrow. There is not much high frequency output above the top or below the bottom of the line, but quite a bit to the sides. Our tweeters have 80 degree horizontal dispersion at 10 kHz (total angle).

Do you have polar response plots you could share or point me to?

There's a set of in-room plots at 320, 2000, and 6300 Hz, where the low frequency curve is a little lumpy as a result. Will this do? Meanwhile, and more generally interesting I think, this topic is covered in detail in acoustics texts. The originals are Olson's Elements of Acoustical Engineering and Beranek's Acoustics. Those who frequent this forum would find tons of stuff in either of these books that would lend truth and clarity to quite a few audio topics that are addressed with varying degrees of accuracy on the web.

This reminds me that I forgot to respond to someone's remark earlier in the thread about how one can get all the info they need about audio on the web these days. Well, unfortunately, the same rule applies to audio info on the web as for everything else.

FWIW, I won't guarantee that I make no mistakes in my own statements, or that I never gloss over second order elements to keep an explanation tidy, but I can say with complete certainty that studying at least some parts of well vetted books like Olson's are the best way to get a clear view of the terrain. Much is presented in an intuitive way that needs no more than basic algebra and trigonometry to get the basic ideas. In no time, you could be estimating the effective widths of tweeters based on their dispersion specs.
 
David,

Thanks for the educational information.

With my schedule, I don't have the time to read the books you recommend. On a practical level, most ESL fans believe the following. Are they myths?

- The bigger the panel the better the sound
ML has been moving away from big panels. You have a fairly small panel. If you used a bigger panel, say 25%, would it sound better, and by how much?

- Full range electrostat is the ultimate. Your speaker sounds great without being one. If you were to design a full range speaker to compete with the clx and soundlab, would it be better or just different from your current design (to appeal to those who think it's better)?

Thanks!
 
Monopole, dipole, bipole radiation patterns -- a must-see web page

I find it very interesting that someone finally had the guts to do an enclosed back large ESL.

Thanks. Guts are us.

I've long held the notion that Dipole is a huge compromise in room interaction to favor low air resistance on both sides of the diaphragm. For frequencies above 300Hz, air resistance is minor and I'm sure there are compensations that designs like the JensZen implement to mitigate.

My 'ultimate theater' design uses ESL's in an infinite baffle mounting alignment to mitigate the dipole nasties.

So I believe we will see more designs doing a monopole config.

I stumbled on a cool page just now about radiation patterns, and while oriented toward HT, I think everyone will find it a nicely done and informative overview: http://forum.blu-ray.com/showthread.php?t=66471
 
. . . most ESL fans believe the following. Are they myths?

- The bigger the panel the better the sound
ML has been moving away from big panels. You have a fairly small panel. If you used a bigger panel, say 25%, would it sound better, and by how much?

First, you have to decide which way to increase the size, which matters somewhat, even for so small a change: taller, wider, both? After deciding and proceeding, each listener would then have a personal opinion about whether the result is an improvement. The primary result would be an almost imperceptible increase in loudness, assuming the additional drive power and voltage is available. There can be other effects, though, depending on how one handles the incidental changes associated with a size change.

Let's get a feel for the ripple effects from such a change on the design and the performance, based on a simple, symmetrical 25% increase in the area. (If you meant a 25% increase in the lengths of all four edges, the result would be the same, just more so.) Note that there is more than one possible result, depending on how the rest of the parameters are handled. I will assume that the priority is on keeping the distortion level and impedance as close as possible to the original.

A 25% area increase would make a panel able to sound about 50% louder (1.8 dB), partly on account of somewhat narrower dispersion, while consuming 25% more power to get the additional SPL. There would be somewhat more lobing at high frequencies. It would require a bit wider electrode gap to keep the same impedance and the same membrane force uniformity, in turn requiring an increase in polarizing and driving voltages to get the same SPL as before at a given power level. It would also shift the natural resonances to slightly lower frequencies, which might allow a slightly lower crossover frequency. The change in ESL efficiency and drive voltage requirements would require re-engineering of the signal step-up transformer, the woofers, the enclosure, and the woofer integration scheme, along with minor changes to the overvoltage protection circuit. If you get all that right, then people can take the new panel for a spin and see if they like the extra 1.8 dB, or whether they notice the increased lobing. Also, if the slight decrease in crossover frequencies is implemented, they can try to detect this and rate its merits. At any rate, we think our original development efforts converged on a good result.

