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In praise of AR3a's


Carlspeak

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His patent is certainly complex enough. The main problem with it is that it appears to require a lot more speakers than most people (certainly spouses) are willing to live with. Even with only six speakers, many home-area situations will not allow for that kind of arrangement. Also, while he claims that Yamaha perhaps stole his ideas, the only thing they appear to have stolen on a provable level is the number of speakers used, and nobody is going to call that a patent violation. Interestingly, Bob Carver had a receiver some time ago that also made optimal use of six speakers, too. I reviewed the thing, and it worked considerably different than the Yamaha approach. Different does not mean better, however, and I still prefer what Yamaha did. Speaker count does not necessarily add up to a processing similarity.

Also interestingly, is that the Yamaha approach at the time of the DSP-1, as well as "soundmixer's" own approach in the patent makes no use of a center channel. Yamaha's early assorted concert-hall and club modes only involve two main speakers up front. However, later on (starting with the DSP-A3090 integrated amp that I both reviewed for a magazine report, and also purchased and used for years) offered a "Classical/Opera" mode that did involve a "derived," Dolby ProLogic style center feed from two-channel sources, as well as a somewhat simplified surround-channel processing program. For me, that mode works better than any of the others (both of my current processor/amps have it, too), simply because of the addition of the center feed. However, to make it work correctly the center level has to be backed off 3 to 4 dB to keep certain parts of the soundstage from being excessively steered to the center speaker. The amount of lateral pinching will vary from recording to recording, depending upon the microphone and mixing techniques used.

In any case, that particular Yamaha playback mode delivers the most realistic soundstaging, spaciousness, and sense of being there that I have encountered with any surround-synthesis or extraction program, including Dolby ProLogic II.

Your EQ patent seems interesting, but it also seems to be more complex than most enthusiasts or manufacturers would care to deal with. I use a basic equalizer with my two systems (3.1 channel Rane THX-44 with the main one and two stereo-only THX-22 units for the smaller one; only one channel of the second THX-22 is used), and with careful use of an RTA in either room I get very flat response at the two (in each room) primary listening locations from all of the front speakers. Going beyond that with some kind of computerized ultra equalizer still would still put the user in the position of having to equalizer individually for each selected seating location. I split the difference and get a generally good EQ at the two most important ones in each installation.

Howard Ferstler

"The main problem with it is that it appears to require a lot more speakers than most people (certainly spouses) are willing to live with."

I regret that the solution to the problem results in something not visually esthetically pleasing to everyone, perhaps even to most people. What should be remarkable IF it works is that it works at all. We've gone from what many would say is impossible to what is esthetically displeasing to look at. I'm disappointed in you Mr. Ferstler, I expected better than such pettiness from you given your reputation. The patent appication made it clear that the example given is only illustrative of the principles and that many variations are possible within the boundaries of the claims. My own original prototype had ten speaker systems each having two drivers in addition to the two main channels. These reverberant channel speakers were wired into 4 quadrants and were designed to radiate in a way that would produce the required sound field geometry. The current prototype has 16 minimus 7 speakers also arranged in 4 quadrants but being unsuited to purpose had to be "adapted." The large number of speakers is a function of the uniformity of the field required to achieve the necessary result. (I think Berkowitz of Acoustic Research used 24 in his patent.) If you can hear where any of them are, the system has failed. A lot of thinking and some experimentation into how direction of sound is determined and how it could be confounded went into this. The current explanations are IMO flawed, there is an entirely different one than is in the literature. I've been readig Dr. Oliver Sach's book "Musicophilia" to gain insight into the current understanding of how sound is subjectively experienced and assessed. Placement and careful calibration of the speaker responses and their relative gain for each individual room to meet the design goals is critical to success of my concept. This alone would make installing it beyond the capabilities of practically all audiophiles and it may not be workable in many types of rooms, especially those that are not symetrical. The closest software available to systematically calculate the optimal design arrangement is the still relatively infant science called CFD, short for computational flow dynamics which predicts among other things air flow within rooms of specific complex geometries and flow restrictions for air conditioning systems in places like data centers. (Obviously the thermodynamic component in that software is not of interest in this application, only the air flow fluidic aspect.)

