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Old 9th June 2007, 06:39 AM   #1001
MBK is offline MBK  Singapore
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Michael,

re: the 3 dB/decade falloff in line arrays, I never properly understood where it comes from. I suspect cancellations from comb filtering.
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Old 9th June 2007, 09:21 AM   #1002
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Default Re: cinema "sound" reality check

Quote:
Originally posted by mige0
To give an even more correct explanation the inverse square law IS valid also with lenses in use. At double the distance you always get the illumination given the SAME lens applied. The size of the picture gets 4 times bigger then of course. (no cylinder wave front with light !)
Yes, right! With the same lens. Here I was giving Lenard a hard time and I was even less clear.

I was thinking more in practical terms. If you need to fill a 30 foot screen, it doesn't matter if the projector is 40 or 150 feet from the screen. Why? Because you'd use a different lens. The closer to the screen, the wider the angle of the lens needs to be, and visa versa. A lot of people will look at the long throw distance and talk about how much light is lost because of the "inverse square law." But of course it isn't lost, because the longer focal length lens is squeezing the light into a tighter beam. So the same amount of light hits the screen either way, minus some small air losses.

The image will look a little different, though. I suppose because the image coming from a long distance has rays that are more parallel than the wider angle lens up close. There is less hot spotting on the screen and more even off axis viewing with the long throw distance.

This may be what Lenard is getting at with his large sound source. How flat the wavefront is may effect how large we perceive the source to be. I certainly prefer large sources. Small speakers always sound small to me, no matter how loud.
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Old 9th June 2007, 12:55 PM   #1003
MBK is offline MBK  Singapore
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Somehow I doubt that the shape of the wavefront has something to do with perception of size. The mechanism that gives us the apparent source width of a low frequency source according to conventional wisdom, AFAIK is the envelope of the LF signal. In other words it is the higher frequency part of a real world signal, and its spread-out arrival time to our ears if the source is large, that conveys the size.

This also explains why stereo sounds larger than mono (directional clues nonwithstanding) and why line sources may seem to sound larger than life. Both have to do with large arrival time differences, in stereo it's L vs. R, in a line source it's the higher vs. the lower part of the line. And even a mono source that is physically large should sound "larger" than a small mono source.

So why would horns sound "large" then? Same thing: mouth diffractions from a physically large mouth, generating arrival time differences for largeness perception. An artefact in other words.
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Old 9th June 2007, 03:46 PM   #1004
Salas is offline Salas  Greece
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Quote:
Originally posted by MBK
Michael,

re: the 3 dB/decade falloff in line arrays, I never properly understood where it comes from. I suspect cancellations from comb filtering.

To my perception the theory of a cylindrical wavefront is not the explanation. Its the geometry of the array's dispersion. When you squeeze your energy vertically to fit it almost only in a horizontal plane in the near field, don't you just boost it by 3dB? You skip a dimension of expansion. Here you go. Then you start losing control vertically when the distance gets longer and you cross to the far field, where you get square law -6dB per doubling of distance spherical expansion business as usual.
The lower the frequency the shorter the -3dB per doubling of distance near field definition. λ divides the near field array length. So you lose 3dB per octave going lower in frequency at a given distance within the near field. About 9-10dB SPL less per decade. Allow for +6dB gain when forming a large enough front baffle after setting the array (+3B addition of sources, + 3dB vertical baffle boundary reinforcement) . You end up to -3dB per decade. Is it coincidental that the rule of thumb is 2 subs per array satellite cabinet in reinforcement practice? Thats +3dB of lows. No wonder all line arrays sound tipped up in balance when playing alone, and mountains of subwoofers are there to support their mid-high long throw avoiding the use of excessive EQ not to lose precious top available SPL. Music for the masses...

See a single EAW KF-730 satellite measurement. It is crossed and processed for fitting in a usually no less than 8 cabinet array.
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Old 9th June 2007, 04:05 PM   #1005
MBK is offline MBK  Singapore
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I can only partially follow.

- The nearfield-farfield transition vs. frequency, good point, but still in a domestic floor to ceiling array the line is supposed to be of "infinite" length (including floor and ceiling mirror images, and those work best for the LF), and all listening at all relevant frequencies is supposed to be in the line array nearfield.
- the +6dB from forming the array in the first place is a baffle step and source addition effect that only occurs once, and likely spread out over a single octave that yields those 6 dB - not a gradual effect over two decades.

It seems more likely that the nearfield-farfield transition is very gradual, and that the "infinite" assumption only partially holds, ending up with -3 dB/decade in practice. But maybe the line array specialists here can give us the official theory on this...
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Old 9th June 2007, 04:49 PM   #1006
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Quote:
Originally posted by MBK
I can only partially follow.

- The nearfield-farfield transition vs. frequency, good point, but still in a domestic floor to ceiling array the line is supposed to be of "infinite" length (including floor and ceiling mirror images, and those work best for the LF), and all listening at all relevant frequencies is supposed to be in the line array nearfield.
- the +6dB from forming the array in the first place is a baffle step and source addition effect that only occurs once, and likely spread out over a single octave that yields those 6 dB - not a gradual effect over two decades.

