Sound Design Live

Build Your Career As A Sound Engineer

How Using Less EQ Can Stop Your Show from Sounding Horrible

By

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 Sound Designer for Broken Chord and Project Design Manager for Sound Associates, Phillip Peglow. We discuss the night broadway was shutdown by COVID-19, whether or not you should go to graduate school, how EQ is ruining your show and what to do about it, why you’ll never beat the room, and why you should give the producer whatever they want.

I ask:

  • What are your concerns about work because of the pandemic?
  • When you get a new system installed and calibrated in a theatre and you want to give it a test drive, what music do you listen to?
  • How did you get your first job in audio?
  • 8 years ago I interview your partners in Broken Chord, Aaron Meicht and Daniel Baker, after I saw their sound design for the pulitzer prize winning play “Ruined”. So how did you meet those guys and what do you like about working in a team instead of working solo?
  • How did you get the job at Sound Associates?
  • Looking back on your career so far, what’s one of the best decisions you made to get more of the work that you really love?
  • What are some of the biggest mistakes you see people making who are new to sound design for theatre?
  • Tell us about the biggest or maybe most painful mistake you’ve made on the job and how you recovered.
  • What’s in your work bag?

You can’t beat the room. Trying to punch your way through with EQ or level is a fool’s errand.

Phillip Peglow

Notes

  1. All music in this episode by HouseFrau and RRound.
  3. Workbag: headlamp,
  4. Books: Sound System, Yamaha Sound Reinforcement Handbook
  5. Quotes
    1. Unless you have that opportunity [graduate school] at zero to you, I wouldn’t do it.
    2. If you really really really really really really really want to be on Broadway then you must move to NYC. It’s not an option.
    3. Use your ears first, before you put pink noise through anything. Start there.
    4. “If you’re making anything more than a 6dB cut, it’s probably time to reevaluate your decisions.” -Jamie Anderson
    5. Don’t ever use a GEQ.
    6. If you’re trying to make narrow narrow cuts, you are probably trying to optimize for a specific point in the room that has not bearing on 3-4 inches away from that position.
    7. You can’t beat the room.
    8. Trying to punch your way through with EQ or level is a fool’s errand.
    9. If I want into a theatre style setup and I have 5 minutes to get it going, I’m going to delay the system before I do anything else.
    10. When the people who sign your checks say, “This is what I want,” then just do what they want. It’s as much a psychological issue as it is an audio issue.

Can you estimate line array splay in the field without software while the riggers are waiting?

By

I have developed, what seems to be, a lesser known method to find target coverage angle and quickly estimate average splay for a line array in the field in relatively few steps. I discovered it by necessity while creating Pro Audio Workshop: Seeing Sound 3 years ago. Recently a student challenged me on a couple of points and it motivated me to take a closer look to see if I could make it more efficient.

Here’s how I have seen other people do it.

Bottom speaker down angle – Top speaker down angle = Target coverage angle

bottom angle
Bottom speaker angle
top angle
Top speaker angle

17º – 6.78º = 10.22º target coverage angle

Target coverage angle
array splay
Result using auto-splay in MAPP

This works fine when you are using modeling software, but I was looking for a solution for the field with a laser disto and a calculator while I have a team of people waiting on me. After playing around with some right triangles for a bit, I discovered a pretty simple method

In short, if you know the array’s rigging height and where the audience starts and ends, you can find the target coverage angle without software.

Find target coverage angle without software

Here are the steps:

  1. Solve triangle Y. You need the length of two sides or one side and one angle. I would go with two sides since that seems to be more reliable.
  2. Solve triangle Z. You can find the length of the opposite side (6.07′) by subtracting the array height from the from the rigging height. You can estimate the array height by multiplying the number of boxes by a single box height.
triangle1

Then plug those numbers into a triangle solver.

triangle2

16.88º – 7.03º = 9.85º

What about inclined audiences?

But that only works for flat audience planes. What if the audience is at an angle?

inclined audience

The process is a similar. To solve triangle Y, we’ll subtract the the height of the end of the audience plane from the rigging height above the audience.

rectangle2

14.8 – 6 = 8.8ft

Solve for the missing angle. 4.19º

We already have the solution for triangle Z (16.88º).

16.8 – 4.19 = 12.61º target coverage angle

inclined
array splay inclined
Result in MAPP using auto-splay

Now what?

With one more step we can calculate average splay.

tar cov ang / available splay angles = average splay

12.61º / 11 = 1.2º

total splay

My speakers don’t offer a 1.2º splay, so I’ll round down to 1º and make up for the loss with a few of the last speakers. Now I have plan to hand the riggers.

angles 1

What is the result using average splay?

avg splay prediction

It’s not great, but in a pinch I’d rather go with this result rather than leave everything at 0º or just guessing.

