Live_Sound_Fundamentals
Live Sound
Fundamentals
Bill Evans
Course Technology PTR
A part of Cengage Learning
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Large-Music Software
Live Sound Fundamentals
† 2011 Course Technology, a part of Cengage Learning.
Bill Evans
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1 2 3 4 5 6 7 12 11 10
To Randy Holland, who knows nothing about audio but who
taught me that—be it knowledge or goodwill —the only way to keep
it is to pass it along.
Large-Music Software
Acknowledgments
There is a reason why some of us have a tendency to just say, ‘‘Thanks, everyone’’
instead of naming names. It’s because we are like that guy at the Oscars a few years ago
who forgot to thank his wife. It is telling that we all remember the incident but not so
much the person involved. Anyway, if I thanked everyone who deserves it, the
Acknowledgments would be a book all by themselves. So, here goes nothin’.
Friends and band mates who were along in early parts of the audio journey, including
Mike Krupka, Mark Lewis, Josh Lober, Julie Prince, Mark Peotter, and Jake Kelly.
To everyone at St. Therese Parish in Alhambra, California, who trusted that I actually
had a clue what I was doing on my first audio gigs.
To the pros who are willing enough to teach that they actually take my calls, including
Dave Shadoan, Big Mick Hughes, Paul Owen, John Cooper, Buford Jones, Tom Young,
Dirk Durham, Dave Rat, Bill Chrysler, Bob Heil, Brian Hendry, David Morgan, and
Mark Dennis.
To my ‘‘team,’’ most of whom I have had the pleasure of working with at GIG, FOH,
and L2P: Baker Lee, Steve LaCerra, Jamie Rio, David Farinella, and the late Mark
Amundson, who I still miss all the time. And to Terry Lowe, Bob Lindquist, and Paul
Gallo for paying me to do it.
To Mitch Gallagher, who recommended me to write this book in the first place.
To the people who started out as ‘‘audio acquaintances’’ and ended up among my closest friends in the world: Larry Hall, Kevin Hill, Paul Overson, and Ken ‘‘Pooch’’ Van
Druten.
For their invaluable help with the chapter on speaker components and enclosures, Mark
Gander from Harman/JBL Professional and Chris Rose from Eminence Loudspeakers.
For their direct input in the chapters on actually doing the gig, Mike Allison from
Clair/Bon Jovi, touring guy extraordinaire Mical ‘‘Mikey’’ Catarina, and the tribe at
ProAudioSpace.com.
Finally, to my wife of 22 years, Linda (who has gone as far as buying me gear she did not
understand and even joining the band when I needed a backup singer), and my daughter, Erin, who have both put up with stupid, crazy hours and a husband and dad who is
often physically in the same room but somewhere else entirely in his mind, and who
long ago stopped asking about the garage full of audio gear because they know the
answers are, ‘‘Yes, I need it all,’’ and, ‘‘It’s gathering dust because I need to fix it.’’
iv
Large-Music
Software
About the Author
Bill Evans has been working in music and audio since...well, let’s just say that it has
been a really long time. He was the editor of GIG magazine in the ’90s and has been the
editor of Front of House (fohonline.com) since 2002. Bill is also the Minister of Propaganda for the Live2Play Network (L2PNet.com) and leads the tribe at the social
network ProAudioSpace (proaudiospace.com).
In his ‘‘spare time,’’ Bill has fronted the band Rev. Bill and the Soul Believers
(revbill.com) since 1984. He likes Little Feat, scuba diving, William Gibson, and
lobbing libertarian political firebombs on Facebook.
Large-Music vSoftware
Contents
Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xi
PART I
THE GEAR
1
Chapter 1
What Is Sound?
3
Good Vibrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
How Loud Is Loud? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Another Kind of Doubling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Chapter 2
Welcome to the Signal Chain
7
End to End . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Power It Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Chapter 3
It All Starts with a Mic
11
What’s the Address? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
It’s All about Heart . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
So What Does It All Mean? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
A Place for Everything and Everything in Its Place . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
The Snare . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
The Hi-Hat . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
Tom-Toms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
Kick Drum . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
Cymbals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Keep It Simple . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Guitar and Bass. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Horns . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Keyboards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
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Chapter 4
Cables and Connectors
vii
21
Shielded versus Unshielded . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
But Wait, There’s More . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
Weird Stuff and Power . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
Chapter 5
The Wonderful World of Wireless
33
Getting Unplugged . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
A Little History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Compression/Expansion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Limited Power. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
And Now the Reality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
Dynamic or Condenser? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
What’s the Format?. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
Auditioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
Time to Taste the Freedom. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
R What? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
The Magic Is in the Motion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
Meat Absorbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
Chapter 6
Snakes and Splits
45
A Pirate Looks at 40 Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
Meanwhile, Back in the Desert… . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
Splitting Sound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Ones and Zeros. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
Chapter 7
It’s Not the Car, It’s the Driver
49
Hostility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
Incompetence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Education . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Chapter 8
The Channel Strip
53
Chapter 9
Console Auxiliary Sends . . . or, What Do the Knobs in the Middle Do?
57
Made to Order . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
Insert Here . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Remember Tapes?. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Now Boarding Group A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
I Wanna Go Home . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
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Chapter 10
Equalization
61
Rule #1: Listen First . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
EQ Bands and Types. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
How Do I Use ’Em? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Rule #2: ’Tis Better to Cut Than to Boost . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Chapter 11
Other Channel Stuff
65
Phantom Power. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Mute and Solo. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
Chapter 12
The Master Section
69
Aux Returns . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Pretty Lights . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
Other Miscellaneous Stuff . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
Splitting It Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
Chapter 13
Gain Structure
73
What Is It?. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
ABC’s of the Signal Chain. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
Audio by Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
The Microphone and Other Delicacies. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Chapter 14
Aux Sends and Returns
77
Hello? (hello . . . hello . . . ) Is There Anybody in There? (in there . . . in there . . . ) . . . . . . . . . . 77
Getting Swishy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
Clap On, Clap Off . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
Dynamics—Compressors and Gates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
When to Use It . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Someone’s at the Gate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Chapter 15
Monitoring
83
The Gear . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
Do You Hear What I Hear? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
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Chapter 16
The Drive Rack
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87
Order, Order, Order! . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
System EQ. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
Getting in Tune . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
HELLO, HELLo, HELlo, HEllo, Hello, hello . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Take It to the Limit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
Chapter 17
Active Speakers
97
It’s the Law . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
Back on Track. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Pros and Cons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Chapter 18
You Gotta Have Power . . .
103
Ins and Outs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
What’s Cooking Inside? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Class A . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Class B. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Class AB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Class D . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
What a Load . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
You Can’t Do That with an Amplifier . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Chapter 19
Loudspeakers
111
The Drivers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
Impedance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
Power Handling and Efficiency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
Box Design . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
Sealed Box. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
Bass Reflex . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
Horn-Loaded . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
Deployment. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
PART II
THE GIG
125
Chapter 20
Getting Your Hands Dirty
127
Kicked by the Wind, Robbed by the Sleet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
Had My Head Stove In . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
But I’m Still on My Feet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
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Chapter 21
Advance and Prep
131
Advancing the Gig . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
The Plot Thickens . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Chapter 22
On the Gig
141
Welcome to the Working Week . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Showtime—No Sleeping In for You . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
How We Roll . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
Getting Pinned . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
Chapter 23
Hello (Hello . . . Hello . . . Hello)—and Welcome to the
World of Delay
147
Chapter 24
Backline Basics
151
Behind the PA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
Getting the Gear Right . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
Chapter 25
Hands on the Knobs
155
Fix It at the Source . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
Find Your Foundation and Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
Triage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Less Is More—Softer Is Louder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Chapter 26
Touring Is Not for the Weak
159
My Day—by Mike Allison . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
Chapter 27
Just Because You Can Doesn’t Mean You Should
165
Dave Shirley . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Jordan Wolf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
Ian Silvia . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
Mike Reeves . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
D.V. Hakes II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
Jeanne Knotts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
Steve McCarthy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
Final Words. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
Index
173
Large-Music Software
Introduction
I still remember my first PA system. . . .
Actually, my first PA was a guitar amp. The gig was a fundraiser carnival for the Crippled Children’s Society, and the year was 1973. My younger brother had succumbed to
leukemia a few months earlier, and that organization had been instrumental in my
family’s financial survival. We took a mic from a reel-to-reel tape recorder, put it inside
my nylon-string guitar, and plugged it into one side of a two-channel guitar amp;
a RadioShack vocal mic went into the other. Add some drums and a trombone trying to
play bass lines, and we had a band.
Fast-forward a few years, and we bought our first real PA from a guy living in one of the
now-razed bungalows behind the Hollywood Bowl. He needed the money to fly to
England to see his guru. Really. You can’t make stuff like this up, and it was still the
’70s.
It was a custom-made 16-channel mono mixer with a separate monitor output. We got
the mixer in a road case, a 150-watt power amp, a pair of Altec horns, two CerwinVega folded-horn bass bins, and a couple of Shure Vocal Master columns that we
turned on their sides and used as monitors. We dubbed the console T.I.M.—totally
intense mixer. And at that time, we had the biggest, baddest PA on our little band
circuit.
Since that time I have been through pretty much every twist and turn and advance in the
world of live performance audio. I was the quintessential ‘‘guy in the band who owned
the PA,’’ and for many years I set up and ran the PA in addition to playing in the band.
Almost 20 years ago, I started renting out my system and mixing other bands. I learned
by watching, listening, and emulating those who were farther up the audio food chain.
The only training I knew about was of the on-the-job variety, and if there was a book
that explained it all, I’d never heard of it.
In the late ’80s—through a series of seemingly unrelated contacts and incidents—I
ended up working for someone who had been in a competing band on that old circuit on
a magazine for musicians called GIG. It was there that my education began on how to
explain what ‘‘all the knobs and buttons do’’ to audio novices. Later—via another
string of ‘‘coincidences’’—I took the helm of Front of House, a trade magazine focused
entirely on the live audio biz. Over the past almost eight years, I have had the pleasure of
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Live Sound Fundamentals
interviewing and learning from some of the top live audio pros in the world. I have had
the honor of meeting audio pioneers including Bill Hanley, Bob Heil, Stan Miller, and a
bunch of others I will regret leaving out after this gets printed.
And I have kept my hand in the more musician-oriented side of things through my
involvement with the Live2Play Network and writing a series of Live Sound 101
columns for our print and online publications. My hope is that this book will serve as an
extension of those articles and will give those who read it a solid grounding in the basics
of live performance audio. (I remember trying to put together written instruction for my
band mates on how to set up the PA for a gig I was going to be late for. When I got up to
11 pages, I started to wish they had all read a book like this.)
I have often said that I know too many really good sound guys to ever claim to be one.
I’m just the guy in the band with the PA who happened to get day gigs where he got to
help explain it all. But you never know where the little bit of knowledge you share will
end up.
A few years ago, I was backstage at a Toby Keith show talking with his longtime frontof-house engineer, Dirk Durham. Later that evening, Dirk would be driving a huge,
state-of-the-art sound system and bringing the sound to some 15,000 screaming fans.
I asked how he got into the audio biz, and he told me about being a rodeo guy who had
friends in bands who told him that if he was going to hang out, he needed to make
himself useful. So he started hauling gear and eventually setting up the PA and finally
mixing the band.
When I asked him how he learned how to set up and run the system, I expected him to
tell me about some guy he met who took him under his wing and showed him all the ins
and outs of audio. I was more than surprised when he said, ‘‘I learned everything I know
about sound from one place. You see, there used to be this magazine called GIG . . . .’’
If you learn anything from this book, be generous and pass it along. You never know
where it will end up.
Large-Music Software
PART
The Gear
Large-Music Software
I
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1
What Is Sound?
T
he anticipation in the room is palpable. The space is already filled with sound as
the conversations of thousands of fans converge into a throbbing, living hum.
Finally, the house lights go down, and the crowd explodes. On stage, all that is
visible are a number of flashlights illuminating the floor. The people in the crowd crane
their necks, stand on chairs, and jump up and down, hoping for a pre-show glimpse.
Finally, you hear the command, ‘‘Cue sound, cue lights.’’ You bring the main faders up
as the stage lights come on, the star of the day asks the crowd how they’re feeling, and
the band launches into its first tune as the crowd roars. And there you are at the console,
controlling it all.
Sounds pretty cool, doesn’t it? And you can get there, but you need to learn the basics
first. Starting at the very beginning: What is sound?
Good Vibrations
At its most basic, sound is vibration. An event occurs—anything from a guitar player
picking a note to the proverbial tree falling in the woods. The event itself is a disturbance
that excites the molecules in the surrounding air, which creates a wave that travels away
from the point of the original event. That wave is going to look something like what you
see in Figure 1.1. That explanation is a little on the dry side, so let’s try it again. Eric
Clapton peels off a blazing blues lick, which causes the molecules in the air to go into a
frenzy and start bashing into each other at different speeds. The speed makes the frequency
or pitch happen, and the force with which they bash into each other equals volume. Cool?
Figure 1.1 A sine wave.
The truth is that air does not have to be involved—the medium could be liquid or even
solid, although there has to be some kind of medium. Sound can’t travel in a vacuum.
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Live Sound Fundamentals
There was an old sci-fi movie called Alien that used the tagline, ‘‘In space no one can
hear you scream.’’ Cheesy but true . . .
Depending on the nature of the original disturbance, the amount of time it takes for that
wave to repeat itself can be measured in terms of distance, because the speed of sound is
constant. At this point it is easiest to look at these waves as a vibration. The speed of the
vibration—that is, the time it takes to complete a cycle as in Figure 1.1—determines
what we humans call pitch. The faster the vibration, the higher the pitch. We measure
the speed of these vibrations in cycles per second, or hertz (named for noted German
physicist Heinrich Hertz). Every musical note produced by any instrument can be described in terms of frequency or hertz (see Figure 1.2).
Figure 1.2 This chart shows how frequency translates to notes. Knowing this and being able to
combine it with the information in the chart in Chapter 10 that shows the frequency range of
various instruments will make your job of crafting a really coherent mix much easier.
An electric guitar with 24 frets can go as high as about 4800 hertz (Hz)—or 4.8 kHz—
and down to about 330 Hz. A bass goes down to about 80 Hz and up to about 200 Hz.
Just in case you are not into the whole metric thing, kilo means thousand. So 5,000 Hz
is 5 kHz.
A couple of things to note: First, these numbers are rounded off and are not exact.
Second, one of the best things you can do as a fledgling sound person is to memorize the
frequency range of the instruments you work with. If you have a tenor sax whose low
end is out of control, then knowing which frequencies to cut will make your job easier.
It is actually more complicated than that, because any acoustic sound source produces the primary frequency as well as a series of sympathetic vibrations at regular
intervals based on the frequency of the primary tone. These are called harmonics.
Large-Music Software
Chapter 1
What Is Sound?
5
One final note on the speed thing: Frequency has nothing to do with the speed of sound.
Sound moves at the same rate regardless of frequency. Things such as temperature and
humidity can affect the speed of sound, but the generally accepted number is 1,125 feet
(or 343 meters) per second in a dry environment at 68 degrees F.
If you were like me, you hated math in school, and your eyes may be glazing over right
about now. But this stuff is important. Understanding how sound is produced and how
it moves will play into everything from designing a system for a small club or church to
setting delay stacks in a stadium. Knowing frequencies will help you kill the feedback
that is screaming through the wedges and threatening your employment status. So
WAKE UP!
How Loud Is Loud?
Sound pressure level (SPL) is measured in terms of decibels, or dB. Why the capital B?
The decibel originated in the Bell Laboratories and was originally part of efforts to
quantify loss in audio levels in telephone circuits. The ‘‘bel’’ part is in honor of Alexander Graham Bell. (Don’t ask me why they dropped one L—it is one of those mysteries
of the universe.) ‘‘Deci’’ is the prefix that denotes 10 in measurement systems and refers
to the fact that the decibel system is a base-10 logarithmic scale. If you know what that
really means, then take a moment to pat yourself on the back for not sleeping through
math class. I personally am not doing any self-patting, but that does not mean I can’t
explain what decibels mean for us sound types.
The decibel system does not really measure absolute sound levels, but rather the difference between levels. In other words, we have all agreed that 0 dB is the point at which
the average human begins to perceive sound. Please note that is average human—not
average Tool fan, whose ears are likely half blown out and whose threshold of perception is going to be substantially higher than someone whose tastes run more along the
lines of Barry Manilow or Norah Jones. From that agreed-upon 0 dB, things get a little
less clear. From a straight physics perspective, a 3-dB increase is a doubling of power.