One innovation of DSP-1 that I contend was "stolen" which may seem obvious in retrospect 35 years later was the introduction of the ability to process FR simultaneously with reverberation, that is inserting equalization into the regenerative loop simulating differing RTs for different frequencies. Beranek says ideal for concert halls is 1.8 to 2.0 seconds at mid frequencies. Data I have for about 200 concert halls shows many have about half to two thirds 1 Khz RT at 8 khz, the hf limit for which this kind of data is usually measured. My own preferences are for moderately longer RTs but I have no explanation as to why. I don't know if it's just my own personal preference as a reverb freak, or having to overcome the short lateral delays the room imposes on the main speakers, or some other factor. In 1974 when this idea struck me, that knowledge was not readily available if it existed at all. All existing reverb units had a uniform passband for all frequencies. Also, the idea of different combinations of delays from different directions was novel for an electronic reverb.

I said there were vestiges of the measurement concept still in the patent. Figure 2 in the patent shows the test source as a spherical array. Obviously the speakers can be wired to individual amplifiers and their gain and equalization adjusted to simulate the spatial propagating characteristics of different musical instruments. Does that mean concert hall acoustics affect different instruments in the same spot differently? Yes, it is complex isn't it. So a tuba whose sound is projected mostly upwards will be affected differently from a clarinet in the same spot whose sound is projected mostly at the floor. And differently from a harp which is more omnidirectional in propagation. Figure 3 shows the array of unidirectional microphones each receiving only echoes arriving from the direction it is aimed at. The reality that the reverberant field is the result of a vector transfer function and not a scalar field was to me the most important realization that led to the invention. In theory at least 3 reverberant field channels are required to achieve this. Practically 4 is the minimum and is specified as such in the abstract. In principle, the more the better. The illustrated example and claims showed a minimum of 6. That is where Yamaha found the loophole I referred to. The nature of the test signal was the key to the measurement and is still proprietary. A pulse will not work because it is not frequency selective, that is you get no information about the spectral content of individual echoes from it and steady state measurements cannot be segregated with sufficient detail to isolate individual echoes. Also, the width of the pulse can overlap and mask closely spaced echoes from the same direction. The goal of the simulator is to recreate the transfer function thereby reconstructing the reverberant field. There are actualy three categories of transfer fuctions that operate simultaneously, the one that gets into the recording as a scalar field, those inherent in the listening room itself, and the one created by the equipment. The last one must be adjusted to take into account the other two.

Adding a center channel is so obvious it is hardly worth talking about. My old HK A500 had a center channel output with an adjustable front panel gain control and that was in 1962. I have an HT setup with a center speaker in a room about 22 feet wide. I'm not convinced it is necessary or even of benefit given the radiating pattern of speakers I've re-engineered except maybe for dialogue in a movie. It certainly is not like true 3 channel sound would be but with more conventional loudspeakers and sound systems, it may fill in the hole in the middle where the distance between speakers is large.

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Well, THAT's certainly a worthy contribution to the discussion. :)

You gave it a thorough read, obviously.

And you were last interviewed when?

http://www.audioxpress.com/magsdirx/voxcoi...a/mowry1008.pdf

And you have how many papers that have been published by the AES?

http://www.google.com/search?q=Geddes+site:aes.org

[He DOES have a few patents, too.... ;) ]

The only thing publishing my theories and ideas in AES or ASA would accomplish would be to give them away for free. The last time my ideas were made public, even protected by a patent they were stolen. Of what benefit to me would it be to publish my discoveries? Lots of people have patents. Not all patents are valuable or even based on valid science. The only kind of device you must supply a working model to the patent office to get a patent for is a perpetual motion machine. That is because such a device is theoretically impossible. Other than that, the criteria for a patent is that an idea for an invention is novel and not obvious to someone skilled in the art.

