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

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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

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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?

Dave Rat’s Simple Problem Solving Mindset for a Long Fulfilling Career in Pro Audio

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In this episode of Sound Design Live I talk with the president of Rat Sound Systems, Dave Rat. We discuss artistic solutions to technical problems, the best mindset for a finding work and a long fulfilling career, and the time Flea shouted in his face for missing an entrance.

I ask:

  • What are some of the biggest mistakes you see people making who are new to FOH mixing?
  • What is the biggest mistake you’ve made on the job and how you recovered.
  • From FaceBook
    • Roshan Malim: Ask him about his sub woofers configurations and which is his favorite and why? And what is his most common sub woofer configurations for gigs? His ideas between flown subs and ground stack subs.
    • Christian Michael Martinez: As an owner of one of the largest production companies, how did you cope with the possibility of failure in the beginning as a small shop? How do you cope with risk now that the company is a success?
    • Maurizio M. D’Errico: Ask him about his technique of tuning a pa using headphones as a reference and if he still feels the same way about it since he made the YouTube video: How to EQ a Live Sound Reinforcement System.
    • Craig Gordon: Balancing touring and family/health.
    • Stephen Mink: Since he has some videos on YouTube about sub placement/patterns/alignment, his thoughts on where in a room to align flown and ground subs.
    • Dave Gammon: Does he really see the M-Force driver as the future
    • Garrick Quentin: Has he seen or heard of anyone else not affiliated with Rat sound using the double hung array as he did with the RHCPs and perhaps any comments on the whole concept in terms of its popularity, or otherwise, within the industry.
    • Davide Bonetti: does he need an assistant? 😇
    • Steve Knots: What’s the worst setup disaster he ever recovered from to pull off a successful show in the end?
    • Marty America: How’s your dog doing?
    • Micah Muth: How’s the Tesla?

The way to be successful is to sell self confidence. Confidence that when you’re there it’s going to be as good as it can possibly be.

Dave Rat

Notes

  1. All music in this episode by The Funk Lives.
  2. See my deconstruction of Dave’s custom end-fire subwoofer array.
  3. Hardware: DBX 12xds Subharmonic Synthesizer
  4. Books: Surely You’re Joking, Mr. Feynman!
  5. Quotes
    1. We didn’t borrow money, take any loans or take on any debt until 1997. So, 17 years of business.
    2. We had no idea. Well, we kind of had an idea. Everyone told us we’d fail. “There’s no way you’ll ever survive doing this.”
    3. If I’m 100% behind it and I’m going to make it happen; it’s not whether it will succeed. It’s how long it’s going to take. So if you just don’t even let failure be an option then it’s just a matter of work.
    4. Avoid silver bullets. They don’t come.
    5. Avoid being giant. You want to be happy, stable, and profitable. Being giant is precarious, stressful, and makes you a target for people to come after you.
    6. Probably one of the most common things for green engineers; the kick drum is so loud! It’s distracting from the core purpose, which is to connect the artist with the audience.
    7. If you act like you have a problem, then you have a problem. If you don’t act like you have a problem, then no one knows you have a problem.
    8. The way to be successful is to sell self confidence. Confidence that when you’re there it’s going to be as good as it can possibly be.
    9. We don’t have any problems here today. We just have a big pile of solutions we haven’t found, yet. It’s Ok, because the ones we can’t fix, can’t be fixed. But if it can be fixed, we’re gonna do it.
    10. It’s that unphasability. Combine that with, “Gimme the tools I need to do the job and everything will be fine,” and you can write your own paycheck.
    11. The sub configuration I prefer is the one that covers the venue as best as possible.
    12. The worst thing you can possible do is run your subs mono.
    13. Having a separate L/R sub aux, I find to be extremely useful.
    14. No where in nature do you have two things that are perfectly in time and phase.
    15. No where in nature do multiple unrelated sounds radiate out of a single point in space.
    16. By trying to make it sound the same everywhere, you loose the ability to make it sound great everywhere.
    17. A world-class sound engineer has the ability to recreate a quality sound consistently throughout a variety of environments and challenges.
    18. One of the ways I kept myself happy and sane was to start an external project. One tour it was blogging. It kept me busy. One tour I decided to learn to weld.
    19. By spreading (the mix) out, you can beat the crap out of the bass guitar into the fifteens of one rig, but the low end of the vocals and the low end of the guitar would stay clean. And the guitar could beat the crap out of the sixes in that one rig, but the vocals would be perfectly clean.
    20. If you desire to learn everything you can.

