Beyond the Ariel

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Stepping back a few steps and looking at the larger picture, the overall goal of the project is low intrinsic distortion and spacious sound. This was the goal with the Amity, Aurora, and Karna amplifiers, and they were met.

It's not so easy with loudspeakers, which usually have low distortion, lots of headroom, or spacious, wide-open sound. Electrostats have low distortion and spacious, almost distant, sound, but there's no headroom. To my perceptions at least, horns have an up-front sound that is thrilling in one way, but not so good at spatial impression (room size, height, ambience, etc.). As an old-time quadraphonic guy, this is important to me, part of what stereo is all about (not a mono fan at all).

The project as it sits almost splits Magnetar's suggestions down the middle: dipole from 70 Hz to 1.2 kHz, and something like the Radian 745PB (3" diaphragm, 1.4" exit) for the high frequencies. But there's a method to this: if we want constant directivity when transitioning from a 12" driver to a 90-degree horn, we can follow Geddes' example and cross where the diameter of the woofer equals one wavelength. Thus, 1.2 kHz (in practice adjusted for best phase match between LF and HF drivers at the crossover frequency).

I'm still considering a rear horn, as well, on the rear side of the baffle, tucked between the midbass and widerange driver. I don't think it'll need to be identical to the front horn, but made from similar diaphragm materials, at a minimum. If the Radian 745PB is on the front, the Radian 465PB on the back might be a good companion, running at reduced level (-6dB), and with a crossover slope and frequency designed to maintain constant power into a sphere. This can be confirmed by measuring off-axis all the way around the speaker, and more importantly, total power into the room, as measured on a flat surface.

You can see where this is going. Low IM distortion, ample headroom, and constant dipole directivity from 70 Hz to 20 kHz. In the below-70 Hz region, stereo subwoofers are essential for maintaining spatial impression, preferably placed as widely as possible, even at the sides of the listener. It might even be desirable to have four subwoofers, at each corner of the room. This would maximize the left-to-right velocity component, which is responsible for LF spatial impression.
 
ScottG said:

The question I have here is:
Precisely why do you like the sealed over the vented servo? (..for a given freq.)
I have done direct comparison of the same "servo" driver with two different servo feedback circuitries so that they will be flat or either sealed or vented box. I have a toggle switch on these servo boards, which I no longer sell them. My original idea was that people can choose either sealed or vented with a toggle of a switch. Later on I realize that most people already have their mind set when they purchase the kits and adding the toggle switch is just a waste of time. But that method has given me a convenient way to build a vented box tuned to 20hz, and then I can also plug to vent and turn it into aperiodic. In this setup, I can hear how the sealed version (in the case is really an aperiodic) sounded cleaner with a more "coherent" sound vs the vented box which sounded just a bit lacking in definition and less coherent. I have matched the frequency response from 20hz up to 80hz between the two. So frequency response is not an issue. Cone control wise, they have about the same effective driver Qts (which is about 0.15), so cone control is not an issue. The only thing different between the two is one is 2nd order and the other is 4th order and the sealed has completely control of the active radiator vs vented box does not have control on the vent resonance. And phase is another big difference. The sealed version has just a bit less than 90 degrees phase shift at 20hz. The wavelength at 20hz is about 50ft and 90 degrees is 12ft (vs in vented box, it can easily have 180 degrees phase shift) and the group delay is just smaller with the sealed box. My experience has been quite consistent though.
 
rythmikaudio said:

The only thing different between the two is one is 2nd order and the other is 4th order and the sealed has completely control of the active radiator vs vented box does not have control on the vent resonance. And phase is another big difference. The sealed version has just a bit less than 90 degrees phase shift at 20hz. The wavelength at 20hz is about 50ft and 90 degrees is 12ft (vs in vented box, it can easily have 180 degrees phase shift) and the group delay is just smaller with the sealed box. My experience has been quite consistent though.

Brian zeroes in on the key point here. To an old Thiele/Small guy that took a class from Robert Ashley in the Seventies, the whole point of using a vented system was a sharp decrease in excursion in the box-frequency region, and a more compact box for a given value of F3 and efficiency. The downside, and it's a real issue, is more group delay errors from a 4th-order highpass filter, and substantially greater sensitivity to tuning issues (because of the high-order filter).