However, doubling the power does not mean doubling the perceived volume. Most
references will set that number at 6 dB, but many audio types maintain that they do not
perceive a doubling of volume at anything less than a 10-dB boost.
For our purposes, we’ll go with the 6-dB standard. When it comes to volume, it is always better to be conservative. Sustained exposure to volume levels above 90 to 95 dB
can result in permanent hearing loss. For someone who seeks to make a living with his
or her ears, this is a crucial concept. (One A-list mixer I know was attending a concert
with his wife, and after two songs he said to her, ‘‘My ears pay the mortgage. I need to
leave.’’ And they left.)
What you need to take away from all of this is the fact that 120 dB is not twice as loud as
60 dB—it is more than 1,000 times as loud. Do the math. If each increase of 6 dB is a
doubling of perceived volume, then the 60-dB difference between 60 and 120 dB
Large-Music Software
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Live Sound Fundamentals
represents 10 doublings of the original level, or 1,024 times the original. Even using the
more lenient 10-dB standard means six doublings, or 64 times as loud.
Just some points of comparison: Normal conversation (again, normal being not between two Tool fans) is about 65 dB. A gas-engine lawnmower at about 3 feet is in the
107-dB range. Your average loud concert can run in the 112- to 115-dB range, and 125
dB is the threshold of pain.
In musical terms, things don’t shake out the way you think they might. I mean, a flute is
quieter than a piano, right? A flute can range between 90 and about 103 dB, and a piano
at normal practice volume is 60 to 70 dB. Even played fortissimo (very loudly), it ranges
between 85 and 103 dB.
OSHA, the federal agency charged with setting safety standards in the workplace, allows for only half an hour a day of exposure to sound at 110 dB. That show you are
mixing is 90 minutes long. Think about that when you are watching the meter hit 112
during the first song.
Another Kind of Doubling
Traditionally, sound was seen as traveling in an ever-expanding sphere centered on the
original source of the sound. Designers of loudspeakers have made huge strides in controlling and aiming sound, as has the military. One got a weapon that can cause damage
by using nothing but very focused beams of sound, and the other got the line array. And
plenty of experienced sound techs will tell you that a line array is a weapon in the wrong
hands. But that all comes later. For now, we stick with the classic physics, which gives
us something called the Inverse Square Law. I am not going to get into the math, because all you really need to know is the result, which says that for every doubling of
distance from the source, the perceived volume drops by 6 dB. There is that 6-dB figure
again . . . .
Practically, what that means is that a sound measured at 90 dB at 1 foot from the source
(for our purposes, usually a loudspeaker) will measure 84 dB at a distance of 2 feet, 78
dB at 4 feet, 72 dB at 8 feet, and 66 dB at 16 feet.
So, if you are trying to get an SPL of 90 dB 40 feet away from the stage, then your level
at 1 foot would have to be more than 120 dB. This is important when it comes to
designing systems, both in terms of the amount of power you need to get the desired
volume at the desired distance and with regard to keeping the audience a sufficient
distance from the speakers to keep from hurting them (the audience, not the speakers—
drunk frat boys have been known to damage speakers from distances as great as
100 feet).
From this very short introductory chapter, you should now understand how sound is
created, how it travels, how loud is loud, and how loud is too loud.
Large-Music Software
2
W
Welcome to the Signal Chain
e have taken a basic look at the nature of sound, how it is created, and how
it moves. Now it is time to get down to the nuts and bolts of setting up and
running a live audio system.
The first thing to remember is that what you are doing is reinforcing the sound created
by whoever is onstage. Over the years, that job has gone from just getting the vocals
audible above the guitars to situations where every conceivable sound source has a mic
on it. But despite the seemingly huge change, the job really remains the same: Make sure
the audience can hear each voice and instrument clearly. In its journey from the source
to the ears of the audience, the sound (which we will often refer to as the signal) passes
through many stages, and we will look at each of these separately. The entire path is
called the signal chain.
End to End
Remember that sound is energy. There are a lot of different kinds of energy, but the kind
we can hear is called acoustic energy. The devices we use to control sound can’t work
with acoustic energy; they need to work with electrical energy. But, you guessed it; we
can’t hear electrical energy, so some transformations need to take place.
The devices that accomplish this are called transducers, and they convert one form of
energy to another form. Both microphones and loudspeakers are transducers. The mic
converts acoustic energy into an electrical signal. That signal travels through the chain
getting adjusted, massaged, and sometimes plain beat up until it reaches the speaker,
where that energy is converted from electrical back to acoustic.
Because these components make these crucial conversions, they are arguably the most
important pieces of the puzzle. On the mic end, we operate on the garbage-in-garbageout principle (and no, I am not referring to the talent of the person or instrument feeding
the mic). We are referring to the fact that if that initial conversion of energy is poorly
done, then you have very little chance of saving it. The cliché of ‘‘fixing it in the mix’’
is—in the case of live audio—usually a lie.
Conversely, no matter how great and pure a signal you present to the loudspeaker, if the
speaker is cheap, there is no way you can get anything out of it that sounds pleasant.
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Live Sound Fundamentals
In an ideal world, the mics and speakers would put out exactly what is put in. In the real
world, that just does not happen. While current technology makes signal ‘‘transparency’’ more achievable than in the past, virtually every stage a signal passes through
affects the tonal quality, timbre, or color of the sound for better or worse. One thing
that separates really good engineers from their less notable counterparts is an almost
encyclopedic knowledge of the gear—not only how each piece affects the sound, but
more importantly, how combinations work together. That knowledge—and the ability
to use their ears—is what allows them to make their act sound good on any system in
any environment.
No Excuses Ken Van Druten—known as Pooch throughout the industry—has a list
of clients that most engineers drool over. He specializes in harder, heavier acts and is
one of the best at what he does. The first two times I heard Pooch mix were at
stadium gigs with huge systems that he was familiar with, and there were no real
issues with acoustics. But the third time was in a Las Vegas nightclub called Rain. The
act was Kid Rock, and the room, while beautiful, was an acoustic nightmare. Round.
All glass, chrome, and rock with a very high ceiling. Add to it the fact that he was
mixing on a rig that was totally new to him. I am not a Kid Rock fan, but the mix
absolutely rocked. I was with a sound-guy friend who is a very good mixer and
knows it. He is also not shy about saying when another engineer is not cutting it.
When he says, ‘‘It was one of the best mixes I have ever heard,’’ it means something.
Pooch is on the A List for a reason. He goes into any venue on any system with any
artist and makes it sound as good as it possibly can. If you are going to be in this
business, that’s your job.
After the acoustic energy (original sound) is transduced or converted into electrical energy, it travels via either wireless transmission or a cable of some kind to a stage box,
also known as a subsnake. This is a box with audio connectors on one end and a bundle
of cables coming out of the other end. Sometimes this stage box converts the signal from
analog to digital for transport over fiber or Cat-5 (computer networking) cable. Other
times, it is split into two or three identical pieces, in which case this box is generally
referred to as a split.
The console is going to be your main tool if you end as a mix engineer, but don’t be
fooled into thinking that this is the most important part of the system. It is important,
but it lies in the middle of the chain. Remember that your most crucial components are
those that do more than adjust the signal—they actually convert it.
The console consists of several parts or sections. First is the input, which is where
the very small signal from the mic is goosed up to something the console can use. The
piece that does the goosing is a preamplifier, also known as a mic pre, head amp, or even
just pre.
Large-Music Software
Chapter 2
Welcome to the Signal Chain
9
The preamp both feeds and is part of the individual channel strips. These include tools
to adjust the signal’s relative volume plus its sonic characteristics. Changing the tonal
quality of a signal is known as equalization or EQ. Depending on the ‘‘level’’ of the
console, it may have anywhere from two to four or more bands of EQ.
These bands of EQ can be as simple as the bass and treble controls on your home stereo
or as complex as three controls for a band that allow you to determine the center frequency, the width of the area surrounding that center frequency that is affected, and
then an amount of boost or cut for the band.
The next set of controls in a channel strip is the auxiliary, or aux, sends. If you think of
the channel strip as a kind of robotic assembly line, the sends are where a signal—first
determined to need extra ‘‘work’’ not available on the console—is sent off to be worked
on before rejoining the assembly line. On a high-end gig, these sends on the main console are pretty much used to send a portion of the signal to some kind of effect—reverb,
delay, chorus, compressor—and then return it to the channel. If you have enough input
channels on your console, you can return the effect, which allows for greater control,
but otherwise you will use one of the returns, which we will get to later. Also, aux sends
come in two flavors—pre-fader and post-fader.
Better consoles will also have subgroups or VCAs (Voltage Controlled Amplifiers),
which allow you to assign groups of inputs and control them all from a single fader. For
instance, you may have all of your drums, backing vocals, or a horn section on a sub or
VCA. You still have the control to tweak an individual instrument in the group, but you
can also bring the entire group up or down without having to muck around with seven
or eight faders.
Following this is the master section, which includes the master fader as well as aux
returns, recording outs, playback inputs, and your talkback mic for communication
with the band onstage without screaming yourself raw.
What comes after the console depends a great deal on the size and type of system. If you
are using powered, or active, speakers, you may very well just go from the console
outputs directly to the speaker inputs with perhaps some kind of processor specific to
those powered speakers in between. If you are using a more standard ‘‘passive’’ system,
then your next stop is the drive rack.
These days, the big, heavy drive racks are disappearing fast. We used to need graphic
EQs, compressors, and maybe a delay unit. Given standard sizes of gear, this could
easily be a 12-space rack. Today, via the magic of digital signal processing (DSP), we
can accomplish all of that and more in one or two rack spaces. Because dbx put out an
actual line of products called DriveRack, we have renamed these devices speaker processors, and nearly everyone who makes any kind of EQ or crossover makes a speaker
processor, including BSS, Carvin, dbx, Sabine, Yamaha, and probably a half-dozen that
I am forgetting about right now.
Large-Music Software
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Live Sound Fundamentals
A moment ago, we briefly addressed the idea of powered, or ‘‘active,’’ speakers. Usually
processes including crossover and time alignment (we’ll get there—patience, Grasshopper . . . ) are handled by the circuitry inside a powered speaker, but not always. For
example, Meyer has designed and built their own processor specifically and only for
Meyer powered speakers.
Power It Up
Once the signal has been ‘‘processed’’ (in other words, split into separate low-, mid-,
and high-frequency signals and each of those optimized), it must be amplified. Up until
this point, the signal that represents the sound is very weak and needs to be boosted
heavily in order to move the speaker cone and again transduce the electrical signal back
into acoustic energy. This process is done by the power amp. The signal goes into the
amp and comes out much stronger but it is still an electrical signal—no sound yet.
Which is where speakers come in—the end of the chain—and that electrical signal gets
goosed up to a point where it can drive a magnetic ‘‘motor’’ in the speaker assembly.
The motor is attached to the voice coil, which is attached to a paper cone or metal
diaphragm. Moving the cone or diaphragm creates an event that excites the surrounding air and—voilà!—we have sound again.
Now, armed with a basic knowledge of the signal chain and which part serves which
function, we can start making some noise—and getting into each part in greater detail.
Large-Music Software
3
It All Starts with a Mic
A
s we agreed earlier, the format of this book follows the signal chain from
beginning to end. And the first thing we need to do is to convert the acoustic
energy of the original sound into electrical energy that the system can use. For
that task we use a microphone. While a mic is not the entire system, it just may be the
most important part.
Years ago, I was having dinner with a rep from Shure—the biggest maker of mics in
the world. He told us about a call they got at customer service from a woman who
said their product was defective because she took it out of the box and sang into it
and nothing happened.
When it comes to mics for live sound, there is a plethora of choices, and one of the things
that separates experienced sound engineers from newbies is the ability to choose the
right mic for the job, be that adding thump to a kick drum or getting the choir loud
enough to fill the room.
Mics come in a lot of different flavors, but for our purposes, we will limit things to dynamics and condensers. (‘‘Yes!’’ he said, with a nod to his ribbon-worshipping friends,
‘‘There are some folks using ribbons’’—including rockers such as Aerosmith’s Joe Perry,
who uses Royer ribbons on his guitar amps—and they sound great. On the downside,
they are expensive and fragile—which makes them a big risk when used for the stage.)
But this is not an article about mics, so here are the bare basics. A dynamic mic uses a
magnet and a diaphragm. The movement of the air caused by the original sound moves
the diaphragm, and the movement of the diaphragm causes changes in the magnetic field
between it and the magnet. This changing field creates a varying, low-voltage signal.
A condenser microphone is similar except it uses a charged plate instead of a magnet. As
a result, the mic needs power to work (typically 48 volts). That power can come from a
battery or an external power supply, but most often it comes from the mixing console.
This is called phantom power, and buying a mixer without this feature is shortsighted.
When it comes to sound, general thinking says a dynamic is more roadworthy but less
detailed, especially in the high end. A condenser is more fragile but puts out a more
‘‘detailed’’ signal. Most dynamic mics also exhibit a trait called the proximity effect,
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Live Sound Fundamentals
which causes the low frequencies to be emphasized as the sound source gets closer to the
mic. This is part of the reason why sound guys bitch about ‘‘mic eaters,’’ although being
too far from the mic is just as bad (not enough energy getting to the transducer), and
some artists use the proximity effect as part of their sound. Condensers also exhibit the
proximity effect, but generally to a lesser degree.
The preceding is a pretty gross generalization, and a lot has changed in mic technology
in the past few years. Condensers have gotten downright tough in comparison to what
they used to be. In my roles as editor of both FOH and the Live2Play Network, I insist
that every mic we review go through a drop test—at least five feet, capsule down onto a
hard surface, such as concrete. We have yet to have a mic fail the test. We have dented
quite a few, but they always work when we plug them in after dropping them.
Before we move forward, I see that I just used a term that we did not explain. The
capsule of the mic is the part that captures the sound. It can be ball-shaped, paddleshaped, or even capsule-shaped, depending on the intended use. The rest of the mic is
called the body. On a handheld or vocal mic, it is a cylinder that easily fits in a singer’s
hand. The body of a mic made for a kick drum may look like a continuation of the
capsule and give the whole thing a kind of oval shape.
A note about vocal mics: The end is usually ball-shaped, although that ball may have a
flat end. Under the usually steel mesh screen is some foam. It can be an integral part of
the screen or a separate piece, depending on the mic. This is called a wind screen and
serves mostly to dampen plosives—heavy sounds emanating from the singer that move a
lot of air and could damage the diaphragm. If you are really looking to dampen the
effects of actual wind, there are big hollow foam balls with a hole in one end that fit over
the top of the mic for that purpose.
What’s the Address?
I once did a gig for a church play that included an actor paying the part of a radio
announcer. We used a condenser mic mostly because it looked right. We would set it up,
and every time he sat down to use it, he would reposition it so that he was speaking into
the top of the mic, because everyone knows that’s how you use a mic—right?
Well, not really. It depends on the position of the diaphragm. The two arrangements
are generally referred to as top- or front-address and side-address. The mic in our
example was a side address, so the actor was making it very unlikely that anyone
would hear him because the diaphragm was facing the desk and not his mouth. (It
also caused us feedback problems.) The majority of the mics you will use in live sound
will be top/front-address but not always. And you need to know before you start setting
them up.
A couple of years ago, I was doing a gig attached to a pro audio tradeshow sponsored by
a speaker manufacturer. It was a weird gig at a club across from the convention center
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with two bands—my 10-piece soul review and a very good Ozzy Osbourne tribute. Oh,
and it was a really loud gig. (Remember, it was put on by a speaker company to show off
their system, which generally means cranking it up pretty hard.)
At sound check we were having feedback problems, and the assumption was that it was
a monitor issue. (My band is usually in-ear, but this gig was all wedges.) But the sound
company owner—who was trying to stay out of it and let his staff take care of things—
knew I was using a condenser mic known for being very hot and pretty wide in its
coverage pattern. The main P.A. was a line array also known for a wide pattern, and
after about 20 minutes of trying to find the feedback in the system, he walked onstage,
unplugged my mic, and replaced it with a very narrow dynamic mic—and the squealing
magically disappeared.
It’s All about Heart
When you are in the studio, you will find mics with many different pickup patterns,
including figure-8 and omni. (The former picks up sound equally from the front and rear
and rejects sound from the sides, while the latter picks up equally from every direction.)
But onstage—except for specialized applications such as choirs and some orchestral
uses—you will find almost 100 percent of mics to be of the unidirectional type. You
would think that unidirectional (meaning one or a single direction) mics would pick up
sound from only one direction. But it is not that simple. What you actually get are several flavors of cardioid, or heart-shaped, pickup. The basic cardioid pattern looks
something like you see in Figure 3.1.
