PDF-Master-Your-Craft-Nathan-Lively

SOUND SYSTEM TUNING:

For Confidence & Consistency

By Nathan Lively

ACKNOWLEDGMENTS

The material in this book comes from the generosity of Bob
McCarthy, Daniel Lundberg, and everyone reading Sound Design
Live and guiding my work for the past four years.

I would also like to thank Elis Bradshaw for editing and
providing clarity where there was chaos and George Kao for
continued support.

INTRODUCTION

Why I Tune

I'll never forget the day I discovered Bob McCarthy's
book, Sound Systems: Design And Optimization. I felt a wave of
excitement and relief. Finally, an empirical system for working with
audio.

As much as we have advanced, so much of pro audio still feels
like black magic. Not because we don't have the science, but
because we refuse to use it.

Honestly, if McCarthy's book had been published and everyone
would have immediately adopted it's guidelines and strategies, I
probably wouldn't have written this book. But I did write it because
it's the least I can do to promote our industry's continued evolution.

How To Use This EBook

You can read this eBook from start to finish, but it also functions
as a manual. Check the table of contents and find those topics that
appeal to you. Interviews are presented in their original dialogue
with my comments and questions in red italics. Some content has
been edited for clarity. Keep in mind that these are transcriptions of
live conversations; this is not a text book.

As an example of style as well as corroboration of my mission,
here is an excerpt from my interview with Larry Crane:

Larry Crane (sound engineer)

Crane is the founder and editor of Tape-Op Magazine. Here we
commiserate on the lack of compelling pro audio journalism, which
lead us both to start publishing on the subject.

The thing, too, that was really apparent is that with something

like Mix Magazine or whomever—EQ, Electronic Musician—and
almost everybody, is that they’re much more likely to talk about the
recording equipment than the techniques and the choices that led to
using that equipment. I call it, The List. “I put a D12 on the kick and
blah blah blah…” I mean, I’m not too interested in reading that stuff
unless it’s something really oddly unique or a brilliant little idea.
[laughs]

I was reading that stuff while recording in my basement. I just
thought, “Yeah, right. I don’t have any of that stuff. I can’t afford it.
How about telling me where to place the snare drum, not telling me
what mic to use?” [laughs]

Yes. I had the same experience with live sound trade magazines.
There are a lot of lists of equipment and stuff that people are using
on tour and unless you’re working on the next Madonna tour, those
lists don’t really help. I want to know why they made those choices.
I want a lot more critical information.

It’s easy to read that the new big tour has a 500 input MIDAS
automated console. I’m sure they do. That’s great. The rest of us are
going, “What? I’ve got a Mackie.” [Laughter] I’ve got a Mackie
that someone poured beer on last week. I mean, I’m a little bit of a
socialist at heart, so to me, when you see that people only discuss
the upper echelons of the business that we’re in, you realize that
there’s something really missing.

WHAT IS SOUND SYSTEM TUNING?

Sound engineers and production staff often confuse sound
system tuning and sound system toning. Tuning a sound system
involves objective goals and measurements. Toning a sound system
is a subjective, artistic endeavor.

Audio can be scary because it's invisible. Therefore, many people
believe all audio engineers have identical wizard-like abilities: they
are expected to walk into any room, anywhere, and create magic
sound with their golden ears.

In reality, system technician and mix engineer are separate jobs.
Although both use their ears and are often performed by the same
person, they require different skills and goals. Let's stop trying to
combine the two jobs into one and get to work. In this post I would
like to share some of the information from Bob McCarthy's original
article, Toning Your Sound System.

Sound System Tuning

Tuning a sound system: Adjust for uniform response over the
listening area, with minimal distortion, maximum intelligibility and

best available sonic imaging.



What's the goal of tuning a system? To have the same sound
everywhere. That means having the same level, same frequency
response, and same intelligibility in every single seat.

Is that attainable? No. Our atmosphere is a harsh place for sound
waves. But do we strive for faithful delivery to all ears? Absolutely.
We know our children will never be perfect, but we love them
anyway.

Tuning can be objectively measured. We have tools that we can
use to measure exactly how close we are to achieving our goal, and
therefore, declare success. With software and hardware such as
SIM, Smaart, Easera, and SATLive, we can perform a transfer
function to measure relative level, delay/phase, and frequency
response. Toning, in contrast, is a subjective process.

