Beyond the Ariel

Bjorn

I quite understand your position and I do not blame you for how other use your data.

I would say the simulations look reasonable given the pure sine nature. Most polars are shown 1/3 octave which washes away a lot of problems (most of them actually). I use critical band smoothing which is about 1/20th octave above 1 kHz. This is still going to smooth out things that you will see with a simulation at a fixed frequency. All in all I would say that your sims - when properly considered - show precisely what the issues are. When taken out of context they can show whatever the poster wants them to show. Its important that people see both sides.

By the way I tried your website but I could not load it. Do you show polar maps for the OSWG. Those are what I know best and could tell if the sims reflect reality or not.

I would have no issue with giving you the dimension of one of my waveguides. I don't remember you asking. I have a ton of very accurate data on them that could be used to validate a model.

I plan to take my measurements up one more notch to better quantify the performance. My hope is that with the next installment I can begin to sort out the HOMs. It's clear that something like that is occurring in your sims as the angular features are not at all constant - a key feature of HOMs. The question that is very hard to sort out is what are caused by the mouth and what are from down into the device. No far field measurement could ever sort that out.

Are you aware that I can reconstruct the velocity at the mouth by a reverse (ala holographic) technique from measured data. It works down to about a 1/2 wavelength now because the boundary conditions are not precise. My plan is to use a large flat baffle which should allow for about a 1/4 wavelength resolution and then possible a reconstruction of the waves inside the device from that. A lot of work but the only way to get to the nitty-gritty of these things.

With compression drivers and direct dome or other direct drivers, in terms of a linear coherent in phase delivery, how close tothis can we achieve so that the horn profile can retain an expanding laminar expanding flow i.e without turbulence as it flow through the horn. Are there always eddies that form andeventually destroy this and produce HOM and other spurious effect. If so are we still seriously limited by the driver delivery even for a perfect shape of horn.

Are direct drivers worse than a CD where the phase plug does more to deliver the right entry and passage through horn or are they inherently more likely to start turbulence with other semi laminar but changing state.
 
Hi Rob

I am not sure what you mean by a "pressure map".

The thing is that I am disinclined to post in this thread since it is clearly an attack where "facts" don't seem to be important. I'd just as soon not get too involved.

Earl,
From where I sit, OS waveguides are a way forward for most applications.
However, if someone has north of $3K invested in custom designed amplifiers with a Damping factor of 4 or less, an OS wave guide with a (relatively)mediocre VSWR may not be the correct engineering solution.
My understanding is that a low VSWR will interact with a Low DF to produce a reverb effect that may be euphonic, but not accurate.
To me, this falls under "horses for courses".
Its all about the design requirements, and the priorities of the design.

Just my 2 cents.

Doug
 
Lynn, thanks for your answer to my comment about the fire bottle amplifier preference you have. I must say that I grew up with many tube amplifiers and it wasn't very hard to get away from some of those after many years. I have had more than one model of Mac amplifier and have also used some of the Cary amplifiers and I must say that those particular models all have harmonic signatures that had much to do with their design goals. So I have not tried to pursue any tube equipment after that. I have seem to many high dollar tube amplifiers at CES shows that just don't cut it as far as I am concerned, but you are in a different class of designer. I imagine that most of what I have heard are in fact Williamson type of tube amps. Now don't even get me going on some of the single ended low power amplifiers, those are to me nothing but tone controls.

The Mac's, although legendary in some circles, have serious design issues. The cathode feedback is an excellent way of linearizing pentodes (which they badly need, since they are so nonlinear), but the dirty little secret is that Mac's operate in nearly Class B, and rely very heavily on multiple feedback loops to achieve acceptable low-level linearity. To me, they are very grainy and congested-sounding, a reflection of not-very-good low-level linearity.

Cary amps are optimized for high distortion, and tend to have very weak driver stages. This results in a fat, heavy, slow sound, similar to the jukeboxes of the Fifties, which used low-cost, primitive circuits. Some people, usually too young to remember the sound of the tube era, like this sound. I don't.

The best Fifties-era vintage amps were the Brit models (Quad, Leak, Radford), the Scott LK-150, and some of the Fishers. The Marantz Model 9, although legendary, is not really better than the Brits, just more powerful.

What's more shameful, or embarrassing, depending on how you look at it, are the American "high-end" tube amps of the Seventies and Eighties. By and large, these are warmed-over Fifties designs, with bad-sounding and instability-prone solid-state B+ series regulation. The worst of both worlds: solid-state failure rates, the necessity for frequent output tube replacement due to over-running the tubes beyond their ratings (more POWAH!), and worse sound than the originals they were copied from. I was startled to find out the single best-selling model of Audio Research was the monstrous Classic 600 ... even in the tube world, more POWAH sells big.

