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Do ground stacked subwoofers really give you 6dB for free?

By Nathan Lively

+6dB
  • Any measurements made at ground plane will always be 6dB louder, despite subwoofer distance.
  • Flown subwoofers give you 6dB for free just like ground stacked as long as a maximum height is respected.

I have always thought that ground-stacked subs would produce 6dB more SPL because they are coupling with the ground. You’ve probably heard people say things like:

  • “Having subs next to a boundary gets you 6dB of additional sensitivity”
  • “The ground is like a mirror, doubling sub energy. “

I was just running a couple of tests for myself to confirm the reliability of this number, but I couldn’t. In fact, all of the models that I tested actually returned lower average level when the subwoofers were on the ground compared to flown in the air.

50ft Room

50ft room model
50ft room measurements

In the low range the flown sub dominates, but later the ground sub takes over.

150ft Room

150ft room model

I also thought that a corner placement would give you even more SPL, but I couldn’t prove that either.

Maybe the room gain from the walls is making the phenonenon harder to verify. Let’s try a bigger room.

300ft Room

300ft room model
300ft room measurements

Rats.

150ft – Outside

Surely if I turn off all of the walls, except for the floor, we’ll get that free 6dB.

150ft outside

Damn it. I want my free money back.

Why is this happening?

I have a two ideas:

  1. I put the mics in the wrong place.
  2. Flown subs get 6dB for free, too, baby!

The Mirror Effect

mirror effect

I had been taking ground plane measurements in the first models to remove the floor bounce, which made sense to me since I assumed an audience filled with people would have the same effect. There are two problems with this:

  1. Below 100 Hz human bodies have little absorption, which is mostly where our subwoofers live. Plus, it’s hard to predict exactly how the bodies will be distributed.
  2. If my measurement is coupled with the floor, it effectively shows half-space loading at any distance due to the mirror effect.

If the listener is located at the boundary he will hear a 6 dB louder direct signal than he would have heard if there was no floor regardless of where the subwoofer is located.

Comments On Half Space, David Gunness

If we zoom in on on the y-axis, it’s hard to tell which is which because any comb filtering is eliminated. No matter how far away the sub gets, measuring at the ground will show coupling.

ground plane only
Ground Sub
flown at ground plane
Flown Sub

If that’s the case then I should try measuring at head height.

Head Height

Let’s simplify the test by removing the walls and ceiling and using a single microphone position so that I can actually get this article done this year. I’ll move the mics up to head height, since that wasn’t working earlier.

Here’s the result from the same 300ft room, this time without walls, ceiling, or a 3-mic average.

Flown subs get 6dB for free, too, baby!

It is shown that a flown subwoofer [will] have a similar far-field efficiency to that of a ground-stacked subwoofer when a maximum subwoofer height is respected. This maximal height is linked to the venue depth via the DHER criterion, and depends on the listening height. For a standing audience, SPL efficiency is recovered for listening distances that are 5 times or more the subwoofer height. The audience benefits from a more homogeneous SPL distribution and an important SPL reduction close to the stage.

AES Convention Paper 10051, On the efficiency of flown vs. ground-stacked subwoofer configurations, Etienne Corteel, Hugo Coste-Dombre, Christophe Combet, Yoachim Horyn, and François Montignies

Pretty cool, right?

So if we want to recover SPL efficiency at ¾ audience depth, then the sub can be as high as 36.75ft and we get the added benefit of an improved front-to-back ratio.

5x rule for sub height

The real benefit of ground stacking has to with the fact that listeners’ ears are not typically on the floor, but four or five feet above it. If the speakers are elevated above the floor, 45 degrees above horizontal from the listeners perspective, the ground bounce will produce its first comb effect notch at about 80 Hz. If the elevation angle is 30 degrees, the first notch moves up to 113 Hz. If the subwoofers are on the floor, then propagation is parallel to the floor and there is no ground bounce. Hence, there is no comb effect.

Comments On Half Space, David Gunness

It’s interesting that our measurement position is less than 10º, putting the first dip from the comb filter at 297Hz, well out of the operational range of this subwoofer. If you wanted to create a null at 125Hz, you would measure at 72ft depth at 23.8º with the sub.

