tryonziess said:
Vega is a star. Its a massive star that burns very hot very fast. My race emigrated to Vega about 100,000 years ago from the Small Magellic Cloud, since Vega is only about 300 million years old and will never be around long enough to create live on any planets. We had to terraform the planet.
As to the loudspeaker system, there is a whole discussion about it here on diy at
http://www.diyaudio.com/forums/showthread.php?s=&threadid=101012&highlight=
The Calipso Array has been developed on a model that includes the following essential characteristics:
1. Three-Way design.
In point source speaker systems, a two way design is often considered to be an inadequate compromise. Much of audio has gone to two way designs with a single sub woofer. Vance Dickason in his 6th Edition of The Loudspeaker Cookbook has indicated that the use of dual woofers that are capable of subwoofer frequencies produces a significantly better sound image than a subwoofer. In this design a large 12 inch high mm Xmax woofer is used with each side, and it crosses to the mid ranges at about 165hz. While mine are not currently in the correct size box, the correct size is a 6-8 cu ft vented box.
2. No Crossover in the most sensitive areas of human hearing.
Rod Elliot at Southwest Audio(http://sound.westhost.com/bi-amp2.htm ) makes the case better than I:
“It is not at all uncommon to see systems where the crossover frequency is set right in the middle of what I call the "intelligence band". This is the range of frequencies from 300Hz to 3600Hz, and is extremely important from a psycho-acoustic point of view.
“It is no accident that this is the range of the telephone system (and has been for many years). If we are only to hear a limited range, then this band of frequencies is by far the most important. Just from this we can recognize a person's voice, which musical instrument is being played (even bass instruments!), and - more importantly - what is being said. It contains nearly all the "intelligence" of the sound, which is to say that if this band is "corrupted", intelligibility is greatly reduced.
“So why do speaker manufacturers insist on placing their crossover frequencies within this band of frequencies? The public address (PA) systems used by many rock bands are a case in point - how often does one find that the vocals are completely unintelligible? Mind you, it may also be the case that the band's lyrics just don't make sense, but that's another story altogether.
“Often this occurs because the system is so loud that the amplifiers are clipping badly, but even at lower levels it is quite common. Place a common-or-garden crossover filter right in the middle of the "intelligence band" and this is exactly what will (and does) happen. With phase aberrations and cancellations, this most important frequency range becomes muddied and indistinct causing loss of intelligibility - not only on voice, but instruments as well.
“The effect is also noticeable with some hi-fi speaker systems, except that it usually less pronounced, and it is far less likely that the amplifier will be driven to clipping. Reviewers will often say of a speaker that the vocals seem veiled, or that there is noticeable coloration of either male or female vocals. These effects are often caused by the effects of phase shift around the crossover frequency, coupled with the fact that the crossover frequency falls right in the middle of the intelligence band.”
What Rod says is very hard to do with point source speakers, but with line arrays, the midrange speakers are often loafing along, and if they have the range, pulling the wagon as a group of horses, they can do the job admirably.
Covering this range adequately above 1600hz requires a speaker that is below 4 inches in diameter and, IMO, needs an Xmax for the lower range of at least 3mm.
3. Use Electrical Crossovers.
Again, passive crossovers are a DATED TECHNOLOGY. Passive’s introduce so much negative into the sound that they should not be there. And the cost of a top of the line passive cross(with the increased cost of copper) and a electronic is almost the same. Read Rod Elliot’s papers on Bi-Amplification and electronic crossovers at http://sound.westhost.com
The only reason for not using them is that you’ve spent years learning how to design them and you aren’t giving that up, or you are in the business of selling speakers and you aren’t willing to sell amps and crossovers too.
4. Separate the midrange speakers from all other influences.
Bad coloration of sound, in my opinion, is caused by other sound that the direct radiation coming through the speakers. This means that you have to eliminate ANYTHING that goes out the back of the speaker, and anything that might be in the box behind the speaker from coming forward through the speaker again and muddying the sound clarity. (Some people don’t believe that this is impactive.)
To do this, this design uses SEALED separate 4 inch PVC tubes for each of the 34 midrange/mid bass speakers. Each tube is physically separates from its peers by ¾ inch of open air space.
On page 35 of the Dickason 6th Edition Loudspeaker Cookbook, there is an empirical study on the efficiency of various stuffing materials. The greatest attenuation was determined to be at the 4 lb/cu ft density level. Frequencies within the range of the mid/midbass speaker are right within the range of attenuation. Only by using tubes can the appropriate density be achieved, since the insulation can be stuff tightly against the walls of the tube and not push up against the speaker itself. The part closest to the speaker is protected by a layer of sheet pillowfill from the fiberglass threads that might otherwise get into the voice coil.