Many of the high-dollar high-end tube amps have egregious design errors, and it's hard to tell if they are intentional or not. For example, a high-powered design might have all four or six power pentodes share a common cathode resistor, or share a common bias point if the circuit uses fixed bias. If one of the power pentodes is just a little more efficient than the others, it will run away and current-hog, with the plates glowing red. Imminent failure is only seconds away; the user doesn't usually catch this until too late.

Now you've got a set of three or five power pentodes in a circuit where all of them must match or it will fail. Guess what? The amp vendor will be happy to sell you another factory-matched set of four or six output tubes. The vendor will tell you it was your fault one of the tubes ran away and cooked itself, or the power supply regulator blew a transistor and wiped out the output stage.

I frankly think this is grossly unethical behavior. Another pet peeve is correcting a design fault, typically with two or three resistors, and making that the "new, improved Mark 3.2 model", and charging a fat fee for "updating" the existing products.

There are well-engineered PP-pentode amps out there, but I don't stay on top of that market. I've heard some pretty decent examples, with sonics not far from the DHT ideal ... but there's almost no correlation between price and quality at all. Things I avoid: driver circuits with less than 8 mA of standing current, circuit boards, which make no sense for simple tube circuits (and create unwanted stray capacitance), and solid-state rectifiers, which have more hard-to-filter HF switch-noise than tube rectifiers.
 
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The only way that this is possible is for the room to be completely dead - an anechoic chamber.
I've been in a couple of homes, of those who were producing equipment, with listening spaces that were made very dead. Very unpleasant places to be in, definitely not for enjoyment of music ...
Rock concerts don't have any venue acoustic since the feed is generally taken directly from the performers and not the room. Played loud enough, the local acoustic kind of goes away and the playback realism becomes quite convincing. Its gone at lower levels, probably because the expectation is for loud.
I would beg to differ, the ones I speak of certainly have some microphone feed from the audience, you can hear the various interjections and murmurings, whatever it is that one tunes into when you 'hear a crowd', it's certainly there. I'm thinking here of one in particular, Jimmy Hendrix Experience, a well done live capture. The level doesn't have to be high, once the system works well enough one's ear can still pick up the little subtleties ...
 
With compression drivers and direct dome or other direct drivers, in terms of a linear coherent in phase delivery, how close tothis can we achieve so that the horn profile can retain an expanding laminar expanding flow i.e without turbulence as it flow through the horn. Are there always eddies that form andeventually destroy this and produce HOM and other spurious effect. If so are we still seriously limited by the driver delivery even for a perfect shape of horn.

Are direct drivers worse than a CD where the phase plug does more to deliver the right entry and passage through horn or are they inherently more likely to start turbulence with other semi laminar but changing state.

Eddies are a static flow situation not an acoustic one. They cannot happen in acoustics. We can get turbulence which is reflected as a higher wave resistance and loss, but not eddies.

How coherent the wavefront at the exit of a compression driver is, is an unknown. We just don't know. Don Keele alluded to some results that he had which indicated that they aren't very good. We may be limited, maybe not. From the data that I see its not a serious issue below 10 kHz.

A direct radiator is not designed to have a planar wavefront so t will never be as good as a compression driver in this regard. That is unless the driver is flat, which is exactly the driver that I used when I built the first OSWG. It acted ideal.
 
Art, you can find captioned figures, details and comments on my web site: Bem Simulations.
Note that these are all simulations, without driver or crossover, intended to show what the horn does. The frequency response will of course depend on the driver, but the directivity will not change much.

Regards,
Bjørn
Bjørn,

Do you have any actual measured horn/driver examples to compare what the differences in directivity are between your simulations and real world response?

Art
 
Earl,
From where I sit, OS waveguides are a way forward for most applications.
However, if someone has north of $3K invested in custom designed amplifiers with a Damping factor of 4 or less, an OS wave guide with a (relatively)mediocre VSWR may not be the correct engineering solution.
My understanding is that a low VSWR will interact with a Low DF to produce a reverb effect that may be euphonic, but not accurate.

Doug

I don't see "damping factor" entering into the problem. If the system is EQ'd flat (with the amps high output impedance taken into account) then there is no "reverb effect". (That or the entire concept of system transfer functions is wrong.) However, this might occur if, for example, the crossover was designed with an assumed zero feed impedance and a tube amp is used. Then the entire system function will have changed and it is no longer correct. I always warn people about this, usually to no avail.
 