Figure 3.1 A cardioid pickup pattern. Image courtesy of Shure Inc.
As you can see, at zero degrees (or straight on), you get the full response of the mic, and
it gradually falls off and dips to a theoretical level of zero at 180 degrees. The idea is to
get the sound you want into the mic and reject the stuff around it. But it only works so
well. Look at that plot again and notice that at 60 degrees off axis, the mic is still picking
up 75 percent of what it does from the front. And for a very long time, this was the
norm.
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But new technology—especially new materials for the magnetic structure of the mic—
allowed for a tighter pattern known as super-cardioid. Figure 3.2 shows what it
looks like.
Figure 3.2 Super-cardioid pickup pattern. Image courtesy of Shure Inc.
As you can see, the response of this mic falls off a lot faster as you move off center. But
nothing comes free. Look at the bottom of the plot, and you will see that at 180 degrees, the
response is actually much stronger than the standard cardioid. So if you are using standard
wedges for monitors, they need to be placed at an angle and not facing the performer
straight on. An even tighter pattern known as hyper-cardioid is also available. It is tighter
off-axis but has an even larger lobe at the back of the mic. With a super- or hyper-cardioid,
straight monitors mean more feedback—exactly the opposite of a standard cardioid.
Figure 3.3 Three examples of monitor placement. The circles represent the mics, and the
squares represent the monitors. In the first, a standard cardioid mic is being used, and the
monitor is placed directly to the rear of the mic. In the middle example, a super-cardioid mic is
being used, and the monitor is placed just off center in order to avoid the lobe at the rear of the
mic’s coverage pattern. In the third example, a hyper-cardioid mic is being used, and the monitor
is placed even more off center to avoid the hyper’s larger rear lobe. Remember, it is all about
putting the monitor where the mic is least likely to ‘‘hear’’ it.
Although they are rarely used in a live setting (with notable exceptions of Danny Leake
with Stevie Wonder’s percussionists, for example), Figure 3.4 shows an example of an
omni-directional mic. Notice that it picks up from any point around the mic.
So What Does It All Mean?
A couple of things. First, vocal mics come in two basic flavors—dynamic and condenser. What you need to know on a practical level is that condensers are generally
thought to sound more open and airy than dynamics, and they generally provide a more
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Figure 3.4 Omni-directional pickup pattern. Image courtesy of Shure Inc.
detailed sound. But for a long time, they were not suitable for live use for two reasons.
First, they were fragile. Drop one, and it would likely not work afterward. Second, they
have a wide response area—at least as wide as a typical cardioid dynamic.
But two things have changed that have made condenser mics pretty common, especially
among lead vocalists. First, they have become a lot more roadworthy, and second, the
move toward in-ear or ‘‘personal’’ monitoring has greatly lessened the possibility of
feedback from a mic with a wide pattern. (Note that this gets more complicated, as
some very smart people are doing actual new development in the mic field. Specifically,
live-sound legend Bob Heil has released a line of dynamics that sound—by all reports—
at least as good as most condensers and have a much tighter pattern.)
So what is the bottom line? It really depends on stage volume. On a loud stage, you need
a tight pattern, and that generally means a dynamic. (One sound guy I know who mixes a
very big Nashville act calls one of the standard industry condenser vocal mics the moving
drum mic because it picks up so much drum sound in the vocal channel.) If you have a
quiet stage or personal monitoring, you may be able to enjoy the generally higher-quality
sound and greater detail of a condenser. What do I use? My own band is on personal
monitors, so lead vocalists get condensers (a mix of Shure, Audix, Audio-Technica, and
AKG, depending on the gig), but my mic locker contains plenty of very tight dynamics as
well, for those gigs where a condenser is too wide. As always, when deciding what to
buy, it comes down to the eternal question: ‘‘What are you going to use it for?’’
A Place for Everything and Everything in Its Place
Where you place the mic in relation to the source of the sound has a huge impact on the
final result. Distance from the source and angle (on-axis or off-axis) are the two things
you are most concerned with. We will go over some very basic principles, but the truth
is that this is a place where experimentation and experience rule the day. The ‘‘old
dude’’ on the crew may not know the minutiae of every digital console out there, but he
sure knows how to mic a kick drum so that it feels like it is kicking you in the chest
without sounding muddy. Watch. Listen. Learn.
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The Snare
Before we get into actually miking drums, make sure this truth is drilled into your head.
Nothing will make more of a difference in your drum sound than a good, well-tuned
drum kit. If you are working with a crappy kit that is poorly tuned with worn heads,
your chances of getting a good sound are about nil. At that point it becomes a case of
minimizing the possible damage to the overall sound.
The most popular type of mic for the snare is a cardioid dynamic with a presence peak,
but many engineers prefer the transient response of a condenser. For years, the go-to
drum mic for snares and toms was a Shure SM57, but pretty much every mic manufacturer makes a mic appropriate for snare. Again, experiment.
As far as placement goes, there are many good systems out there that allow you to
attach the mic directly to the drum without using a stand, which makes for a much less
cluttered look onstage. Start by placing the mic about one inch in and one or two inches
above the head, with it angled toward the spot where the drummer tends to hit and far
enough away from the hi-hat to avoid picking up the rush of air that happens when the
hat closes. Different drummers will require different placement. Although it used to be
just a studio thing, it is not uncommon to see an act with plenty of money and lots of
input channels on the console mike both the top and bottom heads of the snare drum,
with the microphones in opposite polarity. A mic under the snare drum gets the metallic
edge of the actual snares, which pairs nicely with the fuller sound of the mic on top
of the drum.
The Hi-Hat
In situations where channels or the number of mics available are limited, it is not uncommon to forego a hi-hat mic and depend on the snare mic and cymbal overheads to
take care of it. Situations where you do use a separate hat mic call for a condenser placed
about six inches above and pointing down. (Make sure to place it where the drummer is
not going to actually hit the mic.) Some engineers prefer to place them from the bottom
and flip the polarity. (We’ll get into polarity when we talk about channel strips.)
Tom-Toms
When miking the toms individually, the type and placement are very similar to
the snare, with the mic perhaps a bit closer to cut down on leakage. (We will get
into gating drum mics when we get to the processing part of the signal chain.) Again, if
you are working a club or another situation with a limited channel count, you can
‘‘cheat’’ by placing a single mic between pairs of toms. This means having the mic
farther from the drums so that both are picked up, which can negate the proximity
effect that many engineers use to achieve a fuller sound. Mini condenser mics are
also becoming more popular on toms. Besides the sound itself, you need to consider
leakage from cymbals when placing tom mics, aiming the deadest part of the pattern
toward the cymbals.
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Kick Drum
Let’s get one thing out of the way right off the bat—the kick drum is not the lead vocal.
Yes, it is an important part of the foundation of a good overall sound. But I have seen far
too many mixers (both experienced and not) spend more than half of their sound-check
time fiddling with placement and EQ on the kick drum. Remember, 90 percent of the
audience is there to hear the singer sing the songs. If the vocal sounds great, you are
halfway home.
A popular mic for kick drum is a large-diameter, cardioid dynamic type with an extended low-frequency response. But wait, here is another case where lots of channels
and lots of available mics open up options, and two mics on the kick is a popular option.
The idea is to pair a dynamic with a condenser, with the dynamic picking up the thump
of the beater and the condenser picking up the tone of the shell. Great idea, but making
it work can be complicated. Most smart engineers who use two mics have rigged some
bar and clamp system where the drum mics live inside the kick drum so they don’t have
to worry about finding the proper placement. Anytime you use two mics on a single
source, you take the chance of the mics being out of phase and certain frequencies canceling each other out. Remember the picture of the sine wave with its peaks and valleys?
Imagine two waves where one was at the peak at the same time the other was at the
valley, as you see in Figure 3.5.
Figure 3.5 Two sine waves on the same frequency but out of phase. See how one wave is at the
top of its path while the other is at the bottom? When this happens, the two waves cancel each
other out. Illustration by Erin Evans.
Very small changes in placement can affect that phase relationship and drastically affect
the sound, hence the ‘‘get it right and leave it there’’ approach. Another way to get the
two-mic result without the time and hassle of getting the relative placement together is
to use the Audio-Technica AE2500. This gem actually contains both a dynamic and a
condenser element in the same housing with separate outputs for each, giving you all of
the sonic advantage of two mics with none of the phase issues.
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Cymbals
A pair of condenser mics with fairly open patterns on booms above the kit is the most
common setup. But there are other approaches. Jeff Rasmussen, who at the time of this
writing had been with Michael McDonald for well over a decade, has been known to
use a pair of large-diaphragm condensers mounted in front of the kit. And, again, when
channel count and mic availability are no object, you can get really involved. Big Mick
Hughes, who has been mixing Metallica since they were a ‘‘baby band,’’ uses what he
calls underheads with a separate condenser mic for each cymbal. The combination of
lower gain on each mic and proper gating makes for a much cleaner drum sound on
what can be a very loud stage.
In addition to miking cymbals with the ‘‘underhead’’ approach, you can also use X-Y,
overhead, and spaced-pair miking techniques. The X-Y technique uses two matched
microphones. Certain microphones, such as small- and large-diaphragm condensers, are
available in matched pairs. Matched pairs have consecutive serial numbers mainly so
there are minimal, if any, sonic differences between them. Some even come with charts
to show their responses. These matched pairs of microphones are placed next to each
other with the diaphragms facing 90 degrees from one another, with the center between
the two diaphragms facing the source. With the mics facing this way, one mic is panned
left and one right, creating the stereo image. There are also a couple ‘‘stereo’’ mics that
have this technique built in, such as the Shure VP88.
Next is the overhead, also known as the spaced-pair, technique. This is very similar to
the underhead technique, but the mics are placed over top of the cymbals instead of
underneath.
One thing to consider when putting mics in place, especially for cymbals, is the 3-to-1 rule.
The 3-to-1 rule is as follows: When using multiple microphones, the distance between
microphones should be at least three times the distance from each microphone to its
intended source. If the mics are placed too closely to one another, phasing (the unpleasant
kind) will occur, and when listening to both mics together, it will almost sound like the
sound is in a small tunnel. An easy way to try this out is to take two mics of the same
kind—say, an SM58—and turn them both up equally and put one in each hand. Talk into
one mic and then start to bring the second mic closer to the first one.
On a side note, the 3-to-1 rule does not apply to X-Y miking techniques. The X-Y technique will create phasing, but that is part of what makes the unique ‘‘stereo’’ image as well.
Keep It Simple
On quieter gigs—especially acoustic and jazz groups—you can get away with a very
minimal approach. I have seen major jazz artists mixed with just a pair of overheads or
perhaps two overheads and a mic on the kick. On a rock gig, I once used a kick mic and
a PZM (or pressure zone) or boundary mic (typically used as floor mics in stage productions and not in concert settings) mounted to a two-foot-square piece of Plexiglass
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and hung above and behind the drummer on a boom stand. It worked great, and I only
used two inputs.
Guitar and Bass
The rule of thumb is to put the mic where it sounds best. With more open space around a
speaker cabinet than in a drum kit, your options are more open. A couple of tips . . .
Although an off-axis placement can cut the proximity effect and allow the guitar to sit in
the mix better, few things make me crazier than the guitar player who shows up to the
gig with his own mic and simply hangs it over the top of the amp so the body of the mic is
parallel to the speaker. This means that the amount of energy hitting the diaphragm dead
on is somewhere between zero and none, as it all moves across the diaphragm instead of
into it. Often you will find this in club situations where the band is providing their own
gear, and it is generally just laziness. (It means not having to carry an extra boom stand
for the amp.) Always keep a couple of spares handy for this kind of situation.
With more and more guitarists using modeling amps (which are basically big, heavy
computers), there have been a greater number of players running direct. (A direct output
is an XLR connection on the amp itself that goes directly to the PA without a mic. These
are a fairly new development with guitar amps. Bass amps have had them as a standard
feature forever.) Although a direct input theoretically captures all of the tone being
produced, they can sound thin. The speaker itself is a major component of the overall
sound. If you have a situation where the guitar is routed directly to the PA mixer along
with the amp being miked, the main concern is phase. If the sound seems hollow, traditionally you have had two choices. One, move the microphone around until the sound
is more solid. Or two, engage the phase reverse or polarity button on your mixer. But
recently, Radial has come out with a device that takes both the mic and the direct (or two
mics in the case of something like a snare drum) and, instead of the all-or-nothing
approach of a phase reverse on one input, you can ‘‘dial in’’ the phase until you find that
sweet spot. This is a great tool.
Horns
There are myriad options here, and rest assured that almost every mic out there has been
tried. A few guidelines: A horn mic has to be able to take high SPL without distorting.
The level of sound coming out of a trombone or sax can be as much as a kick drum.
Personally, I am a fan of the clip-on horn mics that nearly every manufacturer makes
that are purpose-built for miking horns. They solve two problems. First, you can be
relatively sure you are using an appropriate tool, and second, you don’t have to worry
about the player’s mic technique. If the player moves, the mics moves, too.
Keyboards
Most keyboards will run direct into the PA via either a submixer or a direct box. The
major exception is the acoustic piano, and miking an acoustic piano is an art unto itself.
For tips, read the trade magazines and look for interviews with mixers such as Tom
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Live Sound Fundamentals
Young (Tony Bennett) and David Morgan (Bette Midler), who have to mic acoustic
pianos for every show. There are also systems including those by Helpinstill and
Earthworks that are specifically designed for pianos.
The other big exception is the ‘‘real’’ organ (not a synth) with a Leslie rotating speaker
cabinet. These are usually miked using three inputs—one on either side of the rotating horn
and another lower one in front of the rotating drum in front of the low-frequency driver.
But back to the direct box thing . . . No matter what level you are working at, you should
always have several direct boxes in your workbox. These small boxes can be passive or
powered, and they convert the instrument-level signal of the keyboard (or acoustic guitar or violin with a pickup, for that matter) that comes out of the instrument on an
unbalanced 1/4-inch connection into a line- or mic-level signal on an XLR that can be
jacked into the PA. It is all about cable length. An instrument puts out a high-impedance
signal, which can travel maybe 20 to 30 feet before you start losing signal. A mic signal is
low impedance and can go more than 300 feet without noticeable signal loss. With the
increasing use of computers and iPods playing tracks to augment the actual sound
coming off the stage, there are now specialized direct boxes like you see in Figure 3.6.
Figure 3.6 The Rapco LTI 100 is one of several purpose-built direct interfaces. This one is for
getting a computer, iPod, or other device with an 1/8-inch stereo output into the PA on two
balanced XLR connections. Image courtesy of RapcoHorizon.
Now we have all of our sound sources set up, and it is time to get them into the PA,
which leads us to the next link in the chain—cables, snakes, splits, and wireless.
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S
Cables and Connectors
o, you have your original sound or acoustic energy, which you have converted to
electrical energy via a transducer (also known as a mic). The next step is to
transport that signal to a place where it can be manipulated.
I am sure this sounds so simple that some of you are wondering why we are devoting a
whole chapter to it. The truth is that not only are there myriad cable and connector
types, and you need to know which one is right for which job, but also, when a system
fails, it is almost always a cable or connector that is the culprit.
I used to work with a guy who played trumpet in my band but who was also a killer
guitarist. He used to say that whenever there was a guitar problem, it was always the
cable or the B string. Truth is, he was right more often than not, but his joke points out a
stark fact—cables count. Cables are pretty much defined by construction and connectors. The combination of those things dictates what the cable is used for. Let’s start
with the cable part of it.
Shielded versus Unshielded
Let’s start by taking a look at the simplest cable—your basic unshielded speaker cable
(see Figure 4.1).
Figure 4.1 Standard speaker cable. Image courtesy of RapcoHorizon.
As you can see, what we have here are two insulated wires, or conductors, inside of
an insulated sheath. The sheath is usually rubber or a rubberized plastic, but some
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Live Sound Fundamentals
high-end cable makers have started to use fabric. Rest assured that if the covering is
some kind of fabric, the cable costs twice as much as anything standard.
The reason these are known as unshielded is that they consist of two conductors in a
sheath. Simple. A shielded cable also has two conductors inside a sheath, but the ground
conductor is not insulated and is wrapped or woven around the ‘‘hot’’ conductor.
Sometimes the shield is made of foil. The reason for the shielding is that a long cable is
basically a big antenna. When the signal on the hot conductor is small (instrument, mic,
or line level), outside interference from a number of sources, including radio waves or
any kind of magnetic field, can overwhelm the hot signal, resulting in noise, dropouts,
and even having the local radio station being broadcast through a guitar amp or PA.
