Sound Design Live

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

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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

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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.

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%.

Do all-pass and FIR filters cause delay?

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For a long time I was afraid of all-pass and FIR filters because they seemed exotic and supposedly cause lots of delay, making them unusable for live sound. Turns out this was just an excuse I was using to avoid some mental hurdles.

Do all-pass filters cause delay?

Here’s a measurement of my BLU-160. It’s an output processor from BSS.

Pretty boring.

Here’s that same measurement with 5ms of delay inserted. Let me draw your attention to the Live IR. It’s the exact same shape as in the previous measurement, just pushed 5ms down the time axis.

Let’s take out the delay and insert a second-order 180º APF (all-pass filter) at 100Hz.

Cool.

Wait a second. What’s going on with the Live IR?

Isn’t an APF a frequency-specific delay?

If the half-period of 100Hz is 5ms, shouldn’t we see a 5ms delay in Live IR?

Maybe the Live IR is over represented by high frequency content. Let’s start over and switch to using a using band-limited pink noise (50-200Hz) instead for the signal generator.

I moved the Live IR window over a bit since the peak shifted when I switched to the band-limited pink noise, but I didn’t adjust the delay finder.

Now I’ll insert the APF again.

I see phase shift, but I don’t see delay.

Maybe we just can’t see with enough resolution. Let’s record the wavelet and look at it as an IR (impulse response).

Ah, ha! Now we see some delay. But is it 5ms of delay?

Rats. It’s only 0.042ms.

I have one more idea. What if I record them and look at them in a wave editor.

Here are the waveforms superimposed. It looks like some delay, or maybe a polarity inversion?

But if I invert the polarity…

Instead of the same IR pushed 5ms down the time axis we see…something different; a new waveform.

This is an important distinction. Delay causes delay. It returns a copy of the original, just farther down the time axis. It does not alter the wave shape and there is no frequency dependence.

On the other hand, an APF does not cause delay. Instead, it causes phase shift, which is frequency dependent and returns a new waveform. Phase shift causes the waveform to rotate around the time axis, which can make it difficult to distinguish from delay.

It’s important that we do, though, because if I sent you this new waveform and you wanted to reverse the process you wouldn’t use delay. You would use a complimentary APF.

Problem solving

If we observe summation that looks like this:

We’ll want to fix it with delay so that it looks like this:

If we observe summation that looks like this:

We’ll fix it with a matching APF in the other channel:

Do FIR filters cause delay?

FIR (finite impulse response) filters do not cause delay for the same reasons that APF do not cause delay. Their implementation, though, may result in latency any time they include excess phase or linear phase.

  • Excess phase is any additional phase beyond minimum phase.
  • Minimum phase defines the predictable relationship between magnitude and phase.
  • Linear phase breaks all the rules and removes any relationship between magnitude and phase. 😨

This leads to a very simple rule for a minimum phase network: at a local maximum or minimum in the transfer function, a frequency of maximum curvature of amplitude corresponds to a point of inflection of phase.

Richard Heyser

As long as your FIR filter only includes minimum phase filters, there will be no processing delay. Where there is a change in magnitude, there were will be a corresponding and predictable change in phase. Where there is a change in phase there will be a predictable and corresponding change in magnitude.

Here’s a measurement of a JBL PRX615M.

Here’s the same speaker, but with an FIR filter inserted (800 samples!). Notice the Live IR. The FIR filter is entirely minimum phase so there is no processing delay.

If I were to introduce linear phase filters and make changes to the phase and magnitude independently a filter delay would be necessary and we’d some amount of processor latency.

Why should I care?

To know the right tool for the job, you have to know what it’s called. Imagine a surgeon calling out for the thingy-thing.

  • If you have a time alignment problem, fix it with delay.
  • If you have a phase alignment problem, fix it with an APF.
  • If you want to adjust magnitude with phase, use a minimum phase FIR filter and incur no processing delay.
  • If you need to adjust magnitude and phase independently, use a linear phase FIR filter and incur predictable processing delay.

Trivia: Is Ø the symbol for phase or polarity? Comment below.

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

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In this episode of Sound Design Live I talk with Michael Lawrence who is Document Jockey at Rational Acoustics, Technical Editor at Live Sound International, and a freelance audio engineer. We discuss lots of tips for fighting microphone feedback without a graphic EQ (and without Smaart!) while mixing stage monitors from FOH in a reverberant room.

I ask:

  • How did you get your first job in audio?
  • 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?
  • Running monitors from FOH? Here’s some tips.
    • Like me and many other people, you often work by yourself. You are the system designer, system tech, A1, etc. You’ve developed a lot of processes to efficiently get it all done. Can we go over some of your best tips for running stage monitors from FOH?
    • Why don’t you use Smaart with your stage monitors?
    • Walk me through your process for ring out the monitors.
  • And from Facebook
    • Manuel Elias Costa: What does he know about automix and machine learning algorithms research? It would be interesting to hear more about the developments in mixing and automation.
    • Andrey Andreev: What does he think of the way music industry is developing production-wise and is audio quality still a thing? How are decision been made for speccing a certain sound system? Why is point source so much neglected when it can be so many times better sounding than the usual line array type of systems? (Of course line array has its place but it’s hardly always the right solution)
    • Dave Gammon: Does he see sound being more immersive in the future. Less about right and left but more about a total experience and encapsulating the audience.
    • Garrick Quentin: With the new advancements in line source technology, where do old line arrays go to die? What’s the next game changer in line array technology that we don’t yet know about?
    • Lou Kohley: How do you stay relevant to opposite edges of your market? The novice just starting at a bar gig to working professional to industry veteran.

I hate graphic EQs. I don’t use them unless I don’t have a better choice. You’re talking about 1/3 of an octave. That’s like a C to an F on a piano.

Michael Lawrence

Notes

  1. All music in this podcast by Bionik.
  2. Running Monitors from FOH? Here are some tips.
    1. Verify all outputs with pink noise. If they are all the same model, they should all sound the same. Check settings (line/mic switch, gain).
    2. Practice identifying feedback frequencies. Sing it to yourself.
  3. Hardware: X32, Midas Pro1, LS9
  4. My Results from 30 Days of Ear Training, Download the Aiming Triangles Business Card
  5. Quotes
    1. I treat everything like I’m on tour even when I’m not on tour; doing the same things in the same order every time.
    2. Before you do any test, have an idea of what you expect it to look like.
    3. I always have a cue wedge at FOH. If you don’t have one, you’re guessing.
    4. I hate graphic EQs. I don’t use them unless I don’t have a better choice. You’re talking about 1/3 of an octave. That’s like a C to an F on a piano.
    5. We ignore the polar pattern of the mic. That’s super important. Buy yourself every dB you can get.
    6. I always double patch my money channel.