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How to Set Subwoofer Delay (6 top brands agree)

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You don’t need expensive equipment or a PhD to set subwoofer delay time. In fact, all you need is:

  1. The physical distance from the listening position.
  2. The pre-alignment delay values.

Sound too easy? Consider that top brands like L-Acoustics, d&b audiotechnik, Coda Audio, RCF, Funktion-One and dB Technologies all recommend the same subwoofer phase alignment method. 

  1. Create an alignment preset for two sources that are equidistant.
  2. Modify that alignment in the field, using delay to equalize any difference in distance.

These manufacturers all agree that:

  • Alignment is important. You deserve all of the decibels that you paid for.
  • If an audio analyzer is unavailable or inappropriate, use the two step method described above instead. Some people don’t own an audio analyzer, have not invested the years of practice to master its operation, or don’t have time to set it up. Additionally, some circumstances are not appropriate for an audio analyzer, for example when the reflections in a small room make the data inactionable.

Manufacturers differ on where to get the pre-alignment delay values and the exact workflow, so let’s take a closer look at each of them.

L-Acoustics

Here’s a selection from the L-Acoustrics Prest Guide v18.0. Notice that their method is to find the difference in distance, convert it to a delay time using the speed of sound, and then add the pre-alignment delay.

It includes pages and pages of pre-alignment delay values depending on which speakers you are combining. SubAligner works the same way, but is not limited to a single manufacturer. It will automatically calculate the delay time for any speaker combination.

This alignment method is again recommended in an excellent training video from L-Acoustics.

Never forget that you still have the geometric measurement. If you are in a situation where you cannot ensure the quality of the measurement maybe measuring the geometric distance with a laser meter and calculating the corresponding delay time can give you better results than a bad acoustical measurement.

François Montignies, L-Acoustics

d&b audiotechnik

Here’s a selection from the TI 385 d&b Line array design 10.6 manual. They recommend using their modeling software for alignment, or an audio analyzer, or the same method discussed above converting physical distance to time.

NEXO

In NEXO’s amplifier manuals they start off by warning you of the dangers of a misaligned system, subs on an aux, and additional unnecessary filtering. Then they outline their subwoofer delay setting method, which is to convert the difference in distance from the listening position to time, then verify with an audio analyzer.

Consequences of a badly aligned system
Precautions
delay
nexo alignment

Coda Audio

In the Coda Audio LINUS Control v2.2.33 Time Alignment document, they recommend using an audio analyzer, then defaulting to the same method we have been discussing, which is to add the pre-alignment delay time to that created by the difference in physical distance.

coda audio max perforamance
Flown systems
coda audio tiray

RCF

We can see that the RCF Pre-Alignment Delays v.1.1 – Guide makes very similar suggestions.

Funktion-One

On the Funktion-One website they highlight the importance of choosing the correct crossover frequency range, time alignment, and limiter setting. 

Then they provide table of pre-alignment delay values and crossover settings.

dB Technologies

In this marketing video from dB Technologies they explain that, “The user can easily align the boxes just by using a rangefinder.”

This video should start at 2:31.

How do you measure the physical distance offset?

The short answer: User a laser range finder.

The biggest challenge here is sunlight. Here are some of my favorite tips to get accurate measurements:

  1. Get closer. If you calculate your alignment position with Merlijn’s Sub Align calculator, you will often find that the best alignment position is a lot closer than you thought. This will make a distance measurement a lot easier to do.
  2. 3D design. If you can get the distances from a 3D model ahead of time, that’s the best.
  3. Target. Many laser distos come with a little mirror that you can attach to the speaker. Anything reflective will work.
  4. Camera. My laser disto has a camera on it. This makes it much easier to aim. You still need something shiny to aim at, though, and for longer distances you’ll need a tripod for accurate aim since your hands will never be still enough.
  5. Right triangle solver. If you can’t get a lock on the speaker, you can put the disto on the ground and aim at a target on the ground below the speaker. Then measure the height of the speaker. If it’s an array you’ll need to figure out the distance from the bottom of the array to the center and add that on. Then use a right triangle solver (just google it) to find the hypotenuse. This will also save you if there are obstructions like chairs in the way of your sub measurement.
  6. Range finder. I keep a golf range finder as a back-up. It’s only accurate to within 1m, but it’s better than nothing and can be used to verify questionable laser disto readings.
  7. Alternatives: tape measure, measuring wheel, or proxy loudspeaker. You can use an audio analyzer in conjunction with a proxy loudspeaker to get the time to subwoofer time of flight.