9.5º

The number isn’t black and white, of course. Even with the first null at 297Hz there is a 3dB drop at 140Hz.

floor bounce calculator
Merlijn Van Veen – Floor Bounce Calculator

So we can see that as we move closer to the sub, the difference in distance between direct and reflected sound will affect more of the operational range of the sub. At this time it is my understanding that with flown subwoofers we accept some amount of comb filtering in the front portion of the audience (anywhere before height of sub multiplied by five for standing head height) in exchange for improved front-to-back ratio, coupling with mains, and SPL efficiency in the rear portion of the audience.

What are your experiences on half-space loading? Comment below.

Guess How Many Measurements Positions You Need for Proper Output EQ (it’s more than 1!)

By Nathan Lively

Subscribe on iTunes, SoundCloud, Google Play or Stitcher.

Support Sound Design Live on Patreon.

In this episode of Sound Design Live I talk with the Product Manager at L-Acoustics, Scott Sugden. We discuss the automation built into the new M1 measurement software, why you can’t trust measurement data above 10kHz beyond 80m, and the specific number of measurement positions you need to represent an audience for proper output EQ choice.

I ask:

  • How did you get your first job in audio?
  • What are some of the biggest mistakes you see people making who are new to L-Acoustics systems?
  • What’s interesting for me about the P1 processor is that you have managed to automate some procedures that we would normally do manually. Would you talk about one or two of those procedures and maybe why L-Acoustics decided to pursue this kind of automation?
  • Tell us about one of the biggest or maybe most painful mistakes you’ve made on the job and how you recovered.
  • From FB
    • Kevan Atkins: During a demo I attended for L-ISA, he made a claim that it addresses the problem of comb filtering in system design but didn’t really expand on it. I’d be curious to hear him talk about this in more detail.
    • Haniel Trisna: Explain the idea of boosting high mids in the middle boxes to air compensate for long throws instead of the top box (generated by auto FIR in Sound Vision), and where can we learn more about working with the auto FIR and auto splay.
    • Primož Vozelj: Are they working on vertical processing of their line arrays (like ArrayProcessing, MLA etc.)?
    • Calum Young: I’d love to know what sort of measurement equipment / facilities / testing procedures they go through while developing new units. Is most of the work and decision making done in simulation software pre building prototypes, or is there more extended testing / voicing of units?
    • Steve Knots: If Greek amphitheaters were designed to put the audience in the best place for sound, why are we not creating clubs, theaters and control rooms in similar architectural style?
    • Roy Sputtz: Is the idea of true stereo in live sound a myth?? And also why isn’t anyone making an all weather line array?
    • Ockert Marais: Are they planning on supporting, mic correction curves and z-weighted weighting curves on the P1?
  • What’s in your work bag?

About 8 microphone positions distributed evenly in the center mass of the coverage of a loudspeaker is pretty representative of the overall. The likelihood of a poor EQ choice because of that is pretty low.

Scott Sugden

Notes

  1. All music in this episode by Bodo Felusch.
  2. Justin Vernon and Bon Iver
  3. Cadac J type
  4. L-Isa, Soundvision
  5. Workbag: iSEMcon 7150, DPA 4007, Digigram Cancun 442
  6. Book: The Signal and the Noise: Why So Many Predictions Fail–but Some Don’t
  7. Podcast: 20,000Hz, 538 Politics, Science Vs, Planet Money, Freakonomics, The Flophouse
  8. Quotes
    1. A common mistake made with a lot of systems is the expectation that you can solve your problems of a bad design after you install it.
    2. The lack of knowledge; it’s hard to be aware that you don’t know something. That’s one that only comes with time, experience, and making mistakes.
    3. If you have just one mic at FOH or one at FOH and one 20ft off stage of that, the likelihood of an EQ choice not being representative of the audience is really high.
    4. We have taken measurements outside in atmospheric conditions that are good and found that at 80m from measurement to measurement on average is ±5dB at 10kHz. This means that if you see a measurement with a bump of +1dB at 10kHz, you can’t know if that’s the reality or it isn’t. Even a long average doesn’t help.
    5. We tend to look at measurements and think they are some hard fact, but if you’ve used any measurement software outside at distance, you watch the curve move around and when you think it looks good you hit store.
    6. The best thing we can do is use the modeling environment to find the best result, especially at distance, and then get outside and verify behaviors.
    7. It’s hard for anyone to be an expert at all things. It’s an important part of career growth to identify what you can or want to be an expert at and then support yourself by surrounding yourself with other people that reinforce those skills, not reproduce them.
    8. It’s better to figure out what you like and hone down on that expertise than to try to cover every little thing.
    9. Once you get used to the workflow [of the P1/M1] the savings in workflow time and organization of your data is a 10x increase.