0deg splay

The easiest way to improve this result is to use the the automatic solvers built into your modeling software. The best way to refine the result manually for even more control is covered in detail in Pro Audio Workshop: Seeing Sound.

Warning: Software should always be used to double check rigging points and weight distribution. (Thanks Samantha Potter!)

Have you tried calculating line array splay in the field without software? How did you do it? What were your results?

1 Graph Setting You Need to Change in Smaart for Faster EQ Decisions

By

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 founder of SIA Acoustics and SIA Software and the originator of Smaart©, Sam Berkow. We discuss acoustics, sound system design, and audio analyzer pet peeves.

I ask:

  • How did you get your first job in audio?
  • What’s one of the best decisions you made to get more of the work that you really love?
  • You have managed to build a business that successfully marries acoustic consultancy and system design and integration. It seems like these two jobs would always go hand in hand, but they don’t. Is that because sound system design is a much younger field? Could you talk about what separates and joins the two?
  • What are some of the biggest mistakes you see people making who are new to audio analyzers?
  • Is it cheaper to make a room quieter or make the sound system louder?
  • Tell us about the biggest or maybe most painful mistake you’ve made on the job and how you recovered.
  • From FB
    • Kip Conner: What happened to the Tacoma Dome case study? Can it be reposted?
    • Jason Kleiman: Does he have any advice and/or opinions on using FIR filters in system design and optimization. What is an example use case?
    • Cuauhtémoc Méndez: What are his thoughts on “immersive installations” and their future. Will it last?
    • Aleš Dravinec: Ask him how Kayden is doing.
  • What’s in your work bag?

It’s always cheaper to design it right the first time.

Sam Berkow

Notes

  1. All music in this episode by Wowa.
  2. System toning songs: Ali Farka Toure & Toumani Diabate – Debe, Diane Reeves – One for My Baby, Galactic – Black-Eyed Pea
  3. David Byrne’s American Utopia on Broadway
  4. The Band’s Visit on Broadway with Engineer: Kai Harada 
  5. Workbag: Earthworks, B&K, Studio6 Digital, Hilti laser measure, rubber mallet to bang on walls, a strategy for approaching projects
  6. Podcasts: Live From Here, Wait Wait Don’t Tell Me, TWIT “This Week In Technology”
  7. Books: Love is a Dog From Hell, Sound System Engineering, Sound Systems: Design & Optimization
  8. Quotes
    1. You don’t need to know why gasoline burns to drive a car, but it helps is you understand the fundamentals of how cars work and how they respond.
    2. Noise Criterion a series of curves where you make octave band measurements and what curve you stay under you use as your number.
    3. I think making rooms quiet makes them sound better. But if the show is 100dB then it doesn’t matter.
    4. It’s always cheaper to design it right in the first place.
    5. I’m a big believer in delaying the main system to the backline of rockbands.
    6. People are working to make the audio experience at concert venues like a movie experience.
    7. Because the transfer function inherently at mid and low frequencies is looking at the interaction of the room and the system and at high frequencies is looking at just the system I was hoping that as a tool, Smaart would bring those two things [acoustics and sound system design] together.
    8. The idea of low frequency decay being in some reasonable balance with high frequency decay in a room is critically important and a very important design tool and something that’s easy to measure in Smaart.
    9. My biggest pet peeve is people looking at the screen and not listening.
    10. If you have 80Hz as your crossover point, but your subwoofer is 6-8dB above the full range device, your acoustic crossover will be much lower than if you turn the subs up 10-12dB more. You’ve kept the electronic crossover, but slide the acoustic crossover up by changing the gain. I think you create a lot of mud in those cases, by having the subs go so much higher. I like to add EQ outside the bandpass on the subwoofer to make steeper crossovers and reduce the interaction in those areas.
    11. Complex FIR filters that address low frequencies introduce a lot of delay.
    12. I’m a big believer in delaying the main system to the back line of rock bands. So much sound is coming off of the stage that 7 or 8ms really makes a big difference for the front of the audience. The people up front stop hearing two snare drums.
    13. If you’re going to go out and optimize a system, you should have a step-by-step process in your head.

Can you match speakers through space and time? (🍎to🍎)

By

What if I want to compare my speaker to your speaker to see if they are compatible? Is there any way to actually do an apples to apples comparison?

Hypothesis: Yes!

Let’s look at an example.

Is a PRX615M compatible with a PRX618S? I don’t have either. How will I find out?

