Evan Kirkendall wrote on Fri, 24 April 2009 19:08 |
Just for reference, here's a shot of the stage from about mid ways back: I too noticed the same thing Ivan did. The bass was great at FOH, but really disappeared off to the side. We've tried stacking the subs 2x3 and 3x2 on either side of the stage, but the nodes still exist, only in different places. Today we're trying to delay the outer few subs to hopefully spread the LF out some more! Evan |
Christian Tepfer wrote on Fri, 24 April 2009 13:24 |
Spacing in between the subs eases the beam as well... |
Art Welter wrote on Fri, 24 April 2009 15:27 |
Evan, That is quite an arc going on, how many feet down stage of the subs is forward edge of the array? Have you noticed how much acoustical overlap there is between sub and the 12”s? I have been reading all the responses, but the lack of LF Ivan mentioned behind and above the front mix does not correlate in my mind with “too much of a good thing (directivity)”. Art Welter |
Phillip Graham wrote on Fri, 24 April 2009 14:13 |
It looks like a case of too much of a good thing (directivity). Delay tapering the subs will help immensely, as Gabe said. I personally really like the horizontal line array sub setup in old-style theaters or narrow venues, where the array essentially spans the venue width. This makes for very even coverage. If I had these 12 subs to work with, I would place 4 in the center, and the remaining 4 in a 3/1 cardiod at either end of the stage, angled out towards the audience. Then set the delay time of the side sub arrays along the coverage seam of the center cluster where it intercepts the audience in the stands. This PA looks like it could have used some outfill arrays, too, depending on how far the audience extended to the sides. Another option would be a flown central subwoofer line array... I should clarify that I like the cardiod solution more than the arced and/or progressive delay approaches because those can cause a large lobe of LF to show up right in the center of the stage. If you are in a situation where you need defined coverage in a narrow area (such as multiple stages outdoors for a festival) the spaced horizontal array, these horizontal arrays work well to narrow the LF coverage in the horizontal. |
Too Tall (Curtis H. List) wrote on Fri, 24 April 2009 18:21 |
Hi Phil, Didn't you use to advocate adding a 12dB Butterworth low pass on the the left and right third of a bass sub horizontal line array. Beside only using the center one third of the cabinets to cover the high bass where the line array would cause extremely narrow dispersion the Butterworth filter added the time delay you were looking for the outer boxes. |
Art Welter wrote on Fri, 24 April 2009 19:09 |
Phil, I am aware the boundary causes "virtual boxes" in the floor, creating basically a 2-high subwoofer array. In this case that would only be 90 inches or so, not enough height to impart much vertical directivity below 80 HZ. I’m not sure what you mean by “virtual dipole”, could you explain that? Art Welter |
Phillip Graham wrote on Fri, 24 April 2009 17:33 |
That is why in cardioid sub setups you reverse the box closest to the boundary, and not farthest away. This insures the canceling boxes are closest to the center of the "virtual array", and that the vertical lobe behavior of the array is also the cardioid pattern you seek. |
Jens Brewer wrote on Sat, 25 April 2009 00:29 | ||
Phil, in this example, you're talking about a 2 box vertically stacked cardioid arrangement, right? One sub facing 'forward' and the other faced opposite. I just want to be clear on that since the first thing I think of when I think of cardioid subs is two boxes on the same plane spaced apart by x' with delay added and polarity reversed on the cabinet closest to the audience. Or have I pictured it wrong? |
Christian Tepfer wrote on Fri, 24 April 2009 11:24 |
... Spacing in between the subs eases the beam as well... |
Kevin Windrem wrote on Sat, 25 April 2009 13:35 | ||
MAPP seems to indicate increasing spacing between subs (with no delay tapering) NARROWS coverage (until the spacing gets too large, then you start seeing cancellation). 12 tight packed subs is already pretty narrow. |
Patrick Tracy wrote on Sat, 25 April 2009 21:52 |
If you (or anyone with the tools) had a spare moment to do the predictions I'd love to see an example or two of split subs in a club sized room. |
Mac Kerr wrote on Sat, 25 April 2009 20:24 | ||
What do you mean by "club sized room"? Here is a look at a pair of 700HPs per side 30' apart. This is the same 100'x150' room. |
Mac Kerr wrote on Sun, 26 April 2009 04:59 |
Perfect summation only nets +6dB, but perfect cancellation is infinite. |
Art Welter wrote on Sun, 26 April 2009 11:11 |
Mac, I would think the ceiling height would be as important as the wall relationship as far as LF room response. It appears you can specify room floor dimensions, but what is Mapp deciding to make the ceiling height? Art Welter |
Art Welter wrote on Sun, 26 April 2009 14:11 |
Mac, I would think the ceiling height would be as important as the wall relationship as far as LF room response. It appears you can specify room floor dimensions, but what is Mapp deciding to make the ceiling height? Art Welter |
Nick Aghababian wrote on Sun, 26 April 2009 15:23 |
What are the advantages of a vertical sub array? |
Nick Aghababian wrote on Sun, 26 April 2009 15:23 |
What are the advantages of a vertical sub array? |
Ivan Beaver wrote on Sun, 26 April 2009 19:15 |
...There is no "free lunch". |
Scott Smith wrote on Sun, 26 April 2009 21:47 | ||
Ummm... |
Mac Kerr wrote on Sat, 25 April 2009 17:46 |
How the array behaves is very dependent on frequency. It is also very dependent on the environment. The difference delay makes with no walls is easy to understand, and seems very controlled. When you add in the reflecting surfaces of the room the whole picture changes. It is easy to see why it is not so easy to make this all work in the real world. Mac |
Phillip Graham wrote on Mon, 27 April 2009 15:06 | ||
Hey Mac, Two points of subtlety, even though the point of these graphs should not be lost on anyone. First, MAPP gives no consideration of the third dimension, which can/will change the locations of the nodes and antinodes, The diagrams can be thought of only as accurate in 2d. |
Phillip Graham wrote on Mon, 27 April 2009 15:06 |
Second, MAPP gives you no control of the stiffness and losses of the room boundaries. I don't know the values that Meyer has chosen, but they may or may not reflect reality. Simple (ie unrealistic) boundary conditions is typically computationally expedient, so that might be what Meyer is doing. I am sure Perrin or one of their other modeling guys could chime in on that. |
Mac Kerr wrote on Mon, 27 April 2009 16:07 | ||||
True, I think I mentioned that here. |
Mac goes on | ||
Mapp does give you some control over the surface as shown in the image below. What I have not been able to make it do is make the architectural guidelines be surfaces. As far as I can tell you are limited to the box shape, although you can set the dimensions of the box. Since it is not 3D those details may be irrelevant anyway. Mac |
Kevin Windrem wrote on Sat, 25 April 2009 13:35 | ||
MAPP seems to indicate increasing spacing between subs (with no delay tapering) NARROWS coverage (until the spacing gets too large, then you start seeing cancellation). 12 tight packed subs is already pretty narrow. |
Tom Young wrote on Mon, 27 April 2009 18:16 |
Thanks for pointing that out. It's more akin to Django than (name your favorite 6 or 12 string guitar picker). I'm glad you don't do drugs, Mike |
Ivan Beaver wrote on Fri, 24 April 2009 07:54 |
Evan was running the screaming girls (in the audience) WAAAYYY to hot! |
Bennett Prescott wrote on Tue, 28 April 2009 15:23 |
So one of his requirements is that he not use cancellation techniques (read: cardioid) but then he goes and does exactly that? |
Bennett Prescott wrote on Tue, 28 April 2009 15:23 |
So one of his requirements is that he not use cancellation techniques (read: cardioid) but then he goes and does exactly that? |
Phillip Graham wrote on Tue, 28 April 2009 20:38 |
Well, Dave's implementation is actually an endfire array, with the rear sub at relative zero time. Perhaps he will realized that cardiod is the incorrect term here, and correct his faux pas. |
Mac Kerr wrote on Tue, 28 April 2009 20:53 | ||
That was my first thought as well. No reversed sub polarity reversal. Oh well. Mac |
Dave Rat wrote on Tue, 28 April 2009 23:22 | ||
Hello Phil, Well lets see, my understanding is that 'cardioid' is a mathematical shape that resembles the response patterns of unidirectional microphones where in the maximum rejection is 180 degrees off axis. Resembles is the key word here in that I dot believe anything in the real world actually exhibits a true cardioid pattern. |
Quote: |
By the term 'Cardioid,' in the context of the Coachella sub setup, I am referring to setting up clusters of L- Acoustics SB28 sub-woofers in a the factory 'cardioid' configuration that offers minimum output 180 degrees off axis. Though this is a frequency dependant response, the term cardioid is a useful description. |
Quote: |
Also of note, an end-fire array can be quite easily be setup such that there is minimum output 180 degrees off axis and maximum output on axis which also resembles the cardioid shape. |
Dave Rat wrote on Tue, 28 April 2009 23:50 |
Interesting observations yet based on an incorrect assumption and I personally am not a fan of cardioid subs except... These sub stacks with rear firing subs do implement a reversal in polarity. They achieve that directional sub output with out dumping power and speakers into trying to take sound away from behind. My words were 'destructive sound' and though I purposely skirted being too technical and took some liberties in descriptions, what I was describing is my avoidance of the whole 'noise cancelling concept' where anti-sound is being created as well as staying away from graduated delays that chew up power and sound funky. Simple, clean and maximum summation in the listening area achieved with careful placements and minimum time shifts while realizing rejection benefits onstage and behind the arrays was the goal. And it worked out pretty well as the predictions were quite close to the real world. |
Mac Kerr wrote on Wed, 29 April 2009 16:09 |