That class was the first and only place I'd ever seen a physical copy of Richard Small's doctoral thesis - which became the series of AES articles that kicked off the T/S transformation of the industry in 1973. The interesting thing is that Small had a whole chapter on resistive-vent systems that was never published in the AES Journal - they were 3rd-order highpass filters, much to my surprise. At the end of the class, I asked Bob Ashley why he didn't publish that article in the AES Journal, and the answer was unexpected - because the space-efficiency factor (kn) wasn't any better than a conventional closed box, and excursion was similar.

Closed-box systems always suffered from considerably worse distortion (several times) at the bottom of the working range compared to vented systems, which is why T/S theory led to the renaissance of vented systems - they could be designed accurately for the first time, the boom chased out, and most important, had lower distortion. All this with a few nomograms and a HP scientific calculator. That's why closed-box systems, the traditional favorite of the Fifties and Sixties, fell out of favor by the late Seventies.

But with stable, well-designed servo system, the T/S landscape changes once again. Direct sensing allows arbitrary alignments, as well as real-time sensing and compensation of driver nonlinearities. Much of the vented advantage evaporates, since the distortion figures between the two are now comparable, which leaves behind the disadvantages, worsened group delay and issues with vent turbulence and pipe modes.

One thing I never liked about both closed and vented systems was the presence of a response zero around 1 Hz or so; this is caused by box leaks (Ql), and cannot be completely removed, even with really over-the-top sealing-up measures. It's real enough it has to compensated for when building a test box, otherwise the compliance data will be off by a few percent. One minor advantage of a resistive-vent system is that the zero isn't there - actually, it's moved up in frequency, and becomes part of the 3rd-order highpass alignment, so the "leak" works for the design, instead of against it.

I'm hoping the servo system knows how to compensate for the LF zero, and removes it from the alignment. This zero, along with nonlinearities in the spider, is the reason for the nasty in-and-out "breathing" we see in LF drivers.

To see the breathing, all you have to do is apply continuous heavy drive to a mediocre woofer in a vented system, and watch the cone as it gradually creeps forward or back, then cut off the signal and watch the driver spring back to center. I've seen woofers gradually work themselves almost out of the gap if the spider design is bad enough.

A zero in the response, combined with a nonlinearity that is unpredictably excited by it, is obviously undesirable. I'm expecting the servo system senses this condition, particularly the off-center VC, and compensates accordingly.
 
Hi Brian,

You write about 90 degree phase shift at 20Hz for closed box, and 180 degree for vented.
Either of these phase shifts will sound 'off' (grotesquely yukky to me, which is why I so dislike ABRs) when compared to the phase linear drive which might be experienced via a low impedance voltage driven large baffle.

It is not possible to electronically equalise a LF characteristic (even via a servo arrangement) and retain both a faithful amplitude response on transients and a linear phase response on steady sines.

I would rather have a naturally falling bass amplitude response (when bench measured via steady sines) than have different (confusing) first and second cycle bass responses, the first transient response being unmodified by driver-system characteristics or EQ as the increasing amplitude response builds up during *waveform reproduction time*, before being followed by a delayed boost which provides the 'flat' response, but which is out of phase (not coherent) with the start of the waveform.

If the boost cannot arise in phase (coherently) with input then the bass driver output timing becomes shifted; compensation for this phase shift wrt a listening position then affects the transient response (coherency). And yes this is quite clearly audible below 50Hz, which is why increasing driver cone area with reducing frequency cannot fail to sound better.


Cheers ......... Graham.
 
All loudspeakers have at least a 2nd-order highpass characteristic; there's no way around this, no matter what you do. This is what you get with a closed-box monopole (setting aside for now questions about very low-frequency zeroes caused by box leaks).

All other systems, including dipoles, have higher-order highpass-filter characteristics. A dipole has the 2nd-order highpass resulting from the driver's Fs (which is close to, but not quite the same as, the free-air figure), and an 1st-order resulting from the dipole's 1/f rolloff. Net result, a very spread-out 3rd-order highpass.

Resistive-vent monopoles are 3rd-order, if my memory and quick scan of Richard Small's Doctoral thesis was correct. Classical vented systems, servo-ed or not, are 4th-order highpass filters, and I suspect transmission lines are too.

Given that all loudspeakers are intrinsically highpass filters, what can we do to improve step response? High-order highpass filters always exhibit overshoots - it's just a question of "how much" and "how long". Spreading out the filter, either by cascading 1st and 2nd-order characteristic (as with dipoles), is one way to do it - but "undoing" this 1st-order rolloff with corrective EQ gets us right back where we started.

