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How to take fast impulse response measurements in Smaart© without pissing people off

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Use these settings to efficiently investigate room modes with the impulse response module in Smaart©.

  • FFT: 128k
  • Avg: None
  • Delay: 0 (doesn’t matter)
  • Options > IR > Overlap: 0%
  • Signal Generator: Pink noise, Pseudorandom, Drop IR Data Window
    • Level: +10dB above noise floor

Measuring room modes sucks. It takes too long, pisses people off, and the ROI is debatable.

The last one is user dependent. Let’s work on the first two.

Why does it take too long?

This was user error on my part. I thought that multiple averages were required for sufficient dynamic range and actionable data. This meant that all of my measurements took twice as long because I had the averages set to 2.

While increasing averages does improve dynamic range, it is less important in researching standing waves because you’re going to average together at least 6 of them in the end.

Set averages to 0. Fixed.

Why does it piss people off?

  1. Wrong stimulus
  2. Unnecessarily high level

Use period-matched pseudo-random pink noise

The Pink Sweep stimulus pisses people off. People can get used to music and pink noise, but the sweep is surprising and painful. It’s the most common stimulus to enrage civilians.

While the Pink Sweep does improve dynamic range, the results I observed don’t justify the risk of someone telling you to stop. If you’ve got the room to yourself, though, do it.

From what I can see, using pseudo-random period-matched pink noise will get the job done at the same signal generator level and generate less complaints.

Set level +10dB above noise floor

I was using +20dB because according to the Smaart user manual you need to be at least 10dB above the noise floor, but 20dB is OK, and 30dB is preferred.

noise floor comparison

I ran through six tests at six different measurement positions:

  • A: Pink Sweep at +20dB relative to noise floor
  • B: Pink Sweep at +10dB
  • C: Pink Noise at +20dB
  • D: Pink Noise at +10dB
  • E: Pink Sweep at 0dB
  • F: Pink Noise at 0dB

Then I imported the batches of six measurements into Room EQ Wizard to be level matched at 100Hz, time aligned, and vector averaged. Below are the spectrogram graphs for each of the averages.

pink sweep vs noise at three levels
Spectrogram compare

Even though dynamic range drops as the stimulus in the room is lowered, I don’t see any difficulty in identifying the first four peaks in the first four measurements. Only when the stimulus gets to 0dB compared to the noise floor do I start to have doubts.

Obviously the results will be different in larger rooms compared to my living room, but going through all of these tests gives me the confidence to try a -10dB level moving forward.

Extra Credit

Level

Verify signal generator level by comparing it to the noise floor. Switch over to an RTA graph. Measure the ambient noise floor. Offset it +20dB. Turn on the signal generator and match it at every frequency.

While you’re there, make sure that the input levels (Mic/Ref) are matched and within the yellow target zone.

What about wind?

In any scenario where there might be a possibility of any significant time variance during the measurement period, you would probably be better off increasing the measurement time window and/or using a period-matched stimulus signal rather than upping the number of averages.

Rational Acoustics Smaart v8 User Guide

Signal Generator

Pink Noise – Pseudorando

Use a 2-channel measurement with period-matched signal generator.

At first you might wonder, why would I ever use a single channel measurement? IR measurements go back a long way and can be completed with a balloon and stopwatch. This direct method may deliver mixed results, though, since it might not have an instantaneous envelope or uniform spectral content and it doesn’t tell you anything about the sound system.

The newer indirect method uses a 2-channel transfer function to mathematically estimate the response and combined with a test signal of matched length can simplify our goal of actionable data.

feed the DFT what it really wants to eat: a test signal that either fits completely within the measurement time window or cycles with periodicity equal to the length of the DFT time constant. Signals that meet these criteria can produce deterministic, highly repeatable measurements in a fraction of the time it takes to get comparable results using random signals.

Rational Acoustics Smaart v8 User Guide

Unlike the Pink Sweep, pseudorandom cannot be triggered by the play button so you’ll need to start the signal generator before you press play or Smaart may crash, which is what happened to me the first time I tried it.

Pink Sweep

I was curious to see how the results of a pink sweep might differ so I tried that as well.

Using SATlive here to show IR overlay

Ld = level of direct sound

Ln = level of noise

Ld – Ln = dynamic range of the measurement

The dynamic range of the measurement using pseudorandom noise is -44.73dB. The dynamic range of the measurement using pink sweep is -65dB. That’s an improvement of 20.27dB.

