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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. 

6 Top Brands, 1 Sub Alignment Method

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L-Acoustics, d&b audiotechnik, Coda Audio, RCF, Funktion-One and dB Technologies all recommend the same sub alignment method. It’s no coincidence that SubAligner uses the same method.

I have published several articles discussing this method already, but here’s a quick refresher:

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

With each of these manufacturer guides there are several commonalities:

  1. Alignment is important. You deserve all of the decibels that you paid for.
  2. If an audio analyzer is unavailable or inappropriate, use the relative/absolute method 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. Some circumstances are not appropriate for an audio analyzer, like when the reflections in an arena make the data inactionable.

Interestingly, d&b would like you to start with their modeling software to find alignment, while L-Acoustics eliminates the possibility of LF alignment in their modeling software all together, preferring to provide you with explicit pre-alignment values in their documentation.

L-Acoustics

L-Acoustrics Prest Guide v18.0

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.

d&b audiotechnik

TI 385 d&b Line array design 10.6

NEXO

NXAMP Manual v3.1

Consequences of a badly aligned system
Precautions
delay
nexo alignment

Coda Audio

LINUS Control v2.2.33 Time Alignment

coda audio max perforamance
Flown systems
coda audio tiray

RCF

Pre-Alignment Delays v.1.1 – Guide EN

Funktion-One

Crossover Settings

dB Technologies

VIO series

This video should start at 2:31.

What about you? Have you tried using the relative/absolute method? What were your results?

How to phase align main to sub in Smaart, REW, Open Sound Meter, SATlive, and Crosslite

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The audio analyzer functions primarily as a verification tool. For this reason this article will focus on creating alignment presets, which can then be modified in the field using simple distance measurements.

To fit this into a single article I will offer an overview of a single method for each software. Although the steps with each tool might differ slightly, in general they follow this pattern:

  1. Measure each source solo.
  2. Do whatever is necessary to achieve alignment.
  3. Measure sources combined and verify summation against a target. Listen.

The Setup

  • Ground-plane.
  • Grille-to-grille (coplanar, side by side).
  • Microphone placed equidistant from each LF driver at a reasonable overall distance in order to capture actionable data and still measure the entire loudspeaker as a whole instead of a single driver or port. For subwoofers, this usually means going outside unless you have a very large room. (approx 5x measurement distance)
with permission from Merlijn van Veen

Set Levels

If you are designing an overlap crossover (+0dB), this is easy. Simply match solo measurements to the target and EQ out the summation bump at the end.

If you are designing a unity class crossover (0dB), this is surprisingly one of the most difficult steps because you want the end result to hit the target, not the individual measurements themselves. The goal is to hit the target in a single step. With most tools you’ll be working in the dark, trying to imagine where the sum is going to end up. This is why there’s a whole subroutine in my SubAligner app dedicated to finding the perfect level relationship to hit the desired target. Shout out to SATlive for being the only software that I now of that includes a perfect addition trace so you can set initial levels without worrying about the alignment right away.

For everyone else, you can start by setting levels at -6dB relative to the target and you’ll probably need to do more adjustments in the end once you see the final result.

Where is the spectral acoustic crossover?

For efficiency, it is recommended to focus on the area of interaction at greatest risk of cancellation where magnitude values are within 10dB of each other, aka the combing and transition zones.

Make the pictures match

Use delay, polarity, and filters to achieve your desired result. Either follow manufacturer specifications or get creative and come up with your own path. Maybe create presets for both and see which one your colleagues prefer in a blind listening test.

A common first step is to achieve alignment at a single starting frequency within the crossover region where you have high confidence (coherence). Find the phase offset (ΔPhase) between main and sub, then close the gap. Since the sources are equidistant, you might want to start with filters, but try both ways. Again, if you’re using a manufacturer’s preset, always start by following their guidelines.

If you’d like to use filters:

  • ΔPhase / 45º = Filter order to try. eg. 90º / 45 º = 2nd order (12dB/oct) filter (Butterworth, Bessel Normalized, and Linkwitz-Riley)
  • For all-pass filters (APF): ΔPhase / 90º = Filter order to try.
  • High-pass filters (HPF) will cause positive phase shift.
  • Low-pass filters (LPF) will cause negative phase shift.
  • It may be easier to see this in action on an unwrapped phase plot.

Applying filters is a big topic outside the scope of this article, but if your interested, please see Phase Alignment Science Academy.