You can also choose a gentler rolloff slope - this is similar to spreading out the highpass filter over a broader frequency range - like a Bessel or Gaussian filter. My first subwoofer at Audionics in 1979 was a 4th-order Bessel alignment, which almost popped out of the "unequalized" portion of a 6th-order vented-equalized Butterworth alignment. No free lunch with the Bessel, though - the distortion is almost as high as a closed box, due to box tuning being quite a bit lower than classical B4 alignments.

My own guess is that when we hear "crisp" or "fast" bass, it's not so much transient-perfect as low distortion, combined with freedom from dynamic mistuning artifacts. That was one of the subtle benefits of the vented-Bessel alignment; it was much less sensitive to dynamic Q and Fs variations than the "standard" vented alignments, and that offset steady-state distortion figures that weren't quite as good.
 
Hello Lynn,

I know this is a little off-topic, but it's about the subs. You mentioned at the begining of this thread Garry's Pimm subwoofer design. Is this an option to consider anymore?

The specs where quite amsaing if I remember, ruler flat from 20Hz up. I don't know though what the design was (heavy stuffed U-baffle?) so if there are some references on the web, maybe you could point them out...

The design so far looks great. I followed Magnetar's opinions about ultimate dynamics, and I wonder if a double weaveguide on the widerange driver wouldn't enhance this a little (a 120 degrees WG means around 3-4dB I guess), leaving it a dipole in the same time.
Besides that, with a proper weaveguide shape (maybe EnABL treated), I believe edge difractions would be eliminated. Another plus could be less interactions with the room's walls, without having that focused, projector-like image you dislike horns for.

I hope this makes any sense :)
 
sensing the speaker

Hi


Brian, having not played around with speaker feedback amps yet, I wonder why you have chosen a sense coil wound over the voice coil instead of using a separate coil in a separate magnet structure as outlined in your page ( " Even worse, some authors have even suggested using one coil in a dual voice coil driver as the sensing coil. " ) ? Aren't there some speakers that are especially designed for this ?

From any technical point of view to me it seems to be the worst place to get uncontaminated information about the speed of the VC / diaphragm as the magnet field is heavily affected by the current of the VC ? Furthermore you also should copy all the temperature memory effects right to the sense coil unless you don't use constantan or compensate electronically.

Are there any special compensations effects I am not aware of that make the VC as sensing place a preferable decision ?

Greetings
Michael
 
I wonder if a double weaveguide on the widerange driver wouldn't enhance this a little (a 120 degrees WG means around 3-4dB I guess), leaving it a dipole in the same time.

That exact thought has crossed my mind as well. I actually planned to play with this using 3 mm MDF horn side walls of different length and shape, some time far, far, into the future. Here with experimentation one could choose the double horn parameters in such a way that the loading picks up where the dipole rolloff starts (going down in frequency). Another plus is increased separation distance without cavity resonance, due to the horn profile vs. parallel U-walls. Disadvantage is size and the necessary experimentation.
 
Re: sensing the speaker

mige0 said:
Hi

Brian, having not played around with speaker feedback amps yet, I wonder why you have chosen a sense coil wound over the voice coil instead of using a separate coil in a separate magnet structure as outlined in your page ( " Even worse, some authors have even suggested using one coil in a dual voice coil driver as the sensing coil. " ) ? Aren't there some speakers that are especially designed for this ?

From any technical point of view to me it seems to be the worst place to get uncontaminated information about the speed of the VC / diaphragm as the magnet field is heavily affected by the current of the VC ? Furthermore you also should copy all the temperature memory effects right to the sense coil unless you don't use constantan or compensate electronically.

Are there any special compensations effects I am not aware of that make the VC as sensing place a preferable decision ?

Greetings
Michael

First of all, our patent is not on how the sensing coil should be implemented, rather it is on how the servo feedback and current feedback are used. It is possible to get a sensing coil in a separate gap. However, that introduces an potential failure mechanism. The good thing about sensing coil wound on the same former as the driver coil (and co-center) is so called "self alignement". It is controlling where the force is applied. This is biggest advantage over any other configuration. At high frequency, when there is any distance between the voice coil and sensor, it is going to introduce extra phase shift (because the cone movement can no longer be view as piston motion, any cone break up and such and be a big problem) and it will get worse over time.

The sensing feedback has a input impedance of 1Meg to 51k ohms (frequency dependent and DC coupled, yes our sensing feedback is DC coupled) and the sensing coil resistance is 24ohms. The error introduced by the temperature is -66db (or 1/2000). It is almost neligible.
 