Isn't the system you used exactly what you said you didn't want? The rear facing box is just that, a "noise canceling" signal. By using delay instead of polarity you were able to have control over the frequency range it worked at, but the net result is the same. The rear facing box produces an out of phase (not polarity) signal that cancels the signal to the rear (over some specified range), and adds to the signal toward the front. The same as other cardioid arrays. My original comment was only in relation to the seeming contradiction in your blog post. It seems you came up with a very effective solution. Congrats on the K-1s. Mac |
Dave Rat wrote on Wed, 29 April 2009 15:13 |
I am not sure I agree that the net result is the same. In the out of polarity scenario, the louder the rear facing box is, the quiter it gets in front till you reach a null. Here you have a subtractive scenario where max voulume on axis is achieved by turning the front speakers to full and turning off the rear speaker. With a time aligned rear facing, turning up the rear firing box increases volume in the front. So there is a purely additive scenario where max volume on axis is achieved by turning the front and rear speaker up all the way. Hence the destuctive versus constructive dividing line that I was not very good at clarifying. |
Mac Kerr wrote on Wed, 29 April 2009 20:59 |
In either case you have to use a time offset between the front and rear firing speakers. If all you do is put 1 effectively omni speaker out of polarity with the rest you get no pattern control, and you get cancellation to the front and rear equally as you say. My point was merely that the rear facing speaker is in fact creating pattern control via cancellation. Whether or not the rear facing box is in or out of polarity will have less effect than the time offset between the boxes. Below are 3 models, the first is with the bottom rear facing box out of polarity, and delayed by 4ms. The second is with all 3 boxes with the same polarity, but the forward facing boxes delayed 4ms. the third is with the bottom box out of polarity, but no delay. It is clear this is not useful. This is a half space model at 63Hz, the gain is the same to all speakers. The cancellation and forward gain characteristics will change with frequency, but can be similar in either scenario. |
Dave Rat wrote on Wed, 29 April 2009 19:22 |
So I ask myself, "How can I cover this space, efficiently, with finesse, minimizing brute force (lossy) methods. |
Mac Kerr wrote on Thu, 30 April 2009 00:58 | ||
I'm with ya as far as covering the space efficiently, I guess what I'm missing is exactly what is the "brute force, lossy" method. Most of the steering methods I am aware of (and I admit to not having done it as much as you have) involve using part of the sub array to create a signal that is out of phase the rear direction, and in phase in the forward direction. Since in their useful band most subs are nearly omnidirectional, there will always be a mixing of signals in both the rearward and forward directions. What kind of array (that's not just poorly deployed) only causes destructive combining in the rear direction without also causing constructive combining in the forward direction. I assume we can agree that an array that causes destructive combining in all directions is just not a useful design. It is the combination of the positive forward summing and the negative rearward summing that causes the SPL difference in the coverage. Mac |
Dave Rat wrote on Wed, 29 April 2009 21:42 |
A simple example is a long wide straight sub array wherein it is delayed to simulate a curved array by using gradually increasing delay times. The center sub would be zero in time and each sub outward would have a bit longer delay. While this method can achieve a fairly smooth widening of the coverage, it is a lossy design wherein db's are lost in the coverage area due to the graduated delay times. Conversely, actually stacking the subs in a curve will be a more efficient and better sounding option that is less lossy. Though space limitations can be an issue. Therefore, the simulated curve would need more power and boxes to reach the same output as the actual array and not sound quite as good. Basically brute forcing one shape to act like another shape and to do so we need to throw power and processing at it. It is tempting to use these lossy solutions because they paint pretty picture in the software and often can fit into the environment we operate easier but my experience has been that they rarely work as well or sound as good as they look. Hence the avoidance and the quest for more finesseful solutions. That said, in some situations, these lossy types of solutions can be very effective. It is just about using the right tool for the job. For the Coachella job with some of the best rock and country music engineers coming through, it was important that there was nothing about the setup that could be disliked or pointed to as a flaw. I guess that is what Scott Sugden and I were trying to outline when defining the system design. I very much enjoy this message board chat with you as it helps keep me thinking and refining. Thank you! |
Dave Rat wrote on Wed, 29 April 2009 20:42 | ||||
A simple example is a long wide straight sub array wherein it is delayed to simulate a curved array by using gradually increasing delay times. The center sub would be zero in time and each sub outward would have a bit longer delay. While this method can achieve a fairly smooth widening of the coverage, it is a lossy design wherein db's are lost in the coverage area due to the graduated delay times. Conversely, actually stacking the subs in a curve will be a more efficient and better sounding option that is less lossy. Though space limitations can be an issue. Therefore, the simulated curve would need more power and boxes to reach the same output as the actual array and not sound quite as good. Basically brute forcing one shape to act like another shape and to do so we need to throw power and processing at it. It is tempting to use these lossy solutions because they paint pretty picture in the software and often can fit into the environment we operate easier but my experience has been that they rarely work as well or sound as good as they look. Hence the avoidance and the quest for more finesseful solutions. That said, in some situations, these lossy types of solutions can be very effective. It is just about using the right tool for the job. For the Coachella job with some of the best rock and country music engineers coming through, it was important that there was nothing about the setup that could be disliked or pointed to as a flaw. I guess that is what Scott Sugden and I were trying to outline when defining the system design. I very much enjoy this message board chat with you as it helps keep me thinking and refining. Thank you! |
Charlie Zureki wrote on Thu, 30 April 2009 02:56 |
Hey Dave, How about a double "arc" of subs. The subs set as you describe above and a rear facing line of subs similarly delayed yet with inverse polarity for each "match" and slightly lower powered? Talking about lossy.... and back breaking. Although... I think it might be interesting to see Cheers, HAmmer |
Tom Danley wrote on Wed, 29 April 2009 20:33 |
Hi Dave Do you mean (I think) an “end fire array” in antenna terms? This is not the same as a cardoid array at all as you suggest. The arrays like Mike and Ivan have installed work well but are still limited by the room of course. As I recall, a spacing between each source of about ¼ wl at the highest frequency gave the best array directivity per number box used. These were set up with the front box delayed the most and the rear with no delay etc and are on axis, an additive array. In the concert thread here, this would be a way of reducing the frontal area of the array down to non-interfering dimensions if they could afford to occupy the depth. Best, Tom Danley |
Dave Rat wrote on Wed, 29 April 2009 21:42 |
A simple example is a long wide straight sub array wherein it is delayed to simulate a curved array by using gradually increasing delay times. The center sub would be zero in time and each sub outward would have a bit longer delay. While this method can achieve a fairly smooth widening of the coverage, it is a lossy design wherein db's are lost in the coverage area due to the graduated delay times. Conversely, actually stacking the subs in a curve will be a more efficient and better sounding option that is less lossy. Though space limitations can be an issue. Therefore, the simulated curve would need more power and boxes to reach the same output as the actual array and not sound quite as good. |
Charlie Hughes wrote on Thu, 30 April 2009 16:34 | ||
Hi Dave, I don’t believe this to be the case but am open to input and observations from you and others. A straight array of subs, a curved array of subs, and a straight array of progressively delayed subs can all put out the same sound power for the same number of boxes in each array. The straight array will tend to concentrate this power on-axis, thus increasing the sound pressure, SPL. The curving of the subs, either physically or via delay, will spread this sound power to more evenly cover the off-axis areas. This results in a reduction of SPL on-axis, but an increase of SPL off-axis. I would not characterize this as lossy, just more even distribution. The primary difference that should be seen between a curved array and a straight, progressively delayed array is the sound radiation behind the array. The curved array will actually focus the output at the center of the curve, assuming a constant radius. This will not happen with the straight, progressively delayed array. For an observer behind the array the signal from the outer boxes will arrive later than the center boxes, just as it will for an observer in front of the array. Therefore, the straight, progressively delayed array will have more spread out coverage behind the array as well as in front of it. |
Dave Rat wrote on Wed, 29 April 2009 20:36 |
I think this is a simple version of what you describe. I used 1/4 wavelength spacing of the frequency I desired max rejection behind for the Coachella side and rear sub cluster in the dance tent. These worked really well! First year of 9 that we did not have issues with this 54 sub dance tent stepping on the other stages. Another cool thing is anyone can do this with existing gear and achieve good results. Plus it makes an awesome and stable stacking platform, if you have the depth available. This setup fits all the criteria I strive for, it is additive, efficient, simple and predictable with real world results being in line with expectations. |
