Aleksandar Radisavljevic on RAAL speaker

[mige0]

Thread opened to provide a platform about RAAL speakers - in continuation on the postings to be found here:

http://www.diyaudio.com/forums/multi-way/161299-directivity-dipole-tweeters-2.html#post2416656

No, I don't. Particularly since they'll be even worse vertically and, at $800+ per driver, using them in a line array is really expensive. But lots of folks seem to be happy with designs which neglect directivity.

Hi guys,

a friend told me about this directivity discussion, so I thought I drop in a few facts for your understanding of what influences the directivity. I hope you won't mind as much as I did your conclusion. 🙄

First, in horizontal plane, it's not the width of the baffle that determines it. Its the width in the diaphragm, which in this case is just 15mm, which gives us about 120 deg on 20k.

Second, in vertical plane, even though the diaphragm is a lot longer than the wavelength of 20k, the use of acoustical lens, in form of those two foam pads on the face plate, gives us about 20 deg of vertical coverage without any loss highs compared to mids and no loss of flat tone balance.

With proper baffling, the directivity pattern is almost perfect, rounded, figure-of-eight pattern in a horizontal plane, as the ribbon sprays wider that the open baffle arrangement permits, so the baffle rules that plane, governing the figure-of-eight pattern.
In vertical plane, the foam pads rule the situation, and that behaves exactly the same as closed 140-15D.
Measurements attached, but no polar plots, sorry. Too tedious to prepare for publishing but I'm sure you'll figure out the point from these plots.

I also need to say this: thinking that my designs neglect directivity and that my customers are happy with it, couldn't be further from the truth. The whole point of my designs and of my customers desires, revolves around just the opposite. That is why I felt compelled to explain you gents what's this all about.

On that note, model DIPOLE 140-15D can not be used for stacking into line array, it was designed that way, so the accuracy of a near-point source would not be impaired, as well as symmetry of the front-back radiation, which in turn, allows different usage of acoustic lens front and back, so user could fit the directivity to room acoustics in terms of back wall reflections and stereo depth imaging.

As for the prices, I know, they're expensive, but if it was possible to make them cheaper, we'd do it. If any loudspeaker company out there could make a cheaper magnetically shielded, fully front-back symmetrical, open ribbon with this directivity, efficiency and reliability, they would, trust me, and they would have a market for it. The point is that they can't speed it up more that we did. To retain the quality I aim for, it takes a company of five highly skilled people and a few outside machining services to be able to build 50 of those a month, making them only. Once you put things on paper, add shipping and distribution, that what you end up with for a retail price. I can't fight the way the world of economy works, and nobody can. At least we have a choice, and my choice is this: Hopefully, with ever-growing support of our customers that graciously buy our stuff, RAAL will eventually grow into a large company that will improve the manufacturing efficiency and bring out good new products for affordable money. Right now, we are more of an artisan's shop that does things the way any artisan's shop does.

And so on and so forth...I could talk about decisions I needed to make all day long and positively bore you to death!

I wish you a very happy New Year!
I'm off now to "enjoy" some excessive drinking with friends that won't take no for an answer!
Cheers!

🙂
Michael

 

[mige0]

Hi Michael, John, everyone!

Measurements are done in IEC standard baffle for 8" driver, only the ribbon is flush mounted. Drawing attached for the people that aren't familiar with it.
Smoothing is 1/6th oct, brickwall. In any case, CSD plots can't be smoothed out, so they provide clues to what the manufacturer did to make other plots look nice. It will show if it was tampered with too much.

John, I couldn't agree more!
However...You know it better than I do, that polar plot will look as good as it's measurement baffle shape is, as well as driver's position on it.
We also know, that most people while browsing for a driver to buy, just have a glance at the plots and go on. What happens if they spot a big dip at 1k5? Would they contribute it to a measurement baffle size or to the driver? Will they read the fine print and a lenghty explanation why it is there? What will the polar plot look like when that edge diffraction dip start shifting around with measurement angle and even worse, multiply itself by the number of baffle edges as soon as you move the mike away from the dead center?
Naturally, I would like to present the driver at it's best, but in essence, you are actually suggesting that I should design the baffle shape solution, just like you had to do, to yield the best results with your new speaker. But then, would that shape fit the midrange that someone wants to use? Would it fit their taste in design?
Considering the above, and that I would like to make sure that the driver polar plots will not suffer from the baffle shape, the thing I should do is to measure the polar plots in IEC baffle, but then, they wouldn't really look like anything dipole, or at least, like anything that the user would get in their all-too-different baffles than than mine. Would that make me a liar? Would that situation of users getting different measurements by default than what I published, help in increasing or in decreasing the customer satisfaction in the future? Would RAAL drivers remain respectable in that case?

Whatever the thing may be, whether I'm right or wrong, my decision is that upon request, people that can read that data, will get the data acquired with IEC baffle. The success of the completed speaker itself, rests is upon them, so here it is, PDF attached.
The actual driver that has been measured is regular, closed 140-15D. I hope you will trust me on this, but the DIPOLE 140-15D spits out the same graphs in that same baffle.

I hope this helps and I hope it sheds some light on why driver manufacturers aren't jumping ahead and show all kinds of data by default. Too many misconceptions out there and the ones that possess them are the quickest to judge and tell about it.

Cheers!

 

[mige0]

Rudolf, you're reading my mind. The design of the driver was made with having in mind the wish to use it as bare-naked as possible ( I'll tell my opinion of that a bit later). That throws us back to the question of what is the best baffle shape for linear sources like this one, and I, too, have an opinion of that. So...

In the attached pictures you can see what the driver looks like from behind.

The pear-shaped structure is the iron yoke that closes the magnetic flux. It was designed in FEM simulation and completely closes the flux with a minimal amount of iron. The iron thickness is 16mm. To that, add 2 + 2 mm of front and back plate.

The front plate is just a mounting flange and it also can be made having the same shape, or completely removed. If removed, the mounting will be a problem, but as it has always been, we make custom front plates without surcharge. If clients start asking frequently for that, I even might decide to redesign it and have it eventually offered as such.

Since this driver is not designed to be used lower than 1.6k, and the mean width of this "pear" allows for about 1k5 cut-off, the driver can be used without this rectangular front plate, without impeding it's low-end response.

