Sound Design Live

Build Your Career As A Sound Engineer

3 EQ Snapshots That Will Make Your Corporate Event Mics More Transparent Using Smaart®: SM58, 185, MX412

By

The most common microphones I use on corporate events are Shure wireless SM58, WL185 Lavalier, and wired MX418. I created three EQ snapshots using a mic compare measurement in Smaart for a more transparent starting point.

What is a mic compare measurement?

In the same way that we use the transfer function measurement in Smaart to observe changes to our mix as it passes through speakers and the air, we can also observe changes to the source as it passes through microphones.

To do this, first position the mics so that their capsules are as close as possible to each other. It’s often easiest to place them on-axis with each other with the source at 90º.

Connect the monitor output of your mix console to an input of your audio interface. Create a new transfer function measurement pair using the console’s monitor output as the measurement signal and your measurement mic as the reference signal.

Start the measurement and signal generator in Smaart.

On the mix console, hit solo on the mic channel that you want to measure (sending it to the monitor buss), flatten the EQ, and turn up the monitor output until the measurement trace in Smaart centers around 0dB. For better data on global trends, create an average from several mics like I did.

Creating the snapshot

Adjust the channel EQ to your satisfaction.

Keep in mind that many (most?) microphones include purpose built non-linearities like helpful EQ enhancements. Think every kick mic you’ve ever used.

Here’s an RE320 I measured. I chose to make no EQ changes because I listened to it in headphones and the room and it sounded great.

Shure SM58

Here’s the pre and post EQ measurement for the SM58.

Here’s the manufacturer’s specification.

Here’s the EQ I came up for a more transparent response.

And here’s the EQ I settled on after listening on a show. You can download all of the snapshots here.

Shure 185

Here’s the pre and post EQ measurement for the 185 capsule.

I couldn’t find the manufacturer’s specification. If you have it, let me know.

Here’s the EQ I came up for a more transparent response.

And here’s the EQ I settled on after listing on a show. You can download all of the snapshots for X/M32 here.

Shure MX412

Here’s the pre and post EQ measurement for the MX412.

I couldn’t find the manufacturer’s specification. If you have it, let me know.

Here’s the EQ I came up for a more transparent response.

And here’s the EQ I settled on after listing on a show. You can download all of the snapshots here.

Have you tried a mic compare measurement? What were your results?

Smaart® and the Smaart logo are registered trademarks of Rational Acoustics LLC and are not affiliated with Nathan Lively or Sound Design Live.

Invasion of the Phase Invaders: An Audio Engineer’s Guide to Battle Tactics and Big Scores

By

If you lay awake at night thinking about improving your crossover alignments and winning big at Phase Invaders, then this guide is for you.

What follows are three proven battle tactics from beginner to advanced.

#1 – Beginner

Press all the buttons until you win.

a monkey hitting keys at random on a typewriter keyboard for an infinite amount of time will almost surely type any given text, such as the complete works of William Shakespeare.

Wikipedia

I have to thank my wife for this one. She is often my first beta tester and while not a professional audio engineer, she is driven by a competitive desire to win unlike most people I’ve met. While making random combinations of the delay slider and polarity switch, she keeps track of her score and returns to the combination that returned the highest score.

#2 – Intermediate

Make the pictures match.

The battle tactic I think most audio engineers will start with is the visual one. Naturally, we are all visual learners. One good picture is worth a thousand words. It’s one thing to talk about speaker coverage, but the first time you see its prediction in your modeling software or do your first speaker autopsy, a new level of understanding is reached.

In Phase Invaders, you’ll want to use coherence blanking to remove any noise, zoom into the crossover region, then adjust delay and polarity until the Sum matches the Target. Use the score to fine-tune.

Pro battle tactic: Click and drag to zoom in on the graph. Double click to zoom out.

#3 – Advanced

Find optimum alignment through the crossover region using the phase graph and phase delay formula. (scary!)

Finding alignment on the phase graph may be as easy as sliding the delay around a bit until the pictures match, but many times the data is so hard to read that it can be difficult to tell if you are a half or full rotation away from a better result.