Sound System Toning

Toning a system is the setting of a bank of global equalization
filters at the output of the mix console that drives the sound system.
If you want to set it by ear fine. If you want to set it by 10,000 hours
of acoustical analysis containing mean/spline/root squared/Boolean
averaging then go for it. There is nothing at stake here. Nothing to

argue about. The global equalizer is just an extension of the mix
console EQ. -Bob McCarthy

Sound engineers just love to argue on forums and in trade
magazines about exactly how to do this. The classic advice is to yell
"Check!" into a microphone over and over while moving filters on a
graphic EQ until everyone's ears bleed, and call that success. Other
engineers choose to play songs that they know well. I like pink
noise because it's fast, you can play it quietly, and has no emotional
connection.

The truth is, it doesn't matter. It's not a science and we don't need
to make it one.

Good toning enhances the musical quality, or natural quality of
transmitted sound. Good tuning ensures that the good (or bad)

toning makes it beyond the mix position. -Bob McCarthy

It is important to keep these concepts in perspective because
audio is primarily a subjective and emotional business. We can't see
it, so even those of us who work in the field feel very strongly about
what happens in this invisible space. My guess is that you've seen
an article or two about the advantages or disadvantages of a "flat
sound system." That argument is pointless, because you can't dance
to white noise. Music is all about dynamics, not equal power per
frequency band.

I have to admit that initially there is a temptation to draw straight
lines on the analyzer screen, but it will drive you insane. Besides, so
many other tasks come first in tuning a sound system: checking
driver functionality and polarity, aiming the speakers, adjusting
splay angles and relative level between speakers, setting crossovers,

phase alignment, intelligibility analysis, and treating reflections.
Equalization is the final step in the process.

I once worked with a sound engineer who often repeated a story
of a Meyer Sound technician coming in and "SIMing the sound
system," only to leave it sounding terrible. I'm sure he's right; to him
it sounded terrible. But it probably also sounded terrible at every
seat, because it was properly tuned. His frustration comes from this
breakdown in communication about where system optimization
ends and artistic expression begins. As Bob McCarthy says in our
interview, "Draw me a picture of how you want it to sound and I'll
paint that in every seat. But you can't paint on a canvas with a giant
rip in it. That's no fun."



Here's a common error: Measure at 15 different places
around the room. Average them together. Apply the reverse
settings to your EQ.

How do I know?

Cause I used to do that shit all the time!! I had a hand-held RTA
(real time analyzer) that would allow you to put measurements into
memory, average them, then flip them to make settings to a 31-band
graphic EQ. First of all, an RTA and graphic EQ are never
appropriate for system tuning. Second, it's really important to know
what is happening with your audio in different parts of the room, so
averaging different areas together is misleading. As McCarthy says,
"Why bother to take samples all around the room if you're not going
to do anything about the differences around the room? It's just a
waste of time." Also, the math doesn't work out because
measurements of destructive summation (cuts) are often narrow and

deep while measurements of constructive summation (boosts) are
often wide and short.

Take away message: You don't use an audio analyzer to make
something sound good; you use it to make it sound the same

everywhere.

IS FLAT GOOD OR BAD?

00:00 / 00:00

**NOTE: This is the transcript for an attached video. If the
embedded video will not play, see the separate media included in
this package.

Many of the questions the I receive on the goal of system tuning
are about this word: FLAT. They included:

Is FLAT the real goal when tuning a single box vs tuning a
multiple same box system?
Why does a flat frequency response not sound good,
necessarily?
Does a system have to be flat in order to produce a good
mix?
Is there a target for a frequency curve response? I know
that a flat system doesn't sound good or natural.

The answer is very simple, but it’s also very important, because
the sooner we dispel this myth that we are here to create flat, the
sooner people will start accepting system tuning and measurements
rigs as normal practice.

Here are the two main points to consider:

When I am measuring systems in the field, I’ve never come
across a measurement with 100% coherence. The image you see
here is a measurement I took in a controlled space, on axis, only

about 10 feet away from the speaker and you still don’t have perfect
coherence. There’s always some amount of reflections and comb
filtering and room artifacts that influence the measurement quality.
What I’m trying to say is that there are a lot of practical reasons
why you will never be able to make a sound system measure
completely flat, so stop trying.

