Hi Michael,
Sure I can do that. First I should note that previously I was considering only 1st low pass filters on the additional woofers. because I believed that Lynn indicated that was the approach he was going to take. It can not be done correctly with 1st order LP filters on the additional woofers. Second, the limitation on sensitivity and max SPL is still going to be with all 4 drivers operating at the low frequency cut off. So the question of low frequency dynamics remains.
Never-the-less, here is an example for 4 identical drivers (flat to DC). The dipole = monopole frequency is 100 Hz. Each driver is running separately (like 4 in parallel) so the first driver can increase SPL by 6 dB. Adding the third gives another 3.5 dB and the 4th another 2.5dB. I have added additional filtering to flatten the response of the main driver to 300 Hz. Otherwise there would be a dipole peak of +6dB (96dB) at 350 Hz or so.
The target was a 25 Hz 1st order HP response between 25 and 100 Hz. You can see the contribution of each driver. With 4 drivers the best that can be done is 2 octaves because it gives a max of 12dB increase in SPL (assuming an all parallel connection). Driving this would be difficult for most amps because it would be a 2 ohm load for nominally 8 ohm drivers.
With two drivers, or 4 in a parallel/series connection, I think the best that can be done is one octave below the dipole = monopole frequency since we only have a net SPL gain of 6dB.
Of course, the native driver response has to be superimposed on this to get the net response and this requires all drivers are the same.
So I guess I have to correct myself
and say that the dipole eq at low frequency can be compensated for by adding area but only over an octave with 4 identical driver in a series/parallel configuration (or two drivers in parallel). But this isn't done with a 1st order LP filter on the "boosting driver(s))" which I though was Lynn's intent (and my failing).
Still we are left with the limitation on max SPL. And to get a respectable response will require drivers with the correct Qts, the desired Fs and a baffle shape/size that placed the dipole=monopole frequency roughly an octave above the low frequency cut off.
Thanks for asking Michael as it got me thinking a little more.
Sure I can do that. First I should note that previously I was considering only 1st low pass filters on the additional woofers. because I believed that Lynn indicated that was the approach he was going to take. It can not be done correctly with 1st order LP filters on the additional woofers. Second, the limitation on sensitivity and max SPL is still going to be with all 4 drivers operating at the low frequency cut off. So the question of low frequency dynamics remains.
Never-the-less, here is an example for 4 identical drivers (flat to DC). The dipole = monopole frequency is 100 Hz. Each driver is running separately (like 4 in parallel) so the first driver can increase SPL by 6 dB. Adding the third gives another 3.5 dB and the 4th another 2.5dB. I have added additional filtering to flatten the response of the main driver to 300 Hz. Otherwise there would be a dipole peak of +6dB (96dB) at 350 Hz or so.
An externally hosted image should be here but it was not working when we last tested it.
The target was a 25 Hz 1st order HP response between 25 and 100 Hz. You can see the contribution of each driver. With 4 drivers the best that can be done is 2 octaves because it gives a max of 12dB increase in SPL (assuming an all parallel connection). Driving this would be difficult for most amps because it would be a 2 ohm load for nominally 8 ohm drivers.
With two drivers, or 4 in a parallel/series connection, I think the best that can be done is one octave below the dipole = monopole frequency since we only have a net SPL gain of 6dB.
An externally hosted image should be here but it was not working when we last tested it.
Of course, the native driver response has to be superimposed on this to get the net response and this requires all drivers are the same.
So I guess I have to correct myself
and say that the dipole eq at low frequency can be compensated for by adding area but only over an octave with 4 identical driver in a series/parallel configuration (or two drivers in parallel). But this isn't done with a 1st order LP filter on the "boosting driver(s))" which I though was Lynn's intent (and my failing). Still we are left with the limitation on max SPL. And to get a respectable response will require drivers with the correct Qts, the desired Fs and a baffle shape/size that placed the dipole=monopole frequency roughly an octave above the low frequency cut off.
Thanks for asking Michael as it got me thinking a little more.
Four of these Warrior 12" drivers in parallel on an open baffle with no added efficiency for room gain are (using the t/s they are 90 db at a meter with 2.83 volts each) 102 db sensitive at 2.83 volts/meter at 100 cycles and are 99 db at 55 cycles with the ability to output 120 db at 50 cycles. Add room gain and another pair for safety and you at a whopping $180.00 a channel.