Delay finders struggle with subwoofers because subwoofers reproduce only a fraction of the audible spectrum leaving too little data for the delay finder to lock onto.

The proxy loudspeaker’s sole purpose is to add the frequencies which are missing from the subwoofer, allowing the delay finder to detect arrival time again.

Subwoofer Alignment: The Foolproof Relative / Absolute Method

I wish there were some glasses you could wear that would highlight the laser dot or some kind of scope or sight. I see that the firearm community has a lot of cool toys like this, but haven’t found one that would actually help us.

Where do you get the pre-alignment delay values?

Pre-alignment delay is the amount of delay time necessary to align two sources when they are equidistant. These values are often provided by the manufacturer. If not, SubAligner offers 88,656 combinations or you can generate your own.

To calculate your own pre-alignment delay times, place main and sub speakers on the ground so that they are equidistant with your measurement microphone and then do whatever is necessary (polarity, delay, filters, etc.) to hit your preferred alignment target.

I’ve been using this method to prepare for every new show that I work on. I either go to the shop to measure the speakers and generate and alignment preset or I get it directly from SubAligner.

Sometimes I can even estimate the final delay by measuring the distances in modeling software.

On a recent gig here in Minneapolis I was responsible for deploying a NEXO sound system comprised of S1210 and RS18. To prepare, I:

  1. Designed the sound system in NEXO’s NS1 modeling software.
  2. Calculated the alignment position using Merlijn’s Sub Align calculator.
  3. Measured the physical distances to main and sub from the alignment position in NS1.
  4. Put these distances into SubAligner to automatically calculate the delay time.

Then, when I get into the field and all of the speakers are in place, all I need to do is measure the physical distances and add them to the alignment preset (or just re-run SubAligner).

Final step: verify with an audio analyzer.

Great success. 🙂

What about you? Have you tried using this method for setting subwoofer delay? What were your results?

For the Best Concert Sound, Keep the LF Reverb Time Low

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I initially intended to write a review of this book, but that part is easy. It’s great! Aside from one minor complaint, I loved it and I think everyone should read it.

Instead, I’d like to share some of my takeaways to peak your interest and hopefully start a discussion.

#1 – LF is hard to control

Loud bass is important to rock concerts. It’s also very difficult to control and requires the largest share of electric amplification energy.

Bass is so hard to control because the wavelengths are so long. They go everywhere. This causes various problems.

The LF directivity of loudspeakers is so low that the critical distance becomes very short, puts the majority of the audience in reverberation, and partially masks higher frequency direct sound.

Unfortunately, the presence of an audience doesn’t help much. There’s a common belief that when the room fills with people that the reverb will dry up and even out, but that may only be true at mid and high frequencies.

The presence of an audience does not have a big influence on the low frequency reverberation time of a hall. This is one reason why halls for pop and rock must be designed with a low RT at low frequencies.

Adelman-Larsen, Niels. Rock and Pop Venues. Cham, Switzerland, Springer Nature, 2021.

What’s the solution? Appropriate architecture to control the reverb time (RT) and an optimized sound system to keep sound off of the walls.

It is a fact that the more sound the PA system shoots onto the walls and ceiling, the more the reverberation of the hall is evoked.

#2 – Ideal reverb time is based on volume

The author provides a graph with target reverb times based on room volume and makes this generalization.

The halls investigated in this book are in almost all cases over 1000 m³. With an ideal reverberation time for pop and rock music of 0.6 seconds for that size volume this yields a Schroeder frequency below about 50 Hz.

#3 – Rooms must be designed for low RT at LF

This one factor, low RT at LF, has to be fulfilled in making a good sounding hall for pop and rock music. This is the single most important message of this book.

The 125Hz octave band is the most important to control because:

  • Directivity of loudspeakers increases with frequency, and so does critical distance.
  • Absorption by audience increases with frequency.
  • The 125 Hz band is louder and busier than the 63 Hz band.
  • The human hearing system shows little upward masking in the 63 Hz band.

One octave centered at 125Hz = 88-177Hz. What’s happening in that area? The alignment between main and sub arrays, especially with common haystacking. If you get it wrong, you may end up breaking the first rule of sound system design and spraying the walls with sound instead of the people. For a practical guide to sub alignment, please see 6 Top Brands, 1 Sub Alignment Method.