Eric Snodgrass wrote on Thu, 30 April 2009 19:13 | ||
Dave, how did you do the timing of the subs relative to the top boxes? Did you time each sub stack individually to the top boxes, then delay the front sub stack some more to get the directional sub pattern? |
Dave Rat wrote on Thu, 30 April 2009 20:43 |
And since delaying the V-Dosc would only effect those overlap frequencies, I was not concerned with adding the additional delay time. |
Dave Rat wrote on Thu, 30 April 2009 13:24 |
I agree, they will all put out (output) the same acoustic power and The straight and curved arrays will have relatively the same audible output into the room but concentrated differently, as you mentioned but the array with staggered time delays will have a lower audible output into the room. In order to simplify understanding the concept, lets take it to the extreme, imagine two subs side by side and we then recreate the same three scenarios. 1) they are next to each other facing one direction, no time delay 2) They are curved a bit, no time delay 3) they are flat but one is time delay, no wait, for ease of understanding, lets set the time delay to the worst case scenario for all frequencies. Hence we flip polarity of one sub. Now we analyze the three setups. 1) is somewhat omnidirectional with a soft figure 8-ish shape. 2) is a bit wider in front, but still very omni. 3) has nearly no sound output into the room at all as one speaker uses all its energy cancelling the sound from the other. This is an extreme case but should shed light (sound) onto what happens with time graduated arrays. There is loss associated with the conflicts induced by the time shifts. The closer the boxes are together, the more effectively they are able to conflict and say "bye bye db's and hello extra boxes and amps." |
Charlie Hughes wrote on Thu, 30 April 2009 16:34 | ||
Hi Dave, I don’t believe this to be the case but am open to input and observations from you and others. A straight array of subs, a curved array of subs, and a straight array of progressively delayed subs can all put out the same sound power for the same number of boxes in each array. The straight array will tend to concentrate this power on-axis, thus increasing the sound pressure, SPL. The curving of the subs, either physically or via delay, will spread this sound power to more evenly cover the off-axis areas. This results in a reduction of SPL on-axis, but an increase of SPL off-axis. I would not characterize this as lossy, just more even distribution. The primary difference that should be seen between a curved array and a straight, progressively delayed array is the sound radiation behind the array. The curved array will actually focus the output at the center of the curve, assuming a constant radius. This will not happen with the straight, progressively delayed array. For an observer behind the array the signal from the outer boxes will arrive later than the center boxes, just as it will for an observer in front of the array. Therefore, the straight, progressively delayed array will have more spread out coverage behind the array as well as in front of it. |
Dave Rat wrote on Thu, 30 April 2009 13:24 |
I agree, they will all put out (output) the same acoustic power and The straight and curved arrays will have relatively the same audible output into the room but concentrated differently, as you mentioned but the array with staggered time delays will have a lower audible output into the room. |
Dave Rat wrote on Thu, 30 April 2009 13:24 |
In order to simplify understanding the concept, lets take it to the extreme, imagine two subs side by side and we then recreate the same three scenarios. 1) they are next to each other facing one direction, no time delay 2) They are curved a bit, no time delay 3) they are flat but one is time delay, no wait, for ease of understanding, lets set the time delay to the worst case scenario for all frequencies. Hence we flip polarity of one sub. Now we analyze the three setups. 1) is somewhat omnidirectional with a soft figure 8-ish shape. 2) is a bit wider in front, but still very omni. 3) has nearly no sound output into the room at all as one speaker uses all its energy cancelling the sound from the other. This is an extreme case but should shed light (sound) onto what happens with time graduated arrays. There is loss associated with the conflicts induced by the time shifts. The closer the boxes are together, the more effectively they are able to conflict and say "bye bye db's and hello extra boxes and amps." |
Greg Cameron wrote on Thu, 30 April 2009 21:52 |
How about when the tops are flying vs. a ground stacked on the subs? Kind of hard to get that alignment except for one relatively small place in the venue. I'm sure the dance tent setup was no worse and likely much better then a flown tops scenario even with the offset in alignment. And though I didn't attend, the reports were that it sounded fantastic from several people that I know who where there. That said, it would be interesting to do an A/B test. Greg |
Tom Danley wrote on Thu, 30 April 2009 22:14 |
Hi Mac In theory (which isn’t the same as saying I have tried IIR filters), adjustment of phase / time and amplitude vs frequency, would be the only way to adjust such a system (sources of a fixed physical geometry) so that it had roughly the same shape pattern over a range of frequencies (constant directivity bass then?) It would be easier I think to diddle with the phase settings while looking at a simple polar plot. In other words, one would set up an optimum spaced array at / near the high corner in use. The ¼ wl physical spacing and plain time delay produce the pattern. Then go down say a half octave and tweak the phase / time to get the same shape polar and repeat in steps to the low cutoff noting the requirements. The IIR filter (assuming one were “programming” it in that way) would be a nice way to do it. Best, Tom |
Sebastiaan Meijer wrote on Thu, 30 April 2009 19:50 | ||
Hi Dave, So you mean that time aligning the subs only affects the frequencies i the crossover region? With all due respect, but alignment of subs to tops is worth a centimeter (or inch )-precise approach. The improvements made by having the snap of a kick timing coherently with the umpfh is very important in dance, in my experience. We're even talking phase alignment in that regards. Best! Sebas |
Ivan Beaver wrote on Thu, 30 April 2009 22:31 |
Not that this discussion needs any more "diversions", but here is one. The discussions have been about coverage, not sound quality. Let's say you have an array that has its outerboxes delayed in an attempt to get a more even coverage. And let's also assume the FOH position is in the center of the array-on axis. A low freq impulse response lacks a defined sharp peak, that the higher freq have. And NOW you have even more low freq information arriving at FOH later in time-than if the boxes were not delayed. This will result in a "slurring" of the impact information. ie less detail due to multiple arrivals. So you now have a more even coverage-but the FOH guys says the sub system is "slow"-"sloppy"-"undefined" and so forth. So it really begs the question-what IS more important- more bass for the audience (who pays the ticket prices for the show to go on) or the FOH guy who only cares about his listening position? Is the general audience more concerned with the "quality" of the bass-or the quantity. I would argue the later. The deeper you dig-the more complicated it gets. As Pat Brown says-layers of the onion-there is always another one. |
Sebastiaan Meijer wrote on Thu, 30 April 2009 19:50 | ||
Hi Dave, So you mean that time aligning the subs only affects the frequencies i the crossover region? With all due respect, but alignment of subs to tops is worth a centimeter (or inch )-precise approach. The improvements made by having the snap of a kick timing coherently with the umpfh is very important in dance, in my experience. We're even talking phase alignment in that regards. Best! Sebas |
Dave Rat wrote on Fri, 01 May 2009 01:44 |
...but the time spent stopping all the gas stations may perhaps be better spent doing other things. Sorry for the sarcasm there, I have spent quite a bit of time doing time alignments in cab designs and honestly, it gets really tricky to determine where in time the darn things are. They float around at different frequencies align at different times with slushy readings much larger than a centimeter. And then, even if you get them time aligned, the darn phase is often off so you need to align them wrong to make them kind of right. |
Ivan Beaver wrote on Thu, 30 April 2009 22:31 |
The discussions have been about coverage, not sound quality. Let's say you have an array that has its outerboxes delayed in an attempt to get a more even coverage. And let's also assume the FOH position is in the center of the array-on axis. A low freq impulse response lacks a defined sharp peak, that the higher freq have. And NOW you have even more low freq information arriving at FOH later in time-than if the boxes were not delayed. This will result in a "slurring" of the impact information. ie less detail due to multiple arrivals. So you now have a more even coverage-but the FOH guys says the sub system is "slow"-"sloppy"-"undefined" and so forth. So it really begs the question-what IS more important- more bass for the audience (who pays the ticket prices for the show to go on) or the FOH guy who only cares about his listening position? Is the general audience more concerned with the "quality" of the bass-or the quantity. I would argue the later. The deeper you dig-the more complicated it gets. |
Mac Kerr wrote on Fri, 01 May 2009 07:59 |
Dave's situation, where he is dealing with very large systems outdoors gives him a great opportunity to try new things where the interaction of the walls doesn't wipe it all out. I'm glad he is pushing the envelope, and sharing his experience. |
Dave Rat wrote on Fri, 01 May 2009 01:44 | ||||