That leaves us with this pear shaped yoke, which then plays the role of acoustic baffle, which I believe, is the best possible shape for linear radiators like ribbons. Also, this shape falls in line with this tendency to have double-circle figure-of-eight response extended as much as possible toward highs, but with maintaining the proper low-end (this is a contradictory demand and I'll turn to that later) and without creating too much dips and peaks in the response. It should be a tiny little bit better than naked Neo 3PDR 😀.
Speaking in general now: I mean, c'mon, equidistant edge all around the active surface, are you kidding me?! Show me the naked Neo 3PDR polar plot, horizontal and vertical, please. Better yet, call BG and ask them to send you one. Gee...

later:
Now, to be honest, I never thought that developments of OB design will go towards pushing the ideal, double-circle figure-of-eight all the way up. By definition, you can't have it! Not even with naked Neo3PDR above 2k5. Well, you may get something of a sort, if the active surface size itself is large enough to start beaming, narrowing the kidney, but if it does, it will be frequency dependent and then what? Lobeing and further narrowing toward highs. Double-circle figure of eight is acheivable ONLY below the cut-off frequency of the baffle and that's that.
In short, you can't have double-circle dipole response if your baffle is wider than half of the wavelength at the frequency you are looking at.

Personally, even if it could be done the proper way, I consider it unnecessary, as RT60 times in any room can only have severe nonlinearity and prolongation below 300Hz, and in my book, figure-of-eight (OB woofers) serves as a problem solver in untreated, boomy-bass, rooms by decreasing the excitation of the room modes. I still don't see why it has to be pushed all the way up, as it will only decrease RT60 up there, too, maintaining the RT60 unbalance with low frequencies, solving nothing that OB was originally used for.

To me, whether you have a normal, wide baffle double-kidney or an ideal, double-circle, response of an OB speaker above 3k, isn't something that would make much of a difference, unless you live in one of those ultra-modern, designer apartments that I call "fish tanks" because of the way they sound, because idiots designed them, along with putting the damn kitchen wide open, right in the middle, so you could listen your cutlery rattle with bass and your microwave humming along behind you while you listen to your music waiting on your dinner. If you do live in one of those, my recommendation would be to turn to headphones and forget the high-end speakers based audio altogether, OB or not, and spend more time out of the fish tank.

In any case, you will still have a dipole response all the way up with a null on the side in wide OB's. The main thing is to have that back radiation that reflects from the back wall and gives excellent spatial cues for us to enjoy. A very good reading is Linkwitz's website, and I warmly recommend to anyone that hasn't read it, to do so. A couple of years ago, I've listened to his speaker in his home and from that experience, I can't see what's wrong with having wide baffles. It was THE best stereo imaging I've ever heard in my entire life, and luckily, managed to recreate it or exceed it in my listening room.

Anyway, that was my best shot in making a symmetrical true ribbon tweeter with the best vertical response of all ribbons of the same size out there, with 1k5 cut-off baffle with pretty good acoustical shape, with custom mounting flange when needed. I'm also preparing a small dipole driver, a version of 70-20XR that will be in free sale, not just to manufacturers, which my be better if small pear-shaped baffle ribbon would be better for those design requests mentioned earlier.

Now, as I have a principle of not to use DIYAUDIO forums for business, and things I've already said can be regarded as promotion, I won't be posting any more of anything that has to do with my products. I have my own website for that and any info about new products that I will regard as relevant, will be posted there in due time.

So, when I can join you guys, we can talk principles of operation and such stuff like I tried to do in this post and rattle your cage a little bit 🙂

Cheers!

 

[mige0]

Thank you, gents, I appreciate your welcome. I'll try to be useful, even though I'm coming from omnidirectional with OB woofers camp.

Of course, biased we all are, but we shouldn't forget that we're dealing with objectively measurable stuff and objectiveness will have to unleash us from of our biases once we accept the physical truth and compare it to definitions we claim to follow.

So let's talk principles and definitions we so easily throw around.

For example, cuibono, you do have a fairly nice result, but just at a first glance. It isn't anything near of what it should be.

Let's analyze those results in depth, and someone tell me if I'm wrong. I may be and/or the measurements may be wrong.

Definition: If I remember correctly, almost an ideal dipole response must have about -6 db at 45 degrees, -12 dB at 60 degrees and at least -24 at 90 degrees at all frequencies compared to dead on-axis. Correct me if this is untrue, I'm rusty in math department.

Measured result: That "almost ideal" you mentioned should read "+6 dB to the goal", which is double the energy. Doubling things, or adding 100%, is not a near miss. This happens between the crossover frequency to 6k, above which, things get rough.
Ideal dipole response does not account for such lobeing that manages to make 3dB more on 30 and 45 deg and +1dB at 60 deg. than on axis, turning a response with octave wide -6dB dip on axis into a response with 3dB peak at 60 degrees. That is about 12 to 15 dB more than the goal. That is quadrupling things, which again, can't be called a near miss.

Consequence:
The first reflection off the side wall, which is between 30 to 60 degrees, depending on the room size and speakers-listener layout within it, will on average between 6k to 10k, receive the same energy as your listening position. If anything gives spatial cues, that is this frequency band and if anything, dipole shouldn't do that.

Verdict:
Considering that below 6-7k, dipole is not nearly as fully functional, we can call it a moderate success to the goal that has been set. What we have here is a double-kidney equally obtainable with a wide baffle.

Considering that, centered at 8k, we have a cloverleaf pattern of two lobes front and two back, and depression of 3dB on axis, and that on average it has equal energy dispersion up to 45 degrees, falling off only 11 to 12 dB on average at 90 degrees, we can call it a failure.
9k is a very interesting point, as we have equal energy distribution with only 8-9dB drop at 90 degrees. I also suspect that there is a 45-ish degrees phase-shift break there that allows the 90 deg. 9k peak not to be canceled properly with back radiation. It may be due to slots resonance and/or spacing. Examination of anything that may fall into 38, 19 or 9.5 mm with driver's construction will provide a cause to problems between 6 to 10k.
If it is just edge diffraction related, it is correctable with a baffle/frame of enough width to allow for transposing the dip below the crossover frequency and framing the driver in such a way that the face plate becomes flush with frame front and back surface. If the baffle/frame does not have equidistant edges to the active driver surface (slots area), it may be possible to use the lesser width than what the crossover frequency suggests.

Considering that above 13k we have just simple beaming, but working exactly the dipole below baffle cut-off should, we can consider it a full success.

As far as the room interaction is concerned, all of the above is easily corrected with absorbers on side walls at the point of first reflection. If you don't already have them, that shouldn't be hard to do, so try it out and see what you should be aiming for. Let us know is it better. Follow the speaker-listener-room layout that Linkwitz suggested.

As for the audibility of -6dB at 8k, you certainly hear it, just don't know how to pinpoint it's attributes. A very good exercise is to use headphones and try tampering with any EQ from any media player on your computer.