If you’d like to use delay:

  • ΔPhase / 360 / Frequency * 1000 = time in milliseconds
  • If you need to wrap around the top and bottom of the phase graph then use 360 – ΔPhase. eg. If the measured phase offset between two points is 200º, but the traces are near the top and bottom of the graph and you suspect that they need to wrap around, then 360º – 200º = 160º Δphase.
  • Once you have a single frequency aligned, test out other variations at half and whole cycles away. For half cycles, add a polarity inversion. eg. If you’re aligned at 100Hz then try variations at +5ms INV, +10ms, -5ms INV, -10ms.

If you’d like to consult the Southern Oracle, you must first pass the Sphinxes’ Gate and the Magic Mirror Gate.

Verification

After you have tried several variations, choose the one who’s combined result best matches your preferred target. To break a tie, use the option with less delay or less processing overall. Listen to the result or audition multiple presets to find the one that sounds the best.

Smaart

One of the reliable things about Smaart is that the data will never change after it is stored outside from the quick compare function. This means that any change you care to make must be implemented directly in your output processor and then measured in real time.

  1. Add 10ms of delay to both outputs. The amount of delay is arbitrary, but will save you time in step 6.
  2. Measure the Main solo and capture the trace.
  3. Without changing the compensation delay, measure the Sub solo.
  4. Set the sub level to match your target trace. Capture the trace.
  5. Find the spectral crossover using trace offsets.
  6. Make the pictures match.
  7. Verify alignment and summation. Listen.
  8. Remove any extra delay left over from step 1. 

Here’s an example combining an L-Acoustics X15-HiQ with an SB118. Initial measurements reveal a 38º phase offset between them. We might first attempt to close this gap with 1.16ms of delay on the sub (38º / 360 / 91Hz * 1000), but further tests would reveal an improved alignment with a half cycle of delay and polarity inversion in the main.

Recommendations from SubAligner and the L-Acoustics Preset Guide confirm this result. If you’re a SubAligner user you can open this direct link to the alignment.

Tips: For high quality actionable data I recommend setting temporal averaging to Inf and resetting the averages with each new measurement. Consider downloading measurements from the manufacturer, Tracebook, or SubAligner in order to have some expectations to work against.

REW

The rest of the audio analyzers covered in this article offer functions to simulate output processing. In REW the EQ window allows you to experiment with different filters and then generate a new measurement that includes those filters. Then you can experiment with gain, delay, and polarity using the Alignment Tool and its auto solver options.

  1. Measure Main solo.
  2. Estimate IR Delay. Shift and Update Timing Reference.
  3. Measure Sub.
  4. Find the spectral crossover using Measurement Actions.
  5. Experiment with filters and the Alignment Tool to make the pictures match. Generate an Aligned Sum for each variation.
  6. Compare all of the Aligned Sum variations for alignment and summation. Listen.

Tips: For high quality actionable data I recommend setting the number of measurement repetitions to 8 and the length to 256k.

Open Sound Meter

  1. Measure the Main solo and capture the trace.
  2. Without changing the compensation delay, measure the Sub solo. Capture the trace.
  3. Set the sub level to match your target trace.
  4. Find the spectral crossover using gain changes.
  5. Make the pictures match. You can click on a measurement and adjust its delay and polarity while watching a sum trace calculated with File > Add math source.
  6. Verify alignment and summation. Listen.

In this image you can see me creating the sum trace on the left and then manipulating the main trace on the right to achieve better summation.

SATlive

SATlive includes some of my favorite tools for crossover alignment, which were my inspiration for getting started with SubAligner. The Live Add trace gives you a real time crystal ball preview of what the combination of main and sub will look like. The Perfect addition trace creates a target so you can see how well you are doing. The Delay-Suggestion Tool will run an auto solver and make recommendations for delay and polarity. The Area Of Interaction Tool can be used to visualize the crossover region.

  1. Measure the Main solo and capture the trace.
  2. Without changing the compensation delay, measure the Sub solo. Capture the trace.
  3. Set the sub level to match your target trace while observing the Perfect Addition trace.
  4. Find the spectral crossover using the Area Of Interaction tool.
  5. Make the pictures match with the aid of the Delay-Suggestion Tool.
  6. Verify alignment and summation by comparing the Live Add Trace against the Perfect Addition Trace. Listen.
satlive

Crosslite

Crosslite also includes auto solver functions, but instead of using a brute force iterative approach, it will attempt to align the start or peak of the impulse responses, which can be filtered to focus on the crossover region. One of my favorite tools in Crosslite is the cursor. It can be enabled to find the phase difference between measurements and even converted into time for the alignment. Crosslite also offers various filter options and can be thought of as a full DSP simulator.

