Beyond the Ariel

Thanks for the tip about GR-Research and Rythmik; if they're at the RMAF, I most certainly will give them a listen. Decent subwoofers are kind of rare ducks.

Yes. Please listen. The V1 is a new Gr Research speaker using a pair of servo subs and a 12" pro coaxial speaker in an open baffle arrangement. Actually, it is pretty similar to some stuffs discussed here.
 
The only downside I see with the DR280 is the requirement for time alignment, which would put the AH425 pretty far back - probably actually behind the DR280, on its own stand.

I don't know much about time alignment of horns, but if the objective is to time align the physical location of the speaker magnets, then the speaker magnet of the DR280 is probably about 12" - 15" from the front of DR280, as the horn fires backward, does a U turn, and exists the front.
 
Hmm - all good things to think about.

Going backwards, Point (3): the Zout of the Karna amplifier is slightly less than 1.5 ohms (on the 8-ohm tap). No, I am NOT re-designing the Amity or Karna amplifiers, which have source impedances comparable to many SET amplifiers. That project is done.

The most likely crossover with the horns I have is between 700 and 900 Hz. Referring back to Point (2), that would mean a pretty small baffle, or rather, a pretty small driver - maybe 8" or so? The efficiency is now getting pretty low, not to mention power-handling. The alternative is a much bigger horn for the compression driver - um, no thanks.

Going back to Point (1): Increase Sd - yes, the way to go, solving many problems with distortion, power-handling, and headroom. It conflicts with Point (2), though - a baffle peak at 800 Hz or preferably higher.

The way to square the circle (staying with OB for now) is overlap or frequency-split a large and a small baffle. So - a single efficient 8-incher (this is starting to look like an Orion), with a large (at least 2x15) bass array beneath it. That would cover the desired 80~800 Hz range, at the expense of complexity, since EQ is needed everywhere in the operating range, and the EQ preferably done at the low-level stage ahead of the power amplifier(s).



So it's entirely possible the BFM OmniTop 15 or 2x15 would be a disaster for HiFi applications. Much too thin, much too resonant, too much the PA speaker.

..Their virtues aren't in what they're doing wrong - the horrific response, etc. - but in something they do right.

If the Olimpia Audio W416 is at the RMAF, it should be interesting to audition it - if my low-fi recordings don't get me thrown out the room first (which is what happened - twice - at last year's show).



I didn't realize zout was less than 2. It will react, but not as much as the typical "3 +" of SET's..


I also didn't realize that you are having problems with the (open) baffle at higher freq.s as well. With edge treatment and a much more rectangular shape (than square), will usually remove these issues. You could also go to a 4 driver midrange vertical array. At these low freq.s such an array doesn't act like an array, only a point source, BUT the array tends to average-out the baffles contribution. (..of course again I'm referring to a purpose built mid from around 200 up to 700-900.) The Faital Pro MSN 12-80 should work well for this. Still though, it *is* a lot of effort for what's basically the bandwidth of a small mini-monitor.

In other words for an open baffle config. I'd advocate something like (top-down):

4 x 5" vertical array Faital Pro MSN 12-80 on top - baffle 6" x 22" low-pass crossover near 800 Hz.
Ribbon.
Horn.
2 x 15" vertical array Jenesen neo 15-150's on the bottom - baffle 19" x 32" floor loaded with a low-pass crossover near 200 Hz.

I think that without eq. gain compensation, such a design could hit 100 db 1 watt 1 meter in-room. But I don't think it would look anything like the Orion. ;)



The problem I have with the omni-top isn't the horn per-se, but rather that the horn mouth and/or flair isn't large enough *not* to be audibly noticed. Moreover not only do you have all the additional problems with cabinet vibration, but you also *hear* those idiosyncrasies far more with the horn design, including the quality of the driver (..which may not have anything detectable with modern measurements). While you can hear what they do right, after a while you also can't hear anything but what they do wrong. (..rather like lowthers.) The driver suspension in particular starts to become critical, and NOS drivers start having a significant advantage. I *also* have a problem with any design that puts a bass reflex that near the average that high up in freq.. Delay/Phase goes to "pot" with these designs and damping factor actually is noticeable. With horns like these there is no free-lunch here, in fact it's where I started, but finally gave-up on in frustration. :headshot: (..just trying to save you some agony.. :p )

I think the A7 is similar to the more modern Exemplar bass-bin (that was based on the A7), i.e. around 27"W x 33"H for a single 15" driver. That will get a slightly lower "cut-off" (especially with the floor coupling like the Exemplars), but these designs were at best pushing 102 db near the average with a lower tunning freq. for the vent (when compared to the omni-top), and with the bass region significantly below the average (even considering room-gain).