As you can see, ESL design trade-offs are complicated and interdependent, even when the basic change being considered is simply the size of the panel. Panel size has something to do with maximum loudness, sensitivity, frequency bandwidth, dispersion, diffraction effects, impedance, magnetics interactions, and amplifier loading, but it is not the only factor that affects these characteristics. Other factors include the crossover design, bias voltage, membrane material characteristics, membrane tension, aspect ratio, number, size, and configuration of other transducers in the system, radiation pattern, damping, step-up magnetics, biasing scheme, whether the system is constant-charge or not, and many other things. Also, as a general matter, a wide panel seems to have the potential for creating a "big head effect" that some would say is problematic, whereas a narrow panel would not.

Of course, you never know what someone might mean by "better". At JansZen, we try to make it clear what our speakers do well. We believe their attributes make them most likely of all speakers to let a person experience the music, all of it, nothing more or less. The CAS meeting room, as with most club rooms, with dozens of attendees seated wide and deep in all kinds of positions relative to the apex of an equilateral triangle, created wide variations and asymmetries in audience absorption and wall splash contribution, and of course noise contribution from the air conditioning system that we were lucky to have. Even during the initial portion of the demonstration, however, before switching over to the disc player, with an iPod Nano and its 1 Volt maximum output connected using its 1/8" phone jack through 31 feet of generic cable into a preamp that was turned up all the way, the sound still struck several people as "real".

- Full range electrostat is the ultimate.

ESL woofers were the ultimate at one time, but I would say that the distinction is pretty well gone, now, at least if the right dynamic woofers are applied in the right way. It is now possible to obtain dynamic woofers that sound just as good as ESL woofers when operated well below the frequencies and excursions where they experience cone breakup and other ailments. Once the basic woofer quality problem is put away, the last tricks have to do with mating them to the optimal cabinet design and crossover. If these are not done right, then the bass will not sound as natural as the rest of the spectrum, regardless of the woofer's basic fidelity.

Purists may still have a reason to prefer an ESL woofer, though, especially if properly designed. This is because dipole woofers have an advantage in terms of being more directive at low frequencies than non-dipoles. This excites room modes less and can make the dispersion pattern of the woofers match up better with the other ESL elements, improving low frequency spectral uniformity across a wider area. About cone woofer dipoles created by mounting them in open baffles, in my opinion, it is not possible to integrate them with ESL's, so if dipole woofers are what one wants, then woofers that are electrostatic (or possibly magnetostatic) are the best bet. Who knows, maybe we'll make one someday.

- Your speaker sounds great without being one.

Thanks, David. Technobabble aside, since what we all want is the experience of hearing what was actually recorded, if dynamic woofers can recreate natural sounding bass, then why not use them? They're much smaller and quite a bit less costly. The savings can go into improvements in other areas.

If you were to design a full range speaker to compete with the clx and soundlab, would it be better or just different from your current design (to appeal to those who think it's better)?

I believe the JansZen One, as is, does compete with any full range ESL currently in production, and not just in terms of sound. Try a little straw pole about the JansZen One's appearance among those you know who typically object to speakers on aesthetic gounds, with the grille on, of course.

But if we were to market an ESL woofer, it would reproduce bass more faithfully than any ESL currently on the market, and do it in a smaller, better looking package, with no extraordinary measures. We have the basic technology, already, but are not convinced that it is worth deploying.
 
David,

A few more questions for you:

- Does increasing speaker efficiency improve sound?
- Why do you use 2 crossovers, while ML uses only 1?
- Are crossovers inherently bad, or is this a ESL myth? The coherence of your speaker seems to dispel the myth. Is the problem the crossover or the integration of the slow woofer with the faster panel?
- How did you choose your crossover frequency points? Why 225 vs. 300? Is lower better than higher, or vice versa?
- You added 2 woofers to the speaker. ML summit has 2 woofers also. Other ML models have 1. Why not 3? How much bass is correct or "accurate"?

Thanks
 
David, A few more questions. . .

Efficiency. Clearly, the basic benefit of increased driver efficiency is an increase in SPL for a given electrical power level. Louder can be better in an absolute sense, and high efficiency can let one use a low power amp that has particular merits other than high power.

