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d&b audiotechnik ArrayCalc vs SubAligner

By Nathan Lively

Unlike my previous post about the L-Acoustics preset guide, d&b guide’s its users to perform subwoofer alignment using their ArrayCalc simulation software. I have heard from more than one colleague that they stopped using their audio analyzer after they started using d&b ArrayCalc. With such claims of accuracy I was really looking forward to comparing the results from SubAligner.

Y7P + Y-SUB

At matched level with CUT engaged the center of the crossover region is at 138Hz.

Here’s a prediction at 125Hz.

The result from SubAligner is in good agreement. The minimum delay in ArrayCalc is 0.3ms. 1.35 + 0.3 = 1.7ms.

Here’s a link to this alignment in SubAligner.

J12 + J-SUB

With the sub level offset +6dB from the low-mids and CUT engaged, the center of the crossover region is at 139Hz.

And now the results from SubAligner.

Here’s a direct link to this alignment in SubAligner.

E8 + B6-SUB

At matched level with CUT engaged the center of the crossover region is at 144Hz.

And the SubAligner results.

Here’s a direct link to this alignment in SubAligner.

Have you done any subwoofer alignment with ArrayCalc and then verified it in the field? What were your results?

Know Your Audio Analyzer Averages

By Nathan Lively

After you take several measurements and average them together, what do you expect to see?

If these two measurements are averaged, what do you expect?

Is zero the average of -6dB and 6dB, or something else?

1 octave wide parametric filters at 1kHz

Here are four possible averages you may have guessed, depending on which audio analyzer you use.

tl;dr

  1. Know what kind of averaging your audio analyzer uses.
  2. Collecting more data is more important than the way it is averaged.

Here are demos from a selection of audio analyzers in alphabetical order.

Crosslite+ v2.0.0 8

Along with options for pre and post processing, Crosslite+ offers four different options.

“Arithmetic Average Complex” : Arithmetic mean in complex values.

“Quadratic Average Complex” : Quadratic average or RMS in complex values.

“Arithmetic Average Magnitude” : Arithmetic mean in real values in dB, with phase zeroed.

USER GUIDE CROSSOLITE REV 1.1

Does the magnitude average change with trace offset? Yes. It appears that trace offset in Crosslite is the same as a gain change.

Does the magnitude average change with phase offset? Yes, except for Arithmetic Average Magnitude.

In this test I averaged the response of two microphone cables with a second order APF inserted on one of them at 1kHz.

Se deseja saber a média de fontes separadas coerentes ou de uma curva polar, é bom utilizar a aritmética complexa.

Se é média para curvas de equalização, melhor a de magnitude em dB.

Escolhi os tipos que estão mais presentes na maioria dos softwares que possuem funções de média.

Francisco Monteiro

If you want to know the average of separate coherent sources or a polar curve, it’s a good idea to use complex arithmetic.

If it’s an average for equalization, better the magnitude in dB.

[For Crosslite] I chose to offer the types that are most present in most software that have averaging functions.

Francisco Monteiro

L-Acoustics M1

M1 offers a single kind of average, which appears to be a simple average with phase zeroed.

Does the magnitude average change with trace offset? No. There is no trace offset option.

Does the magnitudeaverage change with phase offset? No.

OpenSound Meter v1.0.5

Open Sound Meter offers vector and polar averaging.

Does the magnitude average change with trace offset? Yes, the results are the same for a measured gain change.

Does the magnitude average change with phase offset? Yes for vector. No for polar.

For in-space averaging I use polar method. For vector (complex) you need to have very close phase responses.

Pavel Smokotnin

REW v5.20

REW offers two options for averaging.

Vector average, which averages the currently selected traces taking into account both magnitude and phase. It can only be applied to measurements that have an impulse response.

RMS average, which calculates an rms average of the SPL values of those traces which are selected when the button is pressed. Phase is not taken into account, measurements are treated as incoherent. This does the same as the Average The Responses button. If the measurements were made at different positions (spatial averaging) it may be helpful to first use the Align SPL… feature to remove overall level differences due to different source distances.

