Line Arrays, directivity and impulse response

But sound waves do load other drivers causing a wide range of effects. This is mostly ignored in most designs, but in arrays it can be an issue. It is not a huge effect, but it exists. It's called Mutual Impedance (as opposed to self impedance)

For example, in a torpedo head with 40 elements we had several blow up at a certain frequency because the mutual impedance went negative. With many drivers things start to get unusual.

Very interesting Earl, pls explain more from speakers perspective, it may be hard to fathom impedance of torpedos, leave alone negative impedance
 
Do you assume they interact close to their mounting position and things can only get worse from there on? Is that what you're saying here?
I think the drivers work together as 1 source (1 driver elongated driver) as long as te CTC spacing is <1/4 wavelength. Then 1 wavefront is created and there will be no interference from the different drivers at the listening position.
There will be interference from the other speaker(s) and indirect (reflected) sound.
With full range or broadband drivers, the CTC spacing will be larger then 1/2 wavelength for higher frequency's. They will behave as separate sources and will interfere everywhere they can; they create dips in the output (like 2 (bass) drivers out off phase) and dips and peaks at every listening position.
So in other words, if two separate drivers create a null it will remain a null from there on forward? As in: the wave stops there and then? :confused:
Yes, just as with active noise control but this happens only to certain frequencies and the rest will spread out in the room.
The subwoofer story seems to confirm that this is your view on things. Do you mind sharing a reference for this theory?
I first started to learn about CTC spacing in relation with subwoofers; place them closer then 1/4 wavelenght of the highest frequency they play or 2 or more wavelenght a part for the lowest wavelenght they play. Otherwise you get nasty peaks and dips at every location in the room.

But not only the spacing between the subwoofers is of importance, also the distances from the subs to the walls and ceiling.
For these reasons the standard 2 subs 2 tops with poles setup left and right of a small stage or dj table/booth, is terrible for the bass. And can be made a lot worse with same distances between the speakers as to the ceiling, sidewall and backwall....

With the lower frequencies and at higher sound levels these cancellations are a lot easier to hear and more disturbing then with higher frequencies as a lot more peaks and dips fit in 1/3 octave. Which seems to be important for what we hear.

Rules for Subwoofer placement and stacking - BillFitzmaurice.info
Is the only site I can remember which informs people correct for setting up subs and there are not many others. To me it is crazy that speaker (especially sub-woofer) manufactures don't have simple instruction on how to set up their speakers correctly.
 
"It's better to have a system which sounds good and measures bad, then one which measures good and sounds average.*"

If such a thing is possible then you are measuring the wrong things. In my world this could never happen because we only measure things that we know correlate with perception.
That is also an explanation for it. :D

In dutch we have a saying: Measuring is knowing, but know what you measure.

I was thinking of: the more you know, the more you know how little you know.

We tend to 'forget' the parts of which we no little or nothing about in our designs or even design for what we can measure so the thing will measure good. :boggled:
 
.......
I wondered if it was due to my wiring. Most line arrays get wired in series-parallel.
It is the easiest way to wire them up.
Meaning 5 drivers in series, 5 sets of that in parallel. I knew of at least one exception, member OPC here had wired them in parallel-series.
....
Great post! With 5P5S or 5S5P I suspect the total output to be the same. But with 3P5S vs 5S3P I wonder.
With batteries you got a similar thing with resistance of the wires and different influence on the battery pack when 1 cell goes bad.

With drivers the biggest difference I found is when blowing a driver. When you first parallel them, you can blow more then 5 (max 21 in theory) of the 25 drivers.
 
Very interesting Earl, pls explain more from speakers perspective, it may be hard to fathom impedance of torpedos, leave alone negative impedance

Think about a simple experiment. You have a circle of several sources around a single source in the center. At some frequency the waves from all the outside speakers will be 180 degrees out of phase of the center at the center if they all play the same signal. The pressure on the cone in the center will actually want to pull the cone in the same direction as the force. At some point this will become unstable.
 
In a reflection all drivers are 'visible' to the ear. The "averaged" out behaviour of the line array is there in the direct sound as well as the reflected sound. A wall or floor or ceiling, all would make the entire array "visible" to the ear at least at lower frequencies. So, there is no benefit as far as reflections are concerned.

The 3db for each doubling may have a detrimental affect, not sure whether it is a problem sonically.
Consider a point source which produces 70dB at listening position at 2 meters and reflected wave travels 4 metres (double, for simplicity) so its at 64 dB.
ow, consider a line array that produces same 70dB. The reflection is stronger at 67dB. It may be an issue.
I don't consider the walls, floor and ceiling small things. They were adressed in earlier posts. It was easy enough to treat first reflection points.

One can look up the resulting differences if interested. My thread is filled with measurements taken at the listening seat and beyond. Measurements taken prior to placing some of the panels and afterwards (only 3 panels were needed to clean up the first 20 ms to my liking, the pré-set goal I wanted to reach). Lots more to see in the array threads from fluid, ra7 and Halair to name a few.
 
