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

By Nathan Lively

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.

3-Step Configuration Hack to Get up and Running Fast with Smaart® v8 Beta

By Nathan Lively

I know you want to use the new beta version of Smaart, but you haven’t found time, yet. This 3-step hack will have you up and running in about 5 minutes.

Steps

  1. In Smaart v8, Config > Manage Configurations > Current Config > Save As.
  2. In Finder, ⇧⌘G then ~/Documents/Smaart v8/Config and copy the config file you just saved.
  3. ⇧⌘G then ~/Documents/Smaart v8/Beta Config and paste it.
  4. In Smaart v8 Beta, Config > Manage Configurations > Stored Configs > Recall.

Now all of the settings you worked for years to customize will be imported and you can start working immediately instead of starting from scratch.

Download the Smaart beta here.

Bonus tip: If you’re ever struggling with Smaart and you just can’t find the problem, save your current config then try Restore Defaults in Config Management. You’ll have to rebuild everything from scratch, but it might be faster then going through every little setting until you find the problem. If it doesn’t fix the problem, you can always restore your saved configuration.

Have you worked with the configurations manager? Have you discovered any helpful hacks?

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

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

By Nathan Lively

FIR Capture spectrograph 2

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.

How to use a custom weighting curve in Smaart® and why I don’t recommend it

By Nathan Lively

At first, using a custom weighting curve in Smaart seems like a brilliant time saver, but it turns a guide into a rule and could make your optimization procedure too rigid.

A custom weighting curve is like a filter for your data, similar to a microphone correction curve. Its intention was to add A and C weighting to Magnitude and RTA graphs, but Smaart makes it really easy to turn your target trace into a weighting curve as well. Then, instead of matching your measurements to a target trace, you can simply match them to the 0dB line.

Here’s an image of me comparing a measurement to a target trace.

And here’s the same measurement with the custom weighting curve.

At first, this saved me time. My normal procedure of capturing multiple measurements at once means that I’m also hiding multiple measurements at once. It usually goes like this:

  1. (G) Start generator
  2. (shift+space) Capture all
  3. (G) Stop generator
  4. (command+shift+H) Hide all traces
  5. Scroll down to find target trace. Unhide it.
  6. Repeat

Using the the custom weighting curve allows me to skip step 4 and work with a cleaner graph. I tried it on a show and it worked great.

Here’s how to create your own custom weighting curve:

  • Load the appropriate target curve. (you can download mine here)
  • Right-click on the name and choose Export as Weighting Curve
  • (option-G) Open Measurement Config and choose one of your measurements.
  • Under Global Settings, choose your weighting curve.

If you want to test this out on saved trace: right click > Info > Weighting

Pretty cool, right?

Here’s the problem.

Your target curve should be used as a guideline and not a rule. The guideline can trigger your intuition when comparing two traces.

Eg. The high shelf at 10kHz reminds you that measurements at different distances will show different amounts of air absorption. The low shelf at 50Hz reminds you that you should expect to see more room gain in the low end when you are inside rather than outside.

Here’s an annotated target curve to give you an idea.

Unfortunately, when you use the custom weighting curve, you don’t have those cues and you can forget that you are looking at filtered data. Plus, if you ever want to review past measurements, you have the added complexity of trying to remember which ones were made with the weighting curve on or not.

I recommend that you try it out because it’s easy and quick, but I don’t recommend that you use it in your normal tuning processes.

Have you used custom weighting curves in Smaart? What were your results?

9 Smaart® shortcuts that will make your life easier

By Nathan Lively

sound-design-live-9-smaart-shortcuts-make-life-easier-align-featured

#1 Spectrum sound check mode

One of the first steps in the Audio Analyzer Verification Checklist is to verify that Smaart is receiving all inputs. You can do this very quickly by pressing play on all inputs and then pressing the 0 (zero) key.

#2 Align all visible traces

Shift-click on any magnitude trace to align all visible traces at that frequency.

#3 Clear All dB Offsets

Now that you’ve made a mess by offsetting all your traces, Kondo your screen by clearing all offsets on visible traces with command+Y. If any hidden traces have offsets, they will be maintained.

#4 Hide all visible traces

When working with multiple microphones you can quickly fill the screen with saved traces. To clean it up, use command+shift+H to hide all visible traces.

#5 Quickly zoom in for detailed work

I recommend you edit your zoom presets to

  1. 30-400 Hz
  2. 200-2000 Hz
  3. 2-10 kHz
  4. 10-18 kHz
  5. Default

Now you can quickly navigate between them with option+1-5. Note that option+5 will return the currently selected graph to default view, but clicking on the perimeter of the Magnitude or Phase graph will return both Magnitude and Phase graphs to default view.

#6 Scroll

This is such a helpful one and a lot of people don’t know it. Most commonly I use it to scroll around the Y-axis of the Phase graph to make it easier to read. Combine this with my earlier post about flattening the phase and you’re in business.

Another common use is scrolling left and right on the X-axis after entering a zoom preset. Maybe I have zoomed into the low end to look at a crossover alignment and I need to scroll a few Hz to the right for more information.

#7 Recapture

A good way to keep your data bar organized is to avoid unnecessary traces. If you realize you’ve made a mistake, instead of saving a new trace, recapture an old one. Select the old trace and press shift+command+space.

#8 Stop the generator

There’s no need to continue running the generator while you are capturing and naming a trace. Quickly hit the G key and then the space bar to stop the generator right before you capture the trace.

#9 Get rid of the clock

Why does every audio analyzer think you need a clock? I guess it’s nastalgia from SIM3.

Use option+k to get rid of it. Simply pressing k will switch back and forth between the clock and the SPL meter.

What’s your favorite Smaart shortcut? Let me know in the comments below.

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