Additionally there are some benefits of the tube’s inability to make odd ordered harmonics, or odd ordered harmonic distortion, but that will not be addressed here.
This design is labor intense but it helps make the 3 inch midrange speaker process viable.
5. Keep the cost down below $1400 for the speakers, and the required electronics.
6. Keep Comb distortion in the mid and high frequency drivers below Griffin minimums, and use dome tweeter SOUND rather than ribbons or planars, for their flatter frequency response.
The only dome tweeters that I know which have a center to center distance small enough, and at the same time can be crossed low enough, are Dayton Neo ND20FA’s. These have their flanges cut so that the c-to-c is .9 inch, and the comb filter distortion doesn’t start until more than half way into the last octave, or more than 15,000Khz.
I prefer dome tweeter sound to the more, IMNSHO, screechy sound of planars or ribbons.
Now you know some of the design requirements.
Zarathu
FYI: See the picture, No combing in RR ids25 array destroying the phantom height impression?
Look at the corresponding point source plots with ideal drivers showing + 6 dB noticeable vertical ripple starting at 1250 Hz and the up-coming severe loobing at lambda (3822Hz) and above.
Wouldn’t a tweeter line insert at 2 kHz or below make a huge performance difference of the soundstage impression?
b
1(1)
Look at the corresponding point source plots with ideal drivers showing + 6 dB noticeable vertical ripple starting at 1250 Hz and the up-coming severe loobing at lambda (3822Hz) and above.
Wouldn’t a tweeter line insert at 2 kHz or below make a huge performance difference of the soundstage impression?
b
1(1)
Attachments
I am sorry, but simple physics will show that the only line array that works outside of the fresnel zone (a mismash of changing energys) is a focused array.
At work i developed a variable focus array that is constantly changing in depth of focus point in materials to evaluate grain boundary conditions (reflected energy at the grain boundry). This works in any point of the time/distance baseline. This is designed for higher frequencies of 2.25 Mhz and higher. However the physics of lower frequencies remains near the same.
ron
At work i developed a variable focus array that is constantly changing in depth of focus point in materials to evaluate grain boundary conditions (reflected energy at the grain boundry). This works in any point of the time/distance baseline. This is designed for higher frequencies of 2.25 Mhz and higher. However the physics of lower frequencies remains near the same.
ron
Clearly, I live in an alternate universe than Ron E, since my line array that his simple physics claims doesn't work, sounds incredible. Maybe there is more Dark Energy in his universe than in mine.
I call my universe the Calipso Universe. The Vega system is part of it.
Anyone wishing to take a jaunt over to my universe is welcome to do so, and I will let them hear how the physics works here. Calipso Universe is a subset of theThe Griffin universe.
Zarathu
I call my universe the Calipso Universe. The Vega system is part of it.
Anyone wishing to take a jaunt over to my universe is welcome to do so, and I will let them hear how the physics works here. Calipso Universe is a subset of theThe Griffin universe.
Zarathu
Clearly, I live in an alternate universe than Ron E, since my line array that his simple physics claims doesn't work, sounds incredible. Maybe there is more Dark Energy in his universe than in mine.
Well maybe in your universe there is a different medium of energy transfer. Maybe the atmosphere is different or the PSI is different.
I am simply stating facts from a physics POV , nothing more , nothing less.
Argue with physics? i have tried for over 40 years and i have found loopholes, but i havent beaten it yet.
ron
Well maybe in your universe there is a different medium of energy transfer. Maybe the atmosphere is different or the PSI is different.
I am simply stating facts from a physics POV , nothing more , nothing less.
Argue with physics? i have tried for over 40 years and i have found loopholes, but i havent beaten it yet.
ron
Amen Ron.
Believe it or not though I suspect you guys might actually be in agreement (or near to it). Zarathu, IIRC, your arrays are based on Jim Griffin's nearfield array paper, right? So you're listening to them within the fresnel zone (or nearfield if you prefer). If you move backward, outside this zone, they'll run into all the problems of lobing / combing discussed here, on other threads & in Jim's own paper. As Ron says, the only array that will work accurately for hifi purposes outside the fresnel zone (i.e. the farfield) is a focussed array.
As for the Russel arrays, I'm not sure what physics were used in their design, but from the remarks here thus far, I've seen nothing that suggests they are not suffering from the issues one would expect from running multiple WR drivers full-range. As we all listen for different things, perhaps these don't bother some people as much as they would others?