REW Help

Does the magnitude average change with trace offset? Yes, but only after the data is permanently changed with the Add offset to data button.

Does the magnitude average change with phase offset? Yes for vector. No for RMS.

RiTA

RiTA offers a single options for averaging traces: Arithmetic Average Complex

Does the magnitude average change with trace offset? Yes.

Does the magnitude average change with phase offset? Yes.

The next version of RiTA will include three options for averaging.

Complex AVG: magnitude estimation is greatly affected when complex averaging is performed. It is useful when you are interested, in close measurements, in knowing the constructive and destructive interference of the sound system.

ABS AVG and dB AVG are intended for spatial averaging of several microphones. Abs AVG tends to give priority to good data and less to data affected by reflections. dB AVG gives equal weight to all data.

By default, RiTA 2.5 uses ABS AVG

Pepe Ferrer

SATlive

SATlive offers a three options for averaging: Create Sum Trace, Complex Add, and Weighted Average.

Does the magnitude average change with trace offset? No

Does the magnitude average change with phase offset? Yes for Complex Add. No for Sum Trace.

SATlive offers 3 different approaches for averaging different measurements.

1. Complex averaging: Will calculate the sum using the amplitude values of each trace and the phase relation between the traces. It is intended to average measurements taken at the same mic position, like Sub/Top time align or interference of different sources. (quick traces -> sum trace complex averaging).

2. Amplitude based averaging: Will calculate the sum by normalizing (center at 0 dB) each trace and afterwards adding the amplitude values only. This is helpful when you want to average traces taken at different mic locations (and in most cases, using the same source).  (quick traces -> sum trace Create Sum Trace).

3. Weighted averaging: This is a special version of 2. where you can assign a weighting factor to each trace (three configurable settings). This was inspired by the Primary/Secondary/Tertiary measurement approach, which I first heard about during my SIM II seminar. In fact, it does not make much sense to add tertiary traces to the result, but it would be possible. (Trace Manager)

Hint: There is a Valid only if all traces valid option for 1. and 2. where you can define wether just one valid result at a certain frequency will be sufficient for a valid result or all traces averaged must contain a valid value to create a valid result.

Which of the options do you recommend to your users for judgement of tonality and EQ operations?

Only option 2. and 3. will make sense here. I rarely work with averaged measurements during EQing. Normally I’d use the Primary Location trace as the base for EQ while the other traces help me to distinguish if the problem is global or just local.

Big differences between the different mic-locations (primary/secondary) indicate a problem that you should fix before applying the eq (redirecting the speaker, additional speaker, speaker with a different directivity pattern).

For overall tonality I’d go for 2 and for Eq-ing for 3

Thomas Neumann

Smaart v8.5.0.2

Smaart offers two options of averaging with the second including built-in proprietary pre-processing.

Decibel spatial averaging, sometimes called arithmetic averaging, is a simple average of decibel magnitudes at each frequency. Spatial power averaging is the average of squared linear magnitudes at each frequency with the result converted to decibels.

Unweighted dB averaging works exactly the same way both transfer function magnitude and spectrum averages. When you select Power averaging for transfer function measurements, however, Smaart automatically adjusts the overall level of all individual measurements going into the average according to their average decibel magnitudes in the range of 225 Hz to 8.8 kHz so that they are all approximately equal in level throughout that range.

Rational Acoustics Smaart v8 User Guide, Release 8.3

Does the magnitude average change with trace offset? No.

Does the magnitude average change with phase offset? No.

Our data is in dB, so we have to decide whether to average linearly or logarithmically, whether to normalize first, whether to weight by coherence (does it make sense that poor-quality data gets as much “say” in the final result as high-quality data?) and of course remembering that FFT math spits out complex data points, not simple integers.