I honestly don’t care what shape you want to make. You like flat,
fine. You like a smiley face, fine. What’s important is that you
come in with a plan and test each step along the way. I have a target
shape that I’m often looking for and after I make a couple of
changes to hit that shape, I play some music, take the filters in and
out, and just listen to see if I’m going in the right direction. I’ll also
speak into a mic to test speech intelligibility. This is why it can be
great to work with a partner, for example, if you are the system tech
working with a FOH mixer, so you can say, “Hey, I made a couple
of changes, would you take a listen if see if that’s what you are
going for?”

So I would invite you to abandon this idea of flat. Live events
and sound systems will never be completely under control. Instead,
prioritize the biggest problems and treat those first.

There is no magic bullet or cookie cutter, you still need an MRI
doctor to look at the brain scans before deciding how to proceed.

WHERE DO I PUT THE
MEASUREMENT MIC?

00:00 / 00:00

**NOTE: This is the transcript for an attached video. If the
embedded video will not play, see the separate media included in
this package.

Another topic that I get a lot of questions on is microphone
placement. They included:

If you are about to tune a system with multiple speakers
what is the most ideal position of your measurement mic to
be sure you will get the best delay times for the system?
What is the best microphone position to align front fill to
main array?
What are the positions of microphones if you have very
little time to adjust a design?
Where should I position the measurement mic when
adjusting subwoofers?
Is there is one point in the audience where the entire sound
systems should be perfectly aligned?
How many mics do I need to set up to get good
measurements of a venue?

First of all, I want to let you in on a little secret. I usually only

take one measurement mic out with me on gigs and it is one of the
cheapest mics on the planet. Now, would it be quicker if I had 4 or
6? Sure, but I like the simplicity and focus of a single microphone
and I’m pretty efficient with it. It does mean more running around
with long mic cables, though.

Second, a good way to think about putting together a sound
system is like sewing a quilt. So you have all of these squares of
fabric that you need to put together so that there won’t be any holes
in the quilt. That place where the pieces come together is the seam.

In a sound system you have different speakers with different
coverage shapes and you have to sew those together so that the
coverage from one to the next flows naturally and doesn’t leave any
holes. So it’s the seams between the pieces that need to most
attention. The seams are what Bob McCarthy calls the crossover
points and that’s where you set the delay and level for your under-
balcony delays and sidefills.

Now some of you are surely asking, How do I find the crossover
point? Now that’s a subject that could take hours to cover
completely, but in general, think of point in space where two
speakers or arrays should meet playing the same source material in
equal level and time. That’s the seam of the quilt, where one
speaker hands over the coverage to another, and it’s generally where
the off-access points of each speaker meet.

For those of you who were asking about where to measure if you
could only measure one spot: It’s pretty much the same answer,
which is the on-axis point of your main speaker or array. This is
what covers most of the audience, right? So if you have time to
measure one spot, go for the point that will affect the largest number
of people and put your mic right were the main speaker is aimed.

But that’s not always true, right? Imagine a situation where you
are 99% sure that everything with the main speakers is fine, but they
just added new under-balcony speakers that no one has taken time

to verify or align. In that case, measuring the crossover point is your
first priority.

For those of you who were asking about mic placement for
subwoofer alignment: Most of the time this is at FOH. The reason
for that is that this is where the mixer is making a lot of decisions
about how much bass to put into the room and I want to help him
make those decisions as best as I possibly can.

PHASE ALIGNMENT: SUBS/MAINS

00:00 / 00:00

**NOTE: This is the transcript for an attached video. If the
embedded video will not play, see the separate media included in
this package.

Alignment between your main speakers and subs is something
that a lot of people want to learn and I want to give you the super

simple method for solving that. There are at least three methods that
I know of, but I think this will be an easy one for you to remember.

1. Measure Mid/High speaker, on axis, and save trace.
2. Without changing the delay or speaker position, measure

the sub.
3. Compare their phase responses and add delay until the

phases align at the spectral crossover frequency. This is set
by the manufacturer, usually between 80 and 120Hz.
4. If that doesn't work, add delay to the Mid/High boxes
instead.
5. Match level at spectral crossover frequency.

If you don’t have delay line or speaker processing gear, try
moving the boxes around.

HOW TO TUNE A SOUND SYSTEM
IN 15 MINUTES

Even professionals often skip sound system setup and go straight
to mixing, because there just isn't enough time. Here's a secret: you
can measure and optimize a sound system in 15 minutes. How? It's
simple.