Should audience depth influence crossover frequency between main and sub?

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Hypothesis: By choosing a lower crossover frequency I can expand the coupling zone between main and sub.

Conclusion: While lowering the crossover frequency does expand the coupling zone between main and sub and this fact may influence the system design, its advantages are secondary to the efficient functionality and cooperation of both drivers.

Coupling zone: The summation zone where the combination of signals is additive only. Phase offset must be <120º to prevent subtraction.

Bob McCarthy, Sound Systems: Design and Optimization

While working on a recent article about crossover slopes I started thinking about main+sub alignment and its expiration. If we know that ⅔ of the phase wheel gives us summation and ⅓ of it gives us cancellation and we know the point in space where the two sources are aligned, then we should be able to predict the expiration date of the alignment, compare it to the audience plane, and consider whether lowering the area of interaction will benefit coverage.

If two sources are aligned at 100Hz and the wavelength of 100Hz is 11.3ft, then a 3.8ft distance offset will create a ⅓ λ (wavelength) phase shift (120º). If we have two sources at opposite ends of a room and they are aligned in the center, then we have a 7.6ft coupling zone. From one edge of the coupling zone to the other is ⅔ λ (240º).

80Hz has a λ of 14.13ft and would give us a coupling zone of 9.4ft, an expansion of 1.8ft.

Lowering the crossover frequency to expand the coupling zone

Here’s a section view of a main+sub alignment where you can clearly see a cancellation at 24ft. The coupling zone is 29ft, which is 65% of the audience plane.

I can lower the crossover frequency and expand the coupling zone by 4ft, which is 71% of the audience plane.

This process can be sped up using Merlijn van Veen’s Sub Align calculator. Here’s the same system design observing the relative level difference at 100Hz.

And here it is at 80Hz. Notice that the checkered pattern indicating the coupling zone has expanded.

Instead of putting every design through every potential crossover frequency, I made a new calculator that shows the percentage of audience within the coupling zone by frequency.

I am now able to quickly compare the potential benefit of selecting one crossover frequency over another by how much the coupling zone will expand or contract. Using the example from above we can see that changing the crossover frequency from 100Hz to 80Hz only provides a 7% improvement. This doesn’t seem significant enough to make a system design decision, but it could be included in other factors in the decision making processes.

Let’s look at another example. In this case the vertical distance offset is reduced and the audience depth is increased.

The calculator reveals that a 120Hz crossover would include 58% of the audience in the coupling zone, but a 75Hz crossover gives us a 13% improvement.

Should I use this calculator to pick my crossover frequency?

No. When it comes to choosing a crossover frequency there are other more important factors to consider like mechanical and electrical limitations. If your design only puts a small portion of the audience in the coupling zone, changing the crossover frequency is not going to save you.

Instead, start by observing the manufacturer’s recommendations, then the native response of each speaker, and the content of the show and its power requirements over frequency.

All that being said, knowing more about the expected performance of a sound system is powerful. I might make design changes based on the calculator’s predictions. I might I do nothing. Either wa,y I walk into the room with fewer surprises during the listening and optimization steps.

If lowering the crossover frequency increases the coupling zone, why not just always make it as low as possible?

I don’t have a great answer for this question. As I mentioned already, there are limitations to how low you can go. One major tradeoff is that your main speaker will need to handle more and more power as the crossover frequency lowers, making it less efficient.

One clear benefit I can see is estimating the viability of an overlap crossover. If you are planning a system with an overlap crossover that goes all the way up to 120Hz and you look at the calculator and see that 120Hz will only be coupling through 50% of the audience, you might decide on a unity crossover to limit the main+sub interaction into those higher frequencies, making it more stable over distance.

What about aligning at 3/4 depth?

Right! I included a phase offset option to test this and it makes a big difference. In the most recent example, if I use a ⅓ λ offset (120º), the portion of the audience in the coupling zone goes up to 88%.

6 Smart, Proven Methods To Control Feedback Onstage (Without EQ)

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sound-design-live-how-to-control-feedback-in-live-sound-evil-monkey

There is nothing worse than spending an entire event struggling with feedback demons. You may have been taught to fight feedback with a graphic EQ, but there is a better way. Actually, that’s not true: there are six better ways. Use my guide to controlling feedback onstage and mix in fear no more.