The reason unshielded cable is often known as a ‘‘speaker’’ cable is because the strength
of the signal traveling between the power amp and the speaker cabinet is such that those
kinds of extraneous signals are unlikely to cause any interference. You may hear that
you should never use shielded cable to carry signal between an amp and a speaker. The
truth is that in a pinch you can get away with it, but assuming you are using a balanced
cable with two conductors inside the shielding, it is possible (though unlikely) that the
two conductors could make contact with the shield, shorting the connection and
blowing your amp.
In the case of a cable with a single conductor, the issue is one of cable size or gauge. The
inner conductor on a guitar-type shielded cable is very thin—probably 22 to 24 gauge. It
carries a signal as strong as what goes between an amp and a speaker—especially over
long distances. When a light cable is used to carry a large signal, it is going to heat up. It
is unable to carry that much signal, and the energy it cannot carry is converted to heat.
(Remember our talk about transducers and energy conversion? Energy cannot be created nor destroyed. However, it can be converted from one form to another. In this case,
the excess electrical energy is converted and dissipates as heat.) The bottom line is that
some of the energy being produced by your amp is being wasted, and if the cable heats
up enough, the insulation could melt, and then we’re back to the shorting-out-the-amp
thing.
The two most common connectors are the 1/4-inch (which comes in two flavors) and
the XLR (which comes in two ‘‘genders’’).
A quick side note: I am really trying to keep this simple, but it is more complex than it
appears on the surface. Just remember that this is an area where you really get what
you pay for. When it comes to premade cables (yes, a lot of us ‘‘roll our own’’),
names such as Whirlwind, Monster, Planet Waves, Link, Rapco, or Horizon are
always safe. Just stay away from molded-on connectors and look for some kind
of strain relief at a minimum. If you are not sure what these terms mean, ask before
you buy.
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A 1/4-inch connector is what many people refer to as a ‘‘guitar’’ cable, and it looks like
what you see in Figure 4.2.
Figure 4.2 A typical guitar cable shown in both straight and 90 angled versions. Image courtesy
of RapcoHorizon.
Figure 4.3 shows another variation.
Figure 4.3 Balanced tip-ring-sleeve (TRS) cable. Image courtesy of RapcoHorizon.
While the two look very similar, there is an important difference. Look at the shaft of
the first example, and you will note a single line separating it into two parts, whereas in
the second there are two separators and three parts. These separating lines are insulators, and each part of the shaft corresponds to a different conductor in the cable.
Sometimes these are called mono and stereo, but more accurate is TS (tip-sleeve)
and TRS (tip-ring-sleeve), or unbalanced and balanced. Mic XLR connectors are
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Live Sound Fundamentals
three-pin and balanced, with the difference being male (see Figure 4.4) and female (see
Figure 4.5).
Figure 4.4 Male three-pin XLR connector. Image courtesy of Neutrik.
Figure 4.5 Female three-pin XLR connector. Image courtesy of Neutrik.
A two-conductor or tip-sleeve or unbalanced connection is easy. One wire carries
the signal, and the other is ground. (If you don’t know what those two terms mean, it’s
time to bone up on basic electrical knowledge, which is not what we are doing here.)
The ‘‘hot’’ or signal wire attaches to the tip and the ground to the sleeve. Simple. So why
three conductors and a balanced connection? Let’s ask Wikipedia.
Balanced audio connections allow for the use of very long cables with reduced introduction of outside noise. A balanced audio connection has three wires. Two of
these are used for the signal, of opposite polarity to the other. The third wire is a
ground and is used to shield the other two. The signal is the difference between the
two signal lines. Much of the noise induced in the cable is induced equally in both
signal lines, so this noise can be easily rejected by using a differential amplifier or a
balun at the input.
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The separate shield of a balanced audio connection also yields a noise rejection
advantage over a typical two-conductor arrangement such as used on domestic hi-fi,
where the shield is actually one of the two signal wires and is not really a shield at all,
but relies on its low, but in practice not zero, impedance to signal ground. Any noise
currents induced into a balanced audio shield will not therefore be directly modulated onto the signal, whereas in a two-conductor system they will be.
Okay, in non-geek this means that the noise in an unbalanced cable is canceled out in a
balanced cable. The truth is that I would use a balanced cable anytime I had a choice.
Every cable in a rack, every input into a mixer, and every line from a mixer to a
processor or amp should be balanced whenever possible. And choose an XLR over a
1/4-inch whenever you can, just because they tend to be more durable.
But Wait, There’s More
Wouldn’t it be nice if there were only two connector types to deal with? But no such
luck. While 1/4-inch TRS and unbalanced and XLR are the ones you will use most
often, there are others that are very important.
First up is the RCA or phono connector (see Figure 4.6). Yes, this is the kind of connector you have seen on the back of your home stereo. On a real pro system, you will
rarely see an RCA connection, but for club and smaller band gigs, churches, and even
small theatres, you will often see a smaller mixer with RCA connectors that say Tape In
and Tape Out that are used to connect consumer-grade CD players, tape decks, and so
on to the board. (There are a few out there already, and by the time you read this, expect
to see more consoles with iPod dock connections right on the face of the mixing surface.)
Figure 4.6 An RCA connector, also known as a phono connector. Image courtesy of Neutrik.
Oh wait, did I say the lowly RCA was not a ‘‘pro’’ connection? Silly me. That was the
case for a long time, but certain digital gear—including smaller and early digital consoles—may include RCA connectors for something called S/PDIF (Sony/Philips Digital
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Interface format). This is a digital format that sends two channels on one cable and
receives two on another. You will also see XLR connectors meant to carry digital signals in what is called the AES/EBU format. The big difference between these and analog
connections using these connectors is the quality of cable needed. Don’t even think
about using a standard AV RCA cable or a standard mic cable; you need to use cable
that is approved for digital signals, which usually means a lower-impedance, higherquality cable.
What do you use them for? Most S/PDIF and AES/EBU connectors are used to transfer
digital signals from something such as a DVD player or computer to the console, keeping it in the digital domain as long as possible. In the case of AES/EBU, it is also found at
the output of consoles and inputs of speaker processors and amps. Again, the idea is to
keep the digital signal digital for as long as possible. Another digital connection you will
sometimes see in pro gear is the ADAT or Lightpipe connector. If you are at Best Buy
and looking for such a cable for your home theatre system (which also uses this connection), it will almost surely be called a Toslink. But there is not an audio pro alive who
does not know about the Alesis ADAT digital recorder, and it was the first piece of gear
that I know about that used this light-over-glass-fiber optical connection to send up to
eight channels of audio on one cable. So don’t call it a Toslink on the gig and expect
anyone to know what you mean.
Weird Stuff and Power
There really was a time not so long ago when all we worried about was the standard
Edison electrical plug, 1/4-inch and XLR. But those days are long gone, and there is
more you need to know about. First, let’s get power out of the way. Most of your gear—
especially in clubs and smaller gigs—will use standard Edison AC. But the cable is rarely
attached permanently to the box. It is usually removable and is almost always what we
call an IEC cable. Figure 4.7 shows what an IEC cable looks like.
Figure 4.7 A standard IEC electrical connector.
One end attaches to the gear, and the other attaches to the electricity. There are two
other kinds of power connections. First, there’s the hated wall wart, which looks
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something like what you see in Figure 4.8, or there’s the ‘‘brick on a rope,’’ which you
see in Figure 4.9.
Figure 4.8 Wall warts are AC adapters that plug directly into an outlet. We hate them because
they are heavy and have a tendency to fall out of the outlet in transport, and they often block
one or more additional outlets on a power distribution unit.
Figure 4.9 Another kind of AC adapter, known by many as a ‘‘brick on a rope.’’ Note there are
two cables protruding from the box. One carries power from the Edison outlet, and the other the
converted power to the unit it is intended to work with. This is better than the wall wart, but it is
still widely hated.
Either way, they plug into the 110v electrical service and convert it to something the
gear needs. Usually, but not always, this is a DC—direct current—signal.
If you don’t know the difference between alternating and direct current, you need to
do some boning up. But it is worth doing if only for the tales of the war between
Thomas Edison (alternating) and Nikola Tesla (direct) over the future of electricity
use.
Why use these annoying wall warts instead of internal power supplies? Like everything
else, it is all about the Benjamins. When you see that familiar UL-approved logo, it
means that a company called Underwriters Laboratories has approved the gear, which
means it can be insured. But getting UL approval is very expensive ($10K+ per
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approval), and the only thing they look at is the power supply. So if you are making
inexpensive gear and most of it uses the same power supply, but you’re making it external to the box, you do one UL approval instead of one for each unit that uses that
power supply. So look at them as a way to save you money.
On real high-end pro gear, you may find a different kind of power connector generically
called a twist-lock but more properly known as a powerCON. (Neutrik, a connector
manufacturer in Lichtenstein, has a patent on the powerCON, so others you see are
probably knockoffs.) A powerCON looks like what you see in Figure 4.10, and the
advantage is that the connector inserts, twists, and locks into place so it cannot be accidently removed.
Figure 4.10 A powerCON receptacle. Note the slot. The connector can only be inserted one way,
and once inserted, a twist locks it in place. Image courtesy of Neutrik.
On things where you often see a bunch of powerCONs used at once, such as power
amps or powered-line array speakers, you will often see daisy-chainable powerCONs.
The power enters one box, and then all of the other boxes are connected to the others
via powerCON cables, and the power moves up the line with a single connection to the
source. It makes for much cleaner cabling.
Finally, we have data connectors. Yes, I said data. Audio is becoming more and more
like computer networking. In fact, a couple of months before writing this, I interviewed
Tony Luna, who was doing monitors for Aerosmith, and he said, ‘‘We have gone
from being audio guys to being network managers.’’ And there is more than a little truth
to that.
The first data connection we all had to deal with was the MIDI cable, which is still in
use. MIDI was a protocol developed by a group of synthesizer makers who wanted their
instruments to be able to talk to each other. And if you really look at the history, MIDI
was the beginning of the change that turned us all into computer geeks instead of
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straight audio geeks. MIDI stands for Musical Instrument Digital Interface, and a connector looks like what you see in Figure 4.11 or 4.12.
Figure 4.11 Five-pin MIDI connector. Image courtesy of Neutrik.
Figure 4.12 Seven-pin MIDI connector. Image courtesy of Neutrik.
The only difference between the five- and seven-pin versions is that the seven-pin can
carry power as well as data. Again, cleaner cabling. MIDI started as a way to use one
keyboard to play the sounds in another. This led to the sound module, which was basically the synth without a keyboard that connected to another synth or keyboard controller to provide more sounds with less stuff. But soon, things such as effects units and
even mixers had MIDI connections that were used to change ‘‘patches’’ on a piece of
MIDI-enabled gear—for example, changing the type or level of reverb mid-song. MIDI
continuous controllers allowed things such as fader moves to be sent to MIDI gear—
you could increase or decrease volume without ever touching the actual audio or speed
up or slow down a delay setting remotely. There are some small mixers meant to do
double-duty, such as the Allen & Heath ZED, which is a real live-sound mixer, but each
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fader is also a MIDI CC device, and the board can be used to mix recordings on your
computer as well—again, with everything remaining digital and nothing ever touching
the actual audio.
The next data connection is one you will be familiar with if you have ever plugged your
computer into a network. The connector itself is called an RJ-45, but they are usually
referred to by the name of the cable type, which is Cat-5. These are becoming more and
more common as audio is transported digitally between the console and the amp or
speakers (or from the stage to the console). There are some units that on this kind of
transport use specialized fiber connections, but they are not standard. Mostly what you
will see is the good old Cat-5. See Figure 4.13.
Figure 4.13 A heavy-duty Cat-5 cable.
But Cat-5 is an issue. It was designed to be used in computer networks, so it is fine in a
permanent installation, such as in a church, club, or theatre, where it is plugged in once
and then only unplugged if there is a reconfiguration of the system or for troubleshooting. But for touring or the kind of local sound gigs you are most likely to do? The
RJ-45 was only designed to be plugged and unplugged a limited number of times, and
that is not a big number. So, they break all the time. You have a few options. The first is
to carry lots of extra Cat-5, because if you don’t have an extra, one will fail. (If you only
have one extra, two will fail. . . . ) What usually happens is that the tab breaks off.
If you don’t want to carry a bunch of extra cable (and you should be carrying at least a
couple extras anyway), then buy a good crimper and a bag of connectors at RadioShack
and learn how to snip off the end and crimp on a new connector. (This is not as easy as it
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sounds. I have a crimper and have done this, but I’m no good at it and fail more often
than I succeed.)
Another option is to only buy gear that uses the Neutrik etherCON connector, which
looks like what you see in Figure 4.14. Now that RJ-45 is protected by a steel shell and
looks kind of like an XLR for the outside. These rock and will last a long time. The
problem is that they are significantly more expensive and take more room than a
‘‘standard’’ RJ-45, so not enough manufacturers are using the female end in their gear.
Figure 4.14 etherCON protects the RJ-45 connector. Image courtesy of Neutrik.
Recently, TMB made a very nice shell protector that they say will protect a standard RJ45 for a long, long time (see Figure 4.15). It protects the tab and even has an optional
cap, so as your less educated crewmates are dragging the cables across the ground, you
don’t even have to worry about dirt getting in the connections.
I use a poor-man’s version of this but will likely switch to the TMB just because it is so
much more pro. But I use cheap RJ-45 ‘‘couplers’’ to attach the cables to each other for
moving and storage.
Figure 4.15 This TMB product protects the RJ-45 but does not require a different connector on
the other end.
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Okay, one last ‘‘data’’ connector, which will lead us to our next subject. (Don’t you love
how that works? It’s called a transition!) The BNC connector is found most often in
digital gear for word clock connections. (Again, this is not a tome on digital audio. If
you need to know about word clock, bit rates, jitter, and dithering, try Sound Advice on
Digital Audio [Course Technology PTR, 2004].) Figure 4.16 shows what one looks like,
and the other place you will find one is as the connection between the antenna and a
wireless receiver (for a mic) or transmitter (for personal monitors).
Figure 4.16 BNC connectors like the one here are used in making word clock connections as well
as connecting antennae to wireless receivers. Image courtesy of Neutrik.
And there is our transition. Cables are not the only way to get a signal into the system.
Sometimes you do it via radio waves. Welcome to the wonderful world of wireless.
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The Wonderful World of Wireless
T
he previous chapter appeared to be about cables and connecters. But what it
really comes down to is moving the signal from the source and into the system.
There is another way to do this that does not involve cabling; it’s called wireless. There are so many flavors of wireless guitar and bass packs, vocal mics, horn mics,
and monitor packs sold at regular old music stores that it is easy to be fooled into
thinking they are simple. This would be a mistake, because wireless is far from simple
and is, in fact, one of the most common reasons for a show to go south.
Although it was once pretty clean and easy, the world of wireless has changed a lot in
the past couple of years as things such as cell phones, wireless PDAs, and wireless laptops have become the norm. The other wildcard here is that depending on what part of
the live audio world you end up working in, you may deal with wireless rarely if at all.
If you are in the backline business (everything onstage that is behind the PA—there is a
whole chapter on this part of the biz later in the book), expect to deal with a lot of guitar
wireless. If you don’t do backline, you will likely not have to deal with it except for very
rare occasions. If you are working for a regional sound company, you will have to deal
with wireless mics and monitor systems regularly. But for every act that wants that
equipment supplied as part of their rental and wants the sound company to deal with it,
there is at least one other act that carries all of their own wireless, and there is a member
of their crew who takes care of running it.
You need to know it not only for those acts that demand it, but also because it makes
you a more valuable member of the crew. The guy who really ‘‘gets’’ wireless and can
make it work and troubleshoot problems is never the first guy to go in a layoff situation.
Lots of guys know how to mix and set up a system. Far fewer really understand wireless.
If you want to make yourself indispensable, then learn this stuff forward and backward.
Oh, and one last thing before we get into the nitty gritty: Even when the wireless is
artist-provided, when it does not work, they will blame it on the sound company. Count
on it.
Getting Unplugged
My first wireless came from W.A.S.P. guitarist Blackie Lawless, purchased in the mid’80s right about the time that infamous metal band was starting to take off. My guess is
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that he was selling it because he got something better, but I didn’t ask. Just standing in
his living room stuffed with skulls, medieval weapons, torture gear, and horror movie
books and paraphernalia was enough for a kid from the suburbs. I gave him the dough
and got out as fast as I could. As I recall, that unit was from Nady, and I used it until I
lost the screw-in antenna. Much like Nigel Tufnel in Spinal Tap (come on, you’ve got to
remember the infamous gig at the Air Force base? If you don’t, then homework for the
night is to go out, find the DVD, rent it, and watch it), I endured dropouts, static blasts,
and picking up other radio broadcasts in exchange for the freedom to move about the
stage at will. Fortunately, the wireless of today bears only a passing resemblance to
those early units—especially when it comes to price/performance ratio. Now, I use
wireless for my guitars and for my personal monitors and generally have between three
and six wireless mics onstage when I play with my own band.