My recommendation is use five or six 16 ohm drivers (like I do) with similar parameters to keep the impedance above 2 ohms -
PS- no EQ needed with room gain. It works, come listen.
My recommendation is use five or six 16 ohm drivers (like I do) with similar parameters to keep the impedance above 2 ohms -
PS- no EQ needed with room gain. It works, come listen.
Hi
JohnK, thanks for doing such terrific simulations.
Basically it confirms that with each Sd doubling - below the baffle peak frequency - you gain another octave of linear FR.
For doing the Sd doubling trick three times we need 8 speakers .
1st one as "fullrange" - 2nd one coming in by XO ~ one octave down the baffle peak frequency - 3rd AND 4th ones coming in by XO ~ two octaves down the baffle peak frequency – 5th AND 6th AND 7th AND 8th ones coming in by XO ~ three octaves down the baffle peak frequency
Its not possible with simple 6 dB filters ? – what else is needed ?
------------
Magnetar, what FR range do you cover by your five 12" and how steep do you cross ?
-------------
12" pro speaker with relative high Xmax I found so far :
Beyma 112NDW ( 13 mm )
18sound 12ND930 ( 13mm )
18sound 12ND830 ( 13mm )
BMS 12N820 ( 9 mm )
Greetings
Michael
JohnK, thanks for doing such terrific simulations.
Basically it confirms that with each Sd doubling - below the baffle peak frequency - you gain another octave of linear FR.
For doing the Sd doubling trick three times we need 8 speakers .
1st one as "fullrange" - 2nd one coming in by XO ~ one octave down the baffle peak frequency - 3rd AND 4th ones coming in by XO ~ two octaves down the baffle peak frequency – 5th AND 6th AND 7th AND 8th ones coming in by XO ~ three octaves down the baffle peak frequency
john k... said:
Its not possible with simple 6 dB filters ? – what else is needed ?
------------
Magnetar, what FR range do you cover by your five 12" and how steep do you cross ?
-------------
12" pro speaker with relative high Xmax I found so far :
Beyma 112NDW ( 13 mm )
18sound 12ND930 ( 13mm )
18sound 12ND830 ( 13mm )
BMS 12N820 ( 9 mm )
Greetings
Michael
Hi
When I did a in-room measurement of the Peerless SLS 12" for the " Fast, fun, Inexpensive OB project "
http://www.diyaudio.com/forums/showthread.php?postid=1340084#post1340084
I was amazed that it goes that deep. If not EQ'd, it would roll off gently at around 50-60 cycles in a moderate large baffle with that very clever XO approach.
Greetings
Michael
When I did a in-room measurement of the Peerless SLS 12" for the " Fast, fun, Inexpensive OB project "
An externally hosted image should be here but it was not working when we last tested it.
An externally hosted image should be here but it was not working when we last tested it.
An externally hosted image should be here but it was not working when we last tested it.
http://www.diyaudio.com/forums/showthread.php?postid=1340084#post1340084
I was amazed that it goes that deep. If not EQ'd, it would roll off gently at around 50-60 cycles in a moderate large baffle with that very clever XO approach.
Greetings
Michael
mige0 said:
Hello, I use a 70 cycle 3rd order high pass because the panel actually has a rising response hump between 50 and 120 cycles od arounf 4 db PLUS I do on occasion expect my system to play at 120 db peaks and I want low IM distortion and stress free sound. The tens are handed off 4th order electronically to the cone loaded horn mid at 400 cycles. The net range is 55 to 400 HZ for five ten inch 16 ohm Warriors.
The inexpensive bass panels sound better than many far more expensive and complicated high efficiency bass systems that I have built including bass horns, Karlson couplers, multi-driver 15" reflex, TAD bass drivers (both direct and horn loaded) and several high end commercial systems including big Magnepans, Dunlavys and full range electrostats.
I use a Crown PSA2 ball buster high current amp to drive the bass. It is a 'keeper' compared to all the other bass amps I've used over the years. Another good amp is the AB Precedent 900 or 1000. Newer prosound amps like the Crown K1, and digital amps are joke in the bass compared to these. I also have a Mesa 200 watt per channel 6550 tube amp (the 400) for the bass it's fat but the tone is fun.
Crown PSA2 Self Analyzing Power Amp
Hi
I expect Lynn to share his income with us by the percentage of postings to this thread – only THEREFOR I stay and keep posting as much as I can...
--------------
Magnetar, roughly a decade ago I was into organising an open air event taking place at a fairly large inner courtyard ( space for 1500+ people ).