Merlijn’s Subwoofer Alignment Method Will Make You Feel like a Jedi Master

By Nathan Lively

merlijn jedi

Subscribe on iTunes, SoundCloud, Google Play or Stitcher.

Support Sound Design Live on Patreon.

In this episode of Sound Design Live I talk with the senior technical support and education specialist at Meyer Sound, Merlijn van Veen. We discuss subwoofer alignment, subwoofer spacing, and M-noise.

I ask:

  • In Subwoofer Alignment: The foolproof relative / absolute method you describe a process of comparing two sources in the near-field when they are side-by-side and measurement conditions are favorable, creating an alignment preset, and then deploying that in the far-field with complementary delay to correct for any distance offset caused by moving the speakers apart relative to the listening position. Unfortunately, most of us attend a seminar where we learn how to align two sources and it seems pretty straight forward. Then we get into the field and the whole thing falls apart. Why can it be so challenging to get actionable data in the field and how did you came to develop the relative/absolute method?
  • You published a series of articles on your site called Mind the Gap, in which you share the performance improvements in directional subwoofer arrays caused by adding an air gap between enclosures. You end the articles with this: “the challenge becomes to determine the minimum required gap size for improved rejection without a noticeable increase in lobing.” Do you have an update for us on this subject and any further information on the minimum gap size?
  • Could you give us a run down of the settings you use in your audio analyzer? smoothing, graph limits, averaging, etc.
  • What is M-noise? Do I need to start using it as my test signal in Smaart?
  • From FB
    • Dave Gammon: If he had hair…. would he have a mullet or ponytail…
    • Swapnil Wakodikar: Accessible software for all which provides stimulation of Line array and subwoofer configuration.
    • Ockert Marais: If you could only teach a single lesson about sound system optimisation for your entire life, What would it be?
    • Thorsten Bunz: Did having your own education site and writing articles help you get the job at meyer? How did it change your career?
sound-design-live-touring-foh-sound-engineer-job-Merlijn_van_Veen

If you ask a violin player to describe their violin, you’re going to get an 8-hour lecture because he knows his instrument intimately. He knows everything there is to know about that instrument because that’s how he makes his money. Ask an engineer to describe the phase response of the loudspeakers that he works with regularly and chances are you will hear crickets.

Merlijn van Veen

Notes

  1. All music in this episode by Derrick Bryant.
  2. Meyer Sound, MAPP XT, M-Noise
  3. Merlijn’s starting audio analyzer settings: 1/48oct resolution, ±30dB with 10dB divisions, MTW FFT resolution, Complex magnitude average type, 16 FIFO or 1sec average
  4. SC0403-A task group
  5. Sound system Design and Optimization: and em Español.
  6. Quotes
    1. It’s notoriously hard to absorb long wavelengths.
    2. If you have really unfavorable conditions, even using a gratuitous amount of smoothing, typically, will not rid you of those fake wraparounds.
    3. If you ask a violin player to describe their violin, you’re going to get an 8-hour lecture because he knows his instrument intimately. He knows everything there is to know about that instrument because that’s how he makes his money. Ask an engineer to describe the phase response of the loudspeakers that he works with regularly and chances are you will hear crickets.
    4. I don’t consider ripple a bad thing. It’s arguably the most important metric that there is in interpreting an analyzer because it gives you an understanding of the degree of interaction and direct to reverberant ratio.
    5. It’s not about wrong or right. If you know what you are doing, anything goes. If you want your analyzer to become an ally, then the analyzer should render the sound as crappy as it sounds, not paint a picture from a data sheet.
    6. It makes no difference which signal we use when it comes to obtaining a transfer function. M-noise does not change my calibration practice.
    7. Calibration is the process of making it sound the same everywhere. Voicing is the process of “How should the sound system ultimately sound?”.
    8. In the absence of a viable alternative, I think MAPP is still the ultimate sandbox to experiment with these things while looking at data that you will run into in the real world.
    9. Vince Lombardi: Excellence is achieved by the mastery of the fundamentals.

Dave Rat’s End-Fire Adjustable Arc Subwoofer Array Explained

By Nathan Lively

During my interview with Dave he explained a subwoofer array that he developed during a Blink182 tour. Here’s what it looks like.

Here’s where Dave explains it in the interview, starting at 45 minutes.

This came out of the sub testing that I did, primarily on the Blink 182 tour and then finished or got farther on the SoundGarden tour. I tried multiple arrays and out of it I came up with this fanned array.