Here are the steps:

  1. Pick a frequency (f) that is near the center of the crossover region, has near matching amplitude on main and sub, and relatively high coherence.
  2. Use the phase delay formula to calculate the delay needed to align main and sub at one frequency. (((Main Phase/360)(1000/f))*-1)-(((Sub Phase/360)(1000/f))*-1). Pop in the four variables and you can paste it directly into Google. As far as I can tell, you can simplify this formula by removing the parenthesis and it still works, but I gave you both just in case. (Main Phase/360*-1000/f)-(Sub Phase/360*-1000/f). For more on this formula read this article.
  3. Use the result as your delay in Phase Invaders. If the phase traces are aligned, Sum is on top of Target, and you have a big fat score, you’re done. If not, try something else. Add or subtract delay to rotate the phase at f by 180º+pol. inv. or 360º. Use 500/f and 1000/f, respectively. Keep going until you find the combination with best alignment, summation, and score.

Pro battle tactic: The phase delay and time period are calculated for you in the cursor read out at the top of the screen.

Walkthrough

Let’s work through it together. Here’s a four-mic average of a measurement I took of a Martin CDD-Live 12 (main) and an SXP118 (sub). The main is 18ft directly above the sub.

The first thing I’ll do is adjust the coherence blanking to get rid of some of the noise.

Now I’ll zoom into the area of interaction and pick a frequency. I’m going to pick 105Hz because it’s near the center of the crossover region, has matching magnitude, and high relative coherence.

If you are already familiar with the phase graph then this probably looks like a polarity inversion. If you’re not, you might look at the cursor readout and see that it says 184.72 °Δ.

(I know the font is hard to read. It seemed clever at first.)

I’ll insert a polarity inversion and we get a near perfect score of 9947396.

This is a pretty clear cut case, but let’s try a couple of other options.

Earlier, with the cursor on 105Hz, we saw that the delta delay was 184.72º and the phase delay was -4.87ms. I’ll take out the polarity inversion and try -4.87ms in the delay.

Right away I can see that while we are aligned at 105Hz, the phase slopes do not match, the sum trace is not on top of the target trace through the entire crossover region, and the score of 9719481 is lower than before.

Our alignment was not improved by going in this direction, but now that we are aligned at one frequency we can easily test other possibilities.

In the cursor readout I can see that the period of 105Hz is 9.48ms and half of that is 4.74. Let’s try 9.48ms.

Our current delay setting is -4.89ms. To get a 360º rotation: -4.89+9.48=4.59ms.

Now we have better alignment of slopes, better summation, and a score of 9913456. That’s still not better than our first score.

Let’s test one more. This time half a rotation at 105Hz with a polarity inversion. 4.59+4.74=9.33ms

This gives us worse alignment and a score of 9619582.

Through efficiently testing a handful of options we have discovered the option with maximum summation and improved our relationship with the phase graph.

Pro battle tactic: Have an empty text document open to keep track of your scores and settings.

Have you tried Phase Invaders? What’s your favorite battle tactic?

Smaart® Beta: Will the new filter control in the delay finder help with your main+sub alignment?

By

The overhauled delay finder in Smaart v8 beta looks like it’s set up to make your main+sub alignments a synch. But will it really save you time?

(Yes, but you can’t throw out the phase graph, yet.)

Key Takeaways

  1. The updated delay finder includes an optional bandpass filter for tracking arrival time by frequency.
  2. It may save you some time, but you’ll still need to verify alignment with the phase graph.
  3. Results are highly variable based on SNR.

Let’s look at how it works.

The new delay finder reminds me of a process I learned at previous Rational Acoustics trainings for observing spectral crossover alignment using the Impulse Response module. It goes like this:

  1. Measure an IR of your main.
  2. Observe the ETC graph filtered at the crossover frequency.
  3. Set delay at peak.
  4. Measure an IR of your sub.
  5. Observe the ETC graph filtered at the crossover frequency.
  6. Find peak.
  7. Time offset = main peak – sub peak

I rarely used it in the field because it seemed slower than other methods, but now these features have been incorporated into the overhauled delay finder making them more accessible.