One of these days, I'll make measurements with this dipole tweeter of mine without front plate and we'll see what is it like. I suspect it will turn out to be allota work to try to correct for anomalies if they appear so I'll give report about it and the correcting methods. Right now I'm too busy, but I'll do it, eventually.

Cheers,

 

[mige0]

Whoa! I'm off by a laaarge margin! Then, very impressive result you have there!

I officially declare the most part of my previous post as **, coming from ignorance and I apologize for being an ****.

Since those are the numbers (thank you), it means that 67 mm wide "baffle" of Neo3PDR will extend the dipole behavior all the way up to at least 6k cleanly.
Now I'm even more interested in the nature of that 9k anomaly.
I strongly suspect the 9k anomaly has mostly to do with Neo3 being bare naked. Can you frame it up somehow, keeping the same overall size, but getting rid of that edge-step and see if it helps? What is that soft, oily, pliable putty that always stays soft and is easily shaped called in English?

I wonder how far it will go, given the size, if the anomaly wasn't there and it makes me wonder how far the dipole ribbon will go, given it's size and pear shape. I thought it won't go far up, but it just might go up enough...I'll do some measurements these days.

You have an interesting observation about 1 to 4k. Most "empty" rooms sound harsh and annoying right there, so having dipole response in that region will certainly help.

Haven't read Toole's book, aside from some papers published in AES Journal, but I know that Linkwitz agrees with side reflections issue.
I tend to agree with that, but only up to an extent, as they tend to smear the focus in the middle of stereo image. I don't know if Toole and Linkwitz specifically mention that, but from my experience, I need to emphasize the importance of having speakers very, very accurately symmetrically positioned, with less than 1cm error between left and right channel. Whatever the distance from side walls, it must be right-left equal, as well as from the back wall. Equally important is the symmetry of furnishing the room. Cabinets, sofas, chairs, tables, anything, wherever it is, especially in region from the wall behind the speakers all the way to the listening position must be symmetrical. Also, move away the table in front of the listener when listening, if it's wide enough to reflect the direct sound towards the listener. Push it all the way to between the speakers, if you can't put it behind you. If you don't, it will force the image down and smear the focus again.

The side wall distance error will easily be heard like sibilants that aren't having the same position in stereo image like the bulk of the vocal. For an inch of positioning error, if the tweeter is about 2 - 2.5 feet from the side wall, the offset is about 4-5 inches in stereo image and it's very annoying.

That is why I consider this 8-9k anomaly very dangerous as it will probably emphasize the positioning error as sibilants are falling right in it.
Omnis are particularly sensitive to symmetry errors in layout, whatever the side wall distance may be, even 6 feet, and you have more of an omni thing going on in sibilants than anything else. (Most people never heard omni speaker accurately positioned in a symmetrically furnished room and attribute the focus smearing to "nature of the omnis". Earl Geddes once dropped in my room at CES with large omnis and didn't like the focus, of course. Neither have I, but he was kinda' deaf to my explanation that because we weren't allowed to move this darn TV cabinet, we had to move both speakers towards right, as well as the listening position, and all symmetry was lost. It still sounded better than having the TV cabinet almost in front of the left speaker. I'll never exhibit anything again at stupid Venetian, BTW).

Dangerous game this is. Many manufacturers when exhibit at shows, place speakers very poorly and furnish the rooms disregarding symmetry. Actually, accurately laid out monopoles will whoop the **** of inaccurately laid out dipoles any day, and you can hear it going on all the time at audio shows.
I know audiophiles spending their entire audiophile "career" listening to poorly laid out rooms and speakers within, spending megabucks on new gear, hoping they'll get lucky, but for some incomprehensive reason to me, fail to listen when you tell them they should play with positioning.
I can't even begin to explain how important the room layout is and how often it's the more decisive factor than the speaker itself.

 

[mige0]

Goes as cos(theta) so half power's at 60 degrees as cuibono mentioned. The point source dipole model is conceptually useful but since actual drivers are acoustically large directivity's better than what you're describing. Like any driver, peak/dip EQ is needed for the dipole peaks and nulls. Once that's applied the non-PDR Neo3 produces consistent directivity within my measurement limits up to about 8.5kHz. The PDR version's a bit less consistent but it's not bad either.

Hmm, looks like entry level pricing's around USD 2500. That's about USD 1000 more than my entire system so it ain't happening for me; even finding a used 8.5 the upgrade to 10 is 999 Euro. Thanks, though.

Hi, twest820!

Right. Cuibono reminded me of it, too, thanks to both.

Even if I was close to the real-life figures, he was referring to ideal dipole, and because of that my discussion is still grossly off target.

As for Clio, if I were you, I wouldn't think twice about it. Maybe it's just me, but I can't use Clio because of it's handling that annoys the hell out of me. I'm a big fan of Praxis, though.

Wow, thank you for the info! 8k5 !!!! Crazy!
That's about 3.3 times it's cut-off frequency!

So, do you have a dip at 8k?
If not because it's EQ'd flat, what does it say off-axis?

 

[mige0]

What we don't want is the typical broadening of the polar response above the crossover point which is typical of dipole speakers using dome tweeter.

I think the manufacture should provide me with the "dipole" format that it is optimal. It then becomes a manor of whether that format suits my needs or not.
.

Hi John,

You're a highly regarded loudspeaker designer and your work is always thorough, it brought so much useful data to wide audience and it was always discovering the good features of affordable components, giving valuable advice to DIY community.
Surely, you must know what would you like to have, regarding dipole tweeters format. You can't just sit back and let me guess what it is. If you're not happy with current offerings, then please define what optimal is. You may be used to manufacturers that don't listen, but I do listen and try to do my best to serve the needs.
Because of that, I have question to ask you, after I point out one thing.

To the difficulties you noted, I would also add the sensitivity issue, which is disregarded here.
Let's not mix apples and oranges, as 90dB/Watt allows a speaker designer a completely different vantage point than 95dB/Watt.

If your choice of amplifier is, say, 25 Watt Firstwatt J2 or any other 15-25 Watt beauty out there, problems start to jump out of the closet. You'd be aware of the potential, but you wouldn't be able to use it with 90 dB/Watt speaker, and that is very frustrating. You'd want at least 95 dB/Watt speaker to match.

For example, closed 140-15 has 95 dB/Watt (not 2.83V), or more, depending on the foam pads positioning. The dipole version, due to increased air loading mass, has about 1.3 dB less with same response curve.
Experience shows that for a good tone balance of the loudspeaker, they both require about 1.5dB more efficient midrange due to dynamic compression issues on the midrange side and due to wider dispersion on the ribbon side.