The way to make it is, find your zero point, your rear sub location, the center of the grille. I would just have someone step on a tape measure there or put a road case on it. Then walk out 6.25ft for a 45Hz maximum rear cancellation. Then I draw an arc with the string.

Then you would place along that arc the front radiating points. One, two, three, however many you want. The 0ms point is your rear sub. Set your front ones at +6.25ft, which is about 5.8ms.

I see, so if you soloed up any of those front subs, they should be arriving at the same time as the rear subs.

Yeah, so the concept is the sound radiates from [the rear sub] and travels 6.25ft to sum with the subs in the front.

The closer these are together the beamier it gets. As you spread them out on the arc, the wider it covers.

[You have to put a sub stack in the back equal to the ones in the front. Each stack has three.] It’s making sound that will actually cancel out everywhere else and sum in the front.

The theory

Dave’s design is a combination of end-fire and physical arc. The end-fire takes care of the cancellation in the rear and the arc controls the coverage width.

End-fire array: an array of multiple subwoofers, placed in a line, one behind the other, with a specific spacing and delay strategy in a timed sequence that creates forward addition and rearward subtraction.

Bob McCarthy, Sound Systems: Design and Optimization

The array achieves summation in the front by delaying the front elements to the rear, causing everyone to arrive at the same time. It achieves cancellation in the rear through delay as well (output processing + distance offset), except in the rear everyone is arriving late causing a chaos of phase relationships.

Physical arced arrays exhibit a coverage angle equal to the segment of the arc segment of virtual circle whose origin lies behind the array.

Merlijn van Veen

A physical arced array turns a line source into a point source. Where all elements were arriving at equal level and time on-axis, they are now steered outward with a virtual origin from behind the array. As arc angle increases so does coverage width as the various spatial crossovers move away from on-axis.

The design

How did Dave come up with the 6.25ft spacing?

spacing = ¼λ

spacing = speed of sound / frequency * 0.25

6.2778 = 1130 / 45 * 0.25

Dave’s design breaks some rules, which is fun. Normally we avoid a two-element end-fire array because it is limited to a single cancellation in the rear and then comes back with a nasty peak an octave up.

It is for this reason that if you design this array in Merlijn van Veen’s Subwoofer Array Designer, it will recommend high and low-pass filters at 29 and 60Hz, respectively.

With preferred filters.

The model

Wide coverage

I haven’t tried this in the field, yet, but we can look at some models in MAPP XT.

I’ll start with the widest spacing to get the widest coverage.

Here’s a prediction at 44.2Hz showing an opening angle of 176º.

In the measurement viewer we have a nice F2B ratio of 50dB at 50Hz.

But why 50Hz? I expected to see it at 45Hz.

It turns out that I spread the subs out so far that the distance offset changed from 6.28ft to 3.77ft from the rear. This changed the combined phase of the front subs to 150º apart at 45Hz and 180º apart at 50Hz, compared to the rear sub.

I reset the placement and delay based on the outside subs instead of the center one and was able to shift the null down to 47Hz. Hurray!

Narrow coverage

Now let’s try the narrow coverage version.

Here’s the prediction at 44Hz. Looks like it’s 10º more narrow than the previous design.

Nice F2B rejection of 31dB at 44Hz.

An alternative?

Dave breaks some rules here, which is fun.

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).

Bob McCarthy

This make me wonder what an in-line gradient array would look like with the same design.

Gradient array: A cardioid configuration commonly used for subwoofer arrays with front and rear elements. The rear element is delayed and polarity reversed to effectively cancel behind the speakers.

Bob McCarthy

The gradient array has frequency dependent coupling in the front in exchange for broadband cancellation in the rear.

You can use the exact same speaker placement, with different output processing.

It looks very similar to the end-fire at 44Hz, but is quite different across its operating range.

Where we had a limited range of cancellation with the end-fire array, we now have broad-band cancellation with the gradient. You may be worried about the dip at 100Hz, but notice that that the expected response is already 20dB down, moving it into isolation if you are running a linear system.

What’s it for?

As we discussed on the podcast, Dave was looking for a way reduce LF interaction in the center of the audience and total LF energy behind the arrays.

Compare a traditional setup of left and right sub stacks…

To two of Dave’s end-fire arcs splayed 90º apart.

If you would like to play with these MAPP designs, you can download them here. Please let me know if you discover any improvements. 🙂

Have you tried something like this in the field. What were your results?

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

By Nathan Lively

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?

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