Nathan Lively's Blog, page 13

Do all-pass and FIR filters cause delay?

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.

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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.

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

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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.

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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.

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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).

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

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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?

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But if I invert the polarity…

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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:

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

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If we observe summation that looks like this:

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

[image error]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.

How To Tune A Sound System In 15 Minutes

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Even professionals often skip sound system setup and go straight to mixing because there just isn’t enough time. Unfortunately, you can’t go directly to your artistic place without first passing through science. The good news is that even the smallest amounts of time can be put to good use. 

How? With a plan.

Simple Sound System Goals

The goal for tuning a sound system is very simple: manage interactions to reduce variance across the listening plane. Put another way: provide the same sound in every seat. Setting the master EQ for perfect sound at the mix position does not meet this goal. Instead, we need an order of operations to help us make changes that will benefit the entire listening area, or at least mitigate damage. The order of operations is:

VerificationPlacementAimEQCrossover alignment

It might seem like you don’t have 15 minutes to spare to check all of this, but the most important items are listed first. Completing a few is better than nothing.

You will need a dual channel analyzer like Smaart, SATlive, SysTune, Tuning Capture, RiTA, Open Sound Meter, etc..

Here are the speakers we need to set up: (2) CQ-1 (wide coverage main), (2) 650-P (2x 18-inch sub) in an uncoupled symmetrical point destination array. It’s your standard left/right mains situation (see diagram below). This is the most common professional sound system setup that I run into; it is not good or bad, just common. 

Our job as a waveform delivery service is to minimize phase distortion that causes comb filtering. Comb filtering makes a swooshing sound in the high frequencies as you move your head and should never be fed after midnight. Unfortunately, any array with speakers facing in towards a destination will produce some amount of combing. We would prefer a single CQ-1 and 650-P flown above downstage center to match the room. This design often doesn’t happen because of hardware and time limitations. I could complain about it and waste your time, but those speakers will still be sitting there, bored as hell.

Download the MAPP XT project if you would like to follow along with each step.

Disclaimer: This is a highly simplified example with minimum microphone positions to give you an idea of the structure for verifying and calibrating a professional sound system. There are many factors at play and details that I do not cover, like how to operate an analyzer. For a more in-depth analysis of this subject listen to my interview with Bob McCarthy.

[image error]Minutes 0-4: Verification

Do you think a lighting technician starts running a show without making sure that each instrument responds at the correct address? No! Better make sure all of your speakers play what they are supposed to play.

Set all outputs to unity.Play pink noise and isolate one speaker at a time. In this setup we are unable to solo individual drivers, but do it if you can.Is the left output playing from the left speaker? If not, track it down. Many times it’s just a case of faulty patching. If you’ve got lines wrong inside of a closed box, you’re going to need more than 15 minutes, so I hope you have a backup. Repeat for each speaker/driver.Listen. Are there any obvious problems like noise, distortion, or Left and Right sounding different?Measure phase response on your audio analyzer at on-axis of each speaker/driver. Confirm matching relative phase. A phase offset of 180° indicates a polarity inversion. Any point in the signal chain could cause a polarity inversion so either track it down or simply invert phase anywhere else so that they all match in the end.

This step is the most important. It will be a sad dance party if your subs aren’t working.

Placement

In this situation there’s not much we can do with placement. We would like to move each speaker closer to the center of its coverage area, but we have a stage in the way and no rigging hardware or points.

Minutes 4-8: Aim

We only have a single measurement microphone, so we’ll need more time on this step to move it between positions. If I were running late and needed to cut one step from this process, I would cut this one and instead estimate the aim with a laser.

Compare Main Left solo at OFFAXL and OFFAXR.Adjust aim until OFFAXL = OFFAXR in the HF (high frequencies).Repeat for Main Right.Minutes 8-12: EQMeasure Main Left solo at ONAX and set output EQ filters to match your target trace.Listen to the filters in and out while playing your reference tracks. Are you going in the right direction?Copy the Main Left output EQ to Main Right output EQ.Measure Main L+R at ONAX and set EQ filters to return system response to your target trace. Listen.Minutes 12-15: Crossover AlignmentMeasure Sub Left solo at ONAX.Compare to Main Left solo. Are phase measurements within 60º through the crossover region? If so, move to step 7. If not, fix it. (for more, see How to verify main+sub alignment in Smaart)Measure MainL+SubL and check the combined response to make sure you have summation throughout the spectral crossover.Apply any necessary combined EQ.Listen to the result with your changes in and out. 

This is a stripped-down example of one of the most common sound system setups that I have encountered in the field. It skips steps and makes assumptions, so use it at your own risk. There is a lot more to do to be thorough, but I wanted to demonstrate that even a small amount of time can be put to good use.

This article How To Tune A Sound System In 15 Minutes appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:3 Phase Alignment Hacks to Make Your Sound System Tuning Easier 6 Most Popular Training Videos on Sound System Tuning Analysis: How to Tune a PA System for Live Sound

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 FacebookManuel 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.[image error]

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

NotesAll music in this podcast by Bionik.Running Monitors from FOH? Here are some tips.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).Practice identifying feedback frequencies. Sing it to yourself.Hardware: X32, Midas Pro1, LS9My Results from 30 Days of Ear Training, Download the Aiming Triangles Business CardQuotesI treat everything like I’m on tour even when I’m not on tour; doing the same things in the same order every time.Before you do any test, have an idea of what you expect it to look like.I always have a cue wedge at FOH. If you don’t have one, you’re guessing.Jazz musicians are much more sensitive than rock musicians.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.We ignore the polar pattern of the mic. That’s super important. Buy yourself every dB you can get.I always double patch my money channel.[image error]

This article Fighting Microphone Feedback WITHOUT a Graphic EQ While Mixing Monitors from FOH in a Reverberant Room appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:FOH Mixing: EQ it till it sounds good Mixing Monitors from FOH: 17 lessons I learned from Grealy at Soulsound Mixing Monitors for Tears for Fears

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

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.