The bassist came with an open baffle speaker of 4 by 4 = 16 10" drivers IIRC.
I smiled slightly when he set up his equipment – thinking, he by no way will have something sounding like serious bass.
After plugging in his bass guitar and after his first lines my jaw dropped.
Plenty of SPL - dry as desert !
Greetings
Michael
Magnetar said:
I expect Lynn to share his income with us by the percentage of postings to this thread – only THEREFOR I stay and keep posting as much as I can...
--------------
Magnetar, roughly a decade ago I was into organising an open air event taking place at a fairly large inner courtyard ( space for 1500+ people ).
The bassist came with an open baffle speaker of 4 by 4 = 16 10" drivers IIRC.
I smiled slightly when he set up his equipment – thinking, he by no way will have something sounding like serious bass.
After plugging in his bass guitar and after his first lines my jaw dropped.
Plenty of SPL - dry as desert !
Greetings
Michael
Magnetar said:
Considering the total lack of commercial interest in the Amity, Aurora, and Karna amplifiers over a ten-year span, it's a safe bet the commercial interest will be exactly zero. The watchword for industry guys is Not Invented Here, especially for a speaker that crosses market categories.
We all know that mass-market and high-end audio is like watching the elephants in a circus, all of the them going from nose to tail, going around in circles. Continuing the circus metaphor, the Big Two and Internet magazines are like the circus barkers, bringing in the rubes to see the Big Show. Since we ain't got no barkers, hardly anyone is aware of the little Free Theatre (or off-off-Broadway) putting on its production in a community-college auditorium.
Neville Theile's industry-changing series of papers languished in an Australian trade journal for ten years before Robert Small wrote his doctoral thesis that expanded and generalized the concept - and at that point was finally picked up by Robert Ashley, teaching at Utah, and also head of the Audio Engineering Society. During that same decade the top designers at Altec, JBL, Electro-Voice, Klipsch, AR, KLH, Tannoy, B&W, and KEF were all blissfully unaware of Theile's work - and Theile had written in English!!!
Since I only live a few miles away from Shred Muzik, I guess I could pay them a call and check out the drivers for myself. I wonder why the efficiency figures are so absurdly exaggerated - maybe that's the norm in the garage-band and DJ sector.
Data from the Shred Muzik site about the Madison drivers. The efficiency was calculated from Fs, Vas, and Qes with this handy site. Note: 1% conversion efficiency is roughly equal to 92 dB/metre/watt into a half-space.
10” Knight
-------------------
Fs=59 Hz
Vas=0.626
Qes=0.79
Qts=0.75
Xmax=0.25”
SD=63.61
Eff=0.44%
$69
10” Warrior
-------------------
Fs=65 Hz
Vas=0.555
Qes=0.915
Qts=0.867
Xmax=0.25”
SD=63.61
Eff=0.45%
$49
12” Executioner
-------------------
Fs=52 Hz
Vas=1.505
Qes=0.569
Qts=0.547
Xmax=0.375”
SD=95.033
Eff=1.0%
$179
12” Knight
-------------------
Fs=50 Hz
Vas=2.25
Qes=0.665
Qts=0.64
Xmax=0.25”
SD=95.033
Eff=1.14%
$89
12” Warrior
-------------------
Fs=41 Hz
Vas=4.12
Qes=0.681
Qts=0.649
Xmax=0.25”
SD=95.033
Eff=1.13%
$59
15” Executioner
-------------------
Fs=46 Hz
Vas=3.69
Qes=0.753
Qts=0.692
Xmax=0.375”
SD=153.93
Eff=1.29%
$229
15” Knight
-------------------
Fs=35 Hz
Vas=4.48
Qes=1.15
Qts=1.006
Xmax=0.25”
SD=153.93
Eff=0.45%
(something funny going on here, probably an error in the other measurements)
$129
15” Warrior
-------------------
Fs=36 Hz
Vas=8.29
Qes=0.888
Qts=0.826
Xmax=0.25”
SD=143.13
Eff=1.18%
$69
Comment: I think the Xmax figures are fictitious, and might be the max-destruction-limit instead. Maybe not even that. The only real way to know is to measure the difference between the gap height and the length of the voice-coil. If these two numbers are the same - which is common in quitar speakers - then Xmax is zero, or in other words, there is no linear region!