One obvious flaw in a constant directivity speaker making it less than ideal for a high fidelity sound system based on my theories is that it has no way to correct for the frequency selective absorption/reflection of the listening room and therefore even if the propagated field is flat, the resulting field the listener experiences which is mostly due to reverberation will not be.

My previous information was not up to date. I've been informed that the line for the world's best speaker now extends beyond the orbit of Pluto. The spiral nebula M31 is just 2 million light years away ;)

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One obvious flaw in a constant directivity speaker making it less than ideal for a high fidelity sound system based on my theories is that it has no way to correct for the frequency selective absorption/reflection of the listening room and therefore even if the propagated field is flat, the resulting field the listener experiences which is mostly due to reverberation will not be.

If you had read the Geddes paper, you would appreciate that fundamental to the concept is minimizing the influence of the room rather than relying upon it and attempting to have it behave as something other than what it is, and working with the real soundstage within the program, instead.

Nobody gives a whit about the concert hall anymore, or striving to generate an artificial one. Today's listeners want to experience the immediacy of the direct field, primarly, with precise imaging, accuracy, and definition, not the diffuse, rolled-off, reverberant field of the 15th row.

Audiophiles and engineers who cater to them today work on the philosophy that they try to reduce or eliminate the effects of room acoustics. They fight the acoustics of the listening room in a battle that is hopeless, it's one they cannot win.

It's easily won, actually, once the burden of max dispersion is eliminated. Manage the room via controlling the dispersion....

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If you had read the Geddes paper, you would appreciate that fundamental to the concept is minimizing the influence of the room rather than relying upon it and attempting to have it behave as something other than what it is, and working with the real soundstage within the program, instead.

Nobody gives a whit about the concert hall anymore, or striving to generate an artificial one. Today's listeners want to experience the immediacy of the direct field, primarly, with precise imaging, accuracy, and definition, not the diffuse, rolled-off, reverberant field of the 15th row.

It's easily won, actually, once the burden of max dispersion is eliminated. Manage the room via controlling the dispersion....

"Nobody gives a whit about the concert hall anymore, or striving to generate an artificial one. Today's listeners want to experience the immediacy of the direct field, primarly, with precise imaging, accuracy, and definition, not the diffuse, rolled-off, reverberant field of the 15th row."

This kind of statement shows remarkable ignorance not only about sound but about music. A concert hall is a room that costs over 100 million dollars to build, takes years to plan, construct, and tune, and often as not is unsuccessful. It is far more than a place of assembly. You could build a high school gymnasium equally large for a tenth that amount. It is an integral part of the sound itself. It's why people who love music as opposed to stereo sets donate huge sums of money to build them and why municipalities, colleges, and other prestigious institutions that have more on their mind than making money are eager to build and have them. The way concert halls work when they are successful, they improve the beauty of sound far beyond anything a raucus harsh sounding stereo system can do, butchering sound so badly it only bears a faint resemblance to real music.

A symphony orchestra reproduced at lifelike sound levels in a home listening environment would deafen the audience and players on the first fff note. Reproduced at less than lifelike levels in a small room, it is a feeble ghost of itself but whether its a 100 piece orchestra or a solo piano, home stereo systems do not produce sound that resembles live music. I wonder how many companies that claim they their products do even own a piano. I wonder how many of them have ever even heard live unamplified music. We get students who come here wanting to study piano, violin, or viola who have never heard a Beethoven Symphony or a Rachmaninoff piano concerto, or serious music by any composer. All they know is trash they hear on their ipods. Music education was pulled out of most public school cirricula decades ago and the results are telling.

BTW, looking at the axial response graphs, I could make a better case for AR3a being a constant directivity speaker than the Summa Cum Lauda speaker. Look at the graphs and see how much more rapidly the treble falls off with angle than the bass and midrange does. There are no musical instruments I'm aware of that propagate sound the way this or any other speaker does. We'll probably never know how really bad they are because they wouldn't dare try one but in a live versus recorded demo, I'd bet this thing would fall flat on its baffle.