I remember first seeing the Exemplars and thinking: "yeah, that's probably about the best you can do for a front-loaded mid-bass horn in a domestic setting that doesn't use the rooms walls". (..the La Scala's and Klipshorns sound wrong in the upper mid.s to me.) The one design that goes beyond domestically acceptable, (but actually does extend below the lowest voice range), is the Living Voice Air Partner.. but not to many want a pair of SubZero-sized horns in their living room. :eek: :D


Olimpia Audio is a very small Italian retailer. Unfortunately I doubt they'll be at any US show, and likely only at one or two Italian shows. :eek:

http://www.olimpiaudio.com/Prodotti/Cleveland/Cleveland.html

CORRECTION - it was the Neo version of the W416 I was referring to.
 
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Yes. Please listen. The V1 is a new Gr Research speaker using a pair of servo subs and a 12" pro coaxial speaker in an open baffle arrangement. Actually, it is pretty similar to some stuffs discussed here.

Yes, I'd provided a link that coincided with removing the rear chamber of the compression driver here:

http://www.diyaudio.com/forums/showthread.php?t=90804&page=40

An interesting design, with the lower freq. dipole panel's use *very* similar to what I'd suggested some where back in this thread. ;)
 
Physical voice coil alignment does NOT mean that the system will be acoustically aligned. The acoustic center and phase of each driver needs to be looked at. In addition, the crossover will also shift phase. If you look at the Danley Sound Labs Synergy horns, you will begin to understand this. The Danley horns are phase coherent through their entire range. This eliminates phase distortion and destroys instrument and voice intelligibility. When I see people go to great lengths to physically “time align” their drivers within a micrometer, I just have to laugh.
 
Physical voice coil alignment does NOT mean that the system will be acoustically aligned. The acoustic center and phase of each driver needs to be looked at. In addition, the crossover will also shift phase. If you look at the Danley Sound Labs Synergy horns, you will begin to understand this. The Danley horns are phase coherent through their entire range. This eliminates phase distortion THAT destroys instrument and voice intelligibility. When I see people go to great lengths to physically “time align” their drivers within a micrometer, I just have to laugh.

Sorry for the typo. See above correction
 
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Funny, I read right over your typo and knew what you meant, didn't even see it.

Acoustic centers are not always at the voice coil, for sure - but how do you measure them? I'm not talking in theory, in practice. I can measure the offset with the auto-align of the DCX2496, but where and how is it measuring? Phase and offset can also be measured with HOLMImpusle ( and other softwares) but how to know where they actually line up?

The practical aspects of measuring phase and time alignment are worthy of a long thread.
 
To give credit where it is due, the math behind much of Gary's work comes from our own John K at this site. http://www.musicanddesign.com/index.html So, if you want to explore other cardioid bass schemes, make sure you get the real information.

Bud


One other minor point to Earl. Monopoles are not unique in excitng room pressurization. Any source which has finite volume displacement below the room fundamental will cause some degree of pressurization. Cardior woofer systems fall into this catagory.


If one would like to side step the dampening material issues with cardioide bass wouldn't it be an option to substitute the dampening material by a second driver firing to the rear.

How would that rear driver for cardioide radiation have to be signaled? Could this be done so to get a smooth transition from cardioide in the lower department smoothly towards diopole some higher up in freqeuncy?

Has this been tried or investigated? Would we *necessarily* end up with the same shortcommings as inherent to the URPS principle (high distortion high power consumption)?

Michael
 
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Hello,
A good tool in order to refine the alignement is CSD or spectrogram.