With guitar speakers, this is actually a plus, not a negative, since the tonal character of the speaker then changes significantly with drive level (T/S measurements are taken at a very low level and may not be affected). Since the Madison site appears to be aimed at heavy-metal bands, it's probably safe to conclude low distortion is not a major design priority.
10” Knight
-------------------
Fs=59 Hz
Vas=0.626
Qes=0.79
Qts=0.75
Xmax=0.25”
SD=63.61
Eff=0.44%
$69
10” Warrior
-------------------
Fs=65 Hz
Vas=0.555
Qes=0.915
Qts=0.867
Xmax=0.25”
SD=63.61
Eff=0.45%
$49
12” Executioner
-------------------
Fs=52 Hz
Vas=1.505
Qes=0.569
Qts=0.547
Xmax=0.375”
SD=95.033
Eff=1.0%
$179
12” Knight
-------------------
Fs=50 Hz
Vas=2.25
Qes=0.665
Qts=0.64
Xmax=0.25”
SD=95.033
Eff=1.14%
$89
12” Warrior
-------------------
Fs=41 Hz
Vas=4.12
Qes=0.681
Qts=0.649
Xmax=0.25”
SD=95.033
Eff=1.13%
$59
15” Executioner
-------------------
Fs=46 Hz
Vas=3.69
Qes=0.753
Qts=0.692
Xmax=0.375”
SD=153.93
Eff=1.29%
$229
15” Knight
-------------------
Fs=35 Hz
Vas=4.48
Qes=1.15
Qts=1.006
Xmax=0.25”
SD=153.93
Eff=0.45%
(something funny going on here, probably an error in the other measurements)
$129
15” Warrior
-------------------
Fs=36 Hz
Vas=8.29
Qes=0.888
Qts=0.826
Xmax=0.25”
SD=143.13
Eff=1.18%
$69
Comment: I think the Xmax figures are fictitious, and might be the max-destruction-limit instead. Maybe not even that. The only real way to know is to measure the difference between the gap height and the length of the voice-coil. If these two numbers are the same - which is common in quitar speakers - then Xmax is zero, or in other words, there is no linear region!
With guitar speakers, this is actually a plus, not a negative, since the tonal character of the speaker then changes significantly with drive level (T/S measurements are taken at a very low level and may not be affected). Since the Madison site appears to be aimed at heavy-metal bands, it's probably safe to conclude low distortion is not a major design priority.
Here is a little more of what I have considered.
It comes down to how you view the dipole roll off. Is it a 1/2 empty or 1/2 full thing? If you look at the roll off as decreasing efficiency as frequency goes down it may appear 1/2 empty. If you look at the dipole effect as efficiency increasing with increasing frequency it's 1/2 full.
I think the basic consideration is just the physics: How much volume displacement is required to produce the required SPL at the low frequency cut off and how are you going to get that. This will not change regardless of how you flatten the response (line level active or passive eq, or rolling off augmentation drivers). This is really a driver/baffle consideration. Properly specified ether approach should result in the same driver/baffle compliment and the same power requirement at low frequency. Then, assuming identical, multiple drivers, rolling off some of the drivers as the frequency rises to equalize the response is really just throwing away the increasing efficiency with rising frequency. It will place greater excursion demands on the main driver as the frequency increases and place greater demand on the power amplifier as well. Running all the drivers full range and using line level eq on the other hand is superior here since it means the increase in efficiency of the multiple driver configuration and the increase in dipole efficiency as frequency rises is retained and less amplifier power is required at the upper bass frequencies to produce the same SPL. With a given driver/baffle complement the power required at the system low cut off will be the same. Since IM distortion is basically related to the low frequency excursion, it will also remain about the same for both configurations. However, running all the drivers full range (i.e. up to 200 or 300 Hz) will reduce excursion in the upper bass range and therefore should result in lower HD at those frequencies. Running all the drivers full range also has the advantage that all drivers are in phase at all frequencies.
On the other hand, the idea of using a low Q driver for the upper ranges and a high Q driver with LP filter to augment the low frequency is really just turning the woofer system into a two-way low frequency system. The lowest frequencies will be reproduced primarily by the high Q driver(s) and the upper bass by the low Q driver(s). Since the low Q drivers won't contribute too much at low frequency this will place greater demands of the high Q driver(s) requiring either greater excursion or even more area. At the same time, the upper bass being handled by only the low Q drivers will requires higher excursions than if all drivers were full range with the possibility of increased HD (just as was the case for identical drivers). But since the low Q drivers will have limited excursion at lower frequencies there is a potential for reduced IM in the upper bass. How the LP filtering of the high Q driver is performed will determine whether advantage is taken of the increasing dipole efficiency with rising frequency.