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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.

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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.

[image error]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.

[image error]Shure SM58

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

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Here’s the manufacturer’s specification.

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

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

[image error]Shure 185

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

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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.

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

[image error]Shure MX412

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

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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.

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

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

This article 3 EQ Snapshots That Will Make Your Corporate Event Mics More Transparent Using Smaart: SM58, 185, MX412 appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:How to maximize gain before feedback of a podium microphone using Smaart What do all of those squiggly lines mean? (a short intro to the graphs in Smaart) How to Update Snapshots on the SD5 in the Middle of a Show

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

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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:

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.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.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.

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

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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.

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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.

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(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.

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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.

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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.

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

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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

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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?

This article Invasion of the Phase Invaders: An Audio Engineer’s Guide to Battle Tactics and Big Scores appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:3 Phase Alignment Hacks to Make Your Sound System Tuning Easier How to flatten the phase for easier main+sub alignment Smaart Beta: Will the new filter control in the delay finder help with your main+sub alignment?

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

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

The updated delay finder includes an optional bandpass filter for tracking arrival time by frequency.It may save you some time, but you’ll still need to verify alignment with the phase graph.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:

Measure an IR of your main.Observe the ETC graph filtered at the crossover frequency.Set delay at peak.Measure an IR of your sub.Observe the ETC graph filtered at the crossover frequency.Find peak.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:

Find delay of main.Filter to crossover frequency.Insert delay.Store trace.Find delta delay of sub.Adjust delay line or physical position.Verify phase alignment.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.

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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.

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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.

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

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

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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.

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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:

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

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

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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?[image error]

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

NotesAll music in this podcast by Nataly.Course 50: How Sound Systems WorkSoftware: GratisVolver, CATT-Acoustic, ReflPhinder, SketchUp, FIR CaptureBooks: Handbook for Sound Engineers, Sound Systems: Design and Optimization Workbag: impedance meter, polarity testerQuotesI 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.The key is to do it enough times to where you don’t have to think about the steps each time.Everyone should have to do retail for a while.The music store makes a great front end for a contracting business.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.”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.You have to minimize the excitation of the room because you are creating your own interference if you are not thinking about that.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.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.

This article If Poor Speaker Choice and Placement Were a Crime, We’d All Go to Jail appeared first on Sound Design Live. Sign up for free updates here.

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

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.

StepsIn Smaart v8, Config > Manage Configurations > Current Config > Save As.In Finder, ⇧⌘G then ~/Documents/Smaart v8/Config and copy the config file you just saved.⇧⌘G then ~/Documents/Smaart v8/Beta Config and paste it.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?

This article 3-Step Configuration Hack to Get up and Running Fast with Smaart v8 Beta appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:How to use a custom weighting curve in Smaart and why I don’t recommend it How to build your own audio analyzer (like Smaart) for FREE 9 Smaart shortcuts that will make your life easier

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

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

Precisely placed notch filters can tighten up your mix, but it’s easy to overdo it. Listen and audition.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: 128kAverages: 2Signal: Pink sweep triggered by IRLevel: 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

Place mic at head height anywhere in the audience. Room modes are not distance dependent.Press play.File > Save impulse response.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.

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

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Here’s what the first measurement looked like.

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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.

File > Import first impulse response (IR). Repeat for all IRs. No time window necessary.Normalize all at 100Hz.Create power average.Observe waterfall plot. If necessary, adjust time window for better resolution.

Here’s all of the IRs imported.

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Here’s the average.

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

[image error]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.

Preferences > Transform size > 131,072 (maximum)Load > Select directory with your IRs.Uncheck any files you don’t want to use.Select the IR with the greatest distance from the source as the reference.Gain > Normalize to a frequency range of 90-110Hz.[image error]Average > Averaging mode > PowerSave > File > Format > WAVSet IR end to maximum (1,365ms).Save[image error]

Back in Smaart…

File > Load impulse response and choose the file you just saved.Calculate Spectrograph at 16kHz with 99% overlap.Adjust upper and lower thresholds to discover room modes.[image error]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. (If you know how, let me know!)

What you can do is generate a spectrogram from the average you created in Averager. It looks like this.

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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.

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

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

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And here it is in Smaart.

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And here it is in REW.

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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.

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?

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

This article How to Measure and Treat Resonances like Room Modes and Standing Waves with Smaart appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:Smaart: Tracking peak frequency without the mouse cursor What do all of those squiggly lines mean? (a short intro to the graphs in Smaart) Smaart is just a tool. You are the analyzer.

INFOGRAPHIC: Create an IEM Link Budget to Avoid RF Dropout and Overload

In this video Stephen Pavlik walks me through how to create a link budget to find out if my IEM system will work…or why it didn’t work.

Download the infographic and get all 3 training videos

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There is no single point solution.

Every point in the RF signal chain can either attenuate or amplify the signal. If it goes too low, we get dropouts. If it goes too high, we get overloads. Through maximizing efficiency at every point, we can avoid dropouts and quickly troubleshoot interference.

Don’t be intimidated! There is no difficult math. The only challenge is researching the loss or gain of each piece of equipment in your signal chain.

This article INFOGRAPHIC: Create an IEM Link Budget to Avoid RF Dropout and Overload appeared first on Sound Design Live. Sign up for free updates here.

Loved this post? Try these:INFOGRAPHIC: How to Avoid RF Dropout and Overload with a Wireless Microphone Link Budget 3 Common Antenna Placement Mistakes and How to Fix Them [Infographic] Real World troubleshooting tips for wireless microphone and In-ear-monitor dropouts and interference