So you can end up with a lot of approaches that are all valid from a mathematical standpoint, but the question becomes “which method gives us the most useful result?” (I could average together the number of socks in my drawer and the number of tires on my car, and even if my math was correct, it’s a meaningless answer for all practical purposes.) So at the end of the day, we want averaging that produces information that’s helpful to the user. If you have a bunch of traces and you average them, we have an expectation of what that final averaged response should look like. How well does it highlight the trends indicated by the individual traces? That’s what we’re looking for when we take an average, and so our averaging is designed with that in mind.

In terms of which to use, just like everywhere else in Smaart: if you’re not sure which setting you need, use the defaults. They’ve been carefully chosen over many years to give good results without the user having to tweak around. I actually reset the software to default configuration every time I use it, and I pretty rarely need to go in and change a bunch of things from that state. The primary advantage of power averaging would be if you’re averaging together a bunch of traces that have severe comb filtering (which hopefully doesn’t happen all that often). The math will give more weight to the peaks and less to the dips, so you end up with something that can be more representative of the overall response in that area and what your ear might tell you. But – in most circumstances, the differences between coherence-weighted dB average and power average end up being very small. If you create both types of average from the same dataset, and lay the two averages traces on top of each other, you’ll see they tend to agree very well. I think you’d have to come up with a pretty contrived situation or have pretty bad-quality measurement data to get a result where the power averaging and the dB averaging disagree.

Michael Lawrence

All together now

Here’s an overview of the different averages being discussed in high contrast. All of these are my own estimations since the math is not exposed and is in some cases proprietary.

Which one should I use?

Please follow the manufacturer guidelines and in most cases stick with the default settings.

The demos in this post average electrical measurements of symmetrical EQ filters in order to clearly expose the calculations being used. I want to be able to see clearly if the average of +6 and -6 is 0 or something else. Measurements of speakers in rooms will feature many wide peaks and narrow valleys instead of this symmetrical behavior.

As I worked through each demo I found myself wondering why I might use one average over another. Being visually inclined and looking at a graph, at first the simple magnitude average made the most sense.

(-6 + 6) / 2 = 0

M1 offers this as its only option and it is the default option in Smaart and SATlive.

Why do the other options exist?

If you had one subwoofer and I gave you another, how much would that be in decibels? You would add 0dB + 0dB to get 6dB.

If I gave you another half a sub you would have 8dB because 20 * log10(1 + 1 + 0.5) = 8.

Following the same process of linear to log conversion, we should calculate the decibel average of -6dB and 6dB like this:

20 * log10((0.5 + 2) / 2) = 1.9dB

Maybe it makes more sense now why some audio analyzers like REW, RiTA, and Open Sound Meter show an average of 1.9dB instead of 0dB.

Interestingly, Bob McCarthy finds even this form of average to be lacking since it does not take psychophysics into account.

Studying summation revealed that 20–40 dB dips are likely to stay down in only a small area, whereas 6 dB peaks may spread over a wide area. Studying perception revealed greater tonal sensitivity to wide peaks over narrow dips. Therefore we should be wary of accepting 0 dB as the best representative here. When samples agree, the averaging builds confidence. When samples differ, the average is suspect. There’s safety in numbers when math averaging is used: get a lot of samples.

McCarthy, Bob. Sound Systems: Design and Optimization: Modern Techniques and Tools for Sound System Design and Alignment (p. 453). Taylor and Francis. Kindle Edition.

In this case “0 dB” refers to the average of a 6dB peak and a -40dB valley.

My takeaway from all of this is that more measurements combined with optical averaging (looking at them all at once) is more important than the specific form of mathematical averaging you choose.

What do you think?

How To Tune A Sound System In 15 Minutes

By Nathan Lively

sound-design-live-professional-sound-system-setup-15-minutes-MAPP-8k

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:

  1. Verification
  2. Placement
  3. Aim
  4. EQ
  5. Crossover 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.

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.

  1. Set all outputs to unity.
  2. 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.
  3. 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.
  4. Listen. Are there any obvious problems like noise, distortion, or Left and Right sounding different?
  5. 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.

  1. Compare Main Left solo at OFFAXL and OFFAXR.
  2. Adjust aim until OFFAXL = OFFAXR in the HF (high frequencies).
  3. Repeat for Main Right.