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"Incidentally, in his book Toole basically writes off floor and ceiling reflections as inconsequential."

He'd be amazed at what his precious Revel Salon would sound like in a room consisting of four walls, an earth floor, and no ceiling or roof. And what a piano would sound like in such a room.

The legitimate purpose of a constant directivity speaker and the reason it was invented is because it will provide the most uniform sound coverage in a sound reinforcement system for a given space with the fewest and therefore most economical number of units and the highest gain before feedback achieveable. And canned computer programs can easily map out where to place them to achieve their goal. That has become pretty much a fairly exact science. For high fidelity use, these guys must be smoking something. But then people will try anything. I did. :)

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Alas, this old horse knows better than to enter this discussion, but HEY!, I just, for a moment, don't know any better.

To reveal for a brief moment before I slip back into being a speaker devotee: I am an architect.

We had, back in the '70s, 2 choices in the design of venues for live music/stage: 1) Pad & P/A or 2) Do it the old way with slanted celings, curved walls, absorbent and/or reflective materials here/there -- time of sound delivery all figured out (analytic stuff) by acoustical engineers, architects intuitively reconfiguring space...sometimes it hit right on, others were, as Frank Lloyd Wright used to say, for planting ivy. The "old way" was / is considered the superior listening environment for live performances. Pad & P/A was the electronic, affordable ersatz substitute for the real thing. All this is back then.

Recently Alice Tully Hall was expensively reconfigured with new materials just to get the sound right. Reports on the audio quality are glowing. Are there gems in this world of live sound: Bernstein conducting live within spitting distance yet with adjustable reflective surfaces overhead. Wagner, crystal clear, from 2 balconies up without a bit of P/A. Joan Baez in a Chicago basement coffee shop in the '50s a mere 20' away. A Mexican guitarist in an old vaulted abandoned monastery cell playing classical works there because, he said, "the acoustices were great." A Portuguese quartet...but I no doubt begin to bore... I vividly remember them all.

There's another world of amplified everything from the get-go, but that's not my world.

Speakers, amps, CD players...all that, are mere substitutes (not necessarily affordable) for the real thing but they fit in the house and, with great care, in the household budget. Since I have rarely been able to sit and enjoy live performances, these substitutes are marvelous.

And now, back to being an ever-lurking AR3a Speaker Novice....

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The only thing publishing my theories and ideas in AES or ASA would accomplish would be to give them away for free. The last time my ideas were made public, even protected by a patent they were stolen. Of what benefit to me would it be to publish my discoveries? Lots of people have patents. Not all patents are valuable or even based on valid science. The only kind of device you must supply a working model to the patent office to get a patent for is a perpetual motion machine. That is because such a device is theoretically impossible. Other than that, the criteria for a patent is that an idea for an invention is novel and not obvious to someone skilled in the art.

.....

You seem to be stuck between a rock and a hard place. You're patent shy now based on past experience with someone you claim stole your idea almost 30 yrs ago and yet, you don't want to publish as you stated above. So, I guess all your years of hard work and effort are going to the grave with you then? That seems a shame.

I recall a guy who published a neat formula (E=Mc sq'd) and shared his theories who lives in imortality.

I have a copy of the first AES Audio Anthology and admire all those who contributed. Admittedly, they are probably not all millionaires now, but at least they made contributions to the technology. Thiele, Small and Heyser and others all come to mind. They will be remembered for their contributions for many years to come.

How do you want to be remembered? :)

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This kind of statement shows remarkable ignorance not only about sound but about music.