The problem is that there aren't much mids out there that are churning out 96.5 or 95 dB/Watt flat responses and the ones that do that, are not small filler drivers, but 8" exotics, like Supravox 215 variety, Seas exotic 8, and such. No planars in the game, either.

The mid will start beaming fairly soon and you can't use a filler driver, because there isn't any variety of 4-5" 95 dB's to choose from.

Just 3dB, let alone 5dB in increased sensitivity demand by the customer, changes the game completely and 50 bucks is not considered a chip any more, due to exponential increase in difficulty of designing and building such a thing. 90 dB/ Watt allows much more liberty and it's easy compared to 95dB/Watt.

So, my question is, if you were to design a dipole 95-96 dB/Watt speaker, having in mind that you'll have to use an 8" midrange, crossing is as low as possible, what would be the tweeter format that you'd like to use?

I think I nailed it pretty close, for what it s intended for, but I'd like to know what would you do, if it ain't a secret.

I'm asking because I may want to provide certain features to the baffle so everybody would be happy.

One possible way to do this is to increase the iron yoke size, but drill a large number of 1/2" to 3/4" holes through it. That way, you can choose which holes to plug and which to leave open, creating different patterns that just might fit any dipole taste, without increasing efficiency.

 

[mige0]

Hello all,

That is the old fashioned engineering point of view but unfortunately it does not make any psychoacoustical sense. What we really need is monotonically widening dispersion somewhere above 3kHz. To minimise the cross talk artefacts of stereo one needs to spatially homogenise the pinna cues and that is most easiest done by providing very wide dispersion at treble, almost omnidirectional above 3kHz.

Also in the midrange below 1kHz one needs to minimise room reflections to be able to provide stereo ITD cues and that is best done with very narrow dispersion.

Clearly it turns out the ideal loudspeaker directivity is not constant, not narrowing but widening!

This radical view was offered for your pleasure by Elias 😀

Since I share this view to the last point, that's why I can't really figure out what would be the benefit of having equal dispersion everywhere, as both rooms and our psyche doesn't accept it equally.
As I pointed out before, all rooms have problems in bass, and some rooms have problems in presence region (centered about 2k5). As cuibono pointed out he prefers dipole at least up to 4k, which falls in line with mentioned room problem.

I just want to add that I stand behind the room treatment and prefer omni down to 200 Hz in such case. Rooms must be treated, or they will ruin a lot of our efforts. However, once appropriately treated according to their size and volume, the results are nothing short of astonishing and the sound is in a different class altogether. Nothing like it, if you want to hear the actual size and feel of the instruments, and horns can't touch it, with their "in your face" presentation. The way I like to put it, they have a big sound, but not big instruments.

One of the people from the whole of audio industry that I admire the most is Amar Bose. He went through the trouble of designing excellent 5" whizzer-less FR drivers and stuffed 9 of them in a clever box, with one shooting forward.
That speaker, with it's dedicated EQ box, kicks *** big time in recreating the proper size of the orchestra and all that it contains.

For me, a wide baffle, big midrange (even 12"), OB is the next best thing in untreated rooms.

 

[mige0]

That is indeed an amazing response - but I'd be concerned that it isn't the whole story. The arrangement of the two drivers makes it look like it may have different responses at different angles (horizontal looks great, but what about vertical, and in between?). Looks easy enough to try though - I guess it is a testament to our laziness 😀.

Aleksander - if you could make a driver dedicated to dipoles, that could hit near LR4 at 1kHz, and had a smooth polar response, and good sensitivity/power handling, you would have a winner, and it would be worth the cost, because no driver can do that yet. It would change the way high-end dipoles are designed. Is there a large-enough market? I don't know.

I second that about Rudolf's work! The same goes to twest820!

I could, but you should know that a good cone driver does things better up to 1k5-2k than a ribbon ever could dream of. That is why electrostats and big planars sound wimpy with drums, percussion and piano.
In midrange, parachute effect, or large air loading, needs to be controlled with greater force per surface area than any ribbon, stat or planar can give.
Trust me, you're much better off with cones in midrange.
If you want to get truly remarkable results in midrange, use 3" FR driver in clusters of 2x4, 3x3 or 6 pieces in hexa and one in the middle for total of 7. Do not use line sources, they don't add together well, as top driver doesn't know what bottom driver does and vice versa, meaning that they don't share the load and sound equally wimpy just like one feeble 3" driver, only you can cross it a bit lower. Sharing the air loading is the key, a such 3" FR drivers clusters are the way to go. They get so much authority and slam that it;s not even funny. The main advantage to a comparable surface area of just one cone driver, is that 3" FR will have first cone breakup problems even as high as 11k, so you'll end up having a piston of a large area all the way up.
Choose only ones without Farraday rings and such tricks. Look for saturated pole plate and central pole piece. BTW, very few good driver use a Farraday ring as it wastes the signal current on showing you a nice impedance and they need 150 Watts to get dynamics and start sounding lively. Only a few bright examples exist, made by people that understand the drawback of it.

 

[mige0]

for sure 🙂




Those amps are not really suitable for OB to say it loud and clear - at least not if you are seeking SPL plus some headroom - and on top of that passive XO possibly.
OB / nude is sheer waste of power to a considerable degree - a luxurious design attempt in any way ...




Use two of these (Scan-Speak 15M/4624G Discovery) - one above the other - and you are almost there..

https://www.madisound.com/store/product_info.php?products_id=8926

IMO pretty sonics for that bargain price too.


Michael

Oh, yes, they certainly are in driving mids and highs! Passive XO between mid and high doesn't hurt it, if designed nicely. It goes without saying that EQ headroom for the low 15" or two, will a big solid state or a D-class.
I assume OB should use EQ, and I assume you need a separate amp for the job that wouldn't be 25 Watts.

As for the number and kind of mids I like to use, please read my previous post.

 

[mige0]

Hola muchachos!

mige0,
I was talking about compression drivers and horns. Having back radiation changes situation dramatically, I believe you. I've been around the block a few times, you know. See what I was playing with back in '95. It was during the war, so there was nothing better to do as economy was ruined. Nothing better than playing with Pass' Zen, SE tube amps, Moscodes, etc. and building speakers, he, he! Not so bad, hey! I did try the front back horn, too. When I mentioned the instrument size, I wasn't referring to HF, as I don't think the instrument size lays in that bush. More of a something below 1k5, but mostly between 200 and 1k. If you dry the room out of that with a dipole or a horn, you may, just may, get cleaner sound in general, but it still doesn't compare to a treated room with omni. Probably because it's not natural.

when i said 3x3, i meant 9, with 2x4 or 2x3, i mean 8 and 6 and with hex+1 in the middle i meant 7, used for midrange above 200Hz.
Of course, beaming gets very tight up there, but as you've said, no free lunch.