Measures taken to increase efficiency tend to have trade-offs. I can only give a few simplistic examples, here, and the existence and degree of trade-offs are not absolute in every case. Adding a horn to a driver improves the impedance match to the air load, which increases efficiency, but has the potential for adding some cavity coloration and diffraction effects. Narrowing the magnetic gap in a cone speaker's motor increases efficiency, but alters its characteristics in other ways, where balancing off the side effects can lead to increased distortion or coloration. Narrowing the electrical gap in an ESL increases its efficiency, but demands a set of compromises that ultimately increase distortion. There are lots of other examples.

JansZen One crossover count. The use of two crossovers is the inevitable result of the JansZen One being a three-way system. :) It is a three way system for one of the two usual reasons: to improve dispersion uniformity across the frequency spectrum. Drivers have a dispersion angle that decreases linearly with increasing frequency. When there are drivers of different sizes, with the smaller ones carrying the higher frequencies, then for the portions of the spectrum that each one carries, the dispersions can be made to overlap advantageously. This works out nicely for all driver technologies, because smaller drivers are generally more appropriate for higher frequencies anyway.

The other usual reason for using more than two driver sizes is not important in the case of JansZen or ML or any ESL maker I am aware of, which is to avoid placing a crossover frequency in the 1 kHz to 3 kHz band where the ear is most sensitive to the slightest problems. In 2-way dynamic systems, driver capabilities usually force the crossover frequency into this band, and athough the sonic interference and phase problems can be largely addressed by the use of complex, 24 dB/octave crossovers, such crossovers have their own problems.

Crossover nefariousness. Generally, the topic is very complex, but crossovers are not inherently bad, although there are many factors that can affect the result. It is easy to get a mediocre result, and possible to get a bad result, I could say, even with all the software available for designing them. Unfortunately, as with most complicated things, for best results, preparation, thought and experimentation are required. Cost tradeoffs can also affect performance. Crossovers that operate on the source signal before amplification offer a lot more flexibility than passive crossovers operating on the amplified signal, and allow a much closer approach in general to ideal behavior, but they require multiple amplifiers, one for each driver.

Some insist that crossovers are bad and the best response is created by not having any crossover at all, while placing all the frequencies together onto one transducer, or onto a compound transducer where the sum of the parts acts as one. This no-crossover concept has intuitive appeal, but the underlying concept is fallacious.

The first problem is that there is always a crossover, except in one very specific case. In an ESL, there has to be something to counteract an ESL's naturally rising response with increasing frequency. The natural increase is 6 dB/octave for an ideal point source, and 3 dB/octave for an ideal line source. This occurs because an ESL is not mass loaded, at least not at audio frequencies, so unlike a dynamic speaker, it does not take increasing amounts of energy to move its membrane as the frequency increases. At the same time, as with any transducer, the dispersion angle decreases with frequency. Without mass loading to attenuate an ESL's output, this narrowing of the angle concentrates the same amount of sound into decreasing volumes as the frequency increases.

To attain a flat response, this characteristic must be counteracted, and the means for doing so is for all practical intents a crossover. It operates on the entire audio band at once, rather than short frequency bands, but it works in essentially the same way. The one exception is an infinite plane ESL transducer, which will not require compensation for varying dispersion, because there is no variation -- there will be zero dispersion at all frequencies. Sanders Sound Systems makes an ESL that approximates such a plane wave, at least at mid and high frequencies, and I presume this allows the elimination of any type of crossover or compensation circuitry above where the dynamic woofer crosses out. Since there is practically no dispersion from the ES transducers, they must be aimed directly at a single listener for full spectrum sound, but I must say that IMO the effect at this position is remarkable.

Then there are the two trade-offs:

1) A full spectrum ES transducer of practical size driven by way of a single circuit will have high capacitance, and thus low impedance at high frequencies. Impedances below 1 Ohm at 20 kHz are not unheard of. The capacitance will also add lag to an amplifier's feedback loop, potentially destabilizing those with modest phase margin. This is somewhat relieved by the response compensation circuitry, i.e., the crossover, but not altogether so.

2) If one is not intentionally designing a plane wave transducer, the extreme narrowing of the dispersion at high frequencies from the large area has to be managed. To get dispersion, the transducer must either be curved or broken up into facets that are arranged at a series of angles relative to one another. When the solution is a curved membrane, the overall width leads to lobing at lower frequencies than one would get from a narrow transducer, increasing the likelihood of their audibility, i.e., changes in the high frequency response and phase as one makes small changes in position might be noticeable. It certainly is with pink noise, but music is another animal. The use of facets has a similar effect, where each facet in the aggregate has wider dispersion for a given frequency than the entire width would. Since each of the individual "beams" overlap by a finite amount, this creates a similar set of diffraction and interference effects as the curved arrangement.