Graham Maynard said:
Hi Brian,

You write about 90 degree phase shift at 20Hz for closed box, and 180 degree for vented.
Either of these phase shifts will sound 'off' (grotesquely yukky to me, which is why I so dislike ABRs) when compared to the phase linear drive which might be experienced via a low impedance voltage driven large baffle.

It is not possible to electronically equalise a LF characteristic (even via a servo arrangement) and retain both a faithful amplitude response on transients and a linear phase response on steady sines.


One thing I would like to clarify is that there is no "linear phase" in HP filters. So all the discussion of group delay is just a notion. It is pretty much determined by the conjugate roots of the transfer function and it is always additive. So eventually, two things help. The lower the order, or the lower the Q, the better. One big Q is all it will take to make it worse and there is no way to get it back. I have seen Ilkka's measurement of REL sub on the other forum and it looked like a first order roll-off starting at 60hz. What a sacrifice :D Our approach is Q=0.5 which is same as two first order group delay added together. That is pretty much the best one can do (and still has -6db at 10hz).

The group delay is merely an indication of how the source will move when the signal gliding on the frequency axis. It is about coherence. It is about how the ambience information will be "correlated" to the source and give us the connection of the sound stage. The absolute phase is really not that important.
 
Brian, thanks again for your comments to this thread. This has been extremely illuminating for me, and answered a lot questions I've always had about servo-feedback systems for loudspeakers.

Readers who've read my comments about feedback on the Nutshell pages might wonder about my enthusiasm for Brian's work - well, my issues with feedback topologies are generic transistor amps with low slew rates, inadequate phase margin with reactive loads, a forward path with high-order nonlinear transfer functions, and slow thermal (feedback) tracking for the Class AB transition on the output devices. This collection of disorders covers most transistor amps, and the problems are not visible with steady-state THD testing. (Note these are almost all problems in the midrange and higher frequencies.)

Servo-feedback subwoofer amps are another animal entirely - the main issue is system stability, and the care and feeding of the voice coil in the woofer. The comment I saw earlier about gradually removing the servo feedback above 100 Hz makes a lot of sense.

Mid and high-frequency servo feedback in loudspeakers has always been a nearly intractable problem, thanks to the "sensing problem" of where to detect the feedback for a diaphragm that is moving differently on different parts of the cone. At least for a low-bass woofer, the voice-coil former has a close relationship with the cone - get much higher in frequency, and parts of the cone start to decouple from the voice coil.

---------

Hmm ... thinking about the waveguide/WR driver comments made earlier. I have a utilitarian view of horns - I'm not really a horn fan, and haven't really warmed to their sound that much. I appreciate the efficiency and low distortion, but don't like the HOM, horn-edge diffraction, and other time/dispersion maladies - and those are very difficult challenges that a lot of smart people have been working on for a long time.

The HF horn is there - well, because aside from the exotic prosound ribbons, there is no alternative if you want 97+ dB/metre efficiency and low IM distortion. I don't see a vertical line array of dome tweeters as a realistic alternative, with their intrinsic problems of non-synchronous arrival times at the listening position.

But a horn for frequencies lower than the HF horn is an esthetic choice, not a technical one. Below 1~2 kHz, arrays of direct radiators can do the same thing, but with somewhat different tradeoffs. The midbass horn will have (much) lower IM distortion, but at the expense of (inevitable) HOM modes and associated transient disorders. The cone array will have better impulse response and broader dispersion, but not as clean IM distortion. You pays your money and you takes your choice.
 
Hi Brian,

I was not refering to phase linearity in high-pass filters, but to the EQ boost at sub-bass.

Where the phase shift of any EQ boost is adjustable, and can be shifted from lagging to leading say through +/-30 degrees wrt to phase linearity, then there is only one setting where reproduction suddenly clicks together. Either side of the correct setting and it starts to sound like simple sub-bass boost applied to a falling driver response which might seem optimum for any system 'as is', yet not actually be correct when compared to a phase linear arrangement; unless it fortuitously hits the sweet spot.

In my experience absolute phase is most important, but I am not going to argue about it.

Cheers ...... Graham.
 
Graham Maynard said:
Where the phase shift of any EQ boost is adjustable, and can be shifted from lagging to leading say through +/-30 degrees wrt to phase linearity, then there is only one setting where reproduction suddenly clicks together.

My argument is there is no linear phase for HP filter so that reference does not even exist. Am I missing something here? A simple example of phase linear is when you double the frequency, the phase shift also doubled. That just cannot happen with HP filter.
 