I don't see "damping factor" entering into the problem. If the system is EQ'd flat (with the amps high output impedance taken into account) then there is no "reverb effect". (That or the entire concept of system transfer functions is wrong.) However, this might occur if, for example, the crossover was designed with an assumed zero feed impedance and a tube amp is used. Then the entire system function will have changed and it is no longer correct. I always warn people about this, usually to no avail.

Agree 100%. The Ariel and the new speaker are designed for amplifiers with output impedances in the 1~2 ohm range. As mentioned in many previous posts, these loudspeakers are not intended for owners of Class AB transistor amps with 20 dB or more of feedback. Those amplifiers are much more powerful, which widens the choice of loudspeakers to include models with very low efficiency (electrostats, magnetic-planars, MBL, etc.).
 
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Lynn,
Thanks for that synopsis of vacuum tube amplifiers. From what you are saying I guess I wasn't wrong in moving away from all of the nonsense in this area. Now I imagine that you must produce your own amplifier given this horrid situation with so called audiophile tube amplifiers. I do remember the old Fisher tube amps but the ones I remember were in large consoles and paired with JBL speakers.
 
In those days audio dealers packaged systems into consoles, Barzilay (?) for instance, with what the buyer selected. There were many choices. I have HiFi handbooks from the late fifties showing just how many brands were vying for the action. The downgrading of the console format came a little later, when it became a vehicle for moving product under a single brand.
 
Lynn,
Thanks for that synopsis of vacuum tube amplifiers. From what you are saying I guess I wasn't wrong in moving away from all of the nonsense in this area. Now I imagine that you must produce your own amplifier given this horrid situation with so called audiophile tube amplifiers. I do remember the old Fisher tube amps but the ones I remember were in large consoles and paired with JBL speakers.

When I finished the Ariels in 1993, I listened at first on my trusty Audionics CC-2, which is still one of the better transistor amps out there. I also know what's inside the amplifier, since Bob Sickler, the designer, walked me through the circuit as he designed it.

Since the Ariel was, after all, designed for DHT-triode amplifiers, I listened to a whole bunch, as well as antiques like the Dyna Stereo 70. To my dismay, the amps all sounded different, and this applied to the transistor amps as well. None sounded alike.

This wasn't true on my previous speaker, the LO-2, which was my last project at Audionics in 1979. The LO-2 was a linear-phase minimonitor-style speaker, but only had an efficiency of 86 dB/meter. The Ariel is a genuine 92 dB/meter.

Well, that made a difference. A big difference. More than I expected. Now I could hear easily hear differences between amplifiers, despite the Ariel being a pretty benign load (conjugate network in the crossover, transmission-line LF loading, etc.).

You have to remember that I was a "speaker guy" since 1975, when Audionics first drafted me into doing it. Until 1993, I was sure that all competently designed amps, particularly solid-state amps, all sounded the same. Some just more power than others, but competent amps should sound alike.

Uh, not so. Kind of embarrassing, really. I was wrong. (Again.)

It's tedious enough designing speakers, now I had to delve into amplifiers and find out what the heck they were doing wrong. If they all sound different, and it's not a simple matter of damping factor (and it isn't), then they are all wrong until proven otherwise.

Most of the SETs were horrible, grossly colored. But a handful were really good. Why? Again, this isn't euphonic distortion. Not when I hear more space, a more realistic perspective, more natural instrumental timbres, better dynamics, and much better resolution than transistor amps (including the CC-2). Just adding gobs of 2nd-harmonic distortion, and a high output Z, won't do that.

Huh. My conclusion, after lots of chats and some pretty exotic measurements with my Tek friends, came down to the devices themselves. They were just more linear ... and importantly, had linear input capacitances. The latter is kind of a big deal; bipolar transistors, JFETs, and most of all, MOSFETs, are known for grossly nonlinear capacitances. You have to resort to extremely fast cascode circuits to really get rid of the nonlinear-capacitance problem, and then stability problems can arise. Back then, the Tek analog engineers were probably the world experts on ultra high-speed analog circuits, so I got a lot of good information on the problem, as well as arcane stability issues.

So I took the brute-force approach of finding the most linear devices, and the forms of loading that would provide the best dynamic range and lowest distortion in the critical 100 Hz to 10 kHz range. In a way, the same device-centric, bottom-up approach I use for loudspeaker design.

I can't claim credit for taking this approach; the "secret sauce" of Tek's most famous instruments was a special lab, Building 50, devoted to R&D of device physics. Tek knew more about the performance of the devices they bought than the vendors themselves.
 