Here attched an example from my mesurements.
In that case the optimal alignment is obtained for the graph labelled "100mm"

Best regards from Paris, France

Jean-Michel Le Cléac'h


Funny, I read right over your typo and knew what you meant, didn't even see it.

Acoustic centers are not always at the voice coil, for sure - but how do you measure them? I'm not talking in theory, in practice. I can measure the offset with the auto-align of the DCX2496, but where and how is it measuring? Phase and offset can also be measured with HOLMImpusle ( and other softwares) but how to know where they actually line up?

The practical aspects of measuring phase and time alignment are worthy of a long thread.
 

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If one would like to side step the dampening material issues with cardioide bass wouldn't it be an option to substitute the dampening material by a second driver firing to the rear.

How would that rear driver for cardioide radiation have to be signaled? Could this be done so to get a smooth transition from cardioide in the lower department smoothly towards diopole some higher up in freqeuncy?

Has this been tried or investigated? Would we *necessarily* end up with the same shortcommings as inherent to the URPS principle (high distortion high power consumption)?

Michael

Yes I did this a long time ago. The rear driver needs to be delayed. This came be done with an all pass delay filter. What you gain is a better cardioid response over the useful rand. But at the cost of an extra driver and the power to drive it. In theory you could design the filter to undergo a transition from cardioid to dipole but you only get 2 to 3 octaves of true dipole or cardioid response form any set up.

Another approach is what I did with my Mini design. The panel (midrange ) is dipole. The woofers are monopole. When correctly positioned the crossover between the dipole panels and monopole woofers yield a smooth transition from dipole mids to cardioid in the crossover region to monopole low frequency. There are a number of ways to attack this in the crossover.
 
Yes I did this a long time ago. The rear driver needs to be delayed. This came be done with an all pass delay filter. What you gain is a better cardioid response over the useful rand. But at the cost of an extra driver and the power to drive it. In theory you could design the filter to undergo a transition from cardioid to dipole but you only get 2 to 3 octaves of true dipole or cardioid response form any set up.

Another approach is what I did with my Mini design. The panel (midrange ) is dipole. The woofers are monopole. When correctly positioned the crossover between the dipole panels and monopole woofers yield a smooth transition from dipole mids to cardioid in the crossover region to monopole low frequency. There are a number of ways to attack this in the crossover.

Thanks a lot – I really hoped you have looked into that at one time.
Sad to hear that bandwidth can not be extended – I was looking for something covering roughly 20- 500Hz or so.
This would have paid for the additional costs of a second driver and amp easily.

Actually I thought we could extend the dipole part up to where usually the limitation of the first dipole peak comes in.

Why is it that you found out 3 octaves as max bandwidth?

Michael
 
Can an accelerometer be used to measure spider/frame resonance of a mounted driver in its cabinet correctly? I realize that the cabinet will be a big influence on this measurement, but curious if this would help to address driver/cabinet resonance to a higher degree of accuracy, or will it matter?

This can help to trace the problem yes, but it doesn't solve it. I have found that the biggest culprit in this frequency region is the spider resonance and/or the magnet against the frame. Frames appear to be quite rigid, but hang a super massive magnet cantilevered off of it and a resonance can occur in the audible spectrum. And spiders are notorious for resonances. I have found that even the best transducer companies don't have a good handle on these problems. So back to the question. Those tests help to find the problem, but short of having your own drivers built they aren't going to help solve it.
 
Acoustic centers are not always at the voice coil, for sure - but how do you measure them? I'm not talking in theory, in practice. I can measure the offset with the auto-align of the DCX2496, but where and how is it measuring? Phase and offset can also be measured with HOLMImpusle ( and other softwares) but how to know where they actually line up?

The practical aspects of measuring phase and time alignment are worthy of a long thread.
It's pretty easy with ARTA to at least get relative offsets if not absolute acoustic centers. It lets you plot excess phase and you can adjust the time you subtract from the impulse response until the excess phase is flat and near zero at the crossover frequency. Do that for both drivers and the difference in the two times is the AC offset.
 

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Funny, I read right over your typo and knew what you meant, didn't even see it.