It seems more likely that the nearfield-farfield transition is very gradual, and that the "infinite" assumption only partially holds, ending up with -3 dB/decade in practice. But maybe the line array specialists here can give us the official theory on this...
My point is that the near field shortens the lower you go in frequency and this correlates with your (right by my reckoning) assumption that the transition is gradual in reality and not infinite. In practice I have never experienced anything infinite or anything mathematically absolute in nature. All stuff averages out in certain tendencies and no absolutes or infinity.
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Old 9th June 2007, 06:20 PM   #1007
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Default parts quality at line vs speaker level

Quote:
Originally posted by Lynn Olson


12 years later, when I was designing the Ariel and its 5 dB greater sensitivity, I discovered that the crossover was exquisitely sensitive to parts quality, especially the capacitors in the HF section. The Sprague 730P and Hovland were the only ones of the day that seemed transparent enough, and the Hovland irritatingly enough required a crossover adjustment to maintain the same subjective balance. I was not happy to make this discovery - parts tweaking is one of my most annoying and least rewarding hifi activities, something I'd rather not do at all.

Nowadays we have many more choices but cap coloration is still an awkward business in a high-resolution speaker - and I found out the hard way that the higher the efficiency, the less forgiving the speaker is of mediocre crossover parts. You can throw junk at a 86 dB/metre speaker and it isn't that audible. 92 dB/metre and up is a different ballgame. I'm expecting 97~100 dB/metre to be completely unforgiving of crossover parts quality - a severe incentive to keep complexity down. These drivers are expensive - the idea of wasting that much money on crossover parts isn't attractive at all.
Do you find the sensitivity to parts quality less of a factor in line-level electronics?
I would be interested, Lynn, in your views on the trade-offs inherent in active crossover implementation vs passive.

Emerald Audio seems to be doing well with the DBX unit used this way, presumably stock :
http://www.dbxpro.com/PA/PA.htm
I wonder whether a little tweaking of the analog stages would produce a better result than more expensive components in a passive EQ.
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Old 9th June 2007, 10:31 PM   #1008
mige0 is offline mige0  Austria
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Hi

Quote:
Michael,

re: the 3 dB/decade falloff in line arrays, I never properly understood where it comes from. I suspect cancellations from comb filtering.


This page gives me some answers. I read it some time ago but forgot. Shame on me!

http://meyersound.com/support/papers...ray_theory.htm
"http://meyersound.com/support/papers/line_array_theory.htm"



Quote:
The pro audio ELS just made me thinking abut some side aspects of highly asymmetric directivity loudspeakers like line arrays, small long ribbons / AMT's or the multiple speakers of the current OB project radiating within the same frequency band.

One of the effects here is that the SPL does not seem to fall at 6 dB with double distance but only around 3 dB du to the cylinder shape of the wave front.
At least for "line arrays ....and the multiple speakers of the current OB project radiating within the same frequency band" I went completely wrong as it seems ! We need quite some height compared to wave length AND very low separation distance between drivers to get that cylinder waveform behaviour.
For long or closely stacked ribbons and AMTs it might work in the upper frequency range.



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Greetings
Michael
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Old 10th June 2007, 03:40 AM   #1009
MBK is offline MBK  Singapore
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Interesting article indeed, although they also have their priorities: as a prosound company they are interested in very long throw systems, and there, especially without boundary reinforcement (stadiums...) the assumptions break down.

In domestic line arrays some of the assumptions may hold "close enough" up to 4-6 m listening distances (as their own table shows: the bass doesn't hold up but at 250 Hz+ the dB loss is close enough to 3 dB / distance doubling). But yes, from all calculations I have seen the line does have to be floor-to-ceiling, with a practical limit of at least 75% floor-to-ceiling if one assumes good reflection by the boundaries, at least for the bass and mids. For the Highs something like 24-36" line length might be enough at 1.5 kHz+, from memory, to preserve line character up to 4 m listening distance.

But all this matters most for achieving the mythical 3 dB/distance doubling SPL loss. If only naturalness and correct source size perception are concerned, then simply making the source larger for lower frequencies, might be good enough.
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Old 10th June 2007, 04:55 AM   #1010
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Default Re: parts quality at line vs speaker level

Quote:
Originally posted by Russell Dawkins


Do you find the sensitivity to parts quality less of a factor in line-level electronics?
I would be interested, Lynn, in your views on the trade-offs inherent in active crossover implementation vs passive.

Emerald Audio seems to be doing well with the DBX unit used this way, presumably stock :
http://www.dbxpro.com/PA/PA.htm
I wonder whether a little tweaking of the analog stages would produce a better result than more expensive components in a passive EQ.
Well, this is where individual perceptions enter in. Earl Geddes, who is a talented speaker designer and sharp theoretician, believes Costco-quality electronics are good enough to exhibit at the RMAF, and anyone who thinks otherwise is fooling themselves, or worse, trying to sell snake oil to the gullible.