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"A" microphone feed, the rest are direct.
Yes, a single microphone feed most likely - but it still picks up enough to 'fill' the mix - including bounce from the instruments. Using my 'mediocre' PC system I just tried John Mayer, live "Where the Light Is" and this certainly is full of ambient clues, wound this up to the limits of the unambitious internal amps - and from down the hallway it sounds like the spill from an open door into a stadium. Walk into the actual room, and there is a pressure wave of sound, a visceral, living presence to it. Moving around the room is like swimming through the sound, it feels as if it's soaking into your body - this is 'you are there' for me ...

To demonstrate that this is all part of of a competent working system, next disk was Naxos, Shostakovich String Quartets - didn't touch the volume control one iota. Lovely string tone, lots of life and smooth vibrato; the sheen, and bite of real instruments was there, with a satisfying acoustic surrounding the players.
 
To the dismay of some readers, I am not interested in constant directivity. There. I said it. Really, I mean it.
I am not dismayed at all and think that's fine, to give context to what I want to say here...

It's not hard to know where the first reflections are coming from. The first will be the floor bounce, in the 1~2.2 mSec interval, depending on driver height from the floor, and to a lesser extent, listening distance. Depending on ceiling height, the next will be the ceiling reflection. The rear and nearest-side walls will come next, followed by a double reflection from rear&floor and nearest-side&floor. There are many more reflections after that.

Most of these reflections are not within the 90-degree cone of a loudspeaker optimized for constant directivity. That point bears repeating. The first rear wall reflection is not within the 90-degree cone. The nearest-side wall reflection is also not within the 90-degree cone. The double-bounce reflections are not within the 90-degree cone. For that matter, the great majority of room reflections do not come from the 90-degree cone.
Hold on now, the shortest path to the nearest surfaces isn't inside that angle, but the first reflections to the listener certainly can be if the room isn't too small, AND a good 90° CD design should really hold up decently for another 10-20° (110-130°), down to wherever it's holding a pattern, and below that you should be looking at mostly treble losses, not anything violent.

Your post went on to discuss further the "hard edge" of CD designs and how your favored horns are still smoothly audible from the side. Have you heard Geddes' speakers or something similar? It really seems like you're describing almost the opposite of how they compare to speakers with patterns that are less smooth and more narrowing off-axis. Like I said, I don't care if you don't care about constant directivity, I just found this whole post confusing enough to drag up from quite a few pages back.
 
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You forgot "looking at performance data", but I think that you would be surprised at how many do this. The ability to personally evaluate a loudspeaker pushes its sales price way up - probably double or more. And this price increase comes with no improvement in performance.
What big spenders often do is allow dealers to setup a system in their listening room, they listen and tell dealers what they don't like, and the dealer just change stuff till it is accepted. At least this is what they do in Taiwan.
 
The AH425 ...on axis response at 8000 is around +11dB, around twice as loud (to our ears) as at 500 Hz. Over a fairly narrow beamwidth, (less than the width of a couch at a 2-3 meter distance) the sound character will change from rather dull to overly bright.
Art


Hello,

You are forgetting that the simulation of the AH424 is based on a theorical source having a constant velocity.

Real measuremenst made on axisymetrical Le CLéac'h horns, never show an increasing slope of the on-axis frequency response curve.

See as an example a real measurement :

Horns

And no JMLC horn sounds overly bright !

Best regards from Paris, France

Jean-Michel Le Cléac'h
 
Eddies are a static flow situation not an acoustic one. They cannot happen in acoustics. We can get turbulence which is reflected as a higher wave resistance and loss, but not eddies.

How coherent the wavefront at the exit of a compression driver is, is an unknown. We just don't know. Don Keele alluded to some results that he had which indicated that they aren't very good. We may be limited, maybe not. From the data that I see its not a serious issue below 10 kHz.

A direct radiator is not designed to have a planar wavefront so t will never be as good as a compression driver in this regard. That is unless the driver is flat, which is exactly the driver that I used when I built the first OSWG. It acted ideal.

I should have edited last night but went to bed. I did use the term eddies with some reservation. Yes, a stiff flat diaphragm on a direct driver and OSWG or Lecleac'h may be as good as it gets if done right.

The compression and phase plug are a tricky compromise for domestic top quality HiFi. It is really like processing the sound like sausage meat. But some of us like sausages. We like the horn sound so we go on, and the faults are not really a problem up to peak normal listening levels. And they can be used to direct the sound where you want it and to reduce room issues.

The fundamental delivery is clearly dependent first on what the air mover i.e diaphragm assembly is doing. And modern high speed photography has provided a solution for driver manufacturers to make genuine high performance diaphragms, and corrsponding linear motors that can excite the diaphragm without any hysteresis or delay in the aural frequencies domain.