It is important to note, however, that the effects of both tradeoffs are not necessarily audible, or when they are, not necesarily objectionable. As usual, let your ear be your guide.

In a dynamic speaker, the mass loading supplied by the cone, coil, bobbin, etc. acts as a continuous crossover. This is in every sense the mechanical equivalent of the electrical circuit needed for ESL's.

On an interesting historical note, there was once an ESL company, Acoustat, proclaiming that crossovers are bad, but still using classical, 2-way crossovers in their speakers. Although they did apply the entire spectrum to their panels, it got there after being split by an ordinary crossover into upper and lower frequency ranges, sent to separate high and low frequency transformers, and then recombined after being stepped up. Electrical recombination is probably better than acoustical, but there is never a need in the first place with multi-way ESL's or ESL hybrids to cross over at a troublesome frequency, since ESL tweeters can produce a very wide spectrum without difficulty. Others may be doing the split transformer thing now, for all I know. Seems like a good idea, especially if one admits it's a crossover, because with a full-range-spectrum-to-all-panels type of ESL, the transformers become easier to make, and the capacitance at the binding posts should be thereby reduced, increasing the impedance at high frequencies and improving the amplifier load.

JansZen One crossover frequency. Already talked about it a bit, but yes, lower is better, down to a point, so to speak. This is because the more spectrum carried by the ESL panels, the better, until one reaches the point where dynamic woofers can do the job just as well, and this also happens to be the point at which floor effects around the speaker become inaudible compared to broader room effects.

Dual woofers. Please refer to my answer to User211's (Justin's) question, near the middle of page 3 of this thread. posted 07-10-2009, 10:31 AM

Bass accuracy. A speaker with a reasonably flat frequency response in a reasonably flat room has at least frequency response accuracy going for it, although it may still be far from accurate. Bass is a part of the spectrum, of course, and ideally should be a part of that flat frequency response. One can then boost or cut bass to suit personal preference. A flat frequency response in the bass, though, can be particularly deceptive, because the accuracy can still be quite poor. Of course, the usual problems are audible, although some will prefer various forms of inaccurate response.

Bass extension. At first, this seems like a question about how deep the bass should go, and the answer is, as deep as you like, but of course there is more to it than that. Some prefer natural (accurate) bass, and others prefer bass that is subject to various distortions or adulterations.

An all time favorite is the extra whump that one can get from a ported enclosure with a low cutoff, which is in addition to the extra extension one gets compared to a sealed enclosure. It is a side effect of poor time domain performance, where the cone motion is poorly controlled near and below resonance, and tends to overshoot.

FWIW, at this time, even when the porting and tuning are done in ways that maximize the time domain performance, we believe that a woofer using a ported enclosure can not be made to integrate convincingly with an ESL.

Another favorite is a bass response that has a broad hump in the vicinity of 100 Hz, which can substitute for deeper bass according to many people's perceptions.

2nd harmonic distortion is pleasing to many people, especially in the bass.

A woofer with 2nd and 3rd harmonic distortion in a certain ratio will create the impression that the fundamental is present at higher SPL than it actually is, i.e., using psychoacoustics to deepen the apparent bass response. There has been at least one iconic speaker that was much smaller than one would expect is necessary, which relied in part on this effect. Teenagers and young adults may be happy with it, but mature listeners who find it convincing will also find it fatiguing.

If one has decided that natural bass is preferable, where smoothness and flatness of response along with a lack of distortion or coloration is the object, then there is still the matter of how deep to go. Our Model One in its standard configuration is flat within 3 dB in-room to about 30 Hz. This is below the lowest note on a bass guitar or double bass, which is E1 at 41 Hz, and matches the lowest note on a 5-string double bass, which is B0 at 31 Hz. Bass is reproduced very convincingly and realistically, although those who are accustomed to boom boom bass may find it lacking.

To radiate the kinds of artificial bass content that some types of music specialize in, such as hip hop, deeper extension and some deep bass boost are needed, but for that, subwoofers are available, although to retain accurate reproduction of musical content, most will have to be modified to cross out at a low enough frequency and at 6 dB/octave. Accuracy is not important in this context, but it is still worth mating the sub properly to avoid humping up the rest of the lower bass spectrum.
 
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