  1. Capture the Main and Sub solo.
  2. User Memories > Functions > Sum > Process Method > Sum Magnitude to generate a perfect addition trace. Adjust the sub level until the Sum Magnitude matches your target trace.
  3. Find the spectral crossover either using Gain or cursors.
  4. Make the pictures match. The most efficient starting point may be found by inserting a peak filter at the input around the center of the crossover region and running the Optimize Time function. Experiment with changing the alignment to rise or peak and the filter from normal phase to phase zero. The best option here may depend on the quality of the measurement data. Always check the phase graph afterwards.
  5. Verify alignment and summation. Listen.

Next Steps

Now that you’ve created an alignment preset, it can be deployed and modified in the field using distance measurements. If you’d like to send me the speaker measurements you took along the way, I’ll add them to the SubAligner app.

How to practice at home without a PA

You can download lots of high quality data from Tracebook to practice with.

Have you tried any of these softwares? What method do you use to optimize phase alignment between main and sub?

Can you remove reflections from live measurements for more accurate alignments?

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In my last article I found some minor success in validating subwoofer alignment indoors through the use of carefully chosen microphone locations and their average. Next up: Can it be done live in real time?

Thierry De Coninck sent me an article he had written outlining a procedure that would allow you to not only create these averages, but also set the delay locator for your transfer function based on distance to avoid any errors from reflections or lack of HF content arriving at ground plane. I also found an AES paper called Measurement of Low Frequencies in Rooms, which outlines a similar method through the post processing of IR measurements.

A new method is outlined for measuring frequency response at low frequencies. This method uses microphones arranged in a low frequency end-fire array to create useful directivity to discriminate against sound waves from rear wall reflections and reverberation. It also operates in the time domain, processing the acoustic impulse response as it arrives at successive microphones – a shotgun microphone writ large.

D. Murphy, “Measurement of Low Frequencies in Rooms,” Paper 9482, (2015 October.).

What interested me most about Coninck’s method, though, was the ability to get results in real time. Here’s the outline:

  1. Deploy two microphones (or more, potentially).
  2. Measure the furthest position setting its delay locator automatically or by distance from the source.
  3. Set the delay locator of the next measurement position manually based on the difference in distance to the source from the previous measurement. (eg. If the second measurement position is 1m closer to the source then offset the delay locator by 1m / speed of sound.)
  4. Verify that the LF phase of both measurements has some agreement.
  5. Create a live average.

Initially I thought I would create a script in MATLAB to automatically calculate the delay locator setting for each transfer function pair using pythagorean’s theorem, but looking at some initial results convinced me that, if possible, setting the delay locator automatically from the IR would be more accurate. Since my goal is to measure and align at head-height then the microphones should receive enough high frequency content to make this possible.

The Setup

Here’s a section view of the sound system at the Sojourn Campus Church here in Minneapolis. Adam Rollin from AVE was kind enough to let me tag along and do some tests during his normal system calibration work.

I put these details into the Sub Align calculator and could see we had a small phase span of 28º. I decided we would do our alignment at the last row and distribute the microphones evenly from there.

Here’s a comparison of all six measurement mics at about mid-depth of the audience after level and polarity match (1/24oct smoothing). You can see that we’ll have our work cut out for us in the low-mids.

Alignment

We used the Relative/Absolute Method to complete the main+sub crossover alignment. Here are the native solo measurements at 1m. The sub trace is offset -6 dB due to half-space loading.

At first I thought the wonky phase around 1kHz was there because the mic was too close, but we found the same thing in the far-field. I’ll have to save the story of that polarity inversion for another time, though.

I wanted to reduce the overlap and improve the alignment without adding a lot of delay. An LR48 HPF was inserted on the main at 42Hz. This left us with the pre-alignment value of 2.4ms of delay on the main and a polarity inversion.

Why the polarity inversion? Compared to other possible variations it allowed for less delay and improved alignment.

Here’s how we deployed the microphones.

Here are all six traces with the average of the main.

At first I though, “Success! We removed the reflections.” Then I noticed the wrap at 200Hz. What the hell?! You’re supposed to disappear!

I can tell it’s not supposed to be there by comparing the average with the near-field measurement.

Where’s it coming from? Is it a floor bounce, a ceiling reflection, or something else?

If it’s a floor bounce, shouldn’t we be seeing a 2.5ms peak in the live IR graph in every measurement? Not necessarily since any peak in the low-mids would be down ~40dB and hardly visible.

There’s one way to rule out the floor-bounce for sure. Here are the ground-plane measurements.

Here is the average ground-plane vs near-field. Good-bye 200 Hz phase wrap.

Why does this matter?

The wrap at 200Hz caused by the floor-bounce causes phase shift through the crossover region and therefore misalignment.

What’s the solution?

Don’t align to a reflection. Do align to direct sound by using the Relative/Absolute Method or SubAligner.