Of course, the final result will depend on the exact driver/baffle configuration but no matter how it is set up the most judicious application of amplifier power is obtained when we take advantage of the rising efficiency of the dipole response and apply line level active or passive eq to minimize the power requirements on the amplifier. If the driver/baffle compliment is correctly chosen for the required low frequency max SPL and excursion limitations the line level eq should always be though of not as boosting the power applied at low frequency but rather reducing the power required as the frequency rises. This is because once the driver/baffle compliment is set the low frequency power requirements are as well.
It comes down to how you view the dipole roll off. Is it a 1/2 empty or 1/2 full thing? If you look at the roll off as decreasing efficiency as frequency goes down it may appear 1/2 empty. If you look at the dipole effect as efficiency increasing with increasing frequency it's 1/2 full.
I think the basic consideration is just the physics: How much volume displacement is required to produce the required SPL at the low frequency cut off and how are you going to get that. This will not change regardless of how you flatten the response (line level active or passive eq, or rolling off augmentation drivers). This is really a driver/baffle consideration. Properly specified ether approach should result in the same driver/baffle compliment and the same power requirement at low frequency. Then, assuming identical, multiple drivers, rolling off some of the drivers as the frequency rises to equalize the response is really just throwing away the increasing efficiency with rising frequency. It will place greater excursion demands on the main driver as the frequency increases and place greater demand on the power amplifier as well. Running all the drivers full range and using line level eq on the other hand is superior here since it means the increase in efficiency of the multiple driver configuration and the increase in dipole efficiency as frequency rises is retained and less amplifier power is required at the upper bass frequencies to produce the same SPL. With a given driver/baffle complement the power required at the system low cut off will be the same. Since IM distortion is basically related to the low frequency excursion, it will also remain about the same for both configurations. However, running all the drivers full range (i.e. up to 200 or 300 Hz) will reduce excursion in the upper bass range and therefore should result in lower HD at those frequencies. Running all the drivers full range also has the advantage that all drivers are in phase at all frequencies.
On the other hand, the idea of using a low Q driver for the upper ranges and a high Q driver with LP filter to augment the low frequency is really just turning the woofer system into a two-way low frequency system. The lowest frequencies will be reproduced primarily by the high Q driver(s) and the upper bass by the low Q driver(s). Since the low Q drivers won't contribute too much at low frequency this will place greater demands of the high Q driver(s) requiring either greater excursion or even more area. At the same time, the upper bass being handled by only the low Q drivers will requires higher excursions than if all drivers were full range with the possibility of increased HD (just as was the case for identical drivers). But since the low Q drivers will have limited excursion at lower frequencies there is a potential for reduced IM in the upper bass. How the LP filtering of the high Q driver is performed will determine whether advantage is taken of the increasing dipole efficiency with rising frequency.
Of course, the final result will depend on the exact driver/baffle configuration but no matter how it is set up the most judicious application of amplifier power is obtained when we take advantage of the rising efficiency of the dipole response and apply line level active or passive eq to minimize the power requirements on the amplifier. If the driver/baffle compliment is correctly chosen for the required low frequency max SPL and excursion limitations the line level eq should always be though of not as boosting the power applied at low frequency but rather reducing the power required as the frequency rises. This is because once the driver/baffle compliment is set the low frequency power requirements are as well.
Of course, direct-radiator drivers are not constant excursion devices. If the response is flat, they are constant-acceleration, and excursion increases at a rate of 12 dB/octave as the frequency is lowered. Thus, excursion in the upper bass and midrange is visibly quite low. Distortion mechanisms then shift from excursion (which dominates at low frequencies) to magnetic and inductive nonlinearities. At the highest frequencies, cone breakup mechanisms dominate.
Thus, three distortion regions, depending on frequency: lowest frequencies, excursion and resulting nonlinearities as the VC traverses the gap structure and its nonuniform magnetic fields, combined with spider nonlinearities. At higher upper-bass and lower-mid frequencies, magnetic nonlinearities (nonlinear Le) dominate, so refined polepiece-shielding methods are beneficial here. At 1 kHz and higher, diaphragm breakup becomes more important, and narrowband peaks in distortion appear.
