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What do sub alignment sound waves look like?

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Have you ever wondered what the sound waves of your sub-alignment look like?

Coupled, Equidistant, Aligned

If your main and sub are right next to each other and phase aligned when equidistant then you can send them the exact same signal and expect the audience to be in the coupling zone.

Here you can see a ripple tank example of two sources encountering an audience represented by a white box. No reflective boundaries have been enabled.

Uncoupled, Unequal, Aligned

If your main and sub get separated, as is often the case with flown mains and ground stacked subs, then they are still phase aligned somewhere, but maybe not where your audience lives.

In this example you can see a power alley (coupling) where arrival times are match along the isosceles triangle and power valleys (cancellation) where the triangle becomes obtuse.

Uncoupled, Equidistant, Aligned

You can equalize the displacement problem with a physical offset or digital delay. Notice how the isosceles triangle rotates down towards the audience. Edit this example here.

Uncoupled, Equidistant, Misaligned 120º

What if you’re not sure if your main and sub are aligned when they are equidistant? You might fix the displacement problem with delay, but unknowingly leave an alignment problem. Now part of the audience is in the cancellation zone.

In this example I have introduce a 120º phase offset into the bottom source.

Uncoupled, Equidistant, Misaligned 180º

In this worst case scenario, you have equalized the distance offset, but accidentally put your audience fully into the cancellation zone. This could happen with an accidental polarity inversion or simply with two speakers that are not designed to be phase compatible when they are equidistant.

The moral of the story is to know your gear. Don’t assume a successful alignment has been created by only equalizing the distance offset.

FOH Engineers: How to sneakily check your system tech’s work

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If you are a FOH sound engineer then you often work on sound systems that someone else has set up for you and you need some way to make sure it was done properly. Whether it’s your best friend in the most famous venue in the world or the local AV provider in small town USA, you’ve got to cover your ass.

Before I go any further, it should be clear that I’m exaggerating here to have a little fun. Open communication is the first best strategy for a safe and successful live event.

No matter the conditions or the people, you always need to verify that things are in place so that you can do your job to the best of your abilities and make your client happy. In this short article I’m going to focus on one of the more difficult things to verify: sub alignment.

We all want a solid low end and for the audience. It’s one of the main reason they even leave the house.

Certain bits of research have found that upwards of around three-quarters of what people determine is high quality sound comes from the low frequency content. It’s really important to get the low end correct at live events.

Adam Hill

So if someone else set up the sound system, how can you check their work without redoing everything yourself or generating mistrust?

How much delay did you put on the mains?

A well-formed question can reveal a lot about someone’s work and can be thrown in as part of a getting to know you conversation.

Take a look at the speaker placement and make a quick estimate about how the sub alignment should play out. In a common situation with ground based subs and flown mains, you can assume that the subs will need some delay to equalize the physical distance offset.

If the mains are 10ft up in the air, you can estimate about 10ms of delay. But instead of asking, “Did you delay the subs 10ms?”, be sneaky and say, “Hey, how did the sub alignment turn out? How much delay did you have to put on the mains?”

You expect them to say, “Mains? You mean subs, right?”

Despite an obvious distance offset, it has happened to me in the field that the mains were delayed 30ms. When I asked why the tech said, “Because that’s what Smaart told me to do.”

I totally get it. I’ve been there. You’re trying to measure with a single mic in some shitty middle school auditorium, and the phase trace is FUBAR. So you do your best with the time and resources and miss the mark. Then later there’s a forehead slap moment when you realize that a visual check renders that solution invalid.

Other examples:

  • Did you have to invert the phase on any subs?
  • I usually high-pass my LR buss, my matrix outputs, and then again in the DSP. Is that OK with you?
  • What’s your favorite crossover filter? I always use Linkwitz Normalized.
  • I’d prefer to use 4th order all-pass filters if possible.

Breakfast measurements

I got this idea from my friend Clint. Imagine that you arrive at the venue in the morning. Nothing has been loaded in yet, and you’re still drinking your coffee. You can take a couple of measurements with your laser disto and either run them through SubAligner or do some quick math.

You’re tapping on your phone and no one is the wiser. A little later you ask the system tech if she can walk you through the output setting. When she does you’ll have a comparison in case a red flag needs to be waved.

Let’s look at two examples.

1: Math

  • FOH to Main = 30ft
  • FOH to Sub = 20ft
  • 30 – 20 = 10ft distance offset
  • 10 * 0.88 ≈ 8.8ms time offset

2: SubAligner

  • FOH to UPQ = 30ft
  • FOH to PRX618S = 20ft
  • Recommendation: Delay the Sub by 2.95ms and invert the polarity

You can even look at the plot and then have some expectation of what the measurements should look like in case that comes up.

Listen to color pulses

Color pulses are great for focusing in on different frequency ranges. Check out Merlijn’s article and download his examples.

You might load them onto a track in your virtual sound check or on the playlist with your reference tracks. Then, when you’re doing your listening tests throw one of those on and try inverting the sub polarity. In a well-behaved system in a well-behaved room, you should hear cancellation.

There are various factors that can break this test (see Don’t Align Your Subwoofer to a Room Reflection), but at least it will provide you with some potential reason for further investigation.

Have you tried any sneaky ways to check a sound system setup? Let me know in the comments.

How to use all-pass filters without a fancy DSP

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All-pass filters can be a very helpful tool to get two speakers to work together, especially when they are from different families or brands. Unfortunately, they are normally only available on higher end output processors and amps like Meyer Sound Galileo, Lap Gruppen PLM series, or Lake.