Well, let me try and add some perspective. If the x-over point is 80 hz which has a wavelength in the 14 foot region, and you are 2 feet off, that equates to about 1/7 of a wavelength in error or about 50 degrees off. So the net effect will be maybe a db or 2 of loss occurring only in the region of maximum overlap. Plus, the stepper the x-over filters, the less the loss That said, if I needed every drop of power maybe, but in that tent with six sound sources all pointing at each other, naw, too many other things going on that over ride the importance, but one must make the best choices with time and energy to achieve the optimum outcome. As far as arrival times, a few feet off at those frequencies is pretty negligible, especially considering that with a normal PA system hang, the error of arrival time between ground stacked subs and flown mains can never be aligned for more than a small percentage of the venue and the errors are much larger. So in this situation, it was not critical, in my humble opinion. As far as a centimeter alignment on subs, well, that may be a bit extreme, maybe like trying to improve gas mileage of your car by carrying around helium balloons because they are lighter than air. Yes, theoretically there is an improvement of some millionth of a gallon per mile resulting in a saving tens of pennies a decade, but the time spent stopping all the gas stations may perhaps be better spent doing other things. Sorry for the sarcasm there, I have spent quite a bit of time doing time alignments in cab designs and honestly, it gets really tricky to determine where in time the darn things are. They float around at different frequencies align at different times with slushy readings much larger than a centimeter. And then, even if you get them time aligned, the darn phase is often off so you need to align them wrong to make them kind of right. |
Ivan Beaver wrote on Thu, 30 April 2009 22:31 |
..The deeper you dig-the more complicated it gets. ... |
Scott Smith wrote on Sat, 02 May 2009 09:40 | ||
Life was so much simpler back when we used to just "plug and play"! How about a giant wall of subs vented through a single output under the front of the stage. Would also serve as crowd control.. |
Martyn "Ferrit" Rowe wrote on Sat, 02 May 2009 15:33 |
Hey Guys, What about this?.... http://www.martin-audio.com/products/ASX.asp |
Phillip Graham wrote on Sat, 02 May 2009 16:52 | ||
Lets put a couple of these together for comparison against the TH221 and TH812. Match them up on a cubic foot basis, or 1 TH vs 2 of the ASX. Now that would be interesting! |
Dave Rat wrote on Wed, 29 April 2009 20:13 |
I am not sure I agree that the net result is the same. In the out of polarity scenario, the louder the rear facing box is, the quiter it gets in front till you reach a null. Here you have a subtractive scenario where max voulume on axis is achieved by turning the front speakers to full and turning off the rear speaker. With a time aligned rear facing, turning up the rear firing box increases volume in the front. So there is a purely additive scenario where max volume on axis is achieved by turning the front and rear speaker up all the way. Hence the destuctive versus constructive dividing line that I was not very good at clarifying. |
Nick Hickman wrote on Mon, 04 May 2009 14:36 |
By "out of polarity scenario", are you describing a classic cardioid sub arrangement (rear box reverse polarity and delayed)? If so, I don't think things are as bad as you say. Having the rear source on can indeed reduce the on-axis level below that of the front source on its own but, in general, only at very low frequencies or at frequencies above the usable range of the system. The basis of this configuration is that, for an on-axis listener, the rear source arrives out of phase with the front source and this phase difference is different for every frequency. The phase difference has three components: 180 degrees from the polarity reversal, an amount that varies with frequency from the physical spacing between sources (i.e. time of flight) and, in a classic system, an equal amount from the delay applied to the rear source. Take the case where the combined front and rear sources (at equal level) give the same on-axis level as the front source only. It's easy to show that a phasor diagram must have the two sources and their resultant arranged in an equilateral triangle (i.e. all three are equal magnitude) and thus that the phase angle between the two sources must be 120 (or -120) degrees. Given the 180 degree polarity reversal, there must therefore be 30 degrees each from the physical spacing and the delay. If F is the frequency, D is the physical spacing between sources, and V is the speed of sound, we have: 30 degrees = 360 * D * F / V If D, by way of example, is 1m and V is 344m/s, then F is 28.7Hz. Below this frequency, the combined output is lower than that of a single source. There are also higher frequencies where the combined output on-axis is the same as a single source. In fact, this happens wherever the physical spacing represents either 30, 150, -30, or -150 degrees. For 1m spacing, the next occasion is at 143.3Hz and, therefore, in the region between 28.7Hz and 143.3Hz, the combined on-axis level is greater than that of a single source. Plotting far-field on-axis level against frequency for 1m source spacing looks like this (where 0dB is the level of a single source): (The blue line represents the phase difference corresponding to the distance between sources.) Looking at all this another way, the effect of the on-axis phase difference varying with frequency is that LF output rolls off at 6dB per octave. If you can afford to EQ that out (and that's a big "if"), you have a system that maintains consistent cardioid pattern control down to arbitrarily low frequency. (Directly behind the sources, the phase difference between sources is 180 degrees at every frequency and thus the only thing stopping a perfect null is level difference between the sources.) A classic "end-fire" arrangement (all in polarity; front source delayed) gives somewhat opposite trade-offs: the response on-axis is flat, but the dispersion pattern is different at every frequency and collapses to omni at low frequency. Predictable caveats apply: this all assumes point-source omnidirectional radiators. Nick |
Nick Hickman wrote on Wed, 06 May 2009 00:03 |
Hi Dave, Thanks for the comments. In your applications, I don't doubt you're making the right choices. My contribution sought to address the narrow issue of the frequency range over which the on-axis level of a pressure-gradient sub system should be greater than that of a single source. Yes, EQing the LF rolloff flat is a big issue and constitutes a big disadvantage in applications like yours where maximum output is a priority. But there are, I think, applications where SPL needs are more modest and where the consistent pattern of the cardioid (pressure-gradient) setup is an advantage. Different trade-offs may be appropriate in different situations. You're also right, I think, about the likelihood of a pressure-gradient system being sensitive to the acoustic environment (i.e. nearby obstructions) to a greater degree than an end-fire arrangement. And the rear null is indeed sensitive to level and EQ differences between the sources. Nick |
Dave Rat wrote on Mon, 04 May 2009 14:03 |
Furthermore, unless front-facing and rear-facing speakers are driven with identical EQ and power, the response of the array will change over the course of the show and and due to changes in the drive levels. Power compression, mechanical nonlinearities and protection limiters all interact to cause a myriad of complex unpredictibilities. |
Jens Brewer wrote on Wed, 06 May 2009 00:34 |
That said, how many BE's chose the traditional omni sub setup vs. directional setup? |
Jens Brewer wrote on Wed, 06 May 2009 05:34 | ||
Dave, I'm not sure why the response would change over the course of a show despite non equal power/eq. Once the forward facing/rear facing relationship is established (however you choose to set it), that characteristic won't change regardless of time of show, or drive level for that matter, until you push into nonlinearity. Just as the effect of a stationary boundary doesn't change over the course of an event, neither will the effect of the reversed sub in a cardioid pair. (And I assuming we're talking about deploying thoroughly tested and tweaked sub arrays from 1st tier manufacturers here, not homebrew; I'm pretty sure most of them have got their ducks straight at this point.) That said, how many BE's chose the traditional omni sub setup vs. directional setup? |
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Put simply, Have you ever experienced a subwoofer system that is powerful and solid when you first start a show and then it gets softer, loses volume and impact as the show progresses? I.E. audible power compression. |
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Well, this same thing happens with cardioid subs that have rear facing out of polarity speakers. Except, unless the rear speakers are exactly the same type, and driven with the same EQ and power levels, the power compression will tend to effect the front speakers more than the rear speakers. |
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So, unlike a conventional sub setup and 'in polarity sub setups' where power compression effects overall level and sound in a fairly uniform way. The reverse polarity rear fire sub setup will tend to become unstable in its coverage pattern when driven hard. |
Phillip Graham wrote on Wed, 06 May 2009 19:41 |
Dave, I have kept on the sidelines for a few pages on thread, and I appreciate you continuing to discuss the topic, but I fear there is some imprecise physics being bandied about here. There is not really such thing as "anti-sound" sound is sound. Whether it be an endfire or cardioid configuration, all of the energy the loudspeakers put out is conserved, and appears somewhere in the directivity lobe around the system. The integrated flux of energy through a sphere, or hemisphere, around the system is conserved. Now that some of that energy may be dissipated as heat, but it did not disappear. Dr. Vanderkooy did the derivation of the heat increase inside of a sealed box at his master class at the September AES in San Fran. I don't think that you me to infer that energy vanishes, but I think the casual reader might draw that conclusion. I am curious why you feel the endfire situation provides any advantage here? After all, if the cardiod array is using the same type box driven at the same levels with only delay and a polarity inversion, where is the differential response on the rear speaker coming from? PS As a point intention, let me state publicly that I think the endfire-type arrays subjectively sound a little better than cardioid, so I am not really opposing what you have done in practice, but trying to make sure the physics remains lucid. |