Although my system is a mishmash of low-end pro and stuff that is just step or two
above entry level, up until about a year ago, I never had a dropout or problem. I give
most of the credit to these rules that I received from people on high who really understand wireless (also called RF for radio frequency). On the surface, wireless microphones, guitar packs, speakers, and earphones are extremely enticing for the performer
who wants to cut the cord and enjoy total freedom of movement. But, as with all technologies, there are certain potential downsides that need to be respected. (Check out the
‘‘Ten Commandments of Wireless’’ sidebar for some important tips that can make life
on stage much better for you and your audience.)
A Little History
Early wireless mics were long antenna VHF (very high frequency) types of units that
were good at preventing dropouts in signal coverage but were clumsy in that they had
large antenna aerials and limited frequencies of operation. Today’s wireless mics and
diversity receivers are mostly UHF (ultra high frequency) with a lot more frequency
agility and high-quality audio to the point that (at least among high-end pro units)
wired and wireless units have no discernable difference in audio quality. With some of
the frequency crowding issues we are seeing, several manufacturers are looking anew at
the largely fallow 900 MHz spectrum for their digital units.
Compression/Expansion
To make the wireless mic experience comparable to that of wired mics, a wide audio
dynamic range is required. Typically, this is around 90 dB signal-to-noise ratio, like
other live audio signal processing. Unfortunately, the frequency modulated (FM)
channels must be as tight or tighter than normal FM radio channels, and that means
about 50 dB signal-to-noise ratio. To obtain the extra 30 dB or more of dynamic range,
an audio compressor circuit is employed to squeeze 90 dB down to 50 dB or less before
frequency modulating the RF signal for the antenna. On the receiver side, once the RF
switch has routed the strongest signal to the FM demodulator, the resulting received
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audio is expanded back to 90 dB and ready to by sent onward to the audio output jacks.
While the diversity receiver has plenty of room and plenty of power to do a high-quality
job of audio expansion, the corresponding compressor on the microphone has very little
battery power and very little circuit board space to do the same quality signal processing. This is why many manufacturers will have different qualities of audio compression/expansion along with RF frequency flexibility for you to choose from. Obviously, a
$1,000 wireless system is expected to have flawless audio quality compared to a $300
system.
Limited Power
Most wireless mics have limited power output capability due to Federal Communications Commission (FCC) regulations for unlicensed operation in the UHF bands. Typically, this is 50 milliwatts of antenna power or less. While radio frequencies diminish
exponentially from the antenna, like audio signals from speakers, the modern diversity
receivers can pick up very weak RF signals from tens to hundreds of feet away from the
microphone. Having said this, most UHF transmission works upon a straight path between the antennas, or ‘‘line of sight’’ communication. This means that a bunch of
people or a masonry wall is not expected between your receiver and the wireless mic’s
antenna. Thus, the best location for wireless mic receivers is off to the side of the stage
or onstage—not out at the mixing console in the back of the audience. Also, both the
wireless mic antenna and the receiver antennae are mildly directional, with minimum
signals coming out the ends of the antennae. So do not point the antennae on the
transmitter and receiver directly at each other, but let them point away so that the
signals coming off the sides of antennae/aerials will allow those invisible RF waves to
expand from antenna to antenna. Obviously, a wild mic-handling vocalist cannot control the transmit mic antenna orientation, but point the receiver diversity antennae
upward and diagonally to catch as much signal as possible. And of course, obey the Ten
Commandments of Wireless for your best wireless mic experience.
Ten Commandments of Wireless
(I) Thou Shalt Have a Wired Backup. Don’t just have it with you, have it in place
and ready to go. In other words, there should be a cable on the floor that you
can be plug into your guitar at one end and your amp at the other in mid song if
needed. True, the wireless units out there today are far better than in the past,
but they can still fail. There is nothing like standing onstage in front of an
audience, futilely trying to get sound out of a dead wireless connection.
(II) Thou Shalt Do Thy Frequency Homework. Know what frequency you are
transmitting on and know how to change it if you need to. I recently heard
about a band that almost got fired from a really good gig because the guitarist’s
wireless was on the same channel as the dude in the metal band next door. You
never know until you get there what the wireless-spectrum situation is at a
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venue. There are venues in Las Vegas where so much wireless is already being
used that plugging yours in requires a presidential decree. If you know what
range you can transmit in, it will make your job—and the sound guy’s—much
easier if you encounter interference.
(III) Thou Shalt Not Insist On Using Thine Own Wireless. This is a companion
concept to the idea that no matter what mic you prefer or carry with you, be
prepared to use the standard-issue dynamic that the venue provides. Plugging
your ‘‘alien’’ instrument into their ‘‘finely tuned’’ system could cause a problem. This is more an issue with wireless mics and personal monitors than with
guitar units, but if you have specified wireless PMs and mics, and the house has
a unit of the same model or similar to what you usually use, use theirs.
(IV ) Thou Shalt Have Diversity. Diversity means more than one signal. At first, that
meant two receivers with two antennae and a microprocessor that monitored
the input to each receiver and then chose the strongest one to send to the
output. Building two units into the receiver was not cheap, so diversity was
reserved for high-end pro gear. Then, someone figured out that they could
accomplish the same thing by using two antennae with the processor picking
the strongest signal and sending it on to a single receiver. This led to the terms
true diversity (the receiver kind) and just diversity, which usually means
antenna diversity. Technology changes quickly, and for a brief period in time
this made a real difference, but, as processing power has increased
exponentially and antenna technology has advanced, the ubiquitous antenna
diversity we see today is at least as good as the true diversity of days gone by.
The bottom line is that if you don’t see two antennae, there is no diversity, so
consider carefully whether it is a good investment. Note that this does not apply
to wireless personal monitor units because they are ‘‘backwards’’ from other
performance wireless. That is, the receiver, not the transmitter, is what is in
your beltpack. I only know of one diversity PM system on the market right now.
(V ) Thou Shalt Not Commit the Sin of Parallel Antennae. If you have a diversity
receiver of any kind, and you set it up with the two antennae on a parallel
plane, you are all but negating any advantage of that diversity. Two antennae
positioned close together and parallel to each other will—with almost total
certainty—pick up the exact same signal. But if you put the two antennae at
any kind of angle to each other, you vastly increase the possibility that one will
pick up stronger signal than the other. The farther apart the antennae are, the
less difference this makes. But, because most MI (musical instrument)–grade
wireless receivers are housed in half-rack units, the antennae are no more than
eight inches apart, and in this case, eight inches isn’t a lot. . . .
(VI) Thou Shalt Understand Companding. To properly transmit any musical performance requires a fairly large signal in terms of bandwidth. The problem is
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that the bandwidth used by performance wireless devices is fairly narrow.
(Though there are some exceptions that are addressed elsewhere in this chapter, most performance wireless devices operate in the space between stations in
the VHF and UHF television bands—a space that is crowded and getting more
so by the day.) To use the least amount of bandwidth, the signal is compressed
before it is transmitted and then expanded on the receiver end. This process of
compressing and expanding is called companding. Why is this important? Because the better high-end pro wireless units now available do their companding
so transparently that only the most golden of ears can really hear it. The
downside is that those units are outside the budget of the typical entry-level
shopper. Less expensive and older wireless units use companding that you can
hear. Generally, a companded signal has less dynamic range than one sent over
a standard cable, so many performers have adapted the use of that compressed
dynamic range as part of their sound. This is especially true of guitar players,
who use that wireless compression to add to the sustain of their solo sounds.
(Some digital wireless units just hitting the market that do not require companding including the Line 6 X2 units. Check the archives at L2PNet.com if you
missed it.)
(VII) Thou Shalt Take Care When Mixing Systems. A frequency is a frequency, and
systems of different ‘‘flavors’’ should work together just fine. In the real world,
however, (barring any endorsement deals) you will almost always find that all
the wireless in a rack is made by the same company. There is good reason for
this. Most companies—especially at the pro end—make software tools that
make setting up multiple wireless systems much less of a chore. They also allow
the engineer to monitor things such as RF, audio level, and even battery condition from the front of house or monitor position. It can save the show if the
engineer can tell you that your battery is dangerously low so you can change it
between songs before it dies.
(VIII) Thou Shalt Practice Proper Transmitter/Receiver Placement. To avoid interfering with things such as TV stations, wireless units are very low power. This
makes locating the transmitter and receiver in a direct ‘‘line of sight’’ very important. Beware of anything that can block the wireless signal, including your
own body. Consider placing the receiver on the floor in front of you, rather
than in a rack behind you, if possible.
(IX ) Thou Shalt Not Lose Track of the Mute Button. This applies primarily to vocalists who insist on taking their wireless mics out into the audience (by definition, in front of the PA speakers). Unless the system has been thoroughly rung
out to allow for such expeditions, this opens the door for big-time feedback. If
you need to take the mic out into the audience, know how to mute that mic the
second it starts to feed back.
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(X ) Thou Shalt Carry Extra Batteries. This goes without saying, but I’m saying it
anyway. Have spares where you can readily access them and be able to change
them out in the dark. While we’re on the obvious, discard your old batteries
ASAP (unless they’re rechargeable), because they have this sneaky manner of
finding their way back into your gig bag. D’oh!
These rules mostly apply to artists, so your best bet is to photocopy these pages and
make sure the artists get them. The other thing to do is carry extra batteries yourself
and charge $5 for the batteries you buy for $2. It can be a nice little source of side
income.
And Now the Reality
This is complicated, geeky, and political all at the same time, but it is something you
need to keep an eye on if you are using wireless anything on your gigs.
Most wireless units operate in the UHF frequency band. For those of us who remember
TV before the days of cable and satellite, that means the channels above 13 on your old
TV. Our wireless gear operates in the open spaces between those TV stations, which has
worked pretty well most of the time for more than 40 years. But the times are a-changin’, and they are changing on multiple fronts all at the same time. First, TV stations
have—by governmental decree—moved from analog to digital signals. It is part of the
whole HDTV thing, but like all things governmental, it also has a lot to do with money.
Digital signals take much less bandwidth than analog, so portions of the VHF and UHF
spectrums will be available for other uses, and the powers that be fully intend to tax and
regulate them.
But the clearing of space that was expected by some in 2009 did not really happen. Until
then, many broadcasters were transmitting both digital (for those with HDTVs) and
analog signals at the same time, which has actually made the spectrum more crowded.
Now that analog has been dropped, one would think there would be more space. That is
a great theory, but broadcasters are looking to use that spectrum to offer additional
premium services, so there’s no new space for us.
Now, to make it really complicated, though we operate in the space between channels,
that space does not really belong to us. In other words, we are kind of wireless squatters,
so when a broadcaster that has paid big bucks for the right to use a certain part of the
spectrum starts using the part that had lain fallow for years (and that we were using
because no one else was), there is not much we can do about it.
This is why what we refer to as frequency agility is so important. You want any wireless
you buy to have a lot of frequency options and not be limited to a small part of
the overall spectrum. Believe it or not, it gets worse. Some very powerful, high-tech
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companies, including Microsoft, Google, Yahoo!, HP, and Dell, have petitioned the
Federal Communications Commission to approve a new class of wireless consumer
devices—likely portable Internet access devices. Because the wireless landscape theoretically opened up when the digital switch was thrown in 2009, and all of those analog
broadcasts went away, they are arguing that the ‘‘white space’’ between channels will
be wasted, and they want to make unlicensed consumer devices that operate in that part
of the spectrum.
The bottom line is that very soon the sound of another broadcast being picked up by
your wireless mic or guitar and coming through your system instead of the wicked solo
you were expecting could move from an occasional annoyance to an everyday occurrence. This is a situation that will eventually require a technological solution, and every
wireless manufacturer is working on some version of a solution right now. In the
meantime, it would not hurt to let your elected representatives know that wireless gear
is important to what you do and that making us go back to all-wired stages would put a
real crimp in a lot of performances. The forces on the other side are huge, with towering
piles of cash that they’re not hesitant to spend. But politicians sometimes listen to voters, and enough calls and letters might buy us the time we need to develop a technology
solution.
Dynamic or Condenser?
Back to the gear . . . Like wired vocal mics, wireless mics come in both flavors. You will
only find a few examples of condensers for less than $1,000, because they tend to be
quite a bit more expensive than their dynamic brethren. The same pros and cons that
apply to wired vocal mics apply to wireless condensers versus dynamics. The biggest is
quality of sound versus controllability onstage. Condensers have become more popular
in live settings in direct proportion to the number of performers using personal monitors. The nature of a condenser mic is that it tends to have a more ‘‘open’’ pickup pattern and is more prone to feedback and bleed from other sources—there is a price to be
paid for that more detailed and airy sound. The bottom line is that, wired or wireless, if
you tend to mix acts on loud stages or use wedges for monitoring, you are probably
better off with a dynamic.
The other issue is power. We had this bite us in the rear on my day gig recently. We
bought a video rig for doing interviews and went all out and got wireless mics with it.
But we went with clip-on mics (called lavaliers), not handheld, and that meant a beltpack like a wireless guitar rig would have. But—surprise—the mics are condensers, and
the beltpacks do not provide phantom power, so we had to find a way to get power to
the mics. In the case of a wireless vocal mic of the condenser variety, the issue is battery
time—the batteries have to provide power to both the condenser mechanism and the
transmitter. A dynamic only has to power the transmitter, so battery life is, in most
cases, longer.
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What’s the Format?
Here we are really talking about physical form. Do you want a rack-mounted receiver
or one designed to sit on a table or the floor that may be able to be racked with an
adapter of some kind? As much as I hate it when people answer a question with another
question, that’s what I’m going to do here. How are you going to use it? If you mix three
or four different bands, then you may want something more portable. If you work with
just one act, then your receiver goes into a PA rack, and you’re done. Keep in mind that
more pro, higher-end models are only available in a rack version. If you need the good
stuff and move between acts a lot, then invest in a small two- to four-space rack. A
power module, your mic receiver, and, say, a personal monitor wireless transmitter, and
that four-space rack is nearly full.
Auditioning
First, do your homework and check out what people in your area are using. Try to
borrow or rent a couple of different models and use them on gigs before you make a
decision. If you can’t do that, then have the guy in the music store hook up several
models and wire them all into the same PA. Set everything flat (no boosts or cuts in the
channel EQ) and compare sound and response that way. Do this with your vocalist. A
mic is a very personal and subjective decision. Comparing different mics in different PAs
or in channels with different settings does not make for an apples-to-apples comparison.
After you have done the level-playing-field comparisons, start dialing in your sound
on the models you liked best in the earlier comparisons. If you still end up with a couple
of models you like, encourage your vocalist to start comparing how the mic feels in his
or her hand. Look at how easy it is to change the batteries (as in, can you do it onstage
in the dark?). How is it built? Remember, metal will always outlast plastic and
take more abuse. If you are still undecided, look at the warranty, the reputation of
the manufacturer, and the recommendations of the person doing the demo for you or—
better—your friendly neighborhood sound-hound. If all else is equal, then it comes
down to price. But don’t rush it. A good wireless mic should last for several years of
even tough gigging. It’s a shame to buy a piece of gear on a whim and figure out a week
later that you don’t really like it or that it does not meet your needs. Be smart.
Time to Taste the Freedom
Some time back, I was doing a guitar sub gig with an ’80s band out on California’s
central coast. I was using the gig as my road test for a number of universal-fit personal
monitor earpieces. To keep from introducing an unknown factor (that would be a
wireless monitor setup) into the equation, I opted to go with a wired beltpack. Knowing
I would be anchored to a cable for the beltpack and remembering that the stage at this
club is pretty small, I opted to leave the wireless guitar rig at home as well. By the third
set of the first night, as I was trying to keep from tripping over the tangled wires coming
from my guitar and to my PMs, I was reminded that going wireless is almost always
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Chapter 5
The Wonderful World of Wireless
41
worth it. If you are already a wireless user, you need no convincing as to why it is cool to
cut the cable. If not, there is nothing I can say or write to convince you. It is just something you have to try in order to really understand why it is such a cool thing. So let’s
leave the ‘‘why’’ part out for now and take a look at the wonderful word of RF.
R What?