Your new alignment process might go like this:

  1. Find delay of main.
  2. Filter to crossover frequency.
  3. Insert delay.
  4. Store trace.
  5. Find delta delay of sub.
  6. Adjust delay line or physical position.
  7. Verify phase alignment.
  8. Verify summation.

Let’s look at this on a real project.

I have a recording of an array of dB Technologies DVA T8 and an S30N. I measured them each solo to find the crossover frequency.

I chose 96Hz because it is in the center of the crossover region, has matched magnitude, and high coherence.

In the delay finder I measured main solo again, this time applying a 1/3-octave bandpass filter at 96Hz. At first, I was getting a different result every time I clicked find. This is the kind of behavior I would normally expect from the delay finder trying to measure a sub. For more, read this article from Merlijn van Veen and this one from Bob McCarthy.

I was about to give up when I remembered that you can customize the delay locator FFT size and averages. In the Smaart manual I read:

The default is a 64K FFT with no averaging, which works out to a time constant of 1365 ms at 48k sampling rate. This is sufficient for finding delay times at distances up to a about 450 hundred feet (140 meters) from a source – a good rule of thumb is that the FFT time constant should be least 3x greater than the expected delay time.

I’m only measuring delays of less than 100ms. If I follow the rule of thumb, I should set my delay time to 300ms, which would be an FFT size of about 16k at 48kHz. I tried the delay finder again with the new FFT size. Still squirrely. I tried increasing the number of averages up to 8. Still no.

But then, when I set the FFT size back to 64k, all of the sudden the results were more consistent. Then I started reducing the averages until the results got squirrely again, finally settling on 5.

Unfortunately, with an FFT size of 64k and 5 averages, it takes about 16 seconds to complete, which is an eternity in production time. Here’s my measurement.

Now I’ll measure the sub solo with the same delay locator settings.

Looks like I am not within 60º through the crossover region, so let’s see if the new delay locator can help me out.

The delta delay says that the sub is arriving 8.39ms early so I’ll pop 8.39ms into the delay line and store a trace.

The phase traces are definitely closer, but I wonder if I can do even better?

At 96Hz the phase measurements are 109º apart. 109º is close to 120º and I know that 120º is ⅓ of 360º. The period of 96Hz is 10.4ms (1/F). 1/3 of 10.4ms is 3.12ms. I’ll subtract 3.12ms from my delay line for a total of 5.28ms for this alignment:

How close am I to perfect summation? I’ll load the measurements into Phase Invaders to find out. 👾

Without making any changes, I have a score of 8823501 and I can see that I am not getting full summation between 97Hz and 115Hz.

If I add 3.48ms to the delay I get a more even balance of summation and therefore a better score. Looks like my previous setting with the delay finder was right after all.

What’s Phase Invaders?? I’ll announce more about it soon to my mailing list. Join here.

Back in Smaart, I take measurement of the sub with the new delay setting to verify what I saw in Phase Invaders.

I don’t have the right recordings to verify summation, but I’ll do that in an upcoming video.

Conclusion: Results will vary widely based on the signal to noise ratio of your measurements, but I’m happy to have another tool in the toolbox to aid my spectral alignments. Just make sure to verify your results using the phase graph and combined summation.

Have you tried the overhauled delay locator in Smaart v8 beta? What were your results?

Smaart® and the Smaart logo are registered trademarks of Rational Acoustics LLC and are not affiliated with Nathan Lively or Sound Design Live.

If Poor Speaker Choice and Placement Were a Crime, We’d All Go to Jail

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 principle teacher at Synergetic Audio Concepts and a co-author of Sound System Engineering, Pat Brown. We discuss the motivation of mistakes, finding clients through retail work, investing in high quality tools, practicing at home, and the biggest mistakes in sound system design and optimization.

I ask:

  • What was the first record you ever bought with your own money?
  • 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?
  • What are some of the biggest mistakes you see people making who are new to sound system design?
  • If you could wave a magic wand and make it so, what is one concept that you wish all sound system designers understood better?
  • Tell us about the biggest or maybe most painful mistake you’ve made on the job and how you recovered.
  • What software do you us in your seminars?
  • What’s in your work bag?
  • What is one book that has been immensely helpful to you?

If the FCC prosecuted sound system designers for poor array design, like you would for a for RF antenna design, they’d be putting us in jail for how we spew energy into rooms.