Minutes 8-12: EQ

  1. Measure Main Left solo at ONAX and set output EQ filters to match your target trace.
  2. Listen to the filters in and out while playing your reference tracks. Are you going in the right direction?
  3. Copy the Main Left output EQ to Main Right output EQ.
  4. Measure Main L+R at ONAX and set EQ filters to return system response to your target trace. 
  5. Listen.

Minutes 12-15: Crossover Alignment

  1. Measure Sub Left solo at ONAX.
  2. 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)
  3. Measure MainL+SubL and check the combined response to make sure you have summation throughout the spectral crossover.
  4. Apply any necessary combined EQ.
  5. 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.

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

By Nathan Lively

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.

Mauricio Ramirez: Trust your ears, not just the audio analyzer

By Nathan Lively

mauricio-ramirez-trust-ears-not-audio-analyzer-featured

Subscribe on iTunes, SoundCloud, Google Play or Stitcher.

Support Sound Design Live on Patreon.

In this episode of the Sound Design Live podcast I talk to Mauricio Ramirez, senior seminar instructor at Meyer Sound. We discuss some of the biggest mistakes he sees made by people who are new to sound system tuning. We also cover:

  • Most popular test tracks.
  • Which country has the most audio engineers who smoke?
  • How did you get your first job in audio?
  • What is the best choice you made to get more of the work that you love? (Hint: It rhymes with dedication.)
  • Why parents shouldn’t teach their kids that they are special and why it’s good to be normal.

mauricio-ramirez-trust-ears-not-audio-analyzer-headshotPeople want to see a beautiful graphic, but what people forget is that we are working with sound. We are not video guys or lighting guys. We are sound guys.

Show notes:

  1. All music in this episode by jACE the Caveat.
  2. Books: Don Davis Audio Encyclopedia, Yamaha Sound Reinforcement Handbook, Modern Recording Techniques
  3. Software: Smaart, SATlive, Systune
  4. The most common sound system tuning mistakes you might be making:
    1. You have an older, cracked version of Smaart without all of the latest features. Buy a license and update.
    2. You are using smoothing. Turn it off (use 1/48 or 1/24 octave smoothing).
    3. You are trying to equalize comb filtering. Move the microphone 1m away. Don’t make decisions based on a single microphone position.
    4. You don’t have enough practice. Watch Jiro Dreams of Sushi.
    5. You trust what you see on the graphic more than what you hear. Trust your ears.
  5. Jiro Dreams of Sushi: “Nowadays, parents tell their children, ‘You can return if it doesn’t work out.’ When parents say stupid things like that, the kids turn out to be failures.”
  6. Wrecking Crew
  7.  Quotes
    1. The big problem is that people smooth the graphic of the analyzer to ⅓. That is the biggest mistake. Smooth is easier to read, but what you are reading is a lie!
    2. Only correct what is common to all measurements.
    3. You need to compare the graphic of the analyzer with something that will be played during the concert. Some people only test the sound system with pink noise and then the graphic looks beautiful and they say, “It’s ready!”
    4. Start to learn what part of the information you can see that you can trust and what part you can see that you can not trust.
    5. Concerts in an anechoic chamber would be horrible because our ears are expecting reflections.
    6. Forget about what you see on the screen. We are sound guys, not video guys.
    7. If you are special, maybe only only special people will enjoy your mixing. But if you are an average guy, 90% of the population will enjoy it.
    8. It’s not important that you are working for famous artists. If people are calling you, you’re doing your job correctly.
    9. I normally prepare two snapshots. One before and one after the EQ. Then play music or speech and change the snapshots. Most people prefer the sound without the filters. So do I.
    10. Listen first. If your brain tells you it’s good, don’t do anything.
    11. If your delay error is less than 20ms, you might not hear an echo, but you’ll have comb filtering.

mauricio-ramirez-trust-ears-not-audio-analyzer-seminar

mauricio-ramirez-trust-ears-not-audio-analyzer-band

Ramirez’s band, Opción Cero, 1997

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