Siegfried Linkwitz is firmly in the solid-state camp, and from his perspective, the better-grade opamps are transparent sounding enough to use in many different locations in the active crossover and equalizer.

I'm pretty much on the other side of debate - and trust my perceptions, having designed audio gear for some thirty-plus years. I find the addition, or substitution, of a single part in the signal chain quite audible. Sometimes the substitution makes the difference between high-quality sound and being completely unacceptable for high-fidelity use.

When I first built the Amity power amplifier in 1997, I was quite disappointed by nearly every linestage I auditioned. It was simple to test whether they were altering the signal, since I had the choice between the DAC output going right into the power amp (without attenuation) and the linestage under test. The Jeff Rowland was one of the very few not to grossly degrade transparency and add annoying tonal colorations.

Most transistor linestages were grainy and two-dimensional, and most tube linestages were bass-heavy with lots of tubey jukebox colorations. Again, this was by flipping a switch and doing a direct A/B comparison - nothing involving multi-hour listening sessions with a long checklist. It was obvious in seconds or at most a minute or two.

It took me a while to come to terms with the brutal transparency of the Amity amplifier - it was forgiving of musical styles, and brought out the best in many speakers, but was utterly unforgiving of poor linestages or substandard DACs (mine was a heavily modified PCM-63K based unit). We're talking of very brief yes-no comparisons here - a linestage was either listenable, or not, nothing too complicated subjectively.

So I'm on the other side of the fence in terms of electronics. I find most electronics unlistenable, and that certainly includes just about all high-priced audiophile products. I didn't design the Amity, Raven, Aurora, and Karna for fun - I didn't care for the sound of commercially available electronics, and was forced to find my own way.

Given how dissatisfied I am with linestages, active crossovers sound worse, much worse. The tube-based ones have gross design errors in the 12AX7 tube section, with tubes operated at far too low of a current to drive the reactive loads of the crossover. The opamps in the solid-state ones are trash, the kind of thing that belongs in boomboxes. I find these things unlistenable, enough so I can't honestly assess the improvements from the EQ and time-alignment.

This seems to be a matter of individual perception. Some people are completely happy with generic opamps with 5V/uSec slew rates and quasi-complementary Class AB output sections. To me, they limit the sound to no better than home-theater receiver quality - barely hifi, down at the MP3 quality level.

When I see slow, generic opamps, with electrolytic coupling caps, I don't usually even bother to turn it on. And unfortunately, that's what you see in $5000 SACD players and "audiophile" linestages, equalizers, and crossovers. Some of us are sensitive to this, some of us aren't. More power to you if opamps and solid-state works for you - you have a lot more choices open to you.

The perception of parts degradation in passive crossovers vs active crossovers is also individual. There's been a big fad in Europe over the last several years of using the Behringer DCX, either modified or stock, in high-end time-equalized horn systems. At the 2004 ETF, I heard some very high-quality horn systems, and when doing a direct comparison between passive and DCX crossovers, I felt the DCX ruined the sound, flattening out all the subtle details and giving an unpleasant metallic coloration to everything. With the passive crossover, at least things sounded more natural.

Now most of the colorations I objected to were at mid and higher frequencies - at lower frequencies, such as 200 Hz or lower, the gizmo sounded fairly benign, and not too offensive. But from 200 Hz on up, it sounded no better than an iPod - and I own one, and know what they sound like.

But I have to keep returning to individual perception. Some people, who I respect as engineers and designers, are completely happy with generic transistor gear. Most of this gear sounds to me like the Bose stereo in my Acura. There is transistor gear I've heard that I liked, but it tends to be rather exotic and well off the beaten path. I'm kind of picky about tube gear as well - Audio Research, Cary, Jadis, VTL - uh, no thanks.

In Europe I heard the DCX with completely replaced, non-solid-state analog sections, and all-new power supplies that don't use switching supplies. All of that work lifted it from iPod to generic CD player quality - OK for bass, most certainly not for full-range use. I've been pretty underwhelmed with the other digital signal benders as well - they sound like generic CD players, no better than that, making it very hard to assess the other sonic benefits. I'd rather listen to a Dynaco Stereo 70 driving an ordinary passive crossover.

By now, I'm sure I sound like a complete nut to people who listen to popular solid-state equipment and like it. What can I say. I've met the reviewers from the audiophile magazines, and heard some of their systems. That is not the direction I am going - but neither am I heading towards 3-watt SET amplifiers and Altec and JBL vintage speakers.

So I can't endorse any of the digital equalizers, at least not in the midband and higher frequencies. For signal conditioning for the bass array and subwoofers, yes, that's what I plan to do myself. The DBX unit looks like a good choice for that application. For the widerange driver and tweeter (either CD or ribbon), it's going to be transformer-coupled triodes from DAC/phono preamp to the 1.2~1.8 kHz passive crossover, just as it is now for the Ariels.
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