Hi Telstar,

Beyma 150 is an interesting beast! here's a plot of dipole 140-15D without acoustic lens. this will make a better comparison to the Beyma tpl150, as they're about the same vertical size. If you're not crossing low, and maybe you don't as you use 4-way, I don't think you'll loose much of the efficiency, if you don't mind the same beaming.
Aaaah, MAD planar. Designed by my first mentor, Dragoslav Colich for HPV. What a jewel that is! It plays so effortlessly relaxed, smooth and soft but with a slam and kick I've never heard out of a planar! Had a pair for a couple of months and living with those was sweet. Are they available now? They used to be $1k a pop...
I've been told great things about BG neo10. I think I'll get a pair.
I need to listen FAL somewhere. I'm very curious.

Well, for one Per Skaaning is the one that knows how not to overdo the ring thing.

Yup, multiplying them little critters will be the center of the new OB speaker I'll do, following that narrow midrange - wide flush HF approach. Not because it's easier done that pushing dipole high up, but because it still makes more sense to me. I know what I've heard and the "big midrange" does the trick for me when OB is in question. Well, if anything is in question, big mids are the ticket. By big midrange driver, I also think cluster of 3"-ers, just to be clear.

Fountek is a phenomenal value for money! I'll shake hands and take a picture with Fountek guys if I ever meet them.

 

[mige0]

I just have to post this!
My first mentor, Dragoslav Colich, co-creator of Monsoon and creator of MAD planars. He's in shorts. We are listening to two panels consisting of 64 MAD drivers on a side, with 24 OB peerless XXLS 12" on a side, with a silly number of amps and power involved.
Now, that was something to remember! That 119dB was the average SPL, slow setting, C weighting. And there was no pain or anything that will make you turn it down.

 

[mige0]

Hi,

Thanks a lot for sharing. Very interesting dirver. Is that 'ladder' at the rear the magnetic circuit?

No, the ferrite magnets are below the plates, those are iron bridges to close the flux. The XO is at 1k8 and it has 90 dB at the front with that assymetrical horn that allowed for very little towing in. It sounds silly today, but I didn't know better back then. Still, the nature of the beast was obvious.

 

[mige0]

Thanks for the compliment, and you should know that I respect RAAL products. But they just aren't my cup of tea. I'm not particularly interested in high aspect ratio radiators, be they tweeters or otherwise. I'm actually toying with the idea of placing a circular wave guide on the Neo3 for the next generation Note. We'll see. I'm not ducking your questions but I really don't know what optimum configuration would be. I know what I would like for the response. I guess if I were to try to design a true dipole tweeter is would be very similar to what Rudolf was proposing many posts ago, back to back domes which radiated as a true dipole but for which directionality took over before the on axis nulls appeared.

I also can not tell you how I would design a higher efficiency speaker. It is not something I would comment on off the cuff. But I can tell you I would not be using any type of gradient woofer system. If I wanted high efficiency across the spectrum I think I would pretty choose to suffer loss of accuracy the much go with a ported woofer. After that I just don't know.

Hi John,

Ahhh, the aspect ratio is killing me for years!!! I know exactly what you mean. Currently, I can't knock it down lower than 3.5 times. That's why I made it rising towards highs and then lensed the thing.
Anyways, along the course of playing with ribbons, I realized that this wide horizontal dispersion is an invaluable asset (to my taste in audio presentation), but I also understand why some don't consider it an asset. In any case, I think that such quality of converting the current to air movement, in HF, could not be achieved with technologies that allow lower aspect ratios. I also realized during playing with speakers that the "sound" of the drivers we use to build speakers is the most important thing and directivity is very secondary matter, and coincidentally, I prefer the presentation with wide flush HF.
That's why, when I get my hands on a good sounding midrange driver, I hold on to it and design around it, following my style in design choices. I actualy think, that for the same reason, you hold on dearly to the very best driver that 57 bucks can buy, because if the task is to present the people the solution that squeezes out the absolute maximum from a low budget, there is no way that it can be done any other way than the way you did it. If I'm right, kudos to you, my friend.

I very much like your answer to my question (thank you!) and I understand what you're saying, which reminds me...at a certain point, once you explored every path in a given field, a playful and relaxed state of mind takes over, that George Carrette, a software engineer, probably explained the best:
“First learn computer science and all the theory. Next develop a programming style. Then forget all that and just hack.”

Hilarious!!!

Let's keep on hacking the thing!

Hi, Telstar, sure you're invited! Impulse response attached.

Hi, Rudolf, I'm not Elias, but I wan't to give my 2 cents on this. Of course, we can't mess with a proper tone balance of direct sound (on axis), so my choice is rising power response at linear on-axis response (more precisely, wide flush tweeter like ribbon, should be attenuated a bit in the whole band, which for a few reasons, feels like linear and this is not just related to increased power response, but also microdynamics, etc.), but the main thing that I observed is that with increasing power response above 3k, is positive to the perception of presented music.
Most rooms, even the emptiest ones, have decreased RT60 times above 3-4k, so increasing energy off-axis up there, does not give disbalanced diffuse sound field in the room. Just the opposite, it equalizes it, which in turn, adds a constant RT60 time of the listening room to any recording we play. Now, that is nice! The highs get more sustain, you hear more of the natural concert hall decay and all of a sudden, everything clicks into place.
I strongly maintain that listening room must NOT mimic a concert hall acoustics with it's faster HF decay, but have linear RT60 up to min. 16k, and if it doesn't, then wide flush HF comes to the rescue. Even if the wide flush HF is used in linear RT60 rooms, no biggie, JUST sit closer and/or increase lateral reflections arrival time, and things are again much better than with equal or monotonically decreased directivity.
It all comes down to this:
We listen and judge by direct sound. All that happens off-axis is solvable, augmented, ruined, whatever, in short: processed and bounced back by room acoustics. Solve the acoustically small room for a given speaker and you've solved the sound. Solve the accoustically large room by a standard, and you've solved the sound for all speakers. Well, just one exception to this rule, you can't help front firing horns with anything. that solution belongs to the past and it is a compromise used to get efficiency. I hope we'll get over it, some time in 21st century.

Hi, pk, I'd go for Fountek. I haven't listened to it, just held it in my hand and examined it the best I could and it give me confidence. In any case, TB does not have saturated pole plate and you don't want unsaturated Iron in the vicinity of the voice coil.