FFT

Unless you are in a very large or noisy room, 128k (2730ms = 2.7s) or 240k (5000ms = 5s) should be plenty. I normally use 128k.

Averages

If we want better dynamic range of our measurements we can either turn up the signal generator in the room, increase the averages, or increase the measurement time. Since my goal is speed, I’m looking for the lowest number of averages and measurement time.

When measuring with period-matched noise or sweeps, averaging is normally set to “None” or 2, although it is still possible that a higher setting could prove helpful if measuring in an extremely noisy environment.

Rational Acoustics Smaart v8 User Guide
averages
AveragesDynamic rangeΔ
0-51.31
2-54.643.33
4-59.995.35

Overlap

Normally this is set to 0% for maximum noise reduction, but can be set higher to save time. Let’s see if there’s any benefit to using more overlap in terms of saving time or improving SNR.

This image compares an IR measurement with 0 averages and 0% overlap in yellow with 2 averages and 99% overlap in blue, which took about the same amount of time to measure.

Doesn’t look like much of an improvement.

Delay Locator

You only need to set the delay locator if you are using random stimulus.

The first is to delay the reference signal to match the timing of the measurement signal, so that the data windows line up. You should always do this when measuring with random signals.

Rational Acoustics Smaart v8 User Guide

We are going to use a period-matched stimulus so we can ignore this step. Yay!

Gratitude

Big thanks to Rational Acoustics support. My laptop crashed a lot during my first tests while “Processing frame”, which really sucked when the signal generator was stuck on. I emailed support. John emailed me back and I found out that an update with a fix was coming out that day. It worked!

How an improperly connected motor cable arced up the chain and almost ended in disaster

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Support Sound Design Live on Patreon.

In this episode of Sound Design Live I talk with the touring FOH sound engineer for Godsmack, Erik Rogers. We discuss how he dealt with various SPL limits while on tour, console setup, and why safety is never a compromise.

I ask:

  • What’s one of the biggest things you’ve learned while working with Godsmack?
  • Tell us about the biggest or maybe most painful mistake you’ve made on the job and how you recovered.
  • From FB
    • Sam: Ask him what kind of issues does he face on the road besides the difference in venues.
    • Andreas: His Routing in his Desk (Input- Groups – Matrix etc)
    • Chris: Always interested in rock vocal chains, and mixing them with an up front guitar sound. It can be difficult to separate them at times. Tips or tricks?
    • Alfons: Does he use gates on all vocals & drums? Does he use peak or RMS compression?
    • Rob: What beard cream is best?
    • David: Compression: heavy or light? Pre or post EQ? What console are you on, what are you doing with bus compression?
    • Mark: Can I train under him ?
    • Jonathan: What tricks does he use to get that epic kick sound?

Safety is never a compromise.

Erik Rogers

Notes

  1. All music in this episode by Godsmack.
  2. Hardware: isemcon 7150, calibrator, never portico 2, shure 91, audix d6, kelly shoe,
  3. workbag: iSEMcon emx7150, calibrator, focusrite scarlet preamp, flashlight, incense, headphones
  4. Books: The Art of Happiness by His Holiness the Dalai Lama
  5. Podcasts: How I built this, Joe Rogan, Serial Killers
  6. Quotes
    1. I don’t have the same direction I did when I was 20.
    2. Sometimes you need to get hit in the face with a brick in order to focus.
    3. I guess that’s the difference between me and a monitor guy; my customer service has a barricade in between us.
    4. You’re an American and you look at 100dB and you’re like, that’s fuck’n crazy.
    5. I calibrate one microphone every day for SPL measurement and then my roaming microphone for system tuning is not calibrated.
    6. Safety is never a compromise.
    7. Anybody who’s motivated to succeed can as long as you don’t give up, and can take a shit load of criticism.

Spot Crossover Slopes in Smaart® and Avoid Falling Sharply

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Key Takeaways

Verify crossover slopes by:

  1. Comparing them to a measurement of your DSP.
  2. Comparing them to my pre-measured templates.
  3. Calculating bandwidth.

I have often heard the advice: Avoid doubling up your crossover filters.

Most modern speakers come pre-baked with high/low-pass filters. If we insert an additional filter in our output EQ near these native filters the acoustic result will be steeper than either of the filters alone.

The danger is that we might already be planning a very narrow unity crossover between our main and sub and then be surprised when the magnitude looks like a brick wall and the phase like a pinwheel.