My first idea is to check the loudspeaker manufacturer’s website to see if they have anything that could help. Maybe they have their own modeling software or GLL files I can compare.

Nope. No software and the EASE data offered is only for standard EASE ($2,330), not EASE Focus ($0). Time to start making some calls.

Good news. I found both, but they are in different locations. My friend Amy in Austin has a PRX615M and my friend Burt in Berkeley has a PRX618S. I ask them to take measurements and send them to me.

So far, they don’t seem compatible. The magnitude and phase don’t match.

Magnitude

I thought about asking them to match their preamp settings, but Amy has an OctaCapture and Burt has a UMC404, which has a totally smooth preamp knob. We could probably figure out a way to match their preamps with a meter, but that would still leave the measurement mic out of the equation. Luckily, I remember that Amy and Burt both have microphone calibrators.

I have them do two things:

  1. Calibrate the MIC channel with the calibrator set to 110dB and the microphone preamp set to -12dBFS on the audio analyzer’s input meters.
  2. Adjust the REF loop so that when the signal generator plays a 1kHz sine wave at -12dB, the audio analyzer input meter also reads -12dBFS.
matching levels
level offset

The magnitude data looks like a closer match now. Things are looking up.

Phase

Let’s figure out what’s going on with these phase traces. Are they really that far off, or is there something I’m missing.

I ask Amy and Burt to send me photos of their measurement setups and realized that their measurement mics are at different distances from the speakers. Amy’s measurement is at 0.83m and Burt’s is at 1.6m.

I could ask them to redo the measurements, but maybe I can fix the timing offset manually. If the sub (PRX618S) was measured 0.77m closer than sub (PRX615M), then its measurement needs to be delayed by 0.77m to be in synch. I’ll use the Phase Invaders to add 0.77ms of pre-delay to the sub.

first try

Closer, but they still don’t line up. That’s when I realize that there’s an important feature of the 2-channel analyzer that could also be out of synch. I compare the delay locator setting used with each measurement. They don’t match.

The delay locator for the main measurement was set at 3.94ms and Speaker B was set at…0ms. 0ms? I asked Burt why he didn’t set the delay locator and he said that he tried it a few times and it didn’t work.

Right! If forgot that the delay locator often doesn’t work on LF drivers. Maybe we can figure it out.

I had already added 2.24ms (0.77m) of pre-delay to the Sub because of the distance offset. If I add 3.94 more, that will make 6.18ms.

How do you add pre-delay in the field? It doesn’t exist, but there is a workaround. Try adding 10ms of delay to all of your outputs before the alignment process. Then you’ll be able to add or subtract delay from any channel. Remove any excess delay at the end of tuning.

second try

They are still not aligned. Further investigation reveals that 3.52ms more pre-delay and a polarity inversion or 8.56ms more pre-delay and no polarity inversion give me two workable alignments. If I needed to set these up fast in the field, I would start with those two presets and modify them using their final distance offset.

aligned

Note: The proposed procedure does not test max SPL, but does compare relative sensitivity. For max SPL testing, see m-noise.

Have you tried comparing measurements through space and time? What were your results?

Questions

Will there be enough headroom?

Maybe not to measure max SPL, but we’re just comparing sensitivity. Once the inputs are calibrated you can adjust the signal generator level all day and the measurement will stay the same.

Is it better to face your subs at the wall?

By

I worked on a show last month where I decided to face the subwoofers at the wall. People were looking at me funny and asking questions…and that was before they even saw the subs. 🥁

So I made a video to investigate the effects of path length differences in this binary scenario. One result is not necessarily better than another, but it’s good to make an informed choice.

You can use this formula to estimate the frequency of the first dip of the comb filter.

c / d / 2 = 1st dip

speed of sound / distance (aka total path length difference) / 2 = first dip of comb filter

Subs facing audience
1130ft/s / 13ft / 2 = 43.5Hz
345m/s / 3.96m / 2 = 43.5Hz

Subs facing wall
1130ft/s / 6ft / 2 = 94.2Hz
345m/s / 1.83m / 2 = 94.2Hz

Pick your poison.

sub wall comparison

Taking a look at the field measurements, it looks like the subs were probably closer to 5ft from the wall because of the dip at 112.8hz, indicating a 10ft round trip distance for the reflection.

reflection comb filter

One side benefit that I forgot to include in the video is that the QSC subs were polarity inverted compared to the Kara mains. Facing them away from the audience meant that I didn’t need the polarity inversion. Tiny victory. 👏

Check out Merlijn’s Low-frequency Acoustic Center Calculator.

Have you tried facing your subs at the wall? What were your results?