Hi, keyser, I think your measurement shows (assuming that you measured in far-field keeping the mike at constant distance to the center of the cone) that, actually, nothing much happens with shifting. There's just some cancellation at 1k6 that makes it look like going down, but it's not. The first null is shifting upwards, but not nearly as much as two point model predicts, so if we take the first null as being closely tied at equal "distance" to the peak, the peak then must be going up, but really slowly. So, as in good old quantum physics saying, you showed that "a good practice, always ruins nice theories"! Theory in this case not being the accuracy of two-point-at-a-distance model, but it's applicability to single vibrating plane reality.
Don't know much 'bout 'custics, but it seems that, at least in case a vibrating plane is within the order of a magnitude of wavelenghts in air it has to wiggle, it becomes a spread parameter model that can't accurately be simplified with two dots and for that, congratulations! Good practice is what we need.

That reminds me, a new era is coming!
Many of you probably know this, but nowadays, there are simulation programs that can accurately predict this type of things, called "multi-physics" modeling software, like COMSOL, for example. The stuff works by setting the environment fluid (usually air or water) physical constants data, as well as to all parts of the model being examined. It can virtually run a jet engine or a speaker, because it works with basic material electrical, magnetic, chemical, thermodynamic, elastic, plastic, all kinds of known properties of any material used, and NOT with mathematical assumptions of how a certain behavior can be modeled. The difference being that you just have to make a 3D drawing of what you want, assign the materials and apply the excitation you want and the thing will "model" itself by interacting the fundamental material properties, just like the real thing does. It'll show you the fluid dynamics, vibration, heating, sound interference field, all that sh..! As much as we are certain of some physical property, taht much the workings of the contraption we drew, will be accurate. Pretty amazing! Too bad it's too darn expensive!

 

[mige0]

Forgot the attachments!

 

[mige0]

Brr... metal cone.
would thisbe an alternative (still in cheap mood to try it out):
Overview of C3N-III Drivers_HiVi,Inc
?

Hi Telstar,
well, this one it the best one yet out of the 3 models proposed, but not because its paper cone...
Here is the explanation of my general criteria for midrange drivers and comments on why:

1) Mechanical Q, as high as possible. My 3"ers have Qms= 9.8

If aluminum voice coil former is used, it's because it will act as eddy current brake and it will help low end control, which was the designer's intention. It works as a compressor, as the faster the jolt is, the better it brakes. If there's anything in this world that I don't want my mid to have, than this is it. I don't want my impulses being braked!
Some aluminum voice coil formers brake more, some less, but kapton doesn't brake at all.

2) Saturated front pole plate and eve more important, central pole piece.

In the first question, whether it was beter to use TB xxx or Fountek xxx, it seemed that Fountek had a better chance in having the saturated pole plate and central pole piece, as it's pole plate is thinner than fountek's. I was confident that it should be able to do it with this 50 mm Neo ring, as i know how I could utilize the ring of that size at 10mm pole plete, but I must apologize now, as I just read it's datasheet and it doesn't even come close to saturation. Both of them don't so I wouldn't use any. That's a pity, wasting so much Neo...
The problem there is when you need to run a very large batch of speakers, very quickly, to get the price down and make your buyers happy, you can't fiddle too much with tight voice coil gaps, especially if they're deep and have such a small diameter. Then you loosen up the tolerances, naturally. Slightly oval voice coils, then quick magnet assembly that doesn't allow you to position the plates too accurately, then fast machining that increases +/- dimensional errors, then very unhappy workforce that doesn't really care much about strict centering of the voice coil, as long as it's not rubbing, and instead of paying them more, you let it go as it's easier for them that way. and so on and so forth. Running a speaker factory is a multidisciplinary art of making people that you work with-happy, then you can make demands...

If I could sublime it: cheap speakers are not cheap because they utilize materials better, but because they don't! Contradiction? Think about it.

But, your proposal is very good as Hi-Vi has a large ferrite that should be able to saturate 20mm pole piece with just 3mm pole plate. Judging from BL, I guess it does, as I don't thing you can get more from small 20mm coil (short wire) in such a shallow gap (3mm). Of course, on can never be sure.

Related to the 2) above:

3) no Faraday ring(s)

Let impedance rise. Its her job when there's a coil of wire over the iron core. Don't force it out of it's job, as you'll just create waste. You'll waste signal energy into inducing current, thus heat, into the shorting ring.
One thing is for sure, that heat will not be transferred into force pushing the cone.
What I'll say now, may induce a debate, but I don't care, since speakers with Faraday rings sound boring anyway.
Yes, they reduce distortion. Why? Because they oppose modulating the magnet material by the voice coil signal current by running current induced in opposite direction and this will keep the net modulation low.
Hmmm, okay, but how the magnet material gets modulated in the first place? Well, because the central pole piece and voice coil around it look like an electro-magnet, lately addressed to as "field-coil" and the signal current tries to induce magnetic field that will eventually close it's path, passing through the magnet circuit, modulating the magnet. This creates 2nd harmonic generation and Alnico magnet types are the easiest to modulate (which is why some like them better), then ferrites, then NdFeB , which are the hardest to provoke(which is why some people that appreciate Alnico "character" don't think much of Neo novelties, as the have no "soul" to them, as I'm told).
Anyhoo, when you start passing magnetic field through the iron, it accepts it well, which is called permeability. At the beginning, all those little crystalline magnetic domains are starting to align themselves, waking up, and soon after, the permeability is at it highest as all the domains are awake and doing their aligning thing by command. If you continue increasing the field, domains will start falling into complete alignment with it and at a certain point, all these little compasses will be pointed north. That is the point of saturation! There is no more room to improve the direction of domains, and you can add field as much as you like, but nothing else will happen. They are at a standstill.
What is the first thing our voice coil current will do in it? It will induce the magnetic field in it and this field will either attract or oppose the static field of the magnet, and this will create force that wiggles this coil back and forth. Now, IF all the domains in the central pole piece are aligned completely by the overwhelming static field, they will all look at the same direction, completely ignoring the voice coil field. Since they aren't able to respond any more, they will effectively protect the magnet from being modulated by the signal current, thus being completely ignorant of what goes on in the coil, or better said, our music.
So, with properly designed magnetic circuit, our voice coil will be just like hanging in the air. it will not see the iron, as iron isn't responding to excitation, as far as magnetic field is concerned, when something isn't responding to it, it must be non-magnetic, thus not responsive to modulation. Also, with properly designed magnetic circuit, all magnets will "sound" the same, and in a good way, as modulating distortion will be gone.
Usually, people that have electromagnets in their speakers, report very obvious gains in sound quality, in detail and microdynamics. That's because when they increase the current feeding the electromagnet, they eventaully get to the point of saturation of all iron parts, so the voice coil doesn't have to waste energy in modulating it.
With the right choice of material, you can have that saturation as low as 0.8 Tesla, and save on magnet size, keep the T-S parameters in the right ballpark even for an 86dB/w speaker, but it'll sound like the best of the best compression drivers or Japanese FR's in microdynamics and detail department. I mention those two groups as they are the most common place to find that approach. The technique was a well kept secret in the industry and a few understood it, as most use Faraday rings as much as they can, and bragged about it. Latest (but not so new, it's been a while) and best solution comes from Billy Woodman of ATC fame. It's easy to do it with overhung voice coils and thin plates, but underhung voice coils with saturated and regulated gap come ONLY from him in his Super Linear (SL) series. And it doesn't use the ring but brought down the distortion by 15dB.