The ideal conditions for concerts include optimized speaker directivity in bigger rooms with more absorption, lower RT, and less sound sources.

This means that a larger share of the audience will enjoy a clear sound when Q [directivity], V [volume], and a [area weighted average of the absorption coefficient] ar increased and when T [reverberation time] and N [number of loudspeakers or clusters] are decreased.

And one final recommendation from the author about placing absorption.

In the design phase, not a floor, not a wall, nor the ceiling must be left without considering low frequency absorption.

Minor Complaint

Let’s get my one minor complaint out of the way.

A challenge with the cardioid configuration can be to match the compensating delay exactly with the physical separation of the subwoofers. The location of the acoustic center of the subwoofer can vary with frequency, making it difficult, if not impossible, to find one delay setting that suits all frequencies. Failure to match delay and physical separation fully can result in significant radiation backward within certain frequency bands, which can represent a problem, for instance for the musicians on stage. All-pass filters can be used here to make phase adjustments at specific frequencies.

Adelman-Larsen, Niels. Rock and Pop Venues. Cham, Switzerland, Springer Nature, 2021.

This is an unnecessary complication of an otherwise simple design and calibration task. I’ve never had a problem calibrating directional subwoofer arrays in the field with delay only. Of course, further optimization could be made, but we’re talking about 1-10º. I asked several colleagues that I trust about this and no one had ever heard of the idea.

Here’s an example alignment of an inline gradient array using delay only. The polarity inversion has been removed to facilitate comparison.

Here’s the same array, this time aligned with an APF.

Here’s the APF I used.

Have you tried building a directional subwoofer array using APF? What were your results?

Stop recommending Linkwitz-Riley filters

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Linkwitz-Riley filters may sum well electrically, but not necessarily acoustically when they are added to a speaker. Better to recommend them as targets to achieve a unity crossover and choose appropriate filters for each individual application.

So why do people always recommend Linkwitz-Riley filters? When we look at a measurement like this and make sense, right? And we look at the sum between them and assuming that we have the necessary polarity inversion, it’s beautiful, right? It seems like a perfect result, but we should really be careful about recommending these filters when they’re used on speakers, because the acoustic result

Will not be the electrical result. 9 times out of 10, it’ll be something slightly different. So what I want to talk about with you today is how you can get started playing with filters investigating the results 

And while you should probably never recommend one specific kind of filter, unless you are fully familiar with all of the circumstances around its implementation. So today I’m going to be doing this demo and Crosslite, which is definitely a piece of software that I recommend, but you can also do this in free software.

So I’m going to do this demo and Crosslite, but then at the end, I’m also going to show you how to do it in REW. So if you want to download REW or you already have it, you can open that up and you can do all of these steps along with me, just on a slightly different platform. Okay. So learning how to use filters. It’s definitely a good idea to start playing around with them in the electrical domain. What is the result? If we have these 2 36 dB per octave Linkwitz Riley filters, what does that look like? What does it look like?If we change the topology, the slope? What if we change the frequency between them investigating all those things in the electrical domain is really good, because then you’re just focusing on the combination of these two shapes.

The next step then is to look at what happens when we start applying these to speakers. And I’ll tell you now that what I teach my students is to start thinking about these electrical filters. Not as specifically what you want to apply, but as the target, this is the result that you might like to achieve.

If you like the shape, if you like the sound of this, if you would like to create a unity class crossover, then this is a great target to pursue. So here in Crosslite, I’ve just created these two separate channels to serve as our targets.

And so what I’m going to do now is I’m going to insert a main and sub speaker that I downloaded from Tracebook, because I wanted you to be able to use the exact same data. So let me show you how I did that. So if you go to chase book.org and you go to the search page, then you can just search for subwoofers. And that’s what I did here. It just grabbed one of the first ones. And then I downloaded the CSV file. That’s just a text file that has magnitude phase and coherence values, all of this stuff that you see in this graph here.

And then I downloaded this speaker because I measured it and I’m familiar. And you’ll notice that these are two speakers from two different brands. They are not designed to work together. And I thought that would make an interesting demo since they are quite different. Okay. So I downloaded the CSV files and then I imported them here into Crosslite. So here’s my main and sub and here they are on top of my targets. So let’s start just working on the main speaker. Okay. So I know this is what, where I want to end up. So what do I need to do? I know this is a 36 dB per octave slope, but if I add a 36 DB per octave filter, is that going to give me the result that I want? Let’s try it. So I’ll change this to Linkwitz Riley 36 dB per octave. We know we’re heading towards 80 Hertz, so I’ll insert that and.