Simple Sound System Goals

The goal for tuning a sound system is very simple: provide the
same sound in every seat. Setting the master EQ 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 this damage. That includes:

Driver functionality and polarity
Speaker aim and splay
Speaker level and crossover
Phase/delay
Filter/EQ

It might seem like you don't have 15 minutes to spare to check all
of this, but remember this universal rule: if we verify every element
of the sound system, nothing will go wrong, but if we skip any one
part, it will fail and hysteria will spread.

Measurement Equipment: dual channel analyzer (SIM, Smaart,
SATLive, EASERA); laser; protractor.

Sound System: (2) CQ-1, (2) 650-P in an uncoupled symmetrical
point destination array (see diagram below). This is the most
common professional sound system setup that I run into; it is not
ideal by any means. Our job as a waveform delivery service is to
minimize phase distortion causing combing. Unfortunately, any
array with speakers facing in towards a destination will produce
combing. We would prefer one CQ-1 flown above downstage center
to match the room and both subs together on one side or in the
corner would help our low-end coverage. These things almost never
happen, and I could complain about it and waste your time, but

those speakers will still be sitting there, bored as hell.

One thing I often like to do in this situation is to mute one of the
subs and apply high-pass filters to the tops. This technique usually
gets more even coverage below 100Hz. One of the subs ends up
being a fancy speaker stand, but no one notices. Download
the MAPP online project if you would like to follow along with
each step.

Disclaimer: This is a highly simplified recipe 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, read my interview with Bob McCarthy.

0-4min: Driver Functionality & Polarity

Do you think a lighting technician starts running a show without
making sure that all of their instruments work? No! Better make
sure all of your speakers are playing 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 output X playing from speaker/driver X? 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. You may need to get your
head close and listen to each one. This is best done with an
assistant to guard and operate the mixing board. Repeat for
each speaker/driver.
4. Measure phase response on your dual channel analyzer at
the on-axis point of each speaker/driver. If the phase trace
is around 0°, that's normal phase. If it's at 180°, that's

inverted. Is output X phase response what you expected?
Polarity inversion is often cause by an improperly wired
cable or by the use of a gender turn-around.

I spend the most time on this step because it's the most important.
It will be a sad dance party if your subs aren't working — don't
make the frat boys cry.

4-8min: Speaker Aim & Splay

As Steve Bush from Meyer Sound likes to say, "Put sound where
there are people."

1. Using a protractor and a laser on top of speaker A, show
horizontal coverage towards the (house) left wall. If you
can avoid the wall completely, great, but in situations like
this where the room is too long and speakers too wide,
shoot for about halfway down the audience area.

2. Point the laser on-axis and find the spot where it crosses

the middle of the audience. Halfway from there to your
speaker is OnAx A.
3. Measure and record OnAx A.
4. Find the edge of the audience (house left), equidistant from
the speaker. This is easy since the delay from the speaker
shown on your analyzer should be exactly the same. This
should be OffAx A.
5. Measure OffAx A. Pay special attention to the HF region.
It should be close to -6dB from OnAx A. If it is not, adjust
speaker position and aim.
6. Copy final aim to speaker B

8-10min: Speaker Level & Crossover

1. Return to OnAx A. Record level at spectral crossover
frequency (100Hz in my case).

2. Without moving the microphone, measure OnAx C. Adjust
output C to match A at spectral crossover frequency (-1dB

in my case).
3. We'll get to speaker B in a minute.

10-12min: Phase & Delay

1. Observe the phase response difference of OnAx A and
OnAx C around the spectral crossover frequency without
changing the compensation delay of the analyzer. Are they
matched?

2. If not, measure OnAx C while adjusting it's output delay to
match OnAx A at the spectral crossover frequency.

3. Once they match, combine A and C and check for uniform
response.

Note: In cases like this where the subs are very close to the tops,
you will often have matching phase response. The manufacturer

has designed them that way.

12-15min: Filter & EQ

1. Set EQ of OnAx A.
2. Add speaker C and EQ.
3. Compare with OffAx A. Make tough decisions (-4dB

between 250 and 500Hz to compensate for pink shift in my
case).

Now for speaker B.

1. Measure OnAx B and match level, delay, and EQ to OnAx
A.

2. Combine A, B, & C and check OnAx B and A again.

This was a very basic example. If you would like this kind of
step-by-step instruction for different system setups, let me know and
I can produce more of them.