“The feedback frequency is determined by resonance frequencies in the microphone, amplifier, and loudspeaker, the acoustics of the room, the directional pick-up and emission patterns of the microphone and loudspeaker, and the distance between them.” –Wikipedia

Method #0 – Psychology

I had to include this step 0 because the more I thought about it and the more I talked to other sound engineers, the more this came up. When it comes to improving your GBF (gain before feedback), start with the beginning of your signal chain and work forwards.

Example 1: Jason works as an AV tech on city council meetings. He was having lots of feedback problems and asked for my help. After we went through everything in the signal chain and made improvements where we could, the most important change we made was simply explaining to the council members the importance of proper microphone positioning. Nothing else we did made more of an impact than getting that first step right.

Example 2: When Brian Adler works as a monitor engineer in situations where he expects the GBF to be an issue, he will purposely start with vocal mikes way too loud in the mix. This will give the performer a little shock and start the sound check off by asking their mix level to be turned down, instead of what normally happens.

Probably the biggest tip I can give in this area is to be proactive and be a pack leader. You don’t want to wait until the stage is all set up and you are halfway through the sound check before you approach the guitarist about potentially moving his amp for a less face-melting experience. Instead, while you’re giving them a hand loading in, mention that “What we normally do here is put the guitar amp on this stand so that you can hear it well and I can get a better mix out front.”

Or for vocalists: “We’ve found that the ideal position for the monitor is with this microphone in this position. If you want it to be somewhere else, I’m totally fine with that, but it might not be able to get as loud, so we’ll have to work around that.”

Method #1 – Microphone Placement

Close Miking

For loud stages and busy rooms, close miking is generally the way to go. It might not always be the best for sound, but for the maximum gain before feedback, you have to kiss the mic. Remember, with each doubling of distance, sound level is cut in half. Plus, if you’re working mostly with Shure SM58 and SM57 microphones, that’s how they are designed to be used anyway.

For corporate audio this usually means teaching your presenter how to handle the mic. For theatre this means adjusting headworn capsule placement. I have seen sound designers successfully mic a play without headworn microphones, but it’s tricky (see How To Mic An 800 Seat Theatre With Floor Mics).

Polar Pattern

sound-design-live-how-to-control-feedback-in-live-sound-polar-pattern
From SoundOnSound

For concert sound you almost never use an omnidirectional mic. Microphones with a cardioid pickup pattern have the most rejection at the rear of the mic capsule, which should be pointed at the stage monitor.

Don’t cup the mic! This will defeat the directional pattern, turning it into an omnidirectional mic.

Corporate and theatre events require specific and stable placement of the microphone capsule. Some sound engineers argue in favor of using omnidirectional capsules on the grounds that they are easier to place and produce more reliable results with the movement of the actor. My experience is that none of that matters when the audience can’t hear the actor because you can’t get enough gain.

I’ve done a lot of musicals and concerts with omnidirectional head-worn microphones in the past, though, and it’s always a struggle. The performers can’t hear themselves, and if the audience starts clapping or singing along, chaos ensues. Why did I do this? Because it was what I had available. These days I try to let directors and event producers know way ahead of time about the limits of working with certain equipment. If possible, I’ll schedule a test so they can hear the difference in the performance space.

Method #2 – Speaker Placement

Stage Monitors

Floor wedges should be placed on-axis and as close to the performer’s head as possible. I’ve heard people suggest moving the monitor away from the performer for better gain before feedback, but don’t do that. That just creates lower sound levels at their ear level, so you’ll have to turn it up louder. Most live stages are loud enough as it is, so anything you can do to lower the stage monitor level will be helpful.

how-to-control-feedback-in-live-sound-hotspot

Have you ever seen those little Hotspot monitors? I haven’t seen them in a few years, but I love the idea. Put a small monitor on a stand and you significantly reduce its distance to the performer.

Sometimes, because of sightline issues or stage layout, you can’t get a monitor right in front of a performer where a cardioid microphone’s off-axis point is. This happens often with drummers and keyboard players whose instruments take up so much space and lead vocalists who want clear sightlines. This is when you need a hyper-cardioid or super-cardioid microphone and this is why many live music venues have a collection of Shure SM58 (cardioid) and Beta SM58A (supercardioid) microphones, or similar.

If you find yourself stuck with a drummer or piano player whose stage monitor is at a 90° angle to a cardioid microphone, try cheating the microphone out closer to 45° to get more rejection. If an artist requests a monitor position that is less than ideal for your microphone selection, go ahead and do it, but warn them that you may run into feedback problems and need to reconfigure the speaker and mic.