For those of you of the less technical persuasion, RF stands for radio frequency and is
the term most often used to label all of the wireless devices in use in a typical musical
performance. It is also seen by even many pretty technical folks as bordering on the
world of black magic. If you really understand RF (and I only know a couple of techs
who do), then you can get why the setup that worked one day is all of a sudden cutting
in and out in the same venue with all of the same settings as before. To most of us, the
best we can come up with is to shrug and say something helpful like, ‘‘Well, it must be
Tuesday.’’ Really. When you get into big productions that use dozens of channels of
wireless, you will generally hire one of the handful of companies around who so get the
RF thing that they have made it what they do. Companies such as Wireless First and
ATK are the ones who get the nod for things such as Super Bowl halftime shows. And
most of us are not sure whether we should approach these guys as fellow techs or as
some kind of high priests with cosmic knowledge that we may never attain.
Now that we have made it clear that you will likely never really understand RF, let’s
look at some of the basics of wireless. If you are using wireless monitors, it is really easy
to understand. What you have is the equivalent of a small, portable radio on your belt
with headphones plugged in. But this radio only picks up one or, at most, a small range
of frequencies that are not generally used for commercial broadcast. For your purposes
here, the ‘‘radio station’’ is the transmitter that is connected to the mixer that is transmitting your mix via the airwaves to the radio on your hip. A wireless mic or instrument
rig is the opposite—the radio station is the beltpack or is built into the handheld mic,
and the radio is the receiver that plugs into the amp or sound system
The Magic Is in the Motion
Whether your artist is wearing/carrying the transmitter or the receiver makes a difference. On the monitor level it is easier because it is a traditional radio model—the
transmitter is stationary while the receiver is in motion. Because the transmitter lives
with the console and uses a constant power source (in other words, it plugs into the
wall), it can transmit as strong a signal as the law allows, which is why a wireless
monitor rig is less likely to suffer from weird dropouts, dead areas on the stage, and
weak signals. A mic or instrument rig, on the other hand, uses a transmitter that not
only moves, its output power is directly linked to its power source—a battery that loses
power over time and eventually dies. This is a big part of why mics and wireless instrument rigs are more prone to dropouts and such, although things such as antennae
placement can go a long way to helping that.
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Now you know the difference between receivers and transmitters, so let’s get down and
dirty. There are a lot of wireless units out there in a huge range of prices. What gives,
and are the expensive ones really better?
Your wireless stuff may seem like just another part of your musical gear, but remember
that it is, in fact, a small radio station serving a single receiver, and as such, it falls under
the authority of the Federal Communications Commission (the FCC).
It works like this: There is a limited amount of space available in any given frequency
band (AM, FM, UHF, VHF, and so on) and lots of competing interests trying to get a
piece of that pie (actually, it is known as bandwidth). So the FCC decides not only how
big a slice of that bandwidth you can have, but also how much power you can use to
transmit in said frequency range. For example, my favorite radio station in L.A. is Jack
FM, which is at 93.1 on the FM dial. If I go to 93.1 in Las Vegas, I get a much less
interesting classic rock station. (Jack is at 100.5 here.) So the FCC has granted two
entities the right to transmit in the 93.1 part of the spectrum, but it limits the power of
each so that they do not interfere with one another. Wireless devices for instruments and
pro audio are always low-power affairs so they can operate in the same frequency range
as, but not interfere with, the broadcast being offered by the person or company with
enough money to hire a lobbyist.
On the compression side, we call it companding because the signal is compressed before
transmission in order to cram all of it into the small slice allowed by the FCC and then
expanded again on the other end. Compress plus expand equals compand—get it? The
problem is that it is just not that simple.
If you want a real hearable test of this, then take a song from a CD you own and rip it to
an MP3 file. Unless your system or ears (or both) are lousy, you will hear a difference
between the two recordings. The MP3, in the process of compressing the data to make a
smaller file, loses some sonic detail and dynamic range. Now take the MP3 and save it
as a standard AIFF digital audio file, like on a CD, and listen. Most of the stuff that was
lost in the MP3 remains lost in the ‘‘new’’ CD-format file.
But there are efforts to combat the effects of companding. There was a very good digital
wireless system made by a company called X-Wire that worked like a big wireless
modem for your guitar and sounded great. (Because digital data takes a lot less bandwidth to transmit than an analog signal, no companding was needed.) X-Wire was
bought by Sennheiser, and the technology pretty much disappeared, though some other
companies are looking to develop digital wireless products suitable for the pro audio and
MI user. Lectrosonics makes an analog/digital hybrid that sounds so good you can use it
with a measurement mic to sonically sample and then ‘‘tune’’ a system to the room in
which it is used. And Shure has come out with something recently called Audio Reference
Companding that uses new algorithms to combat the sonic effects of companding and
does a hell of a good job with it. I recently used their newest wireless on a gig, and it
sounded amazing. Yes, it is expensive, but in this case you get what you pay for.
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The Wonderful World of Wireless
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Meat Absorbers
Some random thoughts on getting the best possible performance from your wireless
gear:
■
When it comes to wireless reception, the biggest impediment is distance. Always set
up your wireless receivers as close to the performer as possible. The second biggest
problem is the performer himself. The difference in power reaching the receiver
when the transmitter is in the line of sight of the receiver and when the performer’s
body is between the transmitter and the receiver is about 30 dB. If you add a distance
of, say, 75 feet (for example, if you put the receivers at the FOH mix position instead
of at the side of the stage), you virtually guarantee dropouts.
■
Diversity, in the wireless world, has nothing to do with being PC. It is a way of
labeling units that can take two versions of the same signal and figure out which one
is stronger and use that, switching inputs as signal strength changes. At first there
was just diversity, and that meant two actual receivers in the box with one antenna
serving each receiver and a microchip determining which signal got sent to the output. As the pressure to drop prices got more intense, designers figured out how to
have the switching occur after the antenna and before the receiver. This meant one
receiver in the box, which meant more affordable boxes. Although this antenna diversity was once seen as an inferior way of doing things, the technology has matured
to the point where true diversity is almost never found in even pro-level gear.
■
Finally, antennae placement. Remember that the transmitter is omni-directional. In
other words, its already paltry FCC-imposed power goes out equally in all directions, so the antennae on the receiver pick up just a small fraction of the total power.
If you have two antennae and you have them parallel, then they are likely both
receiving just about the same signal, and the diversity switching doesn’t really do
anything. By putting them at differing angles, you have a better shot at one picking
up a stronger signal than the other and that changing as the performer’s position
relative to the receiver changes.
Thanks to FOH magazine’s technical editor, the late Mark Amundson, for making this
simple enough that even I could at least pretend to understand it.
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6
Snakes and Splits
S
o far, everything we have looked at has been all about taking sound (or acoustic
energy), converting it into a form the system can use, and transporting it into the
system. So you would think that now we’d get to start looking at the big parts of
the system—and if we were living in the ’60s, that might be true. But as technology has
advanced, so have the demands of those outside of the audience.
What do we mean by “outside of the audience?” Let’s take a look at a state-of-the-art
venue and what looks like a good old-fashioned rock tour for examples.
A Pirate Looks at 40 Channels
Rich Davis and Billy Szocska have been the sound team for Jimmy Buffett for years.
Buffett may not be your cup o’ rum, and his music might lead you to believe that the
production is simple. That would not be a good assumption. You see, there is a lot going
on besides what the live audience is hearing. Monitor engineer Szocska makes sure the
band and the boss are happy, while Davis mixes for as many as four audiences at once.
There is the audience mix, a mix for the video team, a mix for the live radio broadcasts,
plus at least 16 channels of recording, all off a classic analog Midas XL4 console.
And the ability to run that kind of system and that many mixes is a big part of why
Jimmy Buffett is one of the highest-grossing touring artists in the world. You may not
hear it talked about a lot, but those outputs are part of a business model that has made
Buffett rich and kept his crew and sound company of many years (Sound Image, based
near San Diego, California) busy.
Meanwhile, Back in the Desert…
The Maloof brothers make a whole bunch of money as owners of the Palms Resort and
Casino in Las Vegas, the place where celebs and assorted other rich and beautiful people
come to party and spend, spend, spend. When the Maloofs decided to put in a worldclass performance venue, they did not mess around.
When an act performs at the Pearl, located inside the Palms, there are multiple audience
mixes because there are areas for “regular” people plus Vegas-style VIP areas. (There is
even a send to the restrooms and the bar areas.) There is a split to the monitor board for
another mix. There is a split that runs from the stage, up 17 floors, to a world-class
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Live Sound Fundamentals
recording studio, which is tied right into a VIP suite in the hotel. There is another split
available for sending to a broadcast truck.
As shows become more complicated and expectations grow ever higher, the job of the
sound engineer has become much like that of a network engineer. There is one person at
the monitor console and one person at the FOH console, but there are many more than
that on a typical sound crew. And a big part of the job is getting sound from the stage to
where it needs to be in the venue.
Splitting Sound
On all but the simplest gigs, the sources at the stage will go not to a console, but to some
kind of splitter. These are usually referred to by a number—a two-way split or a threeor four-way split. In the days when all consoles were analog, it was not unusual for a
monitor-specific console to include a split, but it is more common for the split to be
separate from the console (although the difference between analog and digital here is
huge).
Check out the pictures in Figure 6.1. This is a smaller analog two-way split. The first
two rows of XLR connectors (all female) take an input directly from a sound source,
such as a mic, or from a subsnake. The signal is then split, with one signal going to the
male XLRs labeled Direct Outs and the second to the blue connectors labeled Iso Outputs. Iso refers to the fact that the last group of outputs is “isolated” from the inputs and
direct outs by use of a transformer. Each output also has a Ground Lift switch, which is
useful for eliminating the 60-cycle hum common in badly grounded electrical systems.
Figure 6.1 A splitter. Image courtesy of Whirlwind.
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The snake allows us to move the audio from the stage to the front-of-house position
over a single cable. The snake may terminate in another box of connectors and need
short cables to get the signal into the console. Snakes generally end in what is referred to
as a fan—several feet of the conductors inside the sheath on the snake exposed and
terminating in an XLR or a TRS connector. Try to imagine lifting a bundle of 56 mic
cables and wrapping it. It’s heavy, dirty work, and if you are lucky, it is where you will
start in the sound biz.
Figure 6.2 shows a small 248 snake. This refers to 24 sends (from the stage to the
console) and 8 returns (from the console back to the stage). When looking at a snake,
the first number is always the number of sends, and the second is always the number of
returns. In a snake of this size, it is possible that the cable assembly is permanently
attached to the box. But more likely, it attaches via a screw-on multiple-pin connector
generally referred to as a mult.
Figure 6.2 A 248 snake made by Whirlwind. Image courtesy of Whirlwind.
Yes, snakes and splits are becoming increasingly digital and, by extension, smaller and
lighter. But trust me, there will be analog copper snakes out there being used for a long
time.
Ones and Zeros
The basic concept of the split and snake remains the same in the digital domain, but it
looks a lot different. It has the same XLR connections on the stage end and does use
another box at the console end. (The first time I used one of these, I quickly realized that
I needed a bunch of three-foot-long cable to go from the box to the console, and not the
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Live Sound Fundamentals
20-footers I usually carry, as the cables piled up behind the console in what had to look
like some kind of viper’s nest.)
The difference lies in the cable. With digital audio transport, an entire 100-foot length
of snake takes up less space and weighs less than maybe six feet of the analog snake.
That’s because instead of individual conductors for each signal, a digital snake converts
the signal from analog to digital at the boxes and transmits it all together over either
fiber optic or standard Cat-5 or Cat-6 cable. The box at the other end converts the
signal back to analog where it goes into the console—that is, if the console is analog. On
a digital console, the connection will be made directly into the console or into its processing unit, depending on the brand and style of digital system.
But we are getting way ahead of ourselves. For now, just note that you will usually need
multiple versions of the input signals sent to various places in the venue, and you use a
splitter to make that happen. The cable that carries all of those signals is called a snake,
and it can be a big, heavy bundle of copper or a single piece of fiber-optic conductor.
Oh, one last thing: subsnakes. It is not unusual to find several smaller versions of the
snake around the stage that can carry anywhere from 8 to 16 signals. These are used just
to keep the stage mostly cable-free. Instead of 12 individual cables going from the drum
kit to the split, you plug each mic into an input on a stage box, and they all travel on
a smaller snake to the split. It’s just neater that way.
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7
It’s Not the Car, It’s the Driver
T
he mixer (also referred to as the console or desk) is almost literally the heart of
the system. Everything comes in, gets swished around, and then gets pumped
back out as a finished product. The only thing left is to deliver that product to
the audience.
Today’s mixing consoles bear little resemblance to the gear we were using when I got
started playing in bands and running sound in the mid ’70s. But somehow, as consoles
got larger, with more channels and more EQ and processing, and finally entered the
digital domain, where they could do anything the most sophisticated recording console
could do, sound at events both large and small got worse, not better. And strangely
enough, the exceptions—that is, the good-sounding shows—are often mixed by the
same people who were kludging together homemade systems in the earliest days of the
performance audio business. They may be driving the most sophisticated system out
there, but they only use the tools they need.
This chapter has nothing to do with gear or technology, yet it is very likely the most
important part of the book. Harsh, but true . . .
A few years back, I was fortunate enough to say yes when a friend asked me to come to a
small town in Utah to help with a festival gig. He had asked another friend, who also
said yes. No one at the small high school where the festival was based had any idea that
the guy with the Aussie accent working with the orchestra in the main auditorium was a
performance audio legend. Howard Page, now the senior director of engineering for
Clair Global, has mixed acts as diverse as Van Halen, James Taylor, and Mariah Carey.
Here is what he had to say about the current state of performance audio.
After being involved in live sound engineering for so long, I am very, very sad to see
the way it has all evolved in the last few years. When did the kick drum become the
lead singer? Show after show, regardless of the style of music, ends up being just a
solid wall of badly mixed, way too loud, over the top, low-end-heavy noise. I have
tried to help and nurture so many young guys over the years to understand what
mixing live shows is all about, and my often-repeated sermon is to make it sound as
close as possible to the recorded material by the artist. If some artists ever came out
front at their shows and listened, I’m sure they would be horrified at how their
performance is being brutalized. True, lately, some artists set out to use the sound
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Live Sound Fundamentals
system to deliberately beat up the audience, but those shows are way beyond any
help.
It comes down to really understanding your role as an audio provider. We refer to what
we do by many names: Live sound, live audio, concert sound, and performance audio
are a few. But I wish we could all get back to the term that really describes our job and
function: sound reinforcement.
Our job is to not be noticed. We are there to help the performer communicate with the
audience. It means giving the performer the means to convey their artistic intent and
emotional vision past the area where they can do so unassisted. It is never about how
cool your gear is or how loud you can make it or how bitchin’ your kick sound is. We
should be invisible and not affect the content of the performance in any way except to
spread it further. Anything else should be considered as a failed gig.
Drew Daniels is an electro-acoustical consultant, studio musician, recording engineer
and producer, and audio technology educator based in Los Angeles. His sound reinforcement experience includes stints with Teac, Fender, JBL, and Disney, where he
filed no fewer than five patents. On his website, he lists four reasons for lousy sound at
live events, and this is the best, most concise list I have ever seen, so I am stealing it.
(Drew used the above quote from Howard Page, which originally ran in FOH magazine, which I edit, so I figure we’re even. . . . Thanks, Drew.)
The List
1. Inadequate technical education
2.
Hostility between sound providers and artists
3.
Inadequate music education
4.
Inappropriate gear or gear being used inappropriately
This book is, hopefully, a beginning point for taking care of #1 and, with a bit of luck,
using the knowledge you glean here and from other sources, you can avoid the pitfalls of
#4. Though it is really outside the scope of this book, we are going to spend a little bit of
time on #2 and #3.
Hostility
I once talked with a good friend, a musician who spent many years touring with a wellknown country act. I don’t remember how it came up, but we got onto the subject of
stage volume—the bane of sound providers at many gigs. When I said that keeping stage
volume under control allowed the sound guy to make it sound better in the house, he
replied, ‘‘Soundmen are the enemy.’’ And unfortunately, this kind of attitude is rampant
with bands. I have personally dealt with acts that were so loud onstage that I could not
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It’s Not the Car, It’s the Driver
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get the vocal up above the guitars, and the only thing still in the system was the vocal. I
have had bands, when I asked them to keep stage volume to a minimum in order to get a
good sound for the audience, tell me, ‘‘Drop it. I play loud.’’ I have seen entire local
concert series cancelled because one band was too loud and refused to turn down.
The act and the sound provider should be a team, so why the hostility? Many reasons.
First, we, as pro audio providers, need to check our egos at the venue door. Remember,
it is not about us; it is about the performer. Many artists have never experienced a
situation in which the sound guy knew and practiced that concept. But I promise you
that every time I have worked with an artist and have been able to communicate that I
understand my role and am only there to make them sound good, I have gotten cooperation. Every time. There have been times when the level of distrust and hostility
was such that I could not communicate that effectively, but I always try.