Dave Rat wrote on Wed, 06 May 2009 18:29 |
If you turn on the front speakers it gets louder in front. If you then turn up the rear speakers it gets quieter in front(and quieter in the rear). Though pattern control is realized but at the expense of suffering a reduced volume level in the forward coverage area. |
Dave Rat wrote on Wed, 06 May 2009 18:29 |
Does that 'additive versus subtractive result in the forward field' concept not make sense? |
Phillip Graham wrote on Thu, 07 May 2009 00:49 |
Both endfire and cardioid are well ahead of a traditional array in output for the same number of boxes due to their substantially higher forward directivity, and the attendant reduction of energy that was previously wasted on areas where there was no audience. |
Peter Morris wrote on Thu, 07 May 2009 07:05 |
Cardioid systems sum in front, cancel at the rear and energy is conserved, but the sound energy from the rear is not transferred as extra sound energy out front, it’s (more or less) lost in heat….sooo they will have the same forward (slight less) output compared to the same number of boxes (front + back) in a traditional array. |
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Then there is my issue of adding two identical non symmetrical acoustic waveforms together – one of which is delayed by approximately 1/2; the period of the dominant frequency and inverted before it’s added to the other……. The resultant wave form’s shape and length is not the same as the original unless it’s a pure continuous tone. |
Peter Morris wrote on Thu, 07 May 2009 02:05 |
Phil said:- “Both endfire and cardioid are well ahead of a traditional array in output for the same number of boxes due to their substantially higher forward directivity, and the attendant reduction of energy that was previously wasted on areas where there was no audience.” Ivan said:- “Both endfire and cardoiod systems are lower in level on axis than with the same number of boxes all piled up in a front firing arrangement. But you will get some addition out front (and lots of reduction in the rear) when the rear boxes are turned on (properly ).” Hi Phil, This is an old discussion for you and me, but I have to agree with Ivan. Cardioid systems sum in front, cancel at the rear and energy is conserved, but the sound energy from the rear is not transferred as extra sound energy out front, it’s (more or less) lost in heat….sooo they will have the same forward (slight less) output compared to the same number of boxes (front + back) in a traditional array. |
I said |
the attendant reduction of energy that was previously wasted on areas where there was no audience.” |
Nick Hickman wrote on Thu, 07 May 2009 08:12 | ||
Hi Phil,
Taken literally, I don't think that can be correct. {snip} Interested in your thoughts. Nick |
Nick Hickman wrote on Thu, 07 May 2009 08:02 | ||
I don't think energy is "lost in heat" (except in the sense of that being its ultimate fate). Any energy radiated goes somewhere, and any spaced arrangement of sources will create a dispersion pattern that puts more energy in one direction and less in another. |
Phillip Graham wrote on Thu, 07 May 2009 14:00 |
That should clarify where I was coming from. The system result is "well ahead" not because of an increase in output relative to a close-coupled source pair, but rather due to an improvement in the behavior of the reverberant field. Since LF problems, in my experience, are 99%+ dominated by the physical room, dumping less energy into the reverberant field of the room is of universal practical advantage. |
Tim McCulloch wrote on Thu, 07 May 2009 15:41 |
Subject line says it all... But there's been more brain food in this thread than anything posted in the last couple of months (maybe longer). Very good stuff to make one think... Have fun, good luck. Tim Mc |
Peter Morris wrote on Fri, 08 May 2009 01:59 |
When I have time what I want to do is apply a drum beat pulse to two inputs of a digital crossover with a 30 to 80 Hz pass band – invert and delay one output channel by 8ms then add those two signal together. Then I want to graphically compare the result with the original signal (+6dB) over the same pass band on an oscilloscope or similar. I’m sure you could do it with mathcad (if I had a copy) Peter |
Ivan Beaver wrote on Fri, 08 May 2009 13:00 | ||
Not the same thing. It is the COMBINATION of physical distance (time of flight) AND (electronic) delay, AND/OR NOT polarity inversion (depending on what type alignment you are trying to do), that makes it work-not just one or two of those. If all you do is delay a signal (and flip the polarity (or not-a DSP has no way of knowing if one output is in front of the other) and mix it with a non delayed signal, you will just get classic combfiltering. There is really no reason to waste your time. Because the results are NOT at all what happens in the real world. |
Peter Morris wrote on Fri, 08 May 2009 06:59 |
@ Nick and Matt – I did say that energy was conserved …and … I was talking about sound energy being (more or less) lost in heat. I was trying to keep the description and terminology simple, but I’m sure Matt’s correct in that the energy is also dissipated in the mechanical and electrical components as a result of changes to the radiation impedance. |
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@ Nick – that last bit of my post is about the rear sub being time delayed and inverted and what happens at the front of the array. (guilty as above) When I have time what I want to do is apply a drum beat pulse to two inputs of a digital crossover with a 30 to 80 Hz pass band – invert and delay one output channel by 8ms then add those two signal together. Then I want to graphically compare the result with the original signal (+6dB) over the same pass band on an oscilloscope or similar. I’m sure you could do it with mathcad (if I had a copy) |
Andrew Welker wrote on Fri, 08 May 2009 16:53 |
Its not just the delay, its the physical placement of the cabinets as well. If you don't have the rear cabinet directly behind the front cabinet, no matter what you do with delay, polarity, or eq you won't get the pattern control you are looking for. |
Andrew Welker wrote on Fri, 08 May 2009 11:53 |
Its not just the delay, its the physical placement of the cabinets as well. If you don't have the rear cabinet directly behind the front cabinet, no matter what you do with delay, polarity, or eq you won't get the pattern control you are looking for. |
Sebastiaan Meijer wrote on Fri, 08 May 2009 23:40 |
Hi Tom, To add even more to the confusion and mind breakers: The applet, and so does about every cardoid solution, assumes that we are dealing with sine waves. The 'back speaker' is delayed (both physically and electronically most often) and therefore can only sum to the front when the second period of the wave is identical to the first one. (Or when using polarity reversal: when the second half of the sine wave is identical but opposite of the first half). Now the question is whether this is a valid assumption for live music. In my experience this is not, thus my preference for the (more) impulse-correct end-fired solution. Another thing to add in this: how do the cabs in front load the cabs in the back by their physical properties? I simply don't know, but have experienced the impact of not even densily packed people near a sub stack. It got me thinking..... Best regards, Sebas |
Ivan Beaver wrote on Fri, 08 May 2009 20:27 |
Wouldn't that be "Amazing UPRIGHT bass"? Funny thing is that they were called upright basses-before there was the "horizontal" (electric) bass. So what was the "other type" back then? |
Peter Morris wrote on Fri, 08 May 2009 19:58 |
EXACTLY !!!! Everyone assumes we are talking about continuous sin waves, and on those wave forms it works. Music is not, the wave forms are asymmetric and impulsive. Perhaps if you consider a signal that makes the front speakers cone more forward (only) from 'zero' and back to 'zero' once……… the rear speaker is inverted so its cone moves backward (after the time delay) and then back to zero. What get out front is a +ve pressure followed by a –ve pressure. The original wave form was just one +ve pulse and 8 ms shorter (4 ms delay + 4ms from position) Peter |
your thought experiment |
Perhaps if you consider a signal that makes the front speakers cone more forward (only) from 'zero' and back to 'zero' once……… the rear speaker is inverted so its cone moves backward (after the time delay) and then back to zero. |
Phillip Graham wrote on Sat, 09 May 2009 02:43 | ||
Peter, This is bad physics and you (should?) know it. There is nothing wrong with not liking a cardioid subwoofer, but don't use bad physics to justify that dislike. I know you know the physics, so I respond to you and not the other poster. |
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1. Any LTI signal can be represented as the sum of periodic functions, e.g sine functions. This is a critical basic from Fourier analysis, and something I am sure you aren't arguing. |
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2. A wave has no meaning until you define the complete period. You cannot speak about an 100Hz tone until you have defined enough time for one period (10ms), because the signal is not periodic until it repeats. Thus the superposition of the acoustic pressure has to be considered on the timescale of the underlying wave period. |
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3. Perhaps if you consider a signal that makes the front speakers cone more forward (only) from 'zero' and back to 'zero' once……… the rear speaker is inverted so its cone moves backward (after the time delay) and then back to zero. |