A distance measurement from the alignment position were almost matched so we decided to leave the pre-alignment delay value unchanged. Here’s the alignment at head-height.

Everything is within the corridor of 60º, but if you wanted to be picky you might decide to use less delay in the main to attempt to close the 30º phase offset at 70 Hz. This may create a misalignment, though, because the floor bounce is creating the extra phase shift.

Attempts to remove the phase wrap at 200Hz by varying the microphone positions over height and width were unsuccessful. A majority of the traces always included a dip around 200 Hz.

Conclusion

I started down this path a year ago in order to prove the results of the Relative/Absolute Method, and therefore, the necessity for a tool like SubAligner. The Catch22 is that the more you need SubAligner, the harder its results are to validate. If the phase trace is inactionable, SubAligner can’t fix that. It just ignores it. And while a measurement at ground-plane will help with visualizing the data, it will hurt the alignment because we don’t have ears on our feet.

Using an audio analyzer to validate SubAligner’s results is like trying to validate the presence of an optical illusion by taking a photo of it. It will only enhance the illusion.

Here are some takeaways from Sojourn:

  1. No amount of measurement data and mic positions can guarantee a reflection-free average in the far-field (especially with poor direct-to-reverberant ratios). The null created by a floor bounce or other boundary may be sufficiently common to each individual trace that it ends up in the average.
  2. Since a measurement cannot be guarantee 100% reflection free, there is always risk for misalignment. Gathering more data in a strategic manner can help reduce this risk, but the only real solution is to abandon the audio analyzer for a more appropriate tool.

How to Estimate Delay and Level Offset Between Speakers in Your 3D Models

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Save time in the field by presetting your speaker output delay and level offsets.

Here’s a model from a recent show where I estimated the level and delay for a pair of relay speakers and line of front-fils.

If you know the difference in distance between two sources, you can accurately estimate their time offset, and therefore, the delay necessary to bring them into alignment. The challenge is finding the correct crossover point.

Drawing About Sound

Crossover (spatial): An acoustic crossover in a spatial domain, i.e. the location where elements combine at equal level.

Bob McCarthy, Sound Systems: Design and Optimization

Finding a spatial acoustic crossover point in the field is pretty simple. Once you have defined the on-axis (ONAX) point for source A and source B, match their levels, then walk in a line between the two points until they sound equal. Place our measurement microphone at this point, then use the audio analyzer for verification.

You can simulate the process in a modeling and prediction software like MAPP XT, but I have been practicing a quick offline method of drawing speaker coverage shapes and estimating their crossover points that you can do on a plane (or even with pencil and paper in 2D).

Here’s a recent show that I worked on using JBL VRX 932 speakers on stands. The distance from the speaker to the end of coverage is 61.66ft. I can divide this distance by the forward aspect ratio of this 100º speaker to find the coverage width.

61.66ft / 1.31 = 47.1ft

For more on forward aspect ratio, please read One Simple Tool to Find the Right Size Speaker for Any Space. These topics are also covered in detail inside of Pro Audio Workshop: Seeing Sound.

This gives me a quick sketch of the coverage shape.

For comparison, here’s a similar speaker’s prediction at 4kHz.

The Steps

Here’s how you would do this in the field with the audio analyzer:

  1. Match solo ONAX levels B1 to A1.
  2. Find XAB (the acoustic crossover between A and B) between ONAXA and ONAXB where A1 and B1 match in level .
  3. Set delay.

Now let’s do those steps in our model with distance measurements.

Set Level

B1 level = 20 * Log10((distance from ONAXB to B) / (distance from ONAXA to A))

In our model that would be -2.4dB.

20 * Log(22.59 / 29.78)

You can drop that formula into google to verify it.

Find XOVR.

Find XAB at the center of their coverage pattern overlap.

Set Delay

B1 delay = (distance to A1 – distance to B1) * 0.9

In our example model that would be 24.17ms.

(48.01 - 21.16) * 0.9

Field results

Once we got into the room the coverage shape of B1 changed due to an obstruction. This changed the level offset, but the delay was very close to my estimate.

Much of the front-fill processing changed because we replaced two of the 928 speakers with 932 models and were able to level set and delay them independently.

Will this work for subs?

Yes for delay. If you have followed Merlijn van Veen’s Relative/Absolute Method, then you already have a preset for the relative relationship of the spectral crossover alignment of your main and sub. Use your 3D model for the absolute step.

No for level. Low-frequency energy will enjoy room gain and I’m less confident that a distance measurement will help you estimate the level. If you try it, let me know.

Have you tried presetting your DSP using distance measurements from a model? How did it go?