What is the cause of the hostility? ’Tis all opinion, so take it for what it’s worth, but
there are too many sound providers who started out as musicians and who are still
carrying a chip on their shoulder about not making it, and they take it out on the acts
they work with. This is very common in local and regional clubs with a house sound
guy. Take a look in the mirror and make sure this isn’t you.
Incompetence
The next reason is just flat incompetence among many house crews. I have seen some
great house crews in my time. I have also, as a performer, had to deal with people who
had no business behind any kind of sound console, had no idea how to operate the gear,
and really didn’t care. Too many venues don’t put enough emphasis on their own
sound. They hire unqualified people just because they will work cheaply. If you were a
performer on tour and had to endure a string of such venues and crews, you would have
your back up, too.
Sometimes the hostility stems purely from the fact that the artist is an egotistical jerk.
But guess what? Even if that is the case, your job is to make him or her sound great.
Refuse the gig the next time they come to town, but if you are there, then you need to do
your job and actually care about how it sounds.
Education
The next item is inadequate musical education. This is a tough one. Many of the best
sound engineers I know have no formal musical education, and they approach audio as
a mix of art and science. Others are very accomplished musicians and know enough of
the science to do the job but really approach it as almost another member of the band
with the sound system as their instrument. (I know this seems to directly contradict the
earlier statement about being ‘‘invisible,’’ but the best way I can explain it is a quote
from Tom Johnston, one of the founders of the Doobie Brothers. In an interview many
years ago, he talked about the playing of his bandmate, Patrick Simmons, saying that
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Live Sound Fundamentals
few people understood how vital he was to the band’s sound, and you often did not
realize he was even playing until he stopped and the song fell apart. Try to apply that to
live sound, and you’ll ‘‘get’’ the artistic approach to sound reinforcement.)
The most important thing you can do with regard to music education is to familiarize
yourself with the artist’s music to the greatest degree possible before the gig. If you are
working in a club that books four bands a night, most of whom you meet for the first
time at sound check, then there is not a lot you can do. But being familiar with and
understanding typical song structure in a given genre will help. Listening to lots of different kinds of music and having a solid understanding of how a great jazz band sounds
versus the vibe of a great rock band is huge.
It comes down to knowing enough about different musical genres to know what is
appropriate. Even that band you meet at sound check is not a total loss. Ask them who
they think they sound like and what kind of music they listen to. This will give you at
least an idea of the direction to start in.
Okay, lecture over. Just remember that it is about the music, not the gear or your ego.
Now let’s look at that console . . .
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The Channel Strip
O
kay, intrepid audio novices. When we last met, we had gotten through the
first parts of the signal chain going from the source converting it via a transducer (mic) or direct box and getting it to the mixing console via either cables
or wireless. So, we are at the console . . . now what?
Let’s start with the inputs. (By the way, our examples will be analog mixers, as hopelessly old-fashioned as that may seem. I firmly believe that a solid foundation in analog
is easer to build on than learning everything digital and then getting to a gig and finding
out that you will be using a 24-channel, 20-year-old analog desk. I have seen young
engineers faint in such circumstances and, yes, I took the time to ROTFL. These are
located on the back of most pro desks, as you see in Figure 8.1.
Figure 8.1 I/O panel of a Mackie Onyx 1640i. Image courtesy of LOUD Technologies.
And sometimes they’re on the top, as you see in Figure 8.2.
Or, they’re on the back in the case of some units that are meant to be rack-mounted.
Inputs come in two flavors: XLR for mic inputs and usually 1/4-inch for line-level signals. (More on this in a minute.) The 1/4-inch can be either balanced or unbalanced.
Again, most pro boards will always be balanced. I explained the difference in the
chapter about cables, so I won’t get into it here except to say that balanced is always
better because it means a cleaner signal and less noise. On some boards you will also
find RCA (phono)–style inputs (like the ones on your home stereo) for hooking up
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Figure 8.2 I/O panel of an Allen & Heath ZED R16. Image courtesy of Allen & Heath.
things such as CD players without taking up a pair of ‘‘regular’’ channels. There are
other jacks back there that we will get to later (inserts and outputs).
Now that we have the signal in the board, let’s take a look at what happens first. It is not
completely accurate, but the easiest way to look at what happens from the time a signal
enters the console until it exits is to think of it like a series of pipes and water flowing
through them. If the input is the source of the water, the first knob at the top is like a
valve that determines how much water gets in to begin with.
That first knob is marked Trim, Pad, or sometimes Input. Note that the first control is
not the fader or volume control on the bottom. That is often the end of the chain for that
individual channel before it is fed to the master section. As I said before, that valve
analogy is not exactly accurate, because it is at this point that a low-level mic signal gets
boosted to a level the board can use, and this is where the magical mic preamps live.
That may seem like a bit of an overstatement, but sound guys will come close to fistfights over mic preamps, and an entire product category has emerged as high-end mic
pre makers have started selling a lot of external preamps to make up for the lack of
‘‘warmth’’ in many digital boards. This is really an argument over what a preamp is
supposed to do. One school of thought says that all a preamp should do is boost the
signal, while others actually change the character of the sound more like an equalizer
might. So what some people hear as brittle or lacking warmth is often just a case of what
comes in going out unaltered.
Part of the reason this whole preamp war is heating up is where the actual preamps are
located in a digital system. Remember, in many high-end digital mixing systems, the
console itself does not pass any actual audio at all. All of the ‘‘mixing’’ and processing
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happens in outboard units, and the console itself is just a control surface—kind of like a
big, complicated mouse with lots of knobs and buttons.
In many of these systems—as well as most digital snakes—the mic preamps reside in a
‘‘stage box,’’ or what we would call the split in the analog world. In other words, all of
the mic signals get boosted right away, so that all signals hit the A/D (analog-to-digital)
converters at the same line level. Now, this may make sense in the computer world (and
remember that a digital mixing system is nothing more than a very specialized computer), but if I am doing what it takes to carry around a 600-pound analog console because
I like the way the preamps color the overall sound, I am not going to be happy bypassing
those preamps in favor of the super-clean ones in that digital snake. It is one of the big
reasons why the most successful digital snakes to date are those that are a part of a
larger digital mixing system.
Okay, back to our analog console. There are a few schools of thought about adjusting
the input level. I am of the school that says you get a ‘‘full’’ signal at the beginning and
then adjust levels later. The method for doing this depends on your console. If you have
a button on the channel marked PFL (pre-fader listen), engage it; it does not really
matter where the fader is. Bring up the trim control until the ‘‘clip’’ light comes on and
then back it down to the point where it stops lighting up on the loudest parts of the
performance. If you have meters, then look to get the signal in the –7 to –5
neighborhood.
Note that this is in Analog World. As I alluded to earlier, gain in the digital realm is
approached much more conservatively just because digital distortion never sounds
good.
Some boards—especially those ones you will encounter on club gigs and other smaller
jobs—will not have that PFL button. In this case—in order to make sure that the relationship between input trim and output gain is correct—you will need to start with all
of your channel faders at unity. This is usually an area along the travel of the fader
marked with an arrow, a zero, or a shaded gray area.
The next section of the board is the EQ—a subject we will spend a chapter on, so hold
tight for now. If there is a button to switch the EQ out of the circuit, keep it engaged
until you have ‘‘level’’ on all of your inputs. If there is not a switch, make sure all of your
EQ is zeroed out—in other words, nothing is being boosted or cut. Also, make sure that
all of your auxiliary sends (auxes) are at zero, or ‘‘unity’’ if they are labeled that way.
We’ll look at what these sends do a little later.
At the beginning of this chapter, I mentioned something called inserts. (Sorry, outputs
will have to wait, too.) An insert is like a detour that a signal takes between the mic pre
and the rest of the channel. Processors such as compressors and pitch correctors go
here—processing that you want the entire signal to get, and not just part of it, like what
happens when you use an auxiliary send. (We’ll get there; we’ll get there!)
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You need to use a special cable for an insert called—who would’ve guessed it—an insert
cable.
It can be either 1/4-inch or XLR, but either way, an insert generally has both the send
(the outgoing part of the detour) and the return (the incoming part of the detour) on the
same jack. The cable is a Y with one connector on one end and two on the other that
jack into the input and output of the device being inserted. On some boards (like most
older Mackies), the insert doubles as a direct out for recording if you insert the connector halfway in.
A last note about insert cables—they change as you move up the food chain. On most
club-quality mixers and even some pro models, you will use unbalanced single-point
inserts, as described above. On higher-end consoles, you will move to dual-point inserts,
which are balanced and require two cables.
By this point you should have mics and DIs set, the snake run, and signal running into
the console, and you should have confirmed that there is an appropriate amount of
signal coming into each channel. That’s enough for now. In the next chapter, we’ll look
at EQ and auxes.
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9
Console Auxiliary Sends . . . or,
What Do the Knobs in the
Middle Do?
I
n my day gig editing a pro audio magazine, we sell T-shirts with the answers to the
questions most often asked of sound engineers by audience members. The first answer on the list is, ‘‘Yes, I know what all the knobs and buttons do, and yes, it took
a long time to learn.’’ (Thanks to James Geddes for that one . . . )
So by this point you have a signal, and it is controllable via a channel on your console. If
the console is analog so is the signal. If it is digital, the signal has been converted
somewhere along the line. The weak mic signal has been amplified, and now we can
manipulate it. And here is where we enter dangerous ground.
Remember some time back I brought up the fact that our job is really to reinforce what
is happening onstage and make it loud enough for a larger group of people? Ideally, you
should not have to touch the EQ at all, but it hardly ever works that way. The same
voice can sound very different through two different mics or speaker systems. Even mics
and systems made by the same manufacturer can have subtle differences that you need
to make up for. But that is EQ, and it is the next chapter. For now, we are going to talk
about auxiliary sends and returns.
The size and ‘‘level’’ of your console will largely determine how many auxes are available and where they appear in the signal chain. Aux sends come in two flavors—prefader and post-fader (and often auxes are switchable between pre and post), and this
position determines what they are used for.
Made to Order
You know how when you are at a bar and you order a drink (be it with or without an
adult component), the bartender pulls out a little gun-type thing from which he can
dispense all kinds of liquid refreshment? Aux sends are kind of like that but backwards.
To continue the plumbing analogy we started some installments back, if sound is like
water, then the console is like a series of pipes and valves that determine what goes
where and how much of it.
An aux is like a valve that sends some amount of the ‘‘water’’ off to another system. In
some cases the ‘‘water’’ is returned to the main system after something has been added,
and in other cases some of it is sent off to do work elsewhere. Whether that ‘‘water’’
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returns to the main system is based on what happens to it while flowing through the
other system.
How you set up the flow at different parts of the system is known as gain structure and
is the source of many arguments in many bars between many audio folks. But there is a
reason for the passion behind something that may seem minor: Good gain structure can
be the single biggest difference between a good-sounding show and a bad-sounding one.
Pre-aux sends are usually used for crafting monitor mixes. (EQ generally comes before
the pre-fader sends in the signal chain, but some consoles will allow you to switch the
EQ in so that it affects those monitor mixes as well as the main output.) Monitor sends
are before the fader because if a change is made to the house mix, you don’t necessarily
want the same change to happen in the monitors (a real issue with digital consoles that
share a mic pre, but again, a subject for another time). Therefore, the sound goes off to
the monitor system before the main fader so that the monitor mix can be crafted apart
from the main mix.
Now, just to make sure there isn’t too much confusion . . . Yes, we talked about splits,
and in a concert-sized rig, one of those splits will go to a separate monitor system. But
most smaller gigs—the kind you will be working with early on—will often require that
you run the mains and a few monitor mixes from the same console. Those extra mixes
are the product of the pre-fader aux sends.
A post-fader aux send is generally used to drive some kind of effect or sound processing.
This allows you to determine how much signal gets that reverb on it, and a separate aux
return or channel determines how much comes back. Most good mixers have aux return
controls in the main output section, but most pro sound engineers will return the effect
through an unused channel so they can easily determine how much of the effected sound
gets back to the system and to the ears of the audience. This is, for example, a great way
to easily kill the reverb on a singer when he or she is talking to the audience. Reverb
sounds cool in judicious amounts while singing, but it almost always makes spoken
communication very hard to understand.
Insert Here
One other kind of send and return is the channel insert. This is like an aux send, but you
don’t get a knob to determine how much sound gets sent. It all goes, and it all comes
back. This is usually used for dynamics processing—compressors, limiters, and gates—
but it can also be used with an outboard EQ or automatic feedback killer. These get
‘‘inserted’’ into the channels where they are needed and are not on an aux.
Your better consoles will have insert send and receive jacks for each channel—usually
balanced XLR connections, but almost all mixers commonly found on small gigs, including clubs, houses of worship, and the like, have a single TRS 1/4-inch jack. To use
the inserts, you will need a TRS send and receive cable. This is a Y cable with a single
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Console Auxiliary Sends . . . or, What Do the Knobs in the Middle Do?
59
TRS on one end and a pair of mono 1/4-inch jacks on the other end. The tip of the TRS
sends the signal to one of the mono 1/4-inch jacks while the ring portion gets the signal
back from the other mono connector. To further complicate things, most pro consoles
have direct outs that send the raw signal from the input off to other places, usually a
recording unit of some kind. But some MI consoles combine the inserts and direct outs.
If you insert a TRS all the way into the jack, it is an insert. If you stick a mono jack into
the halfway point of the first click, it acts as a direct out.
Depending on the size and level (that is, MI, pro, touring) of the mixer, the master
section will be either in the middle (typical of most higher-end models) or on the righthand side. Here you will find your master volume faders along with your aux returns
and usually a master send control. There may also be a two-track in/out with RCA jacks
and a separate mono fader for a center cluster (used in a LCR, or Left, Center, Right,
mix) or to feed a subwoofer. If the mixer includes subgroups or VCAs (voltage-controlled amplifiers), those faders will be placed here as well, and perhaps some rudimentary EQ.
Strange as it may sound, we are going to cover what seems like the simplest part—the
master faders—in a later chapter because it means dealing with gain structure, and that
is the subject of a whole chapter all by itself.
Remember Tapes?
Those RCA two-track jacks are the easiest to get out of the way first. The two-track out
is a mirror of whatever is coming in to the master fader, and it allows for the easy
connection of a recording device for making board tapes. If you have a two-track in, it
allows you to insert a playback device (such as a CD player) without burning up a
couple of input channels. In truth, it is something of a throwback to the days when we
all had a cassette tape machine in the rack. These days, playback is more likely to be an
iPod, and recording is done on a laptop. Some manufacturers now include USB or
FireWire outs to feed that laptop directly. Also, I would not be surprised if, by the time
you are reading this, someone puts an iPod dock connection right on the console as well.
A couple of two-track ‘‘extras’’ can include a separate gain control labeled something
like ‘‘two-track to Aux 3,’’ which allows you to send the playback to just the monitors if
you want to get something like a pitch reference or click track to the band but not have
it in the mains. There may also be a switch that says something like ‘‘two-track to
Channels 15-16,’’ which sends the two-track signal to a couple of input channels so you
can use the EQ and aux sends from the channel if you need to.
Okay, now the next easy part. If your mixer has a small graphic EQ in the master section, it is most likely pretty useless, although there are a couple now available with
digital 31-band EQs at the master. Most likely, you will have a seven- or nine-band
graphic. If there is a switch to bypass these, use it. If not, just flatten it out and ignore it.
An EQ that small is going to boost or cut things that you don’t want to touch as a side
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effect of anything you do want to change. The bands are so wide that it is akin to doing
surgery with an axe when you really need a scalpel.
Now Boarding Group A . . .
If the console has subgroups, here is where the master for those groups will reside. You
assign like inputs (say, drums or horns in a section or backing vocals) to a group. You
can then adjust the level of the entire group of inputs without adjusting individual
channels.
VCAs serve the same purpose, but there is an important difference. In a subgroup, the
audio signal itself routes through the group faders so you may have a direct out or an
insert. With a VCA, no audio is present at the faders. The faders just send control voltage that raises or lowers the output of a channel or group of channels—hence the term
VCA groups.
The advantage to this approach is that it means one fewer set of components for the
audio to travel through, and it can result in a cleaner sound. This disadvantage is the
loss of those inserts that allow you to, say, put a single compressor on a horn section.
I Wanna Go Home
Your aux masters reside here as well. You should remember that there are sends to the
auxes on individual channels. For a pre-fader (in other words, monitor) send, this is the
master volume control for a specific monitor mix. For a post-fader (in other words,
effects) send, it determines how big a signal is sent to the outboard processor. The
amount of the effect you actually hear in the mix is controlled by the aux return.