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Even if your hypothetical system came to rest in such a way, Fourier analysis requires many other frequency components at the trailing and/or leading edge. Because that hypothetical system, by definition, requires higher frequency components to start and stop the signal, it must violate the wavelength spacing requirements cardioid array for these other frequencies. Even if you were to excite the system with a delta function or a square wave, the thought experiment ignore the removal of these higher frequency components from the signal fed the array by the lowpass filter. |
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4. The group delay of this system is already severely affected by both the combined phase response of the high and lowpass filters on the subwoofer, plus the acoustic phase due to the box tuning, so it is overstated to act as if the cardioid is the only thing causing these problems. The single box case already has time domain effects from the non-cardioid processing. |
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-- Vitriol aside, please see that I am not saying this case, and the endfire case are equivalent. Indeed, I AGREE 100% with the improvement the endfire shows relative to the cardioid case. I do not agree with the misappropriation of physics to do so. For the endfire case, the delayed front cabinet is in phase with the rear cabinet on the audience side of the array, but not so with the cardioid array. |
Sebastiaan Meijer wrote on Fri, 08 May 2009 21:44 |
While your snarky reply is quite rude in terms of addressing me, I will try to reply. No, I am not an official physics scientist, but starting with several years in engineering at university gave me some background, and the research done here in our audio lab provides some too. I choose a doctorate in another field however. |
Sebas Replies: | ||
Yes, true too. However, at a certain moment a signal will stop. And then you should not forget that there are 2 acoustic sources. |
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So, now let me provide an example: The front and the back speaker each are fed with a 100Hz continuous sine wave. Now at time T1 we suddenly stop the signal generator, precisely at 0V. The front speaker is exactly in the most forward position, with speed = 0, and the back speaker is at the most inwards position, with speed = 0. Now we are going to listen at a position on axis of the cardiod array where the sine wave sums perfectly. What are we going to hear when the signal generator is stopped? First we hear the HF content arriving from the front speaker associated with the sudden voltage drop at the voice coil, combined with the 100Hz that is still on its way from the back speaker. After the combination of electric and acoustic delay (lets say, 2.5 + 2.5 ms) we hear the HF content from the rear speaker arriving. |
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Now I realise that I should not have said: "This model assumes sine waves" but should have been more explicit to say "This model assumes CONTINUOUS sine waves". |
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Music however is something that starts and stops various frequencies by definition. This does not deny physics but puts constraints to where to apply which model. |
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Yes, but this is not different from any other processed box. What the cardiod thing adds to this is multiple arrival times of the identical signal, which is a problem with non-continous sines / signals called music. Hence the nice rolling electro and dance lows you get with the cardiod solutions but rarely a good kick drum or acoustic bass. |
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As long as we agree on the outcome Sebas |
Peter Morris wrote on Fri, 08 May 2009 22:49 |
Sooo that bass signal that was 100 ms is now 108ms long Regards Peter |
Phillip Graham wrote on Sat, 09 May 2009 05:05 | ||
Please see my post to Sebas, (directly above this one in flat view). You and I both reach the same conclusion about the stretching of the envelope function. What was not elucidated clearly previous to my post above, however, is that the stretched time domain behavior is well within the length of one period of the maximally changing envelope function that the passband allows. This point, to me, is critical for the correct understanding of those reading along here. It appears I owe you an apology for being too harsh, as I spent about an hour drafting the details of a more fully realized version of your conclusion, posted right after your shorter exposition. I feel preciseness is called for on this topic, as it tests the depths of understanding and explanation of this group of knowledgeable contributors. |
Peter Morris wrote on Sat, 09 May 2009 03:36 |
This “stretched time domain” as you call it has some other consequences – to my ear it results in a slight loss of “punch” but when considered in the context of a reverberant field, all bets are off. i.e you could easily gain more than you loose. Peter |
Peter Morris wrote on Sat, 09 May 2009 03:36 |
Hi Phil, At last we seem to agree… on one point at least. |
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This “stretched time domain” as you call it has some other consequences – to my ear it results in a slight loss of “punch” but when considered in the context of a reverberant field, all bets are off. i.e you could easily gain more than you loose. |
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I also wonder if the other technique of delaying the front speaker which will not “stretched time domain” maybe better despite the lack of perfect cancellation at the rear. |
Nick Hickman wrote on Sat, 09 May 2009 11:29 |
If the pressure-gradient system didn't work, we would also have problems with cardioid microphones and we'd be complaining about the late arrival of the out-of-polarity sound as it reached the rear of the diaphragm. (Of course, the distances involved in a microphone are tiny, but the frequencies involved are proportionally higher, and the situations are genuinely analogous.) |
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Further, one could say that every digital filter is achieving its effect using "late arrivals". The source signal is delayed by various amounts and each delayed copy is multiplied by some weight and added together. We don't worry about all these "late arrivals"; we just perceive the effect of the filter. |
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Phil: how does your analysis stack up if you add in a 6dB/oct LPF (with -90 degrees of phase everywhere) before you start? Nick |
Sebastiaan Meijer wrote on Sat, 09 May 2009 14:16 |
Hi Philip, and other posters, Maybe I read to much, thanks for clarifying. |
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The amount of time required for the filter to envelope is however too much in the sub region for live audio. Newer implementations promise to improve here, but here my math-knowledge stops and I leave it to the DSP programmers. |
Sebastiaan Meijer wrote on Sat, 09 May 2009 19:16 |
Hi Philip, and other posters, Maybe I read to much, thanks for clarifying. Your preciseness in definitions is greatly appreciated, and much more elaborate than I can word it. Understanding the concepts versus being able to explain it clearly are definately 2 separate things (especially when you don;t know the English scientific terms from the top of your hat ) I tend to put a lot of emphasis on the impulse response. REgarding the convolution analogy: We research FIR filtering a lot here, and especially the limitations of this technique (You will be surprised how one-dimensional it is implemented in many pro-audio systems right now). With linear phase FIR filters it is possible to create the bandpass filtering that you mentioned. The amount of time required for the filter to envelope is however too much in the sub region for live audio. Newer implementations promise to improve here, but here my math-knowledge stops and I leave it to the DSP programmers. Nick also made fantastic graphs. Regarding the microphone analogy: Yes I personally do favour omni's when possible. Especially in headsets omni's are often way more clear, not only due to the proximity effect, but also the 'chaos' that crosstalk between headsets give. And in the recording world the phase-correct response of omni's is well-known. Let me try to summarize this discussion: Cardiod:
Cons:
End-fired: Pros:
Cons:
Question:
Please add when I missed something. Sebas |
Dave Rat wrote on Sat, 09 May 2009 20:36 |
These may be of interest: http://www.l-acoustics.com/manuels/SB28_UM_ML_1.0.pdf http://www.audio-pro.nl/wx/download/download.php?id=11815967 82 |
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And perhaps some more thought should go into the title "Cardioid vs Endfire-Not the same thing" as though they are not the same, I continue offer that 'endfire' does offer a a cardioid pattern over a certain range of frequencies. |
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http://mathworld.wolfram.com/Cardioid.html |
Dave Rat wrote on Sat, 09 May 2009 20:04 |
[ The summary is cool. Though the terms 'cardioid' and 'endfire' are loose and overlapping, they do serve to differentiate what is being discussed. I am not an expert in the cardioid nuances but would like to add the following. Cardioid: Under cons,the makeup gain that is required can have a serious effect on overall efficiency increasing the boxes and amps required to reach the same volume in front. since the signal sent to front and rear speakers is different in EQ, level or both, the cardioid sub is susceptible to time/volume dependant non linearities in the dispersion. End fire: Under cons, the distance required to achieve pattern control is surprisingly short. I.E. effective pattern control can be realized by merely pointing a cabinet in reverse, next to or below cabs pointed forward. If delay is added to the forward facing cabs to 'wait' for the sound to wrap around front from the rear facing cab(s), a directional sub cluster can be created. Therefore, the additional space required to create a useful, functional and easy to setup endfire array takes exactly the same floor space as a conventional array. |
Ivan Beaver wrote on Sun, 10 May 2009 02:42 |
I have never seen an endfire situation in which the rear cabinet is facing the rear. Yes it can be done-but there is some directionality in the upper part of the passband, so the effective summation out front will not be the same as if both (or more) cabinets are facing forward. |