Hi Brian,

I repeat; I am not writing about the phase linearity in a HP filter, but about sub-boost EQ.

Maybe you are arguing about the method of using a conventional HP filter as way of generating sub boost via a separate channel, which of course cannot be phase linear.

Though of course no matter what sub-bass arrangement is used, there will always be a amplitude/phase compromise through the crossover range.

Cheers ....... Graham.
 
SunRa said:
Lynn,

I've mentioned earlier, in the weaveguide post, but maybe you missed it; I hope I'm not annoying though...

Do you consider any more Gary Pimm's subwoofer design and if yes, where can we find some references about it?

Gary's pulled his audio page off the Web - he's taking a vacation from hi-fi, but I still chat with him on the phone every now and then.

The Gary Pimm subwoofer, if memory serves, was a straightforward W-baffle with Parts Express 15" woofers (2 per box) selected for a Q of 0.6 or a bit higher. The whole thing was separately amplified, of course, and a heavily modified DCX was used for EQ to the room and the dipole rolloff. Gary found the stock op-amps and electrolytic caps were so bad that simply connecting the DCX in parallel with the main system (and the subs turned off) quite audibly degraded the sound. The culprit turned out to be the electrolytic caps, which were "contaminating" the main signal path, even though they were simply connected in parallel. A complete replacement of the audio board with high-quality 1:1 line transformers cured that problem, with the transformers replacing ALL of the opamps and coupling caps, both input and output.

He changed the system from a standard dipole to more of a cardioid by extending the back of the box about a foot and filling the open rear space with recycled cotton filling, the same that is used for non-irritating building insulation. He auditioned it with no filling - thus more of a dipole - and filling, producing more of cardioid pattern - and preferred the cardioid. Both variations equalized flat to 16 Hz with the DCX, of course.

I should add Gary's listening room is fairly small, and we only listened about 6 feet away from the speakers (and the W-baffle subwoofers). The preference for dipole vs cardioid, or no filling vs filling, might easily change in a bigger room.
 
Yesterday I was in Mr. Manger house and we listened bipolar Mangers. He opened the balcony doors and we was walking around with birds singing everywhere. The stereo image was like a hologram - the same from any direction. What's more normal transducers in comparison with MSW have a noise which is annoying and you can easily detect it when clapping your hands. Everything is fine except the Science that cannot explain why two speakers measured similarly (JXR6 and MSW) sounds so different (a good speaker must measure well anyway). There is a new AES article where Schoeps reveal microphone ringing by using MSW as one polarity stimulus (which is required for the task).
 
The Story Continues ...

This is a Quad-12 array, more attractive looking than the previous drawing, and it offers the possibility of one 18Sound 12NDA520 or 12ND710 (the latter available in 8 or 16 ohms) for the widerange driver, one ToneTubby 12" Alnico in 16 ohms, and two ToneTubby 12" Ceramics in 16 ohms.

In terms of connections, the HF and 12" WR share a high-quality amplifier, the 12" Midbass and Bass share an amplifier (possibly a 35~60 watt tube amp), and the subs are independently powered.

If the 16-ohm ToneTubby's are chosen for the MB and B drivers, the Bass drivers can be wired in parallel, low-pass filtered by a single series inductor, and the 2x12 Bass array (with filter) wired in parallel with the single 16-ohm MB ToneTubby. At the lowest frequencies the combined impedance is about 4.5 ohms, the efficiency is about 103 dB/metre, Fs is 78 Hz, and Qts is 0.85. For the cost-conscious, no, it's not a cheap speaker, but the ceramic ToneTubby's are half the cost of the Alnico version - and TT themselves recommend mixing-n-matching the two flavors.

Downsides? I wouldn't expect a lot of linear Xmax from guitar speakers - although I can say from personal experience a single widerange ToneTubby plays really loud, and with none of the usual cone-breakup crud. A 3x12 array of these things would probably put you on stage with the musicians.

P.S. I received an e-mail from Lowther America recently, and they mentioned that there's a version of the Lowther with no whizzer cone - an obvious candidate for a ribbon tweeter to take over at 8 kHz. For those allergic to horns, the Lowther+ribbon would certainly be an alternative in the 1 kHz-on-up range. Now THAT would be an offbeat speaker - a trio or quartet of ToneTubby guitar speakers for the bass, MB, and low-midrange, and a whizzerless Lowther above that.
 

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