I’ve seen some pretty creative microphone and monitor placement that allow for very high gain before feedback. If you are working with acoustic instruments, ask the performers if they have any tips for placement. I used to work with a cello player in Portugal who placed the stage monitor a little behind himself so that it wasn’t pointed at his microphone but it was still aimed at his head. It worked great.

Stage monitor placement for theatre deserves its own article, but my number one tip is to start the conversation early. Explain your limitations to the production team and discuss ways to best accommodate the actors. You don’t want to realize in tech rehearsals that the actors can’t hear the musicians and that the director won’t allow downstage speakers. I often lobby for small downstage monitors straight out of the gate. I also try to make friends with the set director and builder as quickly as possible, alerting them to the fact that I’ll probably need help hiding speakers around the stage.

FOH

Make sure your FOH speakers are covering the house and not the stage. This means checking the speakers’ off-axis angles to make sure they are not spilling onto the stage or creating strong wall reflections. (See also: How To Tune A Sound System In 15 Minutes.) I’ve heard people say that all microphones must be at least six feet behind FOH, but I’ve seen it done many different ways. Some situations call for more separation and control, others less.

Method #3 – Instrument/Source Placement

sound-design-live-how-to-control-feedback-in-live-sound-drum-shield

If you are working with a loud rock band and you place the lead vocalist right in front of the drummer, guess what happens? Your vocal mic will be full of drums and your vocalist won’t be able to hear. This happens all the time, and explains why you see the bands on Saturday Night Live using a drum shield on that very small stage.

Your goal is to balance every source input for the performers and audience. Now let’s talk about the most frequent offenders.

Drums

Drums are loud. Some drummers are interested in harmony and balance, and will change their technique, use brushes, and dampen their instruments. Those drummers are in the minority. Why? Well, have you ever played drums? It’s fun as hell to play loud, and boring as shit to play soft, or so goes my personal experience.

If you’re on tour, you’ll need a rug and a drum shield. If you’re full-time at a venue, put absorption everywhere. Two of the noisiest venues I’ve worked at have pulled the same trick and covered their ceiling and walls with black semi-rigid duct insulation or vinyl that screws right into the wall. It made a big difference.

For more on this topic, see 5 Pro Drummers Explain How to Make a Drum Kit Quieter on Stage.

Electric Guitars

I’m a guitarist, and as such I’m fully aware of how hard it is to hear myself without the amplifier blaring. The only way I was able to handle this in my band was to learn to play without hearing. In the real world, getting a guitarist’s amp as close to their head as possible will help. Put it on a chair or milk crate. Most are open-back, so put a bunch of absorption back there.

In my interview with Larry Crane he mentions a guitarist who built a Plexiglass shield for his amp that redirected the sound upward at an angle so that he could play with feedback and do fancy things with his amp without blasting the stage. Pretty smart.

sound-design-live-how-to-control-feedback-in-live-sound-guitar-amp-doghouse

I worked on a show last year where the guitarist made a shield for his amp from case lids and jackets. This helped it not bleed into other microphones as much.

Buford Jones is famous for doing whole tours mixing from inside a truck outside of the venue. (He’s even more famous for mixing some band called Pink Floyd.) These were large venues where they had little acoustic sound coming from the stage. The guitar amps where all in dog houses off-stage and all of the performers were on IEMs (in ear monitors). Most of us won’t experience that, but it gives you an idea of how far people will go to control sound levels on stage. If you are worried about approaching a guitarist to discuss changing their setup, just remember that asking them to turn down their amp and put it on a stand is nothing compared to removing it from the stage entirely.

Method #4 – Mix

Stage Monitor

Most performers these days are wise to the challenges of microphone feedback on stage and will make specific requests for their monitor mix. I’ve made it a practice to not add anything to a stage monitor mix until expressly asked to, except for vocalists who almost always need reinforcement. When musicians walk in the door saying, “Just give me a mix of everything,” they likely don’t know what they need. Smile and nod.

I’ve made it through entire shows without adding anything to some performers’ stage monitors because the stage layout allowed them to hear everyone. I’ve also worked on shows where the band has skipped sound check then walked on stage expecting a complete mix. I try not to work off of assumptions and I give people only what they need, because the lower your stage volume, the better your FOH mix will be, and everyone will be happier.