4) Suspension and spider.

You will notice with many speakers, inculding the small 3" FR's, that many will show little impedance peaks between 500 and 3000 Hz. Those are cone resonant modes and/or suspension resonance. The less the better. With good suspension material like Supronyl, made very thin and with a small roll, they will dissapear, or move above the intended bandwidth, if the cone stiffness allows for it.
Very soft spiders. I have Fs down to 67 Hz on 3" I use.
My general choice in speakers is towards very soft. I'll control the cone movement with other means, thank you very much, I don't need the coil pushing one way, the stiff suspension pulling the other way, effectively pinching the cone between opposing forces and making it endure unnecessary stress, reducing lifetime and provoke resonances.

5) Diaphragm material.

specifically in case of 3" Fr drivers, I prefer Aluminum cones with hard TiO2 or some other stiff coating, like Al2O3 (thick anodization).
The choice with larger driver is exclusively paper, or properly resined carbon fibre.
With small 3" cones, there is little much to resonate. Voice coil is 1" and that leaves only 3/4"" of the cone that's been edge-damped with surround. Breakup of a small stiff Al cone is very high, like 11k in case of 3"ers that I use. Paper is just fine, too. However, it doesn't allow me to approach the "piston movement" ideal in the whole passband, as it may start to break sooner. Since I can't trust it with humidity and temperature, i avoud it when i can, and so far, I managed to do it only in 3" FR case. Everything else, still seems to be better in paper.

So, I don't know are there any great 3" drivers out there to buy. I ordered mine to be made the way I like them and I wouldn't be able to give advice on every example, but I hope that with this explanations, anyone would know what I'd say about a specific driver.

 

[mige0]

Hi Michael, thanks.

This braking is nonlinear because the the voice coil former ends just below the winding, but extends to the cone above, working in assymetrical stray flux that it picks up, but that's not what I want to talk about.
The main thing is that it depends on the velocity. Highest speaker velocity is at resonant frequency and it's absolute value at a given voltage depends on Qts.
So, it is working better as it approaches resonance.

On 3k, where the cone velocity is very low, it will not do much.
If we look at the single tone at 3k, nothing really happens to it and it's fine.
But let's take a 300 Hz as carrier and 3k riding on top of it.
In highest velocity part of the 300Hz carrier, passing zero, and for example traveling to positive direction, braking will be very good and when 3k sine is superimposed on this, it also will be under braking, but more in the positive direction. When the sine wave approaches peak, the 300Hz velocity will drop to 0, no braking there, and 3k will be freed of influence. When 300Hz strats going back to negative direction, things repeat.
It actually will change the shape and generate, IMD and harmonics of 3k, mixing them with IMD side products, creating more IMD...
Making a good speaker is all about multitone measurement methods.

BTW, speaking of velocity and direction, Doppler distortion is a bogus story altogether and it doesn't exist in speakers in any, even remotely significant way.

I deliberately didn't mention friction, as this thread is already hijacked enough and I didn't want to expand the story too much.
You are right, it is very dangerous. However, it is amplitude dependent in a complex way.
1) Once the cone gets going, it expands the spider and surround, and textile and lossy rubber have increased friction with higher tension.
2) Friction needs a minimum amount of force, or signal power, to overcome it and start the movement of the cone after wasting the signal power on creating heat. It actually eats up small signal, loosing detail and natural sustain.
This is why many speakers can't give much detail when listening softly. You gotta crank it up to get it out, and not just because of friction, but also because of too big Faraday ring, long story again...

Also, for a given listening sound pressure level, with larger midrange speakers, aside from having lower THD and motor nonliearity related IMD, larger speakers will move slower, having less problems with velocity and tension related distortion. That's why the bigger (or more) is better, provided that Qms per used speaker is high.
If we multiply the speakers, we decrease velocity and we don't want to fall into too low velocities where friction takes over.

Luckily, Qms is a very good showcase of frictional components and the higher it is, the better. Speaker should be mostly controlled wit amplifier, not conductive voice coil formers and manufacturers should stop this practice.
They mostly use it for better cooling of the voice coil wire, braking is just an added "bonus". Per Skaaning can give you Qms of almost 20 on an 8" speaker 😉

 

[mige0]

With regard to Elias' and Raal's statements I must ask, does any instrument produces overtones in a manner not at all connected to its fundamental tone and would omnidirectional distribution of those overtones contribute to our stereo sensation ? It seems that this claim might originate from this page: Understanding Ambiophonics . Why stop at 1 kHz, fundamentals extend much higher.

/Erling

Hi Skorpion, Radugazon!

Radugazon, thank you for this brief explanation ILD, ITD and PC.

Skorpion,
I like to approach simply to any problem. One thing is a constant, our hearing mechanism and what habits it picked up while we were growing up.
So, it is what it is and i can't change it.
The other thing is the speaker. I can change that, as obviously, my hearing mechanism isn't satisfied with it.

In short, omnidirectivity will improve the reality of your stereo image, but IMO, not because it will relate in some new way to ILD, ITD and PC, as all the same rules of stereo setup apply, but because of different and more natural relation to your room acoustics, which in turn, will help the processing algorithms of your brain, not directly with sensory gear firmware.
I think that there must be a sensory gear firmware first, and then the processing algorithms after. The algorithms gets learned as we grow up and we usually grow up in rooms.

To be more clear, the necessity for omnidirectivity lies in with how we process the relations of the source (speaker) and the environment.