This is not what we want. We’ve gone a little bit too far, so I could play around with the frequency here, but, we can get this part to match, but not this part to match. So maybe I like this and I could insert a filter here, but I say, you know what, let me keep playing around with this. What if I go to a less steep slope so I’ll go to 24 DB per octave. And now I play around with this and I see, wow, this gets me. Really close. This gets me part of the way there. And now I’m looking at this and I’m thinking, you know what? This is not something I want to EQ because this is just some kind of reflection. There’s a little bit of ripple here, but I am curious to see what this would look like.

Without the ripple. So I’m going to do two things. Number one, I’m going to run the noise reduction function here in Crosslite. I’ve done that already. And here’s what that measurement looks like. So a little bit less ripple here. And now this is totally improper EQ, but I am going to just apply some little filters here, just cause I want to see what this would look like without this ripple, as I mentioned.

So I’m going to put in some little filters here just to make that ripple all go away. And now I have a nice fit here. So ideally it looks like I would have only had to use one filter that Linkwitz Riley 24 DB per octave filter, maybe one other parametric filter here to achieve my target.

So as you can see, we wanted a 36 DB per octave result. But we didn’t use a 36 dB per octave filter to get there because this speaker naturally, already has some amount of roll off. You’re never going to measure any speakers that are just flat from DC to daylight, that they’re just flat all the way across and then you can add filters.

That’s not the way filters work. You’re always adding on top of something that’s already there. And so we can look at the math of that and that’s easy to figure out and we can do the numbers, but it’s also just fun to play with and just add things and see what gets you to the target as efficiently as possible.

And sounds good. All right. Let’s take a look at the sub. Let’s zoom in a little bit here. What do we need to do here? This is sometimes some of the most challenging work because here at the crossover frequency, we are already locked in. By the way, it might help to have some kind of a target here. So over here, I’m going to turn on the target today. I’m just going to use a flat target.

That’s not what I always use, and you don’t have to use the same thing, but that’s what I’m using today. So I have this target. Here’s the top of it. And then I turn on this target Delta so that I can see where the -6 dB down point is. Okay. And I see that my sub is already hitting that point. And so what do I want to do here?

What if I use something super sharp?

Imagining the shape here and I’m thinking what if I come in with a filter that’s super sharp and maybe just push this part down this way a little bit. So let’s try that first. So I put in this 48 DB per octave link what’s Riley filter, and that gets me a fit here, but now I’ll still have to work on this region and this region, and maybe that’s fine, but I’ve played with this a little bit earlier before I started this video and I came up with what I think is. A more efficient solution, which is first applying a six DB per octave first order low-pass filter.

So answer this and I play around with this a little bit and oops, I’m probably moving the wrong filter. Let’s see. I need the red guy. There it is. There we go. So with this first order filter, now I’ve got this top getting closer here. I’m getting closer to the bottom here and now I just need one filter here to make a slightly better match. And now I could keep playing around with these until I find a closer match. Okay. So let’s say that I’m happy with that.

And now let’s look at everything together. so here’s both of my measurements now with the targets. Now I’m going to mute the targets. And now I can look at the sum between these guys and I already want to have kind of a picture in my mind of what I expect. So at this -6 dB point, I expect 6 dB of summation, which should hit up here, and then I’m hoping for a nice flatline transition across here. That is the goal of the unity crossover.

So I turned on the sum, and I see that I have a big problem here. And I remember that’s right. 36 DB per octave target. I’m going to need a polarity and version. And then I take a look at the max sum here, and I know it’s a little bit difficult to see because both of these lines are black, but I’m turning. Max on and off max signifies the total potential summation, if everything was perfectly phased aligned.

So I see that I still have some work to do here in the high end. So what can I do about this? I’ll turn off the sum for a second and let’s place some markers here, just so I can see this is the crossover region that I’m going to focus on for the moment. And I take a look at the phase here and I see, oh, this is slightly problematic here.

By the way, before I go down this path, I should just point out that this some that we had, this could be good enough and I could just move on, but if I want to keep playing with this. I could try different filters. I could try a different combination of filters to see if there’s something that would achieve a better phase compatibility here.