Acoustic centers are not always at the voice coil, for sure - but how do you measure them? I'm not talking in theory, in practice. I can measure the offset with the auto-align of the DCX2496, but where and how is it measuring? Phase and offset can also be measured with HOLMImpusle ( and other softwares) but how to know where they actually line up?

The practical aspects of measuring phase and time alignment are worthy of a long thread.

I don't worry about the actual acoustical center. I really don't know of a way to measure the acoustic center. I just measure the phase and make sure that the two drivers/horns/waveguides/etc. are phase coherent at the crossover point with the crossover components installed. I think of passive crossover components more like transfer function modifiers instead of acoustical filters. I use whatever slopes and orders it takes to get the phase where it needs to be while maintaining a nice flat power response.

Rgs, JLH
 
Hello JLH,

Your method is quite conventional but as phase coherence at the crossover frequency only doesn't insure low phase distortion, the pulse response will not be as optimal as with other methods.

(Note: as only a small part of the people is sensible to phase distortion it may be sufficient for your goals.)

Best regards from Paris, France

Jean-Michel Le Cléac'h


I don't worry about the actual acoustical center. I really don't know of a way to measure the acoustic center. I just measure the phase and make sure that the two drivers/horns/waveguides/etc. are phase coherent at the crossover point with the crossover components installed. I think of passive crossover components more like transfer function modifiers instead of acoustical filters. I use whatever slopes and orders it takes to get the phase where it needs to be while maintaining a nice flat power response.

Rgs, JLH
 
Thanks a lot – I really hoped you have looked into that at one time.
Sad to hear that bandwidth can not be extended – I was looking for something covering roughly 20- 500Hz or so.
This would have paid for the additional costs of a second driver and amp easily.

Actually I thought we could extend the dipole part up to where usually the limitation of the first dipole peak comes in.

Why is it that you found out 3 octaves as max bandwidth?

Michael

I'm basically referring to the useful range below the dipole peak. The first order gradient roll off below this peak means that for flat response you will need roughly 20dB of equalization to extend the response 3 octaves. More if a low Q driver is used and the poles of the driver must be shifted around. You can also look at this a the sensitivity is dropping 6dB/octave below the dipole peak. Some passive designs try to get around this by using higher Q woofers but there is still a lot of power being thrown away in the crossover.
 
Hello JLH,

Your method is quite conventional but as phase coherence at the crossover frequency only doesn't insure low phase distortion, the pulse response will not be as optimal as with other methods.

(Note: as only a small part of the people is sensible to phase distortion it may be sufficient for your goals.)

Best regards from Paris, France

Jean-Michel Le Cléac'h

I would say that phase coherence through the crossover is not the issue. The issue is that the phase variation through the corssover should match that of the acoustic targets. If the phase difference between mid and tweeter is supposed to be X degrees through the crossover, then that is what it should be for proper alignment and correct summing. If you want good impulse response as well then it is necessary to choose a crossover which had that properity to start with, i.e. any of the family of transient perfect crossover, the most common (though must useless) is the 1st order Butterworth.
 
Hello John,

This view of yours only take the mathematics as point of view and is too much restrictive. Psychoacoustics should also be taken in account. I consider myself that low phase distortion below 4000Hz is enough because IMHO over 4000Hz phase distortion is far less audible.

Here in France we call quasioptimal those crossovers showing low phase distortion below 4000Hz (with additional constant "in coincidence" response, a charcateristics we introduced 10 years ago). We know only 4 of those quasioptimal crossovers at the moment (2 of them being "Le Cleac'h crossovers", one being Francis Brooke's crossover and another designed by Dada of the forum Audax).

Best regards from Paris, France

Jean-Michel Le Cléac'h



I would say that phase coherence through the crossover is not the issue. The issue is that the phase variation through the corssover should match that of the acoustic targets. If the phase difference between mid and tweeter is supposed to be X degrees through the crossover, then that is what it should be for proper alignment and correct summing. If you want good impulse response as well then it is necessary to choose a crossover which had that properity to start with, i.e. any of the family of transient perfect crossover, the most common (though must useless) is the 1st order Butterworth.