Almost all drivers we know, fail to some degree as they mostly are not genuinely a technical peak of precision engineering part. Such motor diaphragm assemblies would cost 4 figure sums to make. But this standard of driver would allow for very focussed study of horn throat dynamics and the transition through this into and emitting into free space. Clearly Bjorn and yourself along with others are actively studying horn technology taking this further. The TAD4001 and TAD2001 were in some degree helping the progress but a guarded perhaps none it seems of the older CD.s

Are there good papers that show all the unwanted deviations occuring with real world horns.

We see good observations of the effects of these deviations from ideal, from using graphical 2D and 3D pictorial that can now show an image of the complex issuance of the complex sound wave forms from the diaphragm to the horn exit, going beyond any influence from the horn at least. More developments are needed here to take the horn as far as it can go with the current most modern materials and construction.

Now in the everyday world, where do we get to in the horn versus direct driver debate. It is technically very important for the elite audio guys to know, and we would like a better understanding now. We have waited too long.

The middle ground where you use a wave guide with a super efficient motor gives enough gain to be able to avoid a compression device which in absolute terms is a compromised device. An electrostatic speaker has almost none of these problems and they started out in the 50.s. They do not give a point source for closer listening. That is in mt view, their biggest technical flaw.

This thread is about DHT and other low powered amplifiers requiring a DIYable top end quality speaker. A direct driver replacement for Ariel would be a lot easier but the challenge of a horn remains exciting and could result in a classic like the Ariel has been for many.
 
When I finished the Ariels in 1993, I listened at first on my trusty Audionics CC-2, which is still one of the better transistor amps out there. I also know what's inside the amplifier, since Bob Sickler, the designer, walked me through the circuit as he designed it.

Since the Ariel was, after all, designed for DHT-triode amplifiers, I listened to a whole bunch, as well as antiques like the Dyna Stereo 70. To my dismay, the amps all sounded different, and this applied to the transistor amps as well. None sounded alike.

This wasn't true on my previous speaker, the LO-2, which was my last project at Audionics in 1979. The LO-2 was a linear-phase minimonitor-style speaker, but only had an efficiency of 86 dB/meter. The Ariel is a genuine 92 dB/meter.

Well, that made a difference. A big difference. More than I expected. Now I could hear easily hear differences between amplifiers, despite the Ariel being a pretty benign load (conjugate network in the crossover, transmission-line LF loading, etc.).

You have to remember that I was a "speaker guy" since 1975, when Audionics first drafted me into doing it. Until 1993, I was sure that all competently designed amps, particularly solid-state amps, all sounded the same. Some just more power than others, but competent amps should sound alike.

Uh, not so. Kind of embarrassing, really. I was wrong. (Again.)

It's tedious enough designing speakers, now I had to delve into amplifiers and find out what the heck they were doing wrong. If they all sound different, and it's not a simple matter of damping factor (and it isn't), then they are all wrong until proven otherwise.

Most of the SETs were horrible, grossly colored. But a handful were really good. Why? Again, this isn't euphonic distortion. Not when I hear more space, a more realistic perspective, more natural instrumental timbres, better dynamics, and much better resolution than transistor amps (including the CC-2). Just adding gobs of 2nd-harmonic distortion, and a high output Z, won't do that.

Huh. My conclusion, after lots of chats and some pretty exotic measurements with my Tek friends, came down to the devices themselves. They were just more linear ... and importantly, had linear input capacitances. The latter is kind of a big deal; bipolar transistors, JFETs, and most of all, MOSFETs, are known for grossly nonlinear capacitances. You have to resort to extremely fast cascode circuits to really get rid of the nonlinear-capacitance problem, and then stability problems can arise. Back then, the Tek analog engineers were probably the world experts on ultra high-speed analog circuits, so I got a lot of good information on the problem, as well as arcane stability issues.

So I took the brute-force approach of finding the most linear devices, and the forms of loading that would provide the best dynamic range and lowest distortion in the critical 100 Hz to 10 kHz range. In a way, the same device-centric, bottom-up approach I use for loudspeaker design.

I can't claim credit for taking this approach; the "secret sauce" of Tek's most famous instruments was a special lab, Building 50, devoted to R&D of device physics. Tek knew more about the performance of the devices they bought than the vendors themselves.

Very interesting Lynn. I enjoy hearing the historical eureka s that we all like to have.

With todays minimalist approach one can easily put up the simplest DHT versus discrete solid state and perhaps semi discrete with all the best components. If I had to put money on it I would say the DHT would win.
Has anyone done this out there.