There is a free solution, though, which I’ll be testing out in this post.

The Setup

After my interview with Michael Curtis I’ve been looking at Reaper more and more. For these tests I’ll use the ReaEQ that comes with it.

Although Reaper isn’t technically free, it does come with a 60-day evaluation license. I did do some research into other available VST plugin hosts, but haven’t tested them, yet, so I’ll save that for another post.

A few years ago I published a video called The Poor Man’s Galileo. You can see that I’ve been interested in replacing various parts of my hardware setup with software for years. I wish I was a flute player. I love gear, but I hate how heavy and bulky it all gets. I love tools, but it’s hard to justify carrying them everywhere if I I’m not that I’ll actually need them. I always have my computer, though, which makes the software solution attractive.

I set up two computers with Reaper. One to host the plugin and proces the audio and the other to record. I created two record channels. One that was a simple loop from out to in so that it would experience the same A to D process, effectively removing it from the measurement. The other went to the OCTA-CAPTURE for live processing.

I ran a sine tone for verification and discovered that there was no block size too low. With only the ReaEQ inserted I was unable to interrupt processing with any errors. Once I dropped below 29 samples, though, Reaper stopped reporting a lower latency. In a later test, when I tried inserting an Ozone 9, I had to bring the block size back up to 32 to perform without any audible hiccups.

highest sample rate & low block size
lowest reported latency

Latency

The main thing I wanted to confirm here is that the latency would be low enough for live use. How low is low enough? That depends on everything else in the system and the application. For an IEM monitor rig, 20ms might be too much after other possible latencies in the system, but probably wouldn’t make that much different for a big outdoor gig.

I guess the big question is whether the benefit is worth the expense. If you making a big improvement to a two octave wide crossover region and the latency seems acceptable, it’s probably worth it.

With no plugins inserted I measured a round trip latency of 5ms for my test setup. It stayed the same after inserting the ReaEQ. Just to confirm that I would see a change, I tried inserting an Ozone 9 with 3 modules and the latency jumped up to 75ms.

Conclusion: I would be happy to use this setup as a backup solution in the field. If there’s an internet connection or cellular data connection, I can get Reaper installed on any machine in a few minutes.

Have you tried live processing with plugins system calibration in the field? Let me know in the comments below.

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.

SubAligner vs. L-Acoustics Preset Guide (K2 + SB28, Kiva + SB15m)

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The L-Acoustics Preset Guide is not only a valuable resource for subwoofer alignment, but is also one of my inspirations for creating SubAligner. If L-Acoustics can create alignment presets for their speakers then I should be able to scale the same method to any other speaker pair.

It also provides a quality check for results from SubAligner: Do they match? Why or why not?

A majority of the Preset Guide deals with loading the appropriate preset into the amplified controller for a particular speaker.

The Preset Guide describes the recommended loudspeaker configurations for each system, with the corresponding factory presets and the main resulting acoustic properties.

But there’s a big section at the end called Pre-alignment delay values.

For some loudspeaker enclosure combinations, it is necessary to adjust the delay values for time-alignment. Refer to section Pre-alignment delay values (p.66).

This is where a method for crossover alignment is described to be used “If no acoustic measurement tool is available,” or if I might add, if actionable data is otherwise unobtainable. When might actionable data not be obtainable? Please see Don’t Align Your Subwoofer to a Room Reflection.

The procedure goes like this:

  1. Measure the path difference
  2. Geometric delay = path difference / speed of sound
  3. Alignment delay = geometric delay + pre-alignment delay

Let’s see how the results from SubAligner stack up against the official guide.

K2 + SB28

If you were to attempt to align these two speakers with a distance offset only, you would have a pretty bad day.

k2 with SB28

The preset guide recommends a polarity inversion and 0.5ms of delay in the K2 output.

SubAligner recommends a polarity inversion and 0.92ms of delay in the K2.

A difference of 0.42ms is only about 4% at 100Hz (or 14º). I haven’t interviewed the person who created the preset guide, but I’m happy to see that the human and robot result are in good agreement.

Is there an alternative solution?

If you would like to use less delay or create a crossover region with less overlap you can open the Filters pop-up in SubAligner. The first solution you’ll see is a 4th order Butterworth filter at 73Hz.

Kiva + SB15m

These two speakers are only about 80º apart out of the box so I wouldn’t worry about the alignment too much.

The preset guide recommends 1.4ms of delay on the SB15m.

Meanwhile SubAligner recommends 1.8ms of delay.

It’s interesting to see a difference of 4% again. My assessment, they agree.

SubAligner does give some recommendations for filters, but they provide no significant reduction in delay or overlap.

dV-DOSC + dV-SUB

When equidistant, these guys are ready to party.

Notice that 4% difference again? If I had to guess while splitting some hairs, I would say that the SubAligner results are giving more weight to the lower frequencies of the crossover region while the L-Acoustics recommendations give more weight to higher frequencies. The L-Acoustics result will produce a tiny bit more summation at the top of the crossover region and will allow you to turn your sub up louder in relationship to the main with less misalignment. On the other hand the SubAligner result gives a tiny bit more summation at the low end of the crossover region and allows you to turn the sub down lower in relationship to the main.

It’s also interesting to note that all of these alignments produce a crossover bandwidth of 1.7-2.1 octaves and if you average all 220 L-Acoustics alignments in SubAligner you’ll get 1.44 octaves.

Have you tried any of these presets in the field? What were your results?