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I think the drivers work together as 1 source (1 driver elongated driver) as long as te CTC spacing is <1/4 wavelength. Then 1 wavefront is created and there will be no interference from the different drivers at the listening position.
There will be interference from the other speaker(s) and indirect (reflected) sound.
With full range or broadband drivers, the CTC spacing will be larger then 1/2 wavelength for higher frequency's. They will behave as separate sources and will interfere everywhere they can; they create dips in the output (like 2 (bass) drivers out off phase) and dips and peaks at every listening position.
Yes, just as with active noise control but this happens only to certain frequencies and the rest will spread out in the room.I first started to learn about CTC spacing in relation with subwoofers; place them closer then 1/4 wavelenght of the highest frequency they play or 2 or more wavelenght a part for the lowest wavelenght they play. Otherwise you get nasty peaks and dips at every location in the room.

But not only the spacing between the subwoofers is of importance, also the distances from the subs to the walls and ceiling.
For these reasons the standard 2 subs 2 tops with poles setup left and right of a small stage or dj table/booth, is terrible for the bass. And can be made a lot worse with same distances between the speakers as to the ceiling, sidewall and backwall....

With the lower frequencies and at higher sound levels these cancellations are a lot easier to hear and more disturbing then with higher frequencies as a lot more peaks and dips fit in 1/3 octave. Which seems to be important for what we hear.

Rules for Subwoofer placement and stacking - BillFitzmaurice.info
Is the only site I can remember which informs people correct for setting up subs and there are not many others. To me it is crazy that speaker (especially sub-woofer) manufactures don't have simple instruction on how to set up their speakers correctly.

I'm much more comfortable accepting the principle of superposition of waves, as posted by bbutterfield. This theory lines up with very simple tests I have witnessed in physics class a very long time ago. Plus it agrees with what I measure out at the listening spot.
This does not render the quarter wave rule invalid nor wrong, if one realises the "why" of that rule. It also doesn't invalidate active noise control.
Waves simply don't stop, but pass trough each other without harm.

As for what it does out in the room, this APL_TDA graph taken at the listening position of both arrays firing, should support that claim:
stereo.jpg


Though, in all honesty, the graph is normalised for peak values, so no comb filtering will show up here.

In blue on the right side one can see what the room's response to the speakers is. Those are in-room reflections.
My pure DAC signal, measured with APL-TDA looks like this:
dac.jpg

That would be the ideal shape.

I'm in no way trying to convince you though. Everyone can make up their own mind and take (or leave) from this what they want. :)
 
I'm much more comfortable accepting the principle of superposition of waves, as posted by bbutterfield. This theory lines up with very simple tests I have witnessed in physics class a very long time ago. Plus it agrees with what I measure out at the listening spot.
This does not render the quarter wave rule invalid nor wrong, if one realises the "why" of that rule. It also doesn't invalidate active noise control.
Waves simply don't stop, but pass trough each other without harm.
It took some time, but I think I'm starting to get it; thank you and bbutterfield for the info.
 
I'd have to look it up again, I haven't used APL_TDA in quite a while.
I only used this example to show sound does arrive at the listening spot ;).
It probably is linear though. I have used this program in the past for checking the timing I get at my listening position.
REW now has the same type of view and that is indeed linear scale.
 
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My pure DAC signal, measured with APL-TDA looks like this:
dac.jpg

That would be the ideal shape.

I'm in no way trying to convince you though. Everyone can make up their own mind and take (or leave) from this what they want. :)

As far as I know, there's no consensus about that being the ideal shape. The onset is quite variable with frequency. I suspect a minimum phase version would have a more uniform onset. Although the minimum phase variant wouldn't maintain the phase relationships between frequencies, we aren't very sensitive to such things.
 
Superposition of Waves

"The principle of superposition may be applied to waves whenever two (or more) waves travel through the same medium at the same time. The waves pass through each other without being disturbed."

Since air behaves quite linearly at the sound pressure levels we listen at (changing pressure by only a tiny percentage) we can apply superposition to see how multiple waves sum at any position in space.
I have looked into this in detail and this isn't the whole story which is a lot more complicated as air is a little dispersive and the amount changes with temperature, which has an influence on the speed of sound and thus creates little different results in reality.

The link above also states: It should also be mentioned that this medium is nondispersive (all frequencies travel at the same speed) since the Gaussian wave pulses do not change their shape as they propagate. If the medium was dispersive, then the waves would change their shape.

So we get a little of this dispersive 'smoothing' behavior as well: Waves in a Dispersive Medium - D. Russell

On Speed of sound - Wikipedia i found http://handle.dtic.mil/100.2/ADA076060 which proposes a better equation to calculate the speed of sound, for which it's errors are coming from temperature and the humidity of air influences which have an effect depending on the frequency.

https://classes.soe.ucsc.edu/ams227/Winter16/lecturenotes/Chapter2-part1.pdf talks about non dispersive waves - page 13 "2.1.2 Isothermal vs. Adiabatic sound waves
Physically speaking, the appearance of T in the problem is not particularly
surprising: pressure waves are - by definition - compressional waves, and in
many cases a gas heats up when compressed. The change in the temperature
then changes the density, through the equation of state. "

How big these effects (and the ones we don't know off :p) are combined is the big question.
To me it's logical to say that the more air has to be traveled by sound the more this little dispersion has an influence on the sound.
Turbulence is an other factor which introduces dispersion.
The influence of the little dispersion in air will be of most influence at baffle as it has the most energy and a little turbulence is created by the driver.