I’m not great at spotting these shapes, yet. To help myself practice, I measured a bunch of different crossover filters and stored them in Smaart.

Now I can compare these filter measurements to my acoustic measurements to make sure the slopes actually ended up where I wanted them. For example, here’s an alignment using Linkwitz-Riley 24dB/oct filters at 100Hz.

Comparing them to their corresponding filter measurements, it looks like I got what I paid for.

Going over some of my other previous alignments I did find some asymmetrical filters. In this alignment it looks like I ended up with a combination of 36dB/oct and 48dB/oct filters.

Here it looks like I got a 48dB/oct and a 24dB/oct combination.

Identify crossovers by measuring your DSP

You can measure your output EQ by making it the SUT (system under test). In Smaart, set the REF input of your transfer function to a loopback cable and the MIC input to the output of your DSP.

Store your trace and compare it to your acoustic result. Do they match?

Identify crossovers using these templates

To speed up the process, I already measured 80 common filters for you. Download them here. Drag them into your list of traces in Smaart.

Most of the filters are set to 80Hz so you should be able to accommodate many slopes with a little trace offset. I also measured all of the Linkwitz-Riley filters at 100Hz. If none of those match, you’ll have to measure your own.

Identify crossovers by bandwidth

We don’t normally think of a spectral crossover as having bandwidth, but it does have a clear area of interaction. This can be clearly seen if we load the traces into Phase Invaders.

Notice that the pink Sum trace has a clear starting and ending frequency. Those of you familiar with Bob McCarthy’s summation zones will also be familiar with this concept:

  • Isolation: Magnitude relationship >10dB with low risk of cancellation.
  • Transition: Magnitude relationship of 4-10dB with medium risk of cancellation.
  • Combing: Magnitude relationship of 0-4dB with maximum risk of cancellation.

Let’s look at some example bandwidths.

Linkwitz-Riley

Bessel

Butterworth

This leads me to the following conclusion:

  • 2nd order 12dB/oct filters ≈ 3.26oct bandwidth
  • 4th order 24dB/oct filters ≈ 1.65oct bandwidth
  • 8th order 48dB/oct filters ≈ 0.81oct bandwidth

Assuming your filters are symmetrical, you could use the following shortcut:

  • If f2 ÷ f1 ≈ 9.5 then you may have two 12dB/oct filters.
  • If f2 ÷ f1 ≈ 3.14 then you may have two 24dB/oct filters.
  • If f2 ÷ f1 ≈ 1.75 then you may have two 48dB/oct filters.

Slope considerations

The steeper the slope…

  • The better maintained over distance.
  • The more phase shift incurred.
  • The faster and possibly more unnatural the transition.

Does it work in the field?

How to measure crossover filters in Smaart without an external DSP

Have you learned to identify crossover slopes in the wild (in Smaart)? How did you do it?

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

Sound system design for the largest public event in Philadelphia’s history: The Eagles Super Bowl Parade

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Support Sound Design Live on Patreon.

In this episode of Sound Design Live I talk with Chris Leonard who is Director of Audio at IMS Technology Services. We discuss his sound system design and mix for the largest public event ever in Philadelphia covering 1 mile for the Eagles Super Bowl Parade, dealing with wide dynamic energy coming through a podium mic, and how to build relationships and get more work with Project Managers.

sound-design-live-mixing-monitors-tears-for-fears-chris-leonard-headshot

Yes this is a large scale event, but you’re taking the principles you already know and scaling them up.

Chris Leonard

Notes

  1. Hardware: VTX speakers, CL5, QL5, BSS BLU806, iTech amplifiers, SG300 switches, Rio rack
  2. Outreach
    1. Do
      1. Connect on LinkedIn.
      2. Show that you are actively working with your posts. Make them authentic about the level you’re working at, but focus on more of the work you want to be doing. How organized are you? What equipment are you using?
      3. Follow other freelancers and companies.
      4. Send your blackout dates (vs available) every 1-3 months.
      5. Make your emails personal and interesting.
    1. Don’t
      1. Expect a response to your availability updates.
      2. Email your availability updates more than once a month.
      3. Offer to travel if the request is for someone local.
  3. Quotes
    1. The city learned their lesson and got it right this time.
    2. With Dante over CAT5 you don’t want to go beyond 300ft, but with fiber I can go for miles.
    3. I heavily use LinkedIn.
    4. I need people who can be an A1 for a massive show and some that can be basic breakout AV tech.

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.