This is the excerpt on my paper on our omni speakers, that will reflect my opinion on the subject:
>>>>>>>>>>>>>>>
The requirements:

In order to convey the lifelike presentation of an acoustical event, we need to have the following, properly presented to the listener:

• Depiction of the true scale of an instrument or orchestra and the space it occupies.
• “The angle of view” to a large orchestral body that we can encounter in a live event, keeping the right instrument sizes, but at a smaller “angle of view” because of distance to the orchestral body.
• Pitch-related and distance-related “proportions” of the tonal timbre.
• A multitude of tones, transients and sounds of similar nature must not interfere with each other.
• Directly presented low level detail that requires no “unveiling” concentration whatsoever on the listener's behalf, but with preserved dynamics - no compression of downward dynamics.
• Unlimited large signal dynamics...

Omnidirectivity:

Before we continue, a few words on omnidirectivity as a tool of presentation of the recorded audio events.

The first three of the above requirements rely heavily on speaker-room interaction. In our hearing mechanism, processing the reflections of the direct sound brings very important information to us, like a predator inside or outside the cave and how far, above or on the side to the entrance and so on. Now, in order for a reflection to be recognized as something that relates to the direct sound, and thus, giving us aforementioned info, it must have exactly the same “fingerprint” as a direct sound, or it will be mistaken for something that doesn't relate to the direct sound and it will be disregarded as an unimportant noise that carries no valid and conclusive information. The same “fingerprint”- meaning that the reflection's impulse and frequency response must be the same as in the direct sound, and altered only by the properties of the material of which it bounces off, only assigning it different timbre which depends on the reflective material absorption. The only thing that can be altered to a degree is the spectrum content but without introduced group delay for different frequencies. It is a similar thing with vision. We can recognize people by posture and walk, even seeing their shadows on the wall, but if the shadows are too stretched or skewed, it makes no useful information.
It all may seem too complicated to translate into engineering tasks, but the answer is pretty simple when you think of it. We need omnidirectivity. We need to send out the same information in all directions around the sound source and let our brain take in the info that will bounce off from the walls. Now, there are many seasoned loudspeaker engineers as well as audiophiles which claim that, at the best of their knowledge, introducing excessive number of reflections is derogative. And they are correct in their own right. That definitely applies as true for normal, directive speakers, where the sound emitted in all directions bares little resemblance to the on-axis forward sound, and right there lays the answer to why they are right, but only under that given circumstance of using a directive speaker. Usually, directive speakers require skillful room treatment to balance room acoustics. Paradoxically, omnidirectional speakers are much less fussy. That may sound like nonsense, but it isn't. What matters for an omnidirectional speaker is the symmetry of placement inside the room, with reduced attention to room acoustics and we will have the sharp focus of instrument placement inside the stereo picture and properly set “the angle of view”. The only difference in the pinpoint imaging of an omnidirectional speaker, as opposed to the directive one, is that it sets the true scale of the instruments and events, if those are recorded decently. Sharp but big, to put it simply. With a proper piano recording for example, you can “see” the keyboard stretching with every key sharply in it's place with a naturally large piano body occupying it's natural space. Drum-set is a drum-set and the skin is not the size of a peanut. The “hit” is the size of a peanut, but the presentation is the size of the drum...
What is interesting, as opposed to directive speakers, is that the room acoustic response can vary in much larger scale than it would be normally acceptable. That is because omnidirectivity will equally engage the room reverberation, which will, regardless of the placement of the absorptive and reflective surfaces relative to the speakers, give back the reverberative “output” which is broadly averaged. The directive speakers, and the more directive-the worse, will send out different frequencies in different directions and the room will be selectively engaged, reflecting back a different thing with every different speaker, depending on it's placement and aiming.

So, because we live, walk and talk in rooms our whole lives, our brains can easily relate to their acoustical fingerprint, especially to the ones where we lived for a while, and the difference between us talking and directive speakers playing is that our talk and noises that we make, engage the rooms almost equally in all directions. This is processed in our brain along with the reflections of our rooms added, every time the same way. It tells us that we are not artificial sound sources, the same way that, in the case of directive speaker playing, which don't engage the room equally, it tells us that the fingerprint of the reflection is not the same as the direct sound, rendering the conclusion that the sound source is of artificial nature, since it's fingerprint doesn't belong to our room, bringing down our music enjoyment and emotional engagement as we are subconsciously aware that something is skewed. We call all this “acoustics distortion”. It is not acceptable in the same way other various forms of distortion aren't.

Opponents say that “yeah, but I want to hear the acoustics of the recording room, not add my room to it”. We say, so use headphones then... Unfortunately, that doesn't work because of the way our brains are hard-wired. We must allow the room we are in to bounce back the time-coherent reflections. The only way is sending out to the room the same signal in all directions, i.e. omnidirectivity, or the room will bounce back the skewed shadow of a direct sound that will not just make our brain disregard the event as real, but also, struggle to disregard the reflections, as they will be considered to be background noise that carries no related info in order to enhance intelligibility of the event, leading to fatigue and finally, to loss of enjoyment of paying attention to that artificial event in the first place. The big old truth lies in the standards for the performance halls, which by default, have much longer reverberation time than any of our rooms and there will be no harm whatsoever in adding our decay time to the hall time, but only provided that we engage our rooms in the same way the orchestra engages the hall, hence, in all directions. If that sort of natural fingerprint of our familiar room is recognized, along with proper time and frequency fingerprint of incoming reflections, we'll have no trouble at all to relax and listen to any kind of recorded room acoustics (or none at all) and music, letting our brain be convinced that there may be a real event happening in there. Since that really isn't there, after all, we will know it, but the convincing presentation will be far more enjoyable, as we will definitely feel how excited our sub-consciousness is becoming by listening to it and having no objections to the matter at hand.

Our experiences with Eternity and other omnidirective speakers, one of them being a true masterpiece of electroacoustics as a science and art, we can say that omnidirectivity opponents may have listened to omni speakers but not the right ones and we can definitely state that omnidirectivity, as a principle, is superior and that only the specific executions may be deeply flawed. In short, omni-opponents have listened to the wrong speakers in the wrong set-up and judged accordingly. Omnidirectivity as a principle of presenting the recorded music is far superior to anything else, if executed with properly designed set of loudspeakers in accordance to the specifics of the principle.
>>>>>>>>>>>>>>>

 

[mige0]

Please,

just go back to your own installations and listen just to bass and treble, like in my case 250 Hz to 2 kHz out and report back.
It will be very valuable to see your inputs.

/Erling

Hi Skorpion,

what do you mean. I don't get it?
Try what, because of...?

 

[CLS]

Yes, I'd really like to know more about the Faraday ring(s) in the magnetic circuit... etc.

Thanks in advance 🙂