Pat Brown

Notes

  1. All music in this podcast by Nataly.
  2. Course 50: How Sound Systems Work
  3. Software: GratisVolver, CATT-Acoustic, ReflPhinder, SketchUp, FIR Capture
  4. Books: Handbook for Sound Engineers, Sound Systems: Design and Optimization
  5. Workbag: impedance meter, polarity tester
  6. Quotes
    1. I had just screwed up a system really bad. I wanted to know what I did wrong and was glad to find out I had done everything wrong.
    2. The key is to do it enough times to where you don’t have to think about the steps each time.
    3. Everyone should have to do retail for a while.
    4. The music store makes a great front end for a contracting business.
    5. I get that call all the time: OK Pat, I’m out in the room, I’m got my mic up, I’ve got my USB card hooked up. Now what? And I always say, “Pack it all back up. Go home. Lock yourself in your living room. Get a couple of little sound speakers and learn how to drive the thing.”
    6. If the FCC prosecuted sound system designers for poor array design, like you would for a for RF antenna design, they’d be putting us in jail for how we spew energy into rooms.
    7. You have to minimize the excitation of the room because you are creating your own interference if you are not thinking about that.
    8. I’ve never been impressed by market share. Just because something is the most popular thing out there for doing something; that’s never been a good enough reason for me to use it.
    9. The thing about acoustic modeling programs is that you can be way off. It’s always necessary, if possible, as a sanity check, to compare it to measured data in the room.

How to Measure and Treat Resonances like Room Modes and Standing Waves with Smaart®

By

Room modes can make your mix sound flabby and are most prevalent in small rooms. A few properly placed notch filters can help and in this article I am going to show you how to measure your room and place the filters using Smaart.

Key Takeaways

  1. Precisely placed notch filters can tighten up your mix, but it’s easy to overdo it. Listen and audition.
  2. Smaart will not average multiple IR measurements. You’ll need to do that in another app.

In general, the process is simple. The audibility of room modes is determined by duration so all you need to do is observe a Spectrograph or Waterfall of your room’s impulse response. The tricky part is getting the right impulse response.

The quality of the impulse response is important because notch filters are very narrow. You need accuracy. The more measurements you take, the higher the accuracy.

Once you take all of the measurements, you could simply look at them one at a time and attempt to find the trends among them, but a faster way is to create an average. Unfortunately, Smaart doesn’t have this functionality. Fortunately, I have a workaround for you.

First, we need data. Let’s measure the impulse response.

Measure the impulse responses in Smaart

Without too much explanation for why I have chosen these settings, here’s what I recommend for the impulse module in Smaart.

Settings

  • FFT: 128k
  • Averages: 2
  • Signal: Pink sweep triggered by IR
  • Level: 20dB above the noise floor from 20Hz to critical frequency* (Don’t stress about this. If you were already taking transfer function measurements and getting actionable data through the LF, then your sig gen level is probably fine.)

Steps

  1. Place mic at head height anywhere in the audience. Room modes are not distance dependent.
  2. Press play.
  3. File > Save impulse response.
  4. Repeat six times at six random locations. If the audience and room are symmetrical, you only need to measure half of it.

This should be one of the final steps of your system tuning work. You’ll want the entire system ready to go since you are measuring its interaction with the room.

Here’s the sound system I’ll use for this article.

Here are the mics. They look closely spaced because the audience was only that deep.

Here’s what the first measurement looked like.

Now let’s create the average.

Workaround #1 – Easy/Expensive

The easiest, yet most expensive ($1,400) workaround would be to buy FIR Capture. Pat Brown used it to teach me this process at his OptEQ seminar during InfoCOMM 2019. Let’s go through it.

  1. File > Import first impulse response (IR). Repeat for all IRs. No time window necessary.
  2. Normalize all at 100Hz.
  3. Create power average. (Or optionally, right click on the IR plot and choose Sum Multiple IR to normalize arrival times and preserve the IR)
  4. Observe waterfall plot. If necessary, adjust time window for better resolution.

Here’s all of the IRs imported.

Here’s the average.

And here’s the waterfall plot where I have identified two room modes.