I think it actually represents a fairly realistic assessment of the contemporary audio consumer base. The vast majority of listeners, having never heard live music because they lack the interest to go buy a ticket, are not going to put any priority on reproducing the sound of live music in their homes. Their idea of "accuracy" and "realism" is based on hearing someone play an instrument in a home or in a classroom or on movie theater sound, and it shows in what manufacturers offer in the way of new product and in the number of people who show up here and in other audio forums looking for ways to mod their vintage ARs to make them sound more like Bose Acoustimass.

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AR continued these demos at various hifi show venues well into tne mid-seventies, using the AR-3a and AR-10pi models after the 3 was discontinued. The live music for later demos was a solo guitarist rather than the original chamber music group.

Where or when was the AR-3a used in live-vs.-recorded sessions? To my knowledge it was never in any LVR demonstration. The AR-10Pi was used briefly in the mid-70s for the Neil Grover live-vs.-recorded drum demonstrations (orchestrated by Victor Campos), but that was it after the AR-3 and AR-1. The AR-3 was used in over 75 public Live-vs.-recorded concerts with The Fine Arts Quarter alone, several other concerts with guitarist Gustavo Lopez and in 1966 with the 1910 Nickelodeon. The AR-3 wasn't discontinued until 1973, about six years after the AR-3a was introduced.

--Tom Tyson

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If you had read the Geddes paper, you would appreciate that fundamental to the concept is minimizing the influence of the room rather than relying upon it and attempting to have it behave as something other than what it is, and working with the real soundstage within the program, instead.

Nobody gives a whit about the concert hall anymore, or striving to generate an artificial one. Today's listeners want to experience the immediacy of the direct field, primarly, with precise imaging, accuracy, and definition, not the diffuse, rolled-off, reverberant field of the 15th row.

It's easily won, actually, once the burden of max dispersion is eliminated. Manage the room via controlling the dispersion....

If audio manufacturers can't get their crap to sound like musical instruments, perhaps they could persuade musicians to play inside a box with a round hole cut in one side so their playing will sound like their crap. Finally, realism in high fidelity equipment would arrive. Then the Summa cum expensive will sound like the real thing. :)

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Where or when was the AR-3a used in live-vs.-recorded sessions? To my knowledge it was never in any LVR demonstration. The AR-10Pi was used briefly in the mid-70s for the Neil Grover live-vs.-recorded drum demonstrations (orchestrated by Victor Campos), but that was it after the AR-3 and AR-1. The AR-3 was used in over 75 public Live-vs.-recorded concerts with The Fine Arts Quarter alone, several other concerts with guitarist Gustavo Lopez and in 1966 with the 1910 Nickelodeon. The AR-3 wasn't discontinued until 1973, about six years after the AR-3a was introduced.

Could be I misremembered the 3's as 3a's. It was one of the guitar player demos, and I was a teenager. It's just that I'm pretty sure I recall them saying the demo was for a new speaker, and I didn't think they would have said that about the 3...?

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perhaps they could persuade musicians to play inside a box with a round hole cut in one side

Ever been to the Kennedy Center in DC? It's close. At least it was before they remodeled it, I hear it's better now.

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Ever been to the Kennedy Center in DC? It's close. At least it was before they remodeled it, I hear it's better now.

Couldn't be worse than Radio City Music Hall. My wife and I paid over 100 bucks each and walked out 20 minutes into the concert, the sound was so bad...

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"Nobody gives a whit about the concert hall anymore, or striving to generate an artificial one. Today's listeners want to experience the immediacy of the direct field, primarly, with precise imaging, accuracy, and definition, not the diffuse, rolled-off, reverberant field of the 15th row."

This kind of statement shows remarkable ignorance not only about sound but about music.

Take it up with the engineers doing the recording; their job is to capture the ambience of the hall. If they're successful, it's far better not having the room messing with that.

BTW, looking at the axial response graphs, I could make a better case for AR3a being a constant directivity speaker than the Summa Cum Lauda speaker. Look at the graphs and see how much more rapidly the treble falls off with angle than the bass and midrange does. There are no musical instruments I'm aware of that propagate sound the way this or any other speaker does. We'll probably never know how really bad they are because they wouldn't dare try one but in a live versus recorded demo, I'd bet this thing would fall flat on its baffle.