But for the moment, I’m going to try a solution with just a second order, all pass filter here. And when I move this around, maybe I can find a solution here. A different compromise. So it seems like we have a little bit better response through the entire crossover region there. And now when we take a look at the sum and we turn the max off and on now we have a much closer result.

Okay. So again, I’m not going to say that this is the best solution. This is just what I came up with in the few minutes that I’ve been working on it with you today. If I spent another couple hours on this and tried out a bunch of different variations, I might find something that I liked better.

And then ideally I could deploy all of those as presets and then do listening tests and find the one that I actually liked the best. And doesn’t just look the best on the screen. So you saw here that both with the main and the sub the filters that I applied. Didn’t necessarily have anything to do with the target. So I wanted to end up with this 36 dB per octave result that gave me this nice flat result here through the crossover region, but I just did whatever was necessary. I was just playing around and experimenting with filters until I got to that result.

So this is why it’s probably a bad idea to ever recommend. Just one kind of filter. Don’t tell people, just use Linkwitz-Riley filters, but you could tell them head towards this kind of result, especially if people are just starting out. They’ve never heard of this kind of idea, but. Yeah, totally fine.

To just start with some kind of a template, use Linkwitz-Riley filters to experiment with them in the electrical domain. Look at the result and then play around with that and use that as a target. Okay. So I promised I would also show you how to do this in REW, in case you don’t have Crosslite at home and you want to do this work along with me.

So let’s try that. So I’ve got an REW open and before I actually import those measurements, I want to show you how you can also play around with electrical filters in REW. So you open the EQ tab and then you can just insert filters. Down here. It says crossover filters and I can use the exact same thing I was using in Crosslite here. We see the filter prediction, and now if I want to actually play with that, then I go over here and I click generate measurement from filters, and then I’ll do the same thing for my low pass filter. Generate measurement from filters. There we go. Now I can play around with these and look at the result and compare phase and all that stuff. Just like I was doing in Crosslite. All right. Let’s import those measurements. So I’ll just grab them from the finder and I’ll just drop them here and here they are. So you can see this is this, the raw data. And so now I would like to take the next steps to change the gain between them so that they line up and they look more like this, what I was doing in Crosslite and then start experimenting with filters to achieve this target.

And let me just show you how to take the first steps with that. So here over on the left, do you want to select whichever one you want to apply the filter? And then go to the EEQ window. And then from here, you’re going to choose a target. So I typically choose the speaker driver, and then, we’re going to choose our high pass filter, which is Linkwitz-Riley six. And then set this at zero or three, line it up with your measurement here. And so now we have a target here and now we can start applying filters in pursuit of that target. So I think over in Crosslite we added a fourth order filter. Let’s try that. Yeah, there we go. So that’s getting pretty close.

All of these audio analyzers and modeling and prediction software, they all look slightly different, but the data is all very similar. We see a magnitude response here. We have a target, we have a prediction, so we need some way to practice this stuff. REW is great. Crosslite is great.

There’s plenty of tools out there. So pick one of them and figure out a way that you can play with this stuff at home. And so now that I’ve shown you how to get started here in REW, maybe go back to the beginning of this video and go through step-by-step as I was doing it in Crosslite and you can do the same thing in REW.

Okay. Let me know what questions come up for you. Let me know if you have any suggestions for me and thanks for watching. 

(remote) Home Theater Sound System Calibration with Smaart & Crosslite

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Back in November I had the opportunity to work on a multi-channel home theater sound system built with custom speakers. The client had their own Smaart rig so they measured the drivers and I worked on the EQ and crossover alignment within Crosslite.

What follows are clips from the meeting transcript.

We need for both of these drivers to be captured with the exact same delay locator value. Copy the delay locator value from your HF measurement. Paste it into your TF delay, and this time measure your low frequency driver again, but don’t change the delay locator.

Disable phase smoothing and magnitude smoothing and coherence blanking. Now what you’re going to do is select both of those over in the data tab and export to ASCII.

I’ve marked our crossover region here and our sums looking pretty good because I’ve done a little bit of work already. Let’s see how we’re doing in the phase. Yeah, it’s looking pretty good. So we’re adding both together, the natural row off, plus this electronic filter that I’m implementing. And so the first thing was to just check, what if we just add steeper filters instead of adding delay?