Workaround #2 – Medium difficulty/cheaper

If you don’t want to make the financial investment and the time to learn a new piece of software right now, I have an alternative for you: Averager.

This is an app from Eclipse Audio who you might know if you use FIR Creator. For $50, it’s a nice utility to get this job done.

  1. Preferences > Transform size > 131,072 (maximum)
  2. Load > Select directory with your IRs.
  3. Uncheck any files you don’t want to use.
  4. Optional: Select the IR with the greatest distance from the source as the reference. Delay > Time Align all to Reference.
  5. Gain > Normalize to a frequency range of 90-110Hz.
  1. Average > Averaging mode > Power
  2. Save > File > Format > WAV
  3. Set IR end to maximum (1,365ms).
  4. Save

Back in Smaart…

  1. File > Load impulse response and choose the file you just saved.
  2. Calculate Spectrograph at 16kHz with 99% overlap.
  3. Adjust upper and lower thresholds to discover room modes.
smaart spectrograph
Why is the graph zoomed in to 335ms? This was done before I discovered how to change the transform size in Averager.

This method is more challenging than the previous because the graph is harder to read.

Workaround #3 – Medium difficulty/cheapest

Room EQ Wizard is a free app with some great functionality. Although it can create a power average, it will not generate a waterfall or spectrogram of the average. Your two options are to generate a vector average instead or open the average you created in Averager. You’ll get slightly different results, but essentially the same frequency information. It looks like this.

FirCapture waterfall

I still call this one medium difficulty because of the hoops you have to jump through.

Treat the room modes with EQ filters

Now that you have identified the room modes, treat them with narrow band (Q > 10, BW < 1/6oct) filters and listen to the results.

Here’s what my filters looked like in Vu-Net.

Martin Audio EQ

As you can see, I decided to audition a bunch of different filters.

Martin Audio EQ2

Here’s the average IR of my room post EQ in FIR Capture.

FIR Capture Spectrograph`

And here it is in Smaart.

compare peaks in Smaart

And here it is in REW.

compare peaks

WARNING: YOUR RESULTS MAY VARY

While listening in the audience, I auditioned each filter and discovered how easy it was to overdo it. A little help from the notch filter tightened up the mix, but too much and it lost its life and excitement.

What’s a room mode?

Resonance: wavelengths that “agree” with a volume.

a pressure wave that decays more slowly than those of the surrounding frequencies

daytonaudio.com

Standing wave: non-propagating, it’s “standing” in space because it’s reflecting back and forth between two surfaces or nodes.

In physics, a standing wave, also known as a stationary wave, is a wave which oscillates in time but whose peak amplitude profile does not move in space. The peak amplitude of the wave oscillations at any point in space is constant with time, and the oscillations at different points throughout the wave are in phase.

Wikipedia

Room mode: now we take duration into account. If the standing wave is standing around for longer than its neighbors, it might be a room mode.

Room modes are the collection of resonances that exist in a room when the room is excited by an acoustic source such as a loudspeaker.

Wikipedia

*What’s critical frequency?

Critical frequency is a milestone in the transition of room acoustics from lower density modal behavior to higher density geometric behavior. It can be estimated with by dividing 3,390 by the room’s smallest dimension (3c / RSD).

If you are working in arenas all the time, you’ll never need to worry about it, but if you are working in small rooms it can give you some insight into the behavior of your room.

Also, the first mode can be found dividing the speed of sound by twice the room’s longest dimension (c/2L).

Questions I didn’t answer

Wouldn’t the ceiling need to be 25ft away or shorter to have a room mode at 132Hz?

Yes, because 3,390/25=137Hz. At the time, I neglected to consider this. I never measured the ceiling, but it was probably closer to 30 or 40ft, which would put the critical frequency at about 113 or 85Hz.

Why was I seeing resonances above the critical frequency?

My only idea is that critical frequency is a milestone for a transition not a true/false verification.

What do you think? (read Michael’s response below)

Also, have you tried measuring and treating room modes? What were your results?

Smaart® and the Smaart logo are registered trademarks of Rational Acoustics LLC and are not affiliated with Nathan Lively or Sound Design Live.