The other drawback of a large array going into the lower mid-frequencies, of course, is dispersion. Whether the array is vertical or square, dispersion starts to matter in the 500 Hz ~ 1 kHz region, especially in the crossover region, where lobing can affect tonality and voice quality.
The point of matching the radiating area to the frequency is twofold: offset the very rapid increase in excursion below the baffle peak (particularly if compensating boost EQ is applied), and retain favorable dispersion characteristics above the baffle peak.
Thus, three distortion regions, depending on frequency: lowest frequencies, excursion and resulting nonlinearities as the VC traverses the gap structure and its nonuniform magnetic fields, combined with spider nonlinearities. At higher upper-bass and lower-mid frequencies, magnetic nonlinearities (nonlinear Le) dominate, so refined polepiece-shielding methods are beneficial here. At 1 kHz and higher, diaphragm breakup becomes more important, and narrowband peaks in distortion appear.
The other drawback of a large array going into the lower mid-frequencies, of course, is dispersion. Whether the array is vertical or square, dispersion starts to matter in the 500 Hz ~ 1 kHz region, especially in the crossover region, where lobing can affect tonality and voice quality.
The point of matching the radiating area to the frequency is twofold: offset the very rapid increase in excursion below the baffle peak (particularly if compensating boost EQ is applied), and retain favorable dispersion characteristics above the baffle peak.
Lynn Olson said:
Yes, agreed, and this is one reason, along with IM distortion and others, why we build 3-way systems. In each band we must conside the max SPL required at the low frequency cut off of that band and the dispersion characteristics at the upper limit, along with the impact of distortion. This is true for any system speaker system.
The point of matching the radiating area to the frequency is twofold: offset the very rapid increase in excursion below the baffle peak (particularly if compensating boost EQ is applied), and retain favorable dispersion characteristics above the baffle peak.
I have no problem with this. Again, it's not different than what is done for a conventional speaker. But there are several ways to accomplish the task. One it to combine the drivers in such a way as to achieve flat response without equalization, that is, constant sensitivity vs. frequency (regardless of speaker type, conventional, dipole or other). The other is to design in such a manner as to take advantage of the frequency dependent sensitivity and apply line level equalization to compensate for the frequency dependent sensitivity, yielding flat response. For get dipole systems and consider a simple 2-way box speaker. Maybe the woofer has a 2Pi sensitivity of XXdB and the tweeter YYdB. If we apply full baffle step compensation in a constant sensitivity system the system sensitivity will be XX-6 dB. Since the peak above the baffle step in a conventional speaker is about 3 dB we would be throwing away about 9 dB sensitivity in the midrange around that peak. For the tweeter we would need to add an L pad or some other attenuation throwing away YY - (XX-6) dB of sensitivity. Why? We can retain this sensitivity variation and compensate with line level Eq as I proposed in my Hybrid Design article: http://www.musicanddesign.com/HybridDesign.html and save a lot of amplifier capability in reserve as shown in red (see articale for details).
An externally hosted image should be here but it was not working when we last tested it.
Now some audiophile purists may object to having a line level active (or passive) equalization stage between the amp and preamp, but when considering the dynamic capabilities of the overall system it just makes sense.
Lynn, you keep on referring to the low frequency eq as boost. This was my earlier point about ½ full vs. ½ empty. Once you determine the driver compliment for the SPL requirements at the low frequency cut off consider that as the 0dB point. It is the defining point on the sensitivity curve. Then look at the eq as attenuation, not boost, to compensate for the increasing sensitivity as frequency increases (dipole or baffle step), thus holding amplifier potential in reserve. The same argument applies to using a high efficiency tweeter with lower efficiency woofer (like my design example). you can attenuate the input to the tweeter after the amplifier, dumping power in resistors or what ever, or correctly attenuate the input to the amplifier using line level eq so that on what is required by the tweeter is delivered.
Hi Lynn,
I have not read the whole thread, so I don't know if you would consider building a horn midrange with "out of production" drivers.
Probably the best mass production driver ever made was the the Yamaha JA6681 and it's predecessor. Meyer Sound used them with a "modification" that made them more durable but stole them some fidelity.
I bought 2 of them after speaking with three people that have used them and compared them against other drivers. Two guys could compare them directly to their own TAD 4001 and said the Yam sounded smoother and more natural. It's a 1inch exit driver with
107db 1W1M. It can be fit with a phenolic (6603)or the original aluminum diaphragm. The aluminum diaphragm is not detectable as such, it just sounds uncolored, in the right horn of course. The aluminum version goes to 12000hz and then gradually goes down and the phenolic version goes to 8000hz. What is really amazing is that it can be crossed as low as 350hz in a big enough horn, which as far as I know is not heard of from any other 1inch driver.