Dave Rat wrote on Sun, 10 May 2009 01:36 |
And perhaps some more thought should go into the title "Cardioid vs Endfire-Not the same thing" as though they are not the same, I continue offer that 'endfire' does offer a a cardioid pattern over a certain range of frequencies. |
Joseph Pearce wrote on Sun, 10 May 2009 11:57 |
I guess that this is specifically posted for Nick to answer as it seems to me that he has brought something new to the Cardioid sub setup, a 6db low pass filter...could you please explain this a little more? I do not understand. |
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Does this filter fix this? Is this why some "in-box" cardioid subs sound sluggish? I have deployed end-fire type arrays before and I feel that I have a full grasp for how these work, it is just this last little piece of the "cardioid" puzzle that I am having trouble with. Thanks. |
Phillip Graham wrote on Sun, 10 May 2009 14:39 |
Well, Since Evan K is (indirectly) responsible for one of the most interesting threads in memory here, and for me to think a lot harder about audio than I have in quite a while, I think it is only appropriate to reward him by metaphorically throwing him under the coach he is currently riding in! The beauty (horror?) of mixing a band beloved by teens of the Youtube generation is that they record everything for posterity, and post it online! With the search string "All Time Low live" coupled with 5 minutes of finding (relatively) unmolested audio, I present to you Evan K, king of the Clair center clustered subwoofers, at work: http://www.youtube.com/watch?v=Wa0At0PRMbo http://www.youtube.com/watch?v=kt6jTciFZfk http://www.youtube.com/watch?v=Qy1LoH5q4P0 http://www.youtube.com/watch?v=g2DewGEeMik http://www.youtube.com/watch?v=6qzCKUB6ia4 Cheers to you Evan, a young man who couldn't hide his mixing from the LAB even if he tried! All in good fun, my friend! |
Nick Aghababian wrote on Sun, 10 May 2009 14:54 |
After watching those videos I can see why Evan always complains about not enough sub! I can barely hear it from the camera! |
Phillip Graham wrote on Sun, 10 May 2009 13:39 |
The beauty (horror?) of mixing a band beloved by teens of the Youtube generation is that they record everything for posterity, and post it online! With the search string "All Time Low live" coupled with 5 minutes of finding (relatively) unmolested audio, I present to you Evan K, king of the Clair center clustered subwoofers, at work: |
Phillip Graham wrote on Sun, 10 May 2009 15:06 | ||
Perhaps you are jesting, but in reality there is plenty of subbage if Evan is at the board! Many many of the cheapie camera mics totally crap out on bass/toms/kick in videos of ATL. The ones that don't also don't have any low end. Since I thought no one wants to hear "fart" "fart" "fart" for 3 minutes on youtube, you get videos without much low end |
Milt Hathaway wrote on Sun, 10 May 2009 12:09 | ||
Well, that ruins the "On the internet, no one can hear you mix" sig. |
Peter Morris wrote on Sun, 10 May 2009 13:47 |
length of the pass band’s middle frequency. That way every thing lines up around (say) 62 Hz. This results in a minimal a phase difference between the two signals (one is inverted) over the pass band. It will be close enough at 40Hz and 80 Hz so that every thing sums almost flat, producing maximum SPL. |
Joseph Pearce wrote on Sun, 10 May 2009 16:57 |
I guess that this is specifically posted for Nick to answer as it seems to me that he has brought something new to the Cardioid sub setup, a 6db low pass filter...could you please explain this a little more? I do not understand. |
Phillip Graham wrote on Sun, 10 May 2009 18:26 |
The filter does not "fix" all of the effects, but it does improve them greatly. The applied LPF changes both the relative amplitude and group delay, of each frequency component in the waveform envelope, and this realigns the resultant summation of all the components much closer to the original. I am sure Nick will flesh this out further, including perhaps discussion of a modified filter transfer function that restores the original exactly. |
Ivan Beaver wrote on Sun, 10 May 2009 18:12 |
The top graph is a typical cardioid alignment with the rear cabinet(s) out of polarity and delayed. The bottom graph is endifre with the front cabinets delayed to the rear position. |
Uwe Riemer wrote on Mon, 11 May 2009 00:38 |
I am not sure, if the the pressure gradient model applies to the cardioid array with the backfiring woofer with inverted polarity. |
Matthew Knischewsky wrote on Fri, 24 April 2009 22:26 |
If you have enough DSP, this is what to try next. starting with the center 2 subs at 0ms, apply .5 ms delay to each sub on either side of the center pair. Now add .5 more delay to the next outside pair, (1ms) and so on. until you get to the outsides of the array. you might not even need .5ms per pair to cover the venue, but it's a start. If you don't have enough DSP channels you can physically place the subs in an arc to create delay. Matt |
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And on topic, I was at the 9:30 Club in DC in March, and the d+b rig has cardiod subs, 3 boxes of B2 per side, middle box facing back. After the set, the guys in the band said, "wow, there was a lot of low end on the stage" which really surprised me. |
Nick Hickman wrote on Mon, 11 May 2009 13:24 | ||
Peter:
I'm afraid I don't follow the context. Could you detail the spacings and delays you're advocating? If, in a pressure-gradient case, the delay selected doesn't "match" the spacing of the sources, the result is a cardioid "family member" pattern (i.e. hypercardioid, etc.) rather than actual cardioid. But whatever the pattern is, it's consistent from above the transition frequency on down as low as you want to go. |
Ken Freeman wrote on Mon, 11 May 2009 11:57 |
As Doug noted, the plans do tend to go out the window when you have to deal with a building where you may not have all the info until load in. Ken |
Nick Hickman wrote on Mon, 11 May 2009 08:24 | ||
Ivan:
Despite any shortcomings, that's very helpful. Thanks. If you do repeat the test, it might help to subtract out the raw loudspeaker response so that one can more readily see the level differences caused by the directional system. |
Peter Morris wrote on Mon, 11 May 2009 16:32 |
I think I said “*total delay of the rear box to 1/2 the wave length of the pass band’s middle frequency.” And defined total delay at the bottom as *total delay = path length delay - 4 ft + electronic delay - 4ms By that I meant 4 ft between the speakers with 4 ms of delay 4 ft and 4 ms were approximate values that were some where near the mark for the sake of explanation. (roughly 1ft = 1ms) |
Ivan posted the following measurements on Mon, 11 May 2009: |
Jens Brewer wrote on Wed, 13 May 2009 07:51 |
a) In your first diagram ( http://srforums.prosoundweb.com/index.php/m/0/44778/120/4379 /#msg_433247 ) the blue phase line swings between +/-90 relative to the rear box (which is polarity reversed relative the 'normal' front box). When you get to the theoretical 'corrected' response at the end of the post, we're back to a 'normal' 0 degrees over the relevant bandpass. But that is still relative to the rear polarity reversed box, right? So in effect, the front box is now reverse polarity relative to the original drive signal? |
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b) Is the theoretical 'corrected' response: (native cardioid response + 1st order 125hz LP + 4th order 80hz LP) or just (native cardioid response + 1st order 125hz LP)? Just eyeballing a 6db LP over the first graph looks like it would do it, but I want to be sure I'm not assuming something I shouldn't. |
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c) Was there a reason you used 125hz box spacing (.688m/2ms) but used a fourth order 80hz LP? Since we're dealing with a purely hypothetical sub here, and not tailoring the electrical crossover to a measured response, can I assume that you actually want the subs to crossover around 80hz? Wouldn't we be better off to set the spacing to match (1.07m/3.13ms)? It would push the rolloff slope further down, keeping more of the additive gain in the useable bandpass, and likewise, the first cancellation will occur lower, right about where the mains should start to shoulder the load. |
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d) I've done cardioid sub setups quite a bit and have not measured the drastic 'uncorrected' 6db rolloff that theory would predict. In most cases that I can remember, I didn't get results onaxis that were significantly different than the response of a single box (magnitude excepted). Could this be a function of being indoors for 99% of my gigs, some sort of extra processing that manufacturers have got going on, or perhaps operator error? |
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e) Last question (and this is not directed just to Nick): best practices regarding phase alignment of subs to mains when using cardioid subs. I've been in the habit of having both subs on to set level, kill the rear sub to do the alignment with the mains, then restore and check. The 'time smear' should have an effect on phase since our spacing is only accurate for one frequency, right? How significant, if at all, is the change of the phase slope between one on / both on? |
Nick Hickman wrote on Wed, 13 May 2009 13:29 | ||
Yes, the only change was to illustrate the effect of a low-pass filter. |
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Were your observations based on "DIY" pressure-gradient systems where you did the delay and polarity inversion (and nothing else) or on a manufacturer's "packaged" system? |
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If you align with only the front source, the combined phase will certainly be different when you add the rear source. The phase of both sources combined will be 0 degrees (relative to the front source only) |
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What was your thinking behind aligning with only the front source? |
Jens Brewer wrote on Fri, 15 May 2009 16:24 |
Sorry if my question was not clear Nick, but is it 2 LP filters or just one on the corrected response? |
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If front box=0 and adding the rear box=0, are they not then congruent? They only differ in their arrival times, not phase, in relation to the original drive signal. |