Out of interest, a quick question for the guys who've built one of these RR arrays: at what frequency does the Eq start to be applied to boost the top end?
Regards
Scott
Believe it or not though I suspect you guys might actually be in agreement (or near to it). Zarathu, IIRC, your arrays are based on Jim Griffin's nearfield array paper, right? So you're listening to them within the fresnel zone (or nearfield if you prefer). If you move backward, outside this zone, they'll run into all the problems of lobing / combing discussed here, on other threads & in Jim's own paper. As Ron says, the only array that will work accurately for hifi purposes outside the fresnel zone (i.e. the farfield) is a focussed array.
As for the Russel arrays, I'm not sure what physics were used in their design, but from the remarks here thus far, I've seen nothing that suggests they are not suffering from the issues one would expect from running multiple WR drivers full-range. As we all listen for different things, perhaps these don't bother some people as much as they would others?
Out of interest, a quick question for the guys who've built one of these RR arrays: at what frequency does the Eq start to be applied to boost the top end?
Regards
Scott
i guess it has been a long time since i heard a perfect 50000 dollar set of speakers. but with my reliable mcintosh 2250 amp and a good hi-def recording in am quite satisfied with the way my array, very similar to roger russells, sounds. i like piano, vocals, orchestra, and jazz. if i play really heavy rock music the array is not my first choice. but at the frequency of the music i like i have not heard any shortcomings. you can hear the rustle of the wires over the cimbals and each individual soft squeak of the fingers on the strings and the sharp initial twanq of the harpsicord. i have an active eq and find i use dead flat most of the time. i also, have a set of 12 in woofers connected and x over about 180 which i seldom use. it all depends on how critical you are of the recordings. when i get time i will run a pass through on the computer to see what i am actually listening to but to date me and everyone who listens is quite impressed by the sound. it seems quite strange to me that a respected person like roger would pen his name to product after 45 years that was inferior. he is just not that kind of person. he is a perfectionist at heart and likes music. he always said the final test was the human ear. listen to a pair and make your own judgement. i like mine a lot.
You might enjoy this: http://fullrangedriver.com/forum/viewtopic.php?id=91 A few thoughts Ron put together on the FR forum.
tryonziess said:
What speakers are you using for it and what is the configuration?
I initially ran my 3 inch mid ranges(17 per side) by themselves full range because i didn't have the tweeters hooked up. I thought they sounded wonderful. And then I added the tweeters to the array, and I could hear immediately what I was missing with the wide range three inchers. If I hadn't added the tweeters after running the 3 inchers by themselves I would have never known. I also added the 12 inch woofers later, and found what a differece they made too. However, my system is also tri -amped, with an electronic crossover. The woofers have their own separate 350w/ch amp, the tweeters their own 60 w/ch, and the mids their own 175 w/ch. Electronic crossover Tri-amping takes the whole system into a separate universe.
Zarathu
zaranthu, since i already have the bass element i might give the tweeter array a go. it is by far, per item, the least expensive. my next problem is at what height to start and what height to end. i currently have tang band 3.5 inch full range drivers they have mg/aluminum cones, ie 315d i think. for a first try experiment i am quite pleased. my entire setup is a from the hip approach. no major calculations other than what i proposed would work based on roger russells original article. i honestly believe that if some people actually listened to them instead of reading techno stuff they might find out how the human ear is a lot more forgiving than are calculations on paper. they honestly sound fabulous to those i play them for. any rec's on some tweeters to try.
tryonziess,
...is there somewhere i can go on the internet to decipher what you are showing me with the color graphs…
You may find some explanations and where to find the free ArraySHOW program here:
http://www.diyaudio.com/forums/showthread.php?postid=1209499#post1209499
http://www.prosoundweb.com/install/synaudcon/tt26_3/tt26_3_p1.shtml
See the picture where an ArraySHOW polar curve is compared with Tolvans Xdir and observe the differences, as Xdir is not third-octave filtered but the ArraySHOW is:
http://www.diyaudio.com/forums/attachment.php?postid=1168248&stamp=1174890278
Nice Java applets showing sound source interaction I find quite educational when playing with:
http://www.falstad.com/interference/
http://www.falstad.com/ripple/
b
...is there somewhere i can go on the internet to decipher what you are showing me with the color graphs…
You may find some explanations and where to find the free ArraySHOW program here:
http://www.diyaudio.com/forums/showthread.php?postid=1209499#post1209499
http://www.prosoundweb.com/install/synaudcon/tt26_3/tt26_3_p1.shtml
See the picture where an ArraySHOW polar curve is compared with Tolvans Xdir and observe the differences, as Xdir is not third-octave filtered but the ArraySHOW is:
http://www.diyaudio.com/forums/attachment.php?postid=1168248&stamp=1174890278
Nice Java applets showing sound source interaction I find quite educational when playing with:
http://www.falstad.com/interference/
http://www.falstad.com/ripple/
b
Infinite Line Source Array vs. clusters
Nope. The missing "air" is as a result of HF comb filtering. For any frequency where 1/2 wavelength is smaller than the center to center distance of the drivers there will be peaks and nulls. It can be easily measured and it can be totally sensed with an ear.