FOH

In small to medium venues, you aren’t “mixing” in the classical sense, you are doing sound reinforcement. You are balancing the acoustic energy in the room for a more pleasant musical experience. From my interview with Howie Gordon:

The other thing I hear a lot about [is] guys setting the whole mix base from the drums, and in my opinion that’s the last thing you should do because the thing that immediately suffers is vocals. It’s the one instrument that can’t control its own stage volume. -Howie Gordon

And from my interview with Larry Crane:

How many times have you been blown out of the water by the mains because you’re trying to keep up with the stage? It’s like, “No, no, no! That’s not necessary.” You’re not building the mix up from the kick drum at that point. You’re building the mix down from what’s happening on the stage, and you’re filling in what’s missing, just a little bit. -Larry Crane

If you need definition on the bass guitar, roll off the low end and mix it in. If you are missing the melody from the keyboard, bring up the right hand. If the guitarist is too loud then invert the polarity and lower his volume in the house with deconstructive interference. That’s how noise cancelling headphones work.

(Just kidding! You know I’m kidding, right? If you actually try that and it works, keep it to yourself.)

Compression

Normally, I love compressors, but they raise the noise floor and reduce dynamic range, and therefore reduce gain before feedback. I would really like to use compression on lapel mics during corporate presentations, for example, but I’m often on the verge of feedback and can’t spare the gain.

Method #5 – The Holy Grail

IEMs, e-drums, synths. Done! 😉

Method #6 – Don’t Give A Fuck

“These setups that we’re working on, there’s EQs everywhere. If there’s still feedback, it’s too loud. So lower it or let it ring all night. I don’t give a fuck.” —Dimitris Sotiropoulos

I laugh every time I read this quote, but there is plenty of truth to it. Half of what I write on Sound Design Live is about psychology. People don’t trust sound because they can’t see it. That also means they don’t trust you because they can’t see what you’re doing. Letting the monitor feedback for a second before you bring it down communicates to the artist that it has reached it’s maximum level and that you are turning it down.

Q: But you do use EQ, right?

A: Um, yeah, most of time. At least to attenuate some low end.

#ObligatoryBonus – EQ

This is your last tool in the war on feedback. Use high-pass filters to remove the rumble from guitars and the proximity effect from vocals. Use narrow-band filters on a parametric EQ to surgically remove problem frequencies. Although it’s your last step, it’s also necessary. Temperature, humidity, and performance changes throughout the night will require compensation.

I recently worked with a sound engineer who would cut the low end from all of his vocals up to 200Hz in the stage mix. That’s a lot! But it worked. A few years ago I worked on an outdoor event where everything would be balanced during the afternoon sound check, then explode into feedback at night because of environmental changes.

So I think we can agree that some amount of EQ is necessary, but watch out for assuming too much. There is a process that we sound engineers call “ringing out the monitors” that often takes place before any artists have arrived. We use this process to lessen the amount of time we will need to chase feedback during sound check. I gotta tell you that over years of working on live events I do this less and less. Why? Because if you do it before sound check then you are making a lot of assumptions about the sound that can all be ruined by changing a mic or its placement. You’re also making changes to the speakers’ performance and sound quality without due cause. A better technique is to test for feedback, make note of those frequencies, but hold out on makes changes until you need more gain.

I sort of hate the fact that “ringing out” is supposed to be a normal part of our job. Under normal circumstances, with high-quality equipment and a properly optimized system, you shouldn’t have to do this. The fact that it is a normal part of our job makes me realize that there are a lot of sound systems out there that need your TLC.

Pulling half the bands down on a graphic EQ is like removing a tumor with a wiffle ball bat.

sound-design-live-how-to-control-feedback-in-live-sound-graphic-EQ

If this is the first article you’ve ever read from me, you may wonder what I have against graphic EQs. For system EQ, their fixed frequency, bandwidth, and logarithmic spacing make them unhelpful. They maintain popularity until today because they seem to give you a visual (graphical) representation of the changes you are making. Unfortunately, the visual is misleading. While you appear to be making surgical incisions, you are really making ⅓ octave tonal changes. You can prove this to yourself by measuring one. Here’s a step-by-step guide.

I hate graphic EQs. I don’t use them unless I don’t have a better choice. You’re talking about ⅓ of an octave. That’s like a C to an F on a piano.

Michael Lawrence – Fighting Microphone Feedback WITHOUT a Graphic EQ While Mixing Monitors from FOH in a Reverberant Room

Basically, the only things that graphic EQs are good for are ear training and maybe use in the battlefield that is Monitorland. For more, see my interview with Dave Swallow, my interview with Bob McCarthy, and my review of McCarthy’s book.