Phillip Graham wrote on Fri, 24 April 2009 13:13 |
I should clarify that I like the cardiod solution more than the arced and/or progressive delay approaches because those can cause a large lobe of LF to show up right in the center of the stage. |
Phillip Graham wrote on Fri, 24 April 2009 16:33 | ||
Art, The subwoofers are going to form a virtual dipole by coupling to the floor in the vertical plane. Whenever LF boxes are placed against a solid boundary, its important to remember the boundary causes "virtual boxes" in the floor, creating basically a 2-high subwoofer array. That is why in cardiod sub setups you reverse the box closest to the boundary, and not farthest away. This insures the canceling boxes are closest to the center of the "virtual array", and that the vertical lobe behavior of the array is also the cardiod pattern you seek. If the system was setup as shown, and then aligned in the vicinity of FOH, its not surprising that the relative bass amount other places in the arena was low. Its also possible, but less likely that the slap back from the rear wall of the arena was cancelling some of the forward sub energy at important mix frequencies in the audience. There is always the possibility of catching an unusual room mode in the acoustic space, but my guess is that this was more a function of the alignment sounding right at FOH, but causing insufficient bass in the rest of the venue due to the directivity of the bass array. Physics is physics, and this remains the most likley explanation. PS I don't know any of the system teching details for sure, I really don't want to seem like I am slagging on the CBA guy from my armchair. Just consider it helpful musings. |
Mac Kerr wrote on Wed, 29 April 2009 14:59 | ||
In either case you have to use a time offset between the front and rear firing speakers. If all you do is put 1 effectively omni speaker out of polarity with the rest you get no pattern control, and you get cancellation to the front and rear equally as you say. My point was merely that the rear facing speaker is in fact creating pattern control via cancellation. Whether or not the rear facing box is in or out of polarity will have less effect than the time offset between the boxes. Below are 3 models, the first is with the bottom rear facing box out of polarity, and delayed by 4ms. The second is with all 3 boxes with the same polarity, but the forward facing boxes delayed 4ms. the third is with the bottom box out of polarity, but no delay. It is clear this is not useful. This is a half space model at 63Hz, the gain is the same to all speakers. The cancellation and forward gain characteristics will change with frequency, but can be similar in either scenario. |
Sebastiaan Meijer wrote on Sat, 09 May 2009 13:16 |
Hi Philip, and other posters, Maybe I read to much, thanks for clarifying. Your preciseness in definitions is greatly appreciated, and much more elaborate than I can word it. Understanding the concepts versus being able to explain it clearly are definately 2 separate things (especially when you don;t know the English scientific terms from the top of your hat ) I tend to put a lot of emphasis on the impulse response. REgarding the convolution analogy: We research FIR filtering a lot here, and especially the limitations of this technique (You will be surprised how one-dimensional it is implemented in many pro-audio systems right now). With linear phase FIR filters it is possible to create the bandpass filtering that you mentioned. The amount of time required for the filter to envelope is however too much in the sub region for live audio. Newer implementations promise to improve here, but here my math-knowledge stops and I leave it to the DSP programmers. Nick also made fantastic graphs. Regarding the microphone analogy: Yes I personally do favour omni's when possible. Especially in headsets omni's are often way more clear, not only due to the proximity effect, but also the 'chaos' that crosstalk between headsets give. And in the recording world the phase-correct response of omni's is well-known. Let me try to summarize this discussion: Cardiod:
Cons:
End-fired: Pros:
Cons:
Question:
Please add when I missed something. Sebas |
HarryBrillJr. wrote on Tue, 04 August 2009 21:55 |
Pardon me if this is already addressed later in the thread because I am gradually making my way through this thread as I get time. |
Dave Rat wrote on Sat, 09 May 2009 19:04 |
End fire: Under cons, the distance required to achieve pattern control is surprisingly short. I.E. effective pattern control can be realized by merely pointing a cabinet in reverse, next to or below cabs pointed forward. If delay is added to the forward facing cabs to 'wait' for the sound to wrap around front from the rear facing cab(s), a directional sub cluster can be created. Therefore, the additional space required to create a useful, functional and easy to setup endfire array takes exactly the same floor space as a conventional array. |
Mac Kerr wrote on Tue, 04 August 2009 21:07 | ||
This was a very comprehensive thread. Coming to it this long after it has wound down it mightbe better to read it all and take notes on what you think needs comment. There is no point in repeating information in what is already an overly long thread. Mac |
Ivan Beaver wrote on Sat, 09 May 2009 20:42 | ||
The terms may be overlapping, but the basic concepts are very different. The sub cabinet may both be in the same physical position for both cases-or not. ENDFIRE: Both cabinets are facing the same direction and the front cabinet is delayed to the physical position of the rear cabinet-which is generally 1/4 wavelength of either the highest freq of the ingtended passband OR the center of the intended passband. This is really up to the end user and the patterns will vary with freq-so you have to determine what freq are more important to cancel. There is no natural HF rolloff with enfire vs front radiating. I have never seen an endfire situation in which the rear cabinet is facing the rear. Yes it can be done-but there is some directionality in the upper part of the passband, so the effective summation out front will not be the same as if both (or more) cabinets are facing forward. And the cancellations to the rear will not be the same-because of the actual freq response of the two cabinets involved are not the same. If they are both facing the same direction-the effective freq response of both cabinets in the rear direction is the same-making for a more effective cancellation (due to the physical and electronic delay) The endfire works better if both cabinets are facing the same direction. Endfire can be made up with as many cabinets as you want. It is not uncommon to have 4 cabinets lined up in a row. You cannot do this with a typical "Cardioid" configuration. CARDOID: The rear cabinet is either forward or rear facing (with the driver physically behind the front driver by 1/4 wave length-at whatever freq you want) and is out of polarity with the front cabinet and delayed to the physical distance of the spacing. Rear facing is the "classic" cardioid configuration-but it can be forward facing as well-but the patterns will vary in coverage/cancellation/addition. A Cardioid configuration has a natural HF rolloff of the passband-due to the arrival and spacing issues-as compared to a front radiating array. If you apply different eq to the different cabinets-you will be shifting the phase of that particular cabinet (at that freq)and the end result will probably not be what you might think it is (in all areas). The end result is not always what you might think it is. When I was doing my polar measurements of various endfire/cardioid arrays, I tried applying some eq to what I "thought" would make sense. Well it did make things a bit better out front-but really screwed up the rear rejection in very unfavorable ways. You CANNOT simply make some adjustments "willy nilly" and listen at one position- you HAVE to listen to the overall coverage of the array and have a "before" and "after" measurement in order to be sure it is the overall "correct thing" to do. That is one of the classic mistakes that people often make when doing sound system alignments (at any freq range)-they "fix" one area, while making another area much worse-because they are not "monitoring" what is happening in the other areas. The issue that plagues all sound systems in all freq ranges is the interactions between freq and time-be it time of flight or electronic delay. It is not as simple as is often thought. |
Dave Rat wrote on Sat, 09 May 2009 21:31 | ||
Ahhh, hence the reason I am sharing this info. There has been lots of theory and various other descriptions here. It is very important to keep an open mind as well as keep things in perspective. In my last post I provided links to two premium system manufacturers, L'Acoustics and D & B Audioteknik, that both offer cardioid sub array setups with processing that utilize exactly that, rear facing cabinets to form very effective 'endfire' cardioid sub setups that offer a big shovel full of pro's and a small spoonful of con's. In my opinion, of course. |
Phillip Graham wrote on Sat, 09 May 2009 20:05 | ||||||
Its not clear what l-acoustics is doing from the manual, but clearly you are better positioned to know about the French loudspeaker guys than I! It seems that D&B's implementation, based on reading between the lines in the manual, is indeed endfire by the "phil definition"
As I said to another person offline, one case is pressure gradient, the other is endfire, but both radiation patterns are cardioid(ish). The reason I, personally, stick to my cardioid vs. endfire distinction is that the reversed polarity case owes its lineage directly to pressure gradient microphones, which have been popularly known as "cardioid microphones" for who know's how long.
{Topic Swerve}Mathworld is an amazing website, the breadth and depth of content is unbelievable!{/Topic Swerve} |
Ivan Beaver wrote on Mon, 11 May 2009 12:00 |
\ You can also see how the output level is reduced when using either the endfire or cardioid arrangements-as compared to the front fire arrangement. |
.....bump
Lots of good info here, needs a bump ;-)
Yes awesome thead, However most of the pic's just show a question mark, is it just on my computer or??
mvh
Rasmus