Yeah, like nobody uses constant directivity for live sound reinforcement.

You need to get out more, really, and speculate less.

Spend $168 to build EconoWave and find out. You can even use AR3as for the woofers:

http://www.audiokarma.org/forums/showthread.php?t=150939

The legitimate purpose of a constant directivity speaker and the reason it was invented is because it will provide the most uniform sound coverage in a sound reinforcement system for a given space with the fewest and therefore most economical number of units....

Well, "DUH," precisely what Villchur struggled to achieve, uniform power response!

Clearly, Villchur's renowned live versus recorded demos in highly reverberant spaces are at the core of AR3a mythology, namely, their accuracy. Toole is correct; they are most notably inaccurate, in fact, by contemporary standards, and we do not listen to them that way. Call it "Clever marketing," if you like, but other descriptives come immediately to mind.

Somebody should ask Toole to define the fallacy inherent in the pillow test. It runs counter to his analysis of how thing are with respect to direct vs. reverberant soundfields in small rooms....

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they are most notably inaccurate, in fact, by contemporary standards, and we do not listen to them that way.

I think it could be argued that they were "notably inaccurate" by the standards of their own time as well, and deliberately so. Consider Villchur's memo on AR-3 and 5 level control settings:

http://www.classicspeakerpages.net/library...vel_contro.html

Villchur stated flat out that the "white dot" settings of the 3 and 5 series had been designed to compensate for what AR believed were the excess high frequency levels of recordings of the time. If we presume that today's recordings no longer exhibit this characteristic, then we are indeed not listening to them in the way that AR originally expected them to be most commonly used, and the settings recommended for "uniform energy output at all audible frequencies" - mid and hi pots turned all the way up plus some added boost from amplifier treble controls - are probably called for. This seems to me to be fairly consistent both with the measurements you've been posting and with the way many people here report that they have their level controls set.

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If we presume that today's recordings no longer exhibit this characteristic, then we are indeed not listening to them in the way that AR originally expected them to be most commonly used, and the settings recommended for "uniform energy output at all audible frequencies" - mid and hi pots turned all the way up plus some added boost from amplifier treble controls - are probably called for. This seems to me to be fairly consistent with the measurements you've been posting.

Indeed, but it runs deeper. The objective of AR design was to replicate the response characteristics of the concert hall in the home, with significantly rolled-off highs. All of the measurements in Allison's 1970 paper were taken with the controls set to "Max," and the power response curves roll off significantly, even when averaged over 8 different living rooms, as I noted above.

His conclusion:

In the light of these findings we believe that typical operating settings for loudspeaker high-frequency balance controls should be well below the settings which produce flat acoustic energy output if the objective is a spectrum similar to that produced at a concert hall seat. In view of the variations found in both living room and concert hall frequency balance, and the manner in which these variation occur, we think that home listeners should be encouraged to make more liberal use of amplifier tone controls.

Certainly, in view of current recording practice, flat electrical response in the playback system is more likely to be wrong than right, particularly if the loudspeaker systems used for playback are able to deliver nearly flat acoustical power output and are adjusted to do so.

They did NOT want flat power response, as that does not emulate the concert hall; we'd have to apply ~10 dB of contoured HF boost to actually achieve flat power response with AR3as. It's also why ARs are often characterized as "Dull-sounding" by contemporary listeners....

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AR LSTs or AR3As with the controls set to maximize mid/hi freq, are too bright for me on nearly any music source, regardless of type of music. If I play most pop LPs, CDs, or MP3s with the speakers so set, the highs are ear-hurting bright. There are some recordings that sound great with the controls for max brightness - which based on the referenced info was flat as per AR. But most do not.

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His conclusion:

They did NOT want flat power response, as that does not emulate the concert hall; we'd have to apply ~10 dB of contoured HF boost to actually achieve flat power response with AR3as. It's also why ARs are often characterized as "Dull-sounding" by contemporary listeners....