Greets,
Klaus
I have not read the whole thread, so I don't know if you would consider building a horn midrange with "out of production" drivers.
Probably the best mass production driver ever made was the the Yamaha JA6681 and it's predecessor. Meyer Sound used them with a "modification" that made them more durable but stole them some fidelity.
I bought 2 of them after speaking with three people that have used them and compared them against other drivers. Two guys could compare them directly to their own TAD 4001 and said the Yam sounded smoother and more natural. It's a 1inch exit driver with
107db 1W1M. It can be fit with a phenolic (6603)or the original aluminum diaphragm. The aluminum diaphragm is not detectable as such, it just sounds uncolored, in the right horn of course. The aluminum version goes to 12000hz and then gradually goes down and the phenolic version goes to 8000hz. What is really amazing is that it can be crossed as low as 350hz in a big enough horn, which as far as I know is not heard of from any other 1inch driver.
Greets,
Klaus
Hi
To put this " excursion increases at a rate of 12 dB/octave " into perspective:
Taking a BMS 8N315 for example and inserting Sd=222 and Xmax=8
we get ~ 120 dB at 200 Hz.
( calculating with the SL form / http://www.linkwitzlab.com/spl_max1.xls / D set to 483 )
To keep 120 db at 100 Hz – which is 1 octave down - we would need Sd=1000 at Xmax=8
To keep 120 db at 50 Hz – which is 2 octaves down - we would need Sd=4000 at Xmax=8
Meaning we need 4 times the area each octave down to not run into intermodulation distortion .
Also meaning we need at least 12 db / octave high pass filter to not run into intermodulation distortion when setting the XO point at the lowest possible SPLmax frequency .
Or, if seen from JohnK's perspective – once the SPLmax requirements at the lowest frequency are met, intermodulation ( or linear-max excursion likewise ) isn't a problem for any higher frequency - given you don't go more extreme than 4 times Sd each octave ( which is hardly ever the case ).
Greetings
Michael
Lynn Olson said:
To put this " excursion increases at a rate of 12 dB/octave " into perspective:
Taking a BMS 8N315 for example and inserting Sd=222 and Xmax=8
we get ~ 120 dB at 200 Hz.
( calculating with the SL form / http://www.linkwitzlab.com/spl_max1.xls / D set to 483 )
To keep 120 db at 100 Hz – which is 1 octave down - we would need Sd=1000 at Xmax=8
To keep 120 db at 50 Hz – which is 2 octaves down - we would need Sd=4000 at Xmax=8
Meaning we need 4 times the area each octave down to not run into intermodulation distortion .
Also meaning we need at least 12 db / octave high pass filter to not run into intermodulation distortion when setting the XO point at the lowest possible SPLmax frequency .
Or, if seen from JohnK's perspective – once the SPLmax requirements at the lowest frequency are met, intermodulation ( or linear-max excursion likewise ) isn't a problem for any higher frequency - given you don't go more extreme than 4 times Sd each octave ( which is hardly ever the case ).
Greetings
Michael
Lynn Olson said:
Just a minor point. First, think volume displacement not excursion. Then also think for an open baffle operating in the dipole range it increase at 18dB/octave for flat response. So for 2 octaves at constant excursion the area would have to increase 64 times. Or if we set the area based on 8 mm and 120 dB at 50 Hz then at 200 Hz rhe excursion is down by a factor of 64. But 120 dB at 50 Hz, as I indicated in a previous post, is going to be rather intrusive in size.
Also, the IM is a function of the low frequency excuirsion, not the high frequency and the frequency ratio. That is, for a driver radiating two frequencies, Flow and Fhigh the IM will be dependent on the excursion the driver would experience id reproducing Flow alsone and the ratio of Fhigh/Flow. An article I wrote on IM distortion can be downloaded from my old web site at http://www.geocities.com/kreskovs/Doppler1.html
Lynn Olson said:
I say all the numbers are wrong, not just sensitivity and x max = LOL
I'm glad i got the 16 ohm drivers when they closed them out. Found some new Eminence NEO magnet 12's for really cheap/ I will use 12 per channel in my new panels. Twenty four 12's with a little eq should move some air