Another consideration is where you will insert these EQ filters. Your first idea might be to insert them on the master output buss of the mixing console. Consider that this has global repercussions on the entire mix. You are affecting the system response and mix balance. If possible, scale your changes back to the smallest local change possible. Is the feedback originating from a single microphone to all outputs? Insert your EQ on that microphone’s input channel first. Is the feedback frequency present to varying degrees in all vocal microphones? Insert the EQ on the vocal buss.

In the world of my dreams, I would be able to insert filters on a per-send basis from each input channel for maximum transparency. Unfortunately, the only way I know to accomplish that on modern mixing consoles is to create a duplicate input channel for each send, which is overly complicated.

Ambient Changes

Humidity and temperature changes throughout the night will require compensation, especially if you are outside. My first big lesson in this came will working for the band O’QueStrada in Portugal at an outdoor concert at the Centro Cultural de Belém. I had all of my monitor mixes set just on the edge of feedback, which seemed fine during soundcheck. We came back that night to start the show and as soon as I unmuted the band I also unleashed a storm of microphone feedback. 

At the time I didn’t understand that a rise in relative humidity at that location would result in less high-frequency air absorption. I could have compensated for the change in humidity with a high-shelf filter.

The lesson: Don’t mix your stage monitors to the edge of feedback if you expect a rise in relative humidity and be prepared to compensate with a high shelf filter.

Temperature changes are less obvious. It would take a a 20ºF change in temperature to produce a 2% change in the speed of sound, which may be only enough to shift your acoustic crossover point by one seat. Unless you are working outside with some very large changes in temperature, I wouldn’t worry about its affect on microphone feedback.

Other Tricks To Try

Feedback Eliminator

If you look up reviews for feedback eliminators they are almost equally bad and good. You never see them on professional productions. Part of the issue is that sound engineers don’t like things to be out of their control, but the main problem is that these units just don’t work that well. Everyone who has used them has horror stories.

That being said, sometimes pro audio feels like a war zone, and I will never judge you for using one. Especially for corporate events where you have several lapel mics walking around a stage and you only need to stop one frequency from feeding back on one microphone for three seconds at a time. Or small setups where you are very limited in the way of EQ.

Frequency Training

Imagine the show-stopping seconds you could save if you could identify feedback frequencies immediately without using an analyzer. There are some nice apps out there that will train you to identify frequencies. This is not the same skill as having perfect pitch. It’s pitch memory and anyone can learn it. Most of them train you using the 32 bands of a standard graphic EQ, which isn’t ideal, but is a great place to start, helping you avoid a frequency-wide sweep. I’ve used Audio Frequency Trainer and Quiztones. Read more about my experience here: My Results from 30 Days of Ear Training.

Microphone Splitter

sound-design-live-how-to-control-feedback-in-live-sound-y-cable

Don’t have a digital mixer or a separate mix console for the stage? Try splitting a few channels for more control. Let’s use the lead vocal microphone as an example. Right before it comes into the mixing board, connect a splitter or use a Y cable for the most basic passive version. This will give you two copies of the lead vocal coming into the mixing board. Mix one for the house, mix one for the stage. This will allow you to roll off way more low end then you normally would and make other adjustments to the stage mix without significantly affecting the house.

Polarity and Delay

It has been suggested to me that you can invert polarity or add small amounts of channel delay to get more gain before feedback. I’ve never had success with this. It just moves the feedback to a different frequency, makes it attack slower, or makes no change at all.

Separate Speakers

From the Meyer Sound Design Reference by Bob McCarthy ©1998:

One solution is to double the number of stage monitors and separate them into music and vocal systems. This has the advantage allowing for separate EQ and, in additon, the musicians find it easier to localize their voice and their instruments since they come from different positions.

Thanks to ra byn for tipping me off to that one.

Turn Down

While on tour with Ringling Bros., I found that in some arenas I had GBF for days and in others I could barely get the main vocal up above the band. Our system and performers being the same, I had to accept the fact that my headroom changed from week to week. The audience didn’t know it was different, so as long as the balance was good I could adjust the overall level as necessary.

Conclusion

Your best tool for controlling feedback in live sound is stage layout. That means microphone placement, speaker placement, and instrument/source placement. Then you can work on the mix and if you still can’t get enough gain before feedback, use EQ. If you’re lucky, you’ll work with a synth-pop band (call me, Active Child!). If you’re unlucky, challenges abound, everyone’s a dick, and you just let it ring all night, cause fuck it.

What are your best tips for fighting feedback on stage? Comment below!