My listening experience seems fairly consistent with this. When I have a "concert hall" LP on, every control gets optimized at "white dot" and no treble boost on the amplifier, but for things I'd like to sound as if they're actually in my living room - solo pianists, solo singer + piano, etc. - I want to turn up the mids and highs.

I eventually settled on a default setting of mid and hi pots halfway between "white dot" and full. Amplifier treble control goes a bit down below the "flat" detent for the big symphonies on LP, about 45-90 degrees above it for "small space" stuff on LP and just about everything on CD (my CD player and CDs are the only things in the system that are newer than 25-30 years old). Surprisingly, at least to me, the type of music - classical, jazz, rock, etc. - seems to make no difference in this at all.

I don't know how "accurate" this is, but it sounds "right" to my ears.

BTW, Villchur's note specifically cited the 3 and 5, but I've found this same setup to work about right for the 2ax and 6 as well.

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AR LSTs or AR3As with the controls set to maximize mid/hi freq, are too bright for me on nearly any music source, regardless of type of music.

Consensus appears to be that the AR3a tweeter set at max and the mid at ~ -4 dB (the dot) achieves an optimum balance, as I have verified with the measurements, about as flat as I can set them on-axis.

As others have observed, the highs may still not be "hot" enough, but the mids are adjusted to balance with the woofers using these settings:

http://www.audiokarma.org/forums/showthrea...959#post2570959

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Any thoughts here on the not-unusual phenomenon of a loudspeaker's output sounding closer to what it's attempting to reproduce, when heard from a room or two away?

Does this suggest that the most important aspect of achieving perceived verisimilitude from a distance is reasonable frequency replication and amplitude?

What does this say about the failure to accomplish the same effect, in-room?

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Yes, that's the paper; I'm reading it in the AES anthology. Look at the triptych of 22 locations in 8 living rooms. Even with 1/3 octave smoothing, there's a 6 dB dip at 225 Hz (corresponding to Allison's "discovery," as I recall,) a 3 dB peak at 500 Hz, and a 2.5 dB peak at 2 kHz, with the response falling 13 dB between there and 20 kHz. Even with the mid and high-frequency controls set to Max, there's effectively nobody home above 15 kHz, and it's only "flat" between 600 Hz and 1.2 kHz, one octave. If I listened with 22 ears spread around each of those rooms, that's what I'd be hearing, perhaps, but would you accept that these findings are, well, "contrived?"

In any case, I have auditioned a lot of speakers (and wrote reviews on most), and I can safely say that both wide and narrow dispersion designs can work just fine. I just happen to prefer the room-generated spaciousness of the wide RP jobs, while other people might have tastes that are just the opposite. The primary thing is to get flat, smooth, wide-bandwidth response at the listening position, with opinions about the balance between the direct and reverberant field levels consigned to the realm of taste. My taste says that with most recordings, listened to in most living rooms, wide-dispersion trumps narrow dispersion.

I mostly work with 90° x 50° waveguides. Like you, I find 60° x 40° too beamy for my taste, though it's been a while since I did any critical listening with those.

It's easily seen that if you toe-in 90° CD speakers 45°, the Geddes recommended alignment, all reflections from the proximal sidewall are attenuated at least 6 dB, but they are still sufficient to generate the desired soundfield enhancement without significantly affecting the spectral balance within the directivity window defined by the waveguide beamwidth, no matter how variable the reflectivity of the room. With CD, -6 dB defines the beamwidth, but there is still "spill" outside that pattern, with diminishing SPL at increasing angles. Toe them in less, and you can adjust the relative balance of direct versus reflected soundfield heard by the listener more in favor of reflected, if desired.

In all of this, key to maintaining the flat, smooth, wide-bandwidth response you cite as fundamental is maintaining that requisite spectral balance uniform at all angles; ideally, what is reflected should be the same. If the Beanie Baby collection is at the reflection locus, it's not going to happen, of course, (Soundminded would EQ them out of the room, apparently,) but there's zero chance, even under the best of conditions, that what's coming back from reflections with AR3as is uniform, as what's going out off-axis clearly is not....