The last thing would be to now apply some EQ to make this peak go down because we did an overlap crossover instead of a unity crossover. Now I can just move this filter around a little bit and try to get a nice result there.

Yeah, I’m going to say that that’s 12dB/oct. Our goal is 24. So we need to add another 12dB/oct. So this could be perfect. Okay, let’s find our crossover region. So I’ll look for anywhere where they are 10dB apart. Delta magnitude is ten. The way you can do that in Crosslite is with these cursors. And the way you do that in Smart is you would do a trace offset. So you would offset one of these by ten and then look at this value where they still interact, where they still cross, and then you would go the other way minus ten, and then just put some kind of a marker there.

Oh, shit. It matches already with only a polarity inversion. All right, that was easy. So now we just need a little bit more EQ. Heading back to the input EQ. Okay. Should be pretty good. We can just have a look and see if we like this results. That look pretty good to you?

Let’s deploy these settings into your DSP and verify the alignment.

No delay? Shit!

How to fight power alley using end-fire arrays

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If you don’t like the power alley that results from uncoupled subwoofer arrays and you do have six or more subs and enough real estate, you can try a compromised approach by aiming the left and right energy away from the center, improving isolation and lowering variance across the audience.

I should make it clear that the result appears to reduce the power alley only in contrast to the rest of the audience. There is still interaction in the middle, it’s just lower in level compared to on-axis with the sub arrays.

A reduction of 4dB at 63Hz is found at the center of the audience.

Download the MAPP3D file and run your own tests.

Why six or more?

The 2-element end-fire array is a one note wonder. It cancels at a single frequency in the rear. A more efficient option would be the gradient array, although there are exceptions.

from Subwoofer Array Designer

How do you design an end-fire array?

Space the elements in a line so that their operating frequency range fits nicely in between the preferred filters recommended in Subwoofer Array Designer.

What’s the least amount of cabinets that can be used effectively?

As the number of cabinets goes up, so does the range of cancellation and consistency of coverage.

  • 2 cabinets: Cancellation at 1 frequency. Could be useful for a fighting a single resonant frequency on stage. Otherwise prefer gradient array.
  • 3 cabinets: Cancellation at 2 frequencies. Better an 2. Option to convert to 3-element inverted gradient stack.
  • 4 cabinets: Cancellation at 3 frequencies. Now we’re talkin’.
  • 5 cabinets: Cancellation at 4 frequencies. Even better.
  • 6 cabinets: Cancellation at 5 frequencies. Begin to approach the point of diminishing returns.

Four elements is the most common end-fire quantity because it is effective and reasonably practical. Economizing to three units sharply reduces the randomization in the rear, leading to frequency-dependent reduction. Never end-fire with just two elements. It’s a one-note-wonder on the back side. Use the gradient in-line instead (same physical, different settings). We don’t have to stop at four, bearing in mind that the horizontal pattern narrows with quantity. Get crazy! RF antennas will end-fire 10+ deep.

McCarthy, Bob. Sound Systems: Design and Optimization: Modern Techniques and Tools for Sound System Design and Alignment (p. 321). Taylor and Francis. Kindle Edition.

A more consistent polar pattern?

I had never considered this before Tamas asked about it and I was excited about the possibility. Unfortunately, my experiments do not reveal a significant improvement using 2nd order all-pass filters over pure delay.

FrequencyOpening Angle w/DelayOpening Angle w/APF
40Hz172º160º
50Hz152º152º
63Hz134º144º
80Hz120º116º
100Hz106º100º
100 – 40Hz66º60º

One interesting side effect was the development of a MATLAB script to calculate the ideal frequency and Q parameters for each APF. Let me know if you’re interested in hearing more about that and I can update the article or send you the script.

How do I use SubAligner with end-fire arrays?

Measure the distance to the main array as you normally would, but measure the distance to the furthest subwoofer. All of the other subs in the array are aligned to it as well.

Distance measurements
SubAligner recommendation
SubAligner plot

Here’s a direct link to this alignment if you’d like to use it in SubAligner.

Resulting phase alignment in MAPP 3D
Prediction at 63Hz

Don’t we need to add 4th order filters, as suggested in Subwoofer Array Designer?

Normally, yes, but in this case there is already a native low-pass slope of 24dB/oct.

Have you tried end-fire arrays on your shows? What were your results?