These are some of the methods I've come up with to deal with
this all-too-popular professional sound system setup. What are some
of your tuning tricks for this scenario? Email me.

HOW TO FIND SPEAKER

COVERAGE IN ONE STEP

In sound system design, step one is to define speaker coverage.
Lucky for you, Daniel Lundberg and I have created a great new
speaker coverage calculator that makes this initial step quick and
easy. The calculator will give you a consistent coverage result for
any scenario, but unfortunately it's not very smart. It doesn't know
about walls and other speakers, so it works best for single-speaker
applications like balconies, side fills, and smaller venues.

Consistent coverage means that SPL will not vary more than
6dB. So if the highest measured level in the audience is 100dB SPL,
then no part of the audience may be lower than 94dB SPL. This
calculator is designed to give you the least amount of level variation
across the listening area, but there are many more variables at play

here. With that in mind, let's proceed.

Step 1 And Done

Put your measurements into the Speaker Coverage
Calculator (also included in media folder).

Example 1 - Balcony

Let's take the room from How To Tune A Sound System In 15
Minutes, add a balcony, and raise the ceiling. I used visual aids to
show a four-foot seated audience height covered by four speakers:
Main, Balcony, Sidefill, and Under Balcony. Download my MAPP
Online Pro project to follow along.

Put your measurements into the speaker coverage calculator
above. It recommends a nominal vertical coverage of 51° and a
UPQ-1P aimed 25.3° above the front. You can manually work out
the angle in MAPP, or use the speaker coverage calculator. Input
the angle of your architectural guide (the one showing distance to
front) and the calculator will tell you exactly what to put into the
Rotation About CDRM box in MAPP. I put in a 139° front angle

and got -15.7° for the speaker aiming angle.

Here's a prediction at 8kHz. Did we meet our criteria for
consistent coverage? Just looking at the graph I would say there's no
more than a 3dB difference from on-axis to off-axis, and consulting
the virtual SIM I see there is no difference in average SPL and only

about a 4dB difference in the high end. Great success!

Example 2 - Side Fill

The speaker coverage calculator recommends a UPA-1P aimed
21.2° above the front. Virtual SIM tells me that there is about a 8dB
difference from on-axis to off-axis. This doesn't meet our
requirements, but I found that if I aim it lower I can can get more
even coverage and meet the crossover point with the main speaker.



Example 3 - Under Balcony

With limited height, it's going to be hard to cover this listening
area with one speaker. The speaker coverage calculator recommends
subdivision, but it also keeps recommending a giant JM-1P (see
prediction below). Now I know how Dr. Frankenstein feels: I asked
for consistent coverage and I got it, but I also got a mob of angry
townspeople. I tried a UPM-2P here aimed at the last row and that
worked a lot better.

Example 4 - Small Venue

Let's take a look at Ashkenaz, one of the first music venues I
worked at when I moved to Berkeley. When I first put the triangle
lengths into my speaker coverage calculator (9.5', 28.5', 25'), it gives

me some crazy results and recommends that I subdivide the array.
Let's assume that I use a down fill to cover the first ten feet of the
audience. With new results, I try a UPQ-1P and see that it provides
even coverage. I also observe that aiming it further down will get
me less consistent coverage, but will let me send more of those
expensive decibels into the audience, providing more bang for the
buck.

To Subdivide or Not To Subdivide

How do we determine when we need to subdivide our coverage
into more than one segment? Here's a nice rule of thumb that I
learned from Bob McCarthy's seminar: measuring from the speaker,
take the ratio of the distance from the first row to the last and

subtract one. So in our example the first row is 9.4ft from the
speaker array and the last row is 28.5ft from the same array. That's a
ratio of 1:3. 3 - 1 = 2. So a two-segment array would probably work
better for this coverage area. How many speakers in each segment?
That's up to you and your budget. For more on this topic, read
McCarthy's article, Array Or Not To Array.

Finding Speaker Coverage When Front And Back Are
Equal

The horizontal coverage plane of a sound system often looks like
an isosceles triangle (two equal sides and angles) and the
loudspeaker nominal coverage can be found with basic
trigonometry. Remember soh cah toa from trigonometry? No?! Me
neither. Lets watch this video! To find speaker coverage angles in
the field, find the lengths of the triangle for the area you are trying
to cover and solve for the inside angles. There are also plenty of