Some other temperature related effects are atmospheric absorption and refraction
https://en.wikibooks.org/wiki/Engineering_Acoustics/Outdoor_Sound_Propagation#Refraction[17][18]
Outdoor_Sound_Refraction.png


I just measured the temperature of my floor and ceiling in my living room and there already is a 2 degrees Celsius difference, but that can be a lot more on warm days or when my heater is on. So this 1 more of the probably many other factors which influences the sound at the listening (or any) position.

There seem to be so many little factors at play when measuring speakers.
I have to let this sink in and hope you guys can truthfully tell me that the little dispersion in air is of little effect or that I'm completely wrong.
 
Think

These effects are insignificant for our purposes. Air dispersion is a factor only at very high frequencies and then only over fairly large distances. For example, in an auditorium there might be a few dB loss at the back rows relative to the front rows at 10 kHz. Below 5 kHz the effect is gone. (This explains why our hearing is limited to < 10kHz as outdoors there is not much sound > 10 kHz.) In a home listening room this loss is negligible. The wave bending that you show due to thermal gradients would take several miles to be significant.

Assuming a completely linear medium is very accurate for audio in a home. This means that superposition holds as stated before.
 
OK, I will tell you. the temperature differences within a room make little difference.
The equation for the speed of sound in air uses absolute temperature. On that scale, 0 degrees C is about 293 degrees absolute, if I recall correctly. So a 3 degrees C temperature change across a room is only 1% change and so is unlikely to lead to an audible effect.
 
As far as I know, there's no consensus about that being the ideal shape. The onset is quite variable with frequency. I suspect a minimum phase version would have a more uniform onset. Although the minimum phase variant wouldn't maintain the phase relationships between frequencies, we aren't very sensitive to such things.

I should have spend a bit more words on that.
-If output is supposed to be input, this is about as close as it gets-

Though my goal for my speakers is to have a minimum phase output over the bandwidth the speaker plays. Which is what that measured graph from my speaker showed.
 
Think

These effects are insignificant for our purposes... In a home listening room this loss is negligible. The wave bending that you show due to thermal gradients would take several miles to be significant...

This is not meant to be contrary in any way but to share an extreme example:

Several years ago we had some Danley proaudio gear out on the pavement here in LasVegas in July or August. It was about 110F with pavement temperature in the sun hovering about 145F.

We had a BC812 sub, a J2 I think and a J4 31 HF super horn, the one with 64 compression drivers that will do 150dB SPL continuous from 3k to 18kHz. 30 degree wide and 10 degree vertical. This thing will absolutely light your hair on fire if you’re in the pattern.

We were running the system hard and on axis 350-360 feet from the speakers when the J4 was toggled on and off we could not even tell. It’s entire output was thermally steered over our heads. With a high delta T it doesn't take miles. :)

To be sure this was not how this system would ever be deployed, it was convenient for some fun.

Barry.
 
Apologies for commenting so late. Going way back to post #10, the EQd IR at 3m: there's a lot of "hash" extending to 6ms and probably beyond. I may be wrong, but if one compared a spectrum of the FT windowed at about 0.5ms to 6ms, it may have an average level within 10dB of the spectrum windowed before that. IOW, the IR is not highly "resolved."

And this brings me to a fundamental concern about line arrays in general and CBT arrays in particular: we can EQ the FR to the nth degree, but since the temporal arrival of the radiation from all the drivers usually fails a mininum-phase condition at higher frequencies, we can still get an FR that looks impressive but yields a poor IR, which would relate to the IFT not being unique for a given transfer function magnitude. Correct?

And if that's all true, how much does it matter? (!)
 
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I guess that was one of my measured IR's. Let's see two gated graphs, no smoothing applied:

First up, 0.5 ms gate:
gate0-5.jpg


Next up, a 6.0 ms gate:
gate6-0.jpg


I think you need a new theory ;). I don't see a huge difference here.... This is from my unshaded full range array.
My current shaded array will perform better on the top end, showing no combing. Does this answer your curiosity?
I wouldn't cal the IR poor, in fact, I'd call it pretty good. It corresponds well with the FR, as it's the same thing, just shown differently.
It was measured at 3m in a living room.

As far as how much it matters, to me it didn't matter much as it wasn't holding me back to upgrade the drivers from Vifa/Peerless TC9 FD18-08 to Scan Speak 10F 8414 G10. I wouldn't have done that if I weren't satisfied with the results I achieved.

Here's that last graph smoothed 1/48:
smoothed6ms.jpg
 
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