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Any thoughts here on the not-unusual phenomenon of a loudspeaker's output sounding closer to what it's attempting to reproduce, when heard from a room or two away?

My take on that is you are hearing the integrated sum of what the system is delivering to the room, with the influence of the room fully integrated as well....

For you, a live-music simulation is not the goal. The goal is something "beyond" what is possible with live acoustic-instrument music. In that case, why not just opt for headphones?

There's no space, no depth. That has to be artificially generated via matrixing. See Part Two: Binaural Techniques here:

http://www.harman.com/wp/pdf/HowManyChannels.pdf

Toole also contends that constant directivity affords the best opportunity for realistic and accurate reproduction, both in multi-channel as well as convential two-channel stereo....

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There are photos of my smaller system there, too, with more conventional speakers that I made myself.
Perhaps you are also a candidate for participation in the $168 EconoWave constant directivity experience....? ;)
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If they were "notably inaccurate" as you say they simply would not have performed as well as they did in those live-vs-recorded sessions. Even if they did not do a perfect job (and we have to remember that the AR-3 did have a 1 kHz woofer/mid crossover point, which is a deficiency the later AR-3a dealt with), they did a good enough job to not warrant Toole's comments as you indicate. The fact that Toole's measurements of the speaker showed deficiencies by his standards indicates that there is something wrong with the Toole approach to measuring speakers and correlating those measurements with how they sound to listeners.

Not really; as I now understand it, AR's "Max dispersion" design approach is intended to exploit spaces in which the reverberant field is dominant to mimic typical concert hall response. When compared to a live musician's performance in such a space, if they got the integrated acoustic energy correct, the reflective character of the space would normalize both in the same way.

In more conventional listening spaces, where the direct field dominates, it doesn't work quite so well, apparently:

http://www.classicspeakerpages.net/library...llison_to_hoff/

AR similarly rationalized CBS Lab's analysis of LST in 1971:

http://www.classicspeakerpages.net/library...ort_on__12.html

Toole's metric, like those of CU and CBS labs, relies upon anechoic response measurements, and the power response derived therefrom is primarily indicative of how faithfully off-axis response retains the axial spectrum balance, and all that implies. Ideally, the axial response is smooth and flat, and the power response a similar approximation thereof.

AR's "Acoustic Power Output," on the other hand, is measured in a reverberant chamber and represents an integrated summation of both direct and reflected response therein, masking the true direct response. Other than in the Allison paper cited and as independently measured by others, AR judiciously avoided disclosure of the actual frequency response of their systems....

I am going to assume that you did take time to go look at the photos, and that your comment is just a friendly joke.

I assumed, erroneously, perhaps, that as a long-term audio professional, despite now being retired, you might have a continuing interest in exploring the technology.

Now with 5000 posts, 750 attachments, 200,000 views, and 40 systems "Officially" completed thus far by AK DIYers, EconoWave is by no means a joke....

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AR LSTs or AR3As with the controls set to maximize mid/hi freq, are too bright for me on nearly any music source, regardless of type of music. If I play most pop LPs, CDs, or MP3s with the speakers so set, the highs are ear-hurting bright. There are some recordings that sound great with the controls for max brightness - which based on the referenced info was flat as per AR. But most do not.

That's probably because you have most of your original tweeters functioning and your listening room is not an anechoic chamber. Try feeding the sound of a typical DVD into your system and you'll probably find that even with your speaker pots set as they are you are turning down your treble control a lot (I find myself turning both my treble and my bass a lot for DVD sound).

If my ship ever comes in and I can afford a home with a dedicated home theater with carpeted floors, heavy drapes and ceilings covered with Alphasorb, I may finally have a use for speakers that measure flat in an anechoic chamber. ;)

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