Exactly.
A major cause of even lower frequency comb filtering is from floor and ceiling reflections from line source arrays where the drivers do not continue completely to the ceiling and floor.
Basically, with a line array that goes from (reflective) floor to (reflective) ceiling, an infinite line source is made--just like in Figure 6 of James Griffin's doc. Here's is the link again for the few of you that don't have it bookmarked:
http://www.audiodiycentral.com/resource/pdf/nflawp.pdf
Now, mentally extend that picture to include reflections of reflections. So now there is an infinitely tall line array.
(Maybe imagine the drivers as point sources of light and the floow are ceiling as two parallel mirrors.) Certainly the floor and ceiling reflections are attenuated by the surface--it will not be perfect reflections)
But, if the drivers are not continued fron floor to ceiling, there are now gaps in the array.
In most embodiments, which have a tight cluster of tweeters in the middle of the driver, this becomes a repeating cluster of tweeters (in our infinite line array) , so there are effectively WIDELY separated sources.
hifryer said:
Nope. The missing "air" is as a result of HF comb filtering. For any frequency where 1/2 wavelength is smaller than the center to center distance of the drivers there will be peaks and nulls. It can be easily measured and it can be totally sensed with an ear.
Scottmoose said:
Exactly.
A major cause of even lower frequency comb filtering is from floor and ceiling reflections from line source arrays where the drivers do not continue completely to the ceiling and floor.
Basically, with a line array that goes from (reflective) floor to (reflective) ceiling, an infinite line source is made--just like in Figure 6 of James Griffin's doc. Here's is the link again for the few of you that don't have it bookmarked:
http://www.audiodiycentral.com/resource/pdf/nflawp.pdf
Now, mentally extend that picture to include reflections of reflections. So now there is an infinitely tall line array.
(Maybe imagine the drivers as point sources of light and the floow are ceiling as two parallel mirrors.) Certainly the floor and ceiling reflections are attenuated by the surface--it will not be perfect reflections)
But, if the drivers are not continued fron floor to ceiling, there are now gaps in the array.
In most embodiments, which have a tight cluster of tweeters in the middle of the driver, this becomes a repeating cluster of tweeters (in our infinite line array) , so there are effectively WIDELY separated sources.
Would it be possible with an active equalizer to build a pair of arrays, one tweeter the other mid-range, where the cone size and spacing with eq. make comb filtering non exixtent. the center to center spacing and frequency of combing could be calculated so crossover is below combing for each particular array. Tweeters are roughly 1 inch and mids 3 inch. Is this practical. Tad
EQ out a comb filter?
The answer is "no" but I can't just leave you with such a simple answer.
Actually, for a single point in listener space, you probably could EQ an acceptable level. (for one ear that can't move more than about 1/4 of an inch) That is why it looks like it works for the microphone (sometimes).
The queston usually implies at least a partial misunderstanding of what a comb filter is, or how constructive and destructive interference works.
To start with, take a look at this:
http://www.mcsquared.com/wavelength.htm
Play with it for a few minutes.
Then meet me back here.
I have seen this question get asked many times in different ways--including an assumpton that it can be done where an engineer takes a single reading using a calibrated microphone and then adds a boost or cut using a parametric equalizer.tryonziess said:
The answer is "no" but I can't just leave you with such a simple answer.
Actually, for a single point in listener space, you probably could EQ an acceptable level. (for one ear that can't move more than about 1/4 of an inch) That is why it looks like it works for the microphone (sometimes).
The queston usually implies at least a partial misunderstanding of what a comb filter is, or how constructive and destructive interference works.
To start with, take a look at this:
http://www.mcsquared.com/wavelength.htm
Play with it for a few minutes.
Then meet me back here.
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.