Turning up the return combines more of the wet (effected) signal with the dry input. If
you have too much reverb in the mix, it likely means that the effect returns are dialed up
too high.
There may be another control labeled something like ‘‘return to Aux 3,’’ which puts the
effect in the monitors as well. Be careful with this one. Even if the singer likes a lot of
grease on the vocal, the more reverb in the wedge, the greater the chances are for
feedback.
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I
Equalization
n talking about the console, we have been concentrating on the individual channel
strips, and that will continue here. But there are other places in the system where
EQ will come in again. Just so you know . . .
Rule #1: Listen First
You know how when you go to dinner with friends, there is always someone who picks
up the salt shaker and starts salting away before they have even tasted what is on the
plate? I hate that.
I also hate it when I see an engineer start making EQ adjustments before there is sound
in the system. Every system is different, and you have to listen to get the most out of it.
The most important skill you can develop when it comes to running sound (at any level,
from a small rehearsal room, to a club, to an arena) is to learn how to listen. With all of
the new high-tech toys available, I find far too many sound guys who spend more time
looking at laptops, touch screens, and processor menus than they do listening to
the band. They are, in effect, trying to mix with their eyes. This doesn’t work very well.
It’s important not to get too tied up in where the knobs are pointing. Adjusting
the EQ based on what you are hearing is far more beneficial than making sure a particular frequency band is knocked down by 6 dB like the guy in some magazine says it
should be.
I remember watching one sound guy, whose ears I admire, adjusting a system. He did
not even look at the knob—in fact, his eyes were closed. He turned it until it sounded the
way he wanted it to. That is a great approach.
EQ Bands and Types
A typical MI (musical instrument)–quality mixer will have anywhere from two to four
bands of EQ on each input channel. Two is easy—one is high and the other is low—just
like the bass and treble controls on your home stereo. As we add bands, we get into the
midrange, and that is where things can start to get confusing.
Let’s start with EQ types. First, you need to know whether you are looking at a true cutand-boost filter or a simple roll off. With a roll off, all of the frequency content of a
particular frequency band is present when the control is dialed all the way on. Dialing it
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back ‘‘rolls off ’’ the content of that band. A true cut and boost is at zero—or flat—when
the knob is at 12 o’clock. There is often a notch in the knob’s rotation at that point
called a détente. Dialing the knob up or down either boosts or cuts the content of that
frequency band. Both types of EQs are centered at a specific frequency and have a specific width (how many adjacent frequencies they affect) called the Q. These frequency
centers and filter widths are a huge part of what makes one mixer sound different from
another.
The other kind of EQ or filter is called a parametric or semi-parametric. These are also
referred to as sweepable and are usually found in the midrange. A good console—for
me, anyway—will have four bands of EQ including two sweepable mids.
A fully parametric EQ consists of three adjustments. First is the center frequency, next is
the amount of boost and/or cut applied to the band, and finally, the Q control that
adjusts how wide the band actually is. An EQ that includes all three of these controls is
referred to as true or full parametric. Most of the sweepable controls you will find on
MI mixers will leave out the Q control (the width of the filter is fixed) and are properly
referred to as semi-parametric.
There is another member of the EQ family that you need to know about. It’s called a
high-pass filter. This is not a knob, but a switch that allows frequencies above a set point
to pass and steeply rolls off anything below that. (A low-pass filter does just the opposite.) High-pass filters are found on most pro consoles and can work wonders for
cleaning up a mix. Just engage the filter for any source that does not have content below
the point at which the filter is set (typically about 100 Hz). You’ll find that this is most
of your mix.
How Do I Use ’Em?
The first thing I do with any board is to zero it out by setting all of the channel faders,
auxes, and EQ controls at their zero settings. Remember, on a true cut-and-boost EQ,
the zero setting is usually at the 12 o’clock position. As you gain experience and get a
feel for your system, your mics, and the players, you will find yourself making the same
cuts pretty much all of the time (such as cutting at 120 Hz to take the mud out of a kick
drum or cutting 1.25 kHz from a vocal mic). When you get to that point, it is tempting
to just make those adjustments automatically before really listening to the system. In my
world, that is just a bad idea. Start flat and listen before you start adjusting.
Rule #2: ’Tis Better to Cut Than to Boost
When it comes to EQ, it is always better to cut than it is to boost. Remember our
plumbing analogy from Chapter 9? Well, assuming your main pipe is pretty full to start
with (as it should be if your channel trim is set right), then adding EQ is like adding
water to that pipe, which could overload it. In the audio world, that means distortion
and maybe feedback. When in doubt cut, don’t boost. So, how do you get more bass, for
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example? Try cutting everything except the lows and then boosting the overall signal a
little bit to get the same effect as just boosting the bass.
Table 10.1 shows the frequency bands of various instruments. (This is a little like giving
away the secret of the ages for sound guys. Keep this info close to the vest, lest it fall into
the wrong hands. . . . ) Knowing the frequency ranges of various instruments can make
the job of EQing a lot easier. Keep this chart handy until you have it burned into your
brain.
Table 10.1 Frequency Bands of Several Instruments
Frequency Range (Hertz)
Instrument
Minimum
Nominal
Maximum
Kick Drum
130
164
196
Floor Toms
220
-
440
Rack Toms
350
-
700
Snare Drum
650
784
1000
Cymbals
500
-
40k
Bass Guitar
31
-
500
Guitar
82
-
700
Piano
27
-
4186
B3 Organ
32
-
5920
Tenor Sax
110
-
587
Violin/Fiddle
196
-
2093
Blues Harps
196
-
2959
Baritone Voice
110
220
392
Tenor Voice
146
261
440
Alto Voice
196
392
698
Soprano Voice
261
523
1046
Tommy Rat is a legend in the live sound business and the survivor of literally thousands
of shows—many of them of the punk variety, which can be very tough. He is also a true
mentor to a generation of sound guys. And he has a great system for learning what a
given frequency ‘‘sounds like.’’
One of the ways I teach people to reference frequencies is to sit them down with a
microphone, a high-powered speaker, and an EQ. If you do this, and I suggest that
you do, you can turn up the volume until you get a controlled feedback. Once this is
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achieved, you can utilize the equalizer to find the frequency that is feeding back. By
positioning the microphone in different places, you will be able to create different
feedback scenarios and hone your skills for recognizing tones and applying them to
their corresponding numbers. EQing is basic math, and every tone has an assigned
number.
It is all about ‘‘ear training.’’ A good experienced engineer can almost instantly identify
a frequency that is feeding back. Honestly, I am not very good at it, so maybe
I appreciate the ability more than most.
If your housemates or neighbors are not excited about you blasting feedback to attain
this knowledge, and if you have an iPhone, look for an app called Dog Whistler. It is
made as a dog-training tool, but it generates frequencies and identifies them, which
makes it a good sound-guy training tool as well. Whatever tool you use, really knowing
your frequencies will make you much more valuable on any crew out there.
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D
Other Channel Stuff
epending on how ‘‘pro’’ the console is, there may be some other controls that
you need to understand.
Phantom Power
Depending on the level of the board, phantom power may be switchable on a perchannel basis in two or more groups of channels, or a single switch may turn on phantom power for the whole board. Generally, it is controlled by a switch labeled þ48.
As we discussed back in Chapter 3, condenser mics need power to work, and this power
is supplied to the mic from the console. (Active direct boxes also need phantom power.)
There are myriad issues with phantom power, mostly in the form of lack of standardization. Some boards don’t put out a true 48 volts, and most mics can use less. On lowerend, less expensive mixers, you are likely to find the phantom power switchable in
groups or globally.
You may have heard that using phantom power on a dynamic mic can damage it. Kinda
sorta, but that’s not really true. It can damage an unbalanced dynamic mic, but you are
unlikely to run into one of those. Same with ribbon mics. But the truth is that pretty
much any mic made in the past 30 years should be fine. But more things than just mics
go into your system.
Anything with a line-level output can be damaged by phantom power, and that includes
keyboards, drum machines, some bass amp direct outputs, and especially consumer
devices, such as CD players. Phantom power is only sent to the mic input, so using the
line input should negate any damage issues. Problems arise when someone tries to get a
1/4-inch (almost always line or instrument level) into an XLR mic input. Why? Well, if
the interface to the console is via a stage box and snake, it is an easy leap to use an
adapter to change that 1/4-inch connector into an XLR and run it right into the stage
box, right? Wait, what’s that noise? And what’s burning? And . . . well, you get the idea.
This is where a good direct box comes into play. The DI takes a line-level signal and
takes it down to an appropriate level for the mic input. It matches impedance and converts the line-level unbalanced signal to a balanced signal that can run much greater
distances without noise or RF (radio) interference. It also uses a transformer to isolate
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the voltage coming up the line and keep it from reaching the outputs of your line-level
device. There are a few companies making specialized direct boxes for taking an 1/8inch stereo mini jack (such as the headphone out on your iPod), converting it into two
separate signals (left and right channels), and sending them on their merry way over an
XLR connection.
Most direct boxes you will use are of the passive or unpowered variety. But for some
applications, an active (powered) DI is called for. The active DI gets its power from a
battery or from the console’s phantom power. The big difference between the two is
that the input impedance of the active DI is much higher than that of the passive DI, and
that higher input impedance preserves all of the upper-register harmonics that a passive
DI can choke out of a signal. This makes them especially good for things such as
acoustic guitars, where you want that top-end ‘‘sheen’’ to the sound.
Mute and Solo
It’s pretty straightforward—mute stops the signal from a channel from reaching the
output. Solo is also referred to as solo in place, and you need to be careful. It mutes any
channel that does not have a Solo button pressed. This can be useful during sound
checks and line checks to isolate a single input, but if you hit it during the show, the
audience will get treated to not the sound of a full band but, say, a really cool bass solo.
If you have buttons labeled either AFL or PFL, they are a lot more useful. AFL stands for
After Fader Listen and will isolate the signal after the fader so you hear exactly what is
going to the main section of the console. PFL stands for Pre Fader Listen, and it lets you
hear what is happening without regard to the actual fader level. The difference here is
that in most cases, the AFL or PFL will have no effect on what is hitting the master
section of the board. Rather, it is routed to the headphone output, which allows you to
listen to the isolated signal over the headphones. This can be very useful for adjusting
EQ or effects on a single input during a show.
One other thing: Many mixers feature direct channel outputs most often used for recording. Increasingly—especially in midsized mixers—these outputs are digital over
USB, FireWire, or ADAT Lightpipe. When those kinds of outputs are available, there
may be some kind of button or switch that lets you determine at what point in the signal
chain the direct out sends—pre- or post-EQ, pre- or post-fader, and so on.
Finally, the control that is (in my humble opinion) the most abused, misunderstood, and
ignored on the strip is the pan control. Too many sound operators are under the impression that because it is really only possible to get a true stereo sound in certain parts
of the room, they should just run in mono. Likewise, people playing stereo instruments
have a bad tendency to try to give you a mono signal. All of this is a mistake.
First, let’s address the instrument issue. The sounds programmed into a stereo keyboard, guitar processor, or whatever are made to be heard in stereo. Yes, they may have
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a ‘‘mono’’ out. (Usually either the left or the right output will also be labeled mono, and
when there is nothing jacked into the other output, you will get some kind of ‘‘summed
mono’’ signal that should have all of the audio info on one output.) But in reality, it just
doesn’t work that way—especially with sounds that include some kind of modulation
effects. You can’t get the swirling, motion-filled sound of a stereo chorus or even a
Leslie speaker in mono. Not gonna happen . . . It is like you are starting out with one
hand tied behind your back. Yes, if you don’t have stereo channels on your board, it
means burning two channels for one instrument. That is why you get the most inputs
you can afford when buying or specifying a console.
On the whole ‘‘sweet spot’’ issue: Yes, it’s a problem, and yes, only a small part of the
audience will get a ‘‘true’’ stereo experience. Big acts have gone to major expense to try
to alleviate this.
One approach is to do a standard left and right at the front of the stage, and as the
system moves to stage left (your right when you’re facing the stage) there is another
stereo set, but this one is reversed, with the right channel on the left and vice versa.
Why? People sitting closer to the corner of the stage might hear sound from both the
main speakers at the front of the stage and the ones at the side. By repeating the right
channel, that part of the image stays coherent as it wraps around the stage.
Using this approach, the image is flipped on the sides of the stage, but most of the
audience gets some kind of stereo sound. However, this is expensive and requires serious system design and acoustic chops that are way past the scope of this book. So why
should you bother with stereo signals and panning even on a simple system? It is all
about opening up the center of the sound image for the important stuff, such as the lead
vocal. It is not about extreme panning, just moving things around a little bit.
This is very much a matter of opinion, and there will be sound guys who will violently
disagree, but on a typical gig I tend to pan things a little bit based on where the player is
onstage to keep some kind of integrity to the overall sound image. Take a look at the
stage and try to move your pan knobs so that they basically represent the direction from
which the source sound is coming. If the lead guitar is standing to the right of the lead
vocalist, pan that channel a bit to the right. Pan your toms so that they move across the
soundstage as they are played. Ditto drum overheads, but not hard panned—just
somewhat off center. Stereo inputs such as a keyboard or stereo guitar processor get
panned hard left and hard right. So, what is straight up the middle? Lead vocal (or
primary instrument in the case of a non-vocal act), kick drum, and maybe snare. Everything else is panned just a little bit. Again, it opens up the middle for the ‘‘money’’
channel.
Another way that system designers and venues make sure the vocal is really out front is
to implement an LCR (left, center, right) system. This format allows you to place supporting instruments into the left and right outputs and reserve the center cluster for your
money channels.
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The Master Section
S
o you have all of your inputs in all of their individual channels at the right level,
EQ’d as needed and being sent where they need to go. So you’re finished, right?
Not quite. Next comes the master section, which can be thought of as a channel
all on its own. In other words, it is not just those master level faders that you need to
worry about.
Aux Returns
Any aux send that has the potential to run post-fader will generally have a return control in the master section because they have the potential to be used for effects. These
returns are often just level controls and determine how much of the signal that was sent
to the effect gets returned to the master section. In the case of, say, a reverb, the return is
the overall reverb level control. The channel sends determine how much of the dry signal goes to the reverb. All of those signals hit the reverb and are processed according to
the settings on the reverb unit. The output of the reverb goes to the return, and the
setting of the return determines how much of the effected signal goes to the master, not
back to the individual channel. If you have an effect that you want to send and return
directly to a channel, you use the channel insert, not the aux send and return.
Most mixers will, given sufficient channels, bypass the effect returns and send the effected signal back to another individual channel. There are several advantages to this
approach. The effected signal is another input that can be EQ’d and so on. Crucially, the
effected signal can be muted when the person at the mic is speaking between songs. That
killer reverb plug-in may sound great when the person is singing, but when he is
speaking, it tends to turn the voice into incomprehensible mush.
We have already established that your early gigs—at least the ones on which you are
allowed near the console—will not be with big touring acts. They will be with local acts,
maybe regional bands in clubs and such. We need to be reminded of that now because
on those kinds of gigs, you will rarely have someone else to mix monitors from a separate console, and you will be providing as many monitor mixes as you have pre-aux
sends for on your console. Instead of a return for these sends, you will find a master send
level that determines how much total signal gets sent to those monitors. Sometimes you
will get a switch or, if you are lucky, a fader labeled something like Aux 1 to Aux 3. This
allows you to put a little ‘‘grease’’ in the vocal monitor. But use this very carefully. If
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you are mixing for the house and monitors, you have your hands pretty full. That lead
singer may like the way his or her voice sounds drenched in reverb, but if you are using
stage wedges, just a little too much reverb in a wedge will put the feedback cycle into
gear.
Pretty Lights
The VU meters of old have given way almost completely to LED-based meters that go
from green to yellow to red as the signal they receive gets hotter. Any addition to the
signal level—be it from an individual channel, an aux return, or the level of the main
output—will affect this meter.
And this is one of the places where we have to think very differently depending on
whether we are mixing analog or digital. On an analog mixer, you want to mix at as
close to 0 or ‘‘unity’’ as possible. Given the dynamic nature of performance, this means
that the meter will inevitably hit the red a bit from time to time, and that is acceptable.
But when the mixer is digital, the standard is closer to –6 dB from unity as your average
point to ensure that digital clipping never happens. Why the difference? Analog distortion is simply not as hard on the ears as digital is. In fact, in the analog world, what
many musicians and engineers refer to as ‘‘warmth’’ in the system is really just a little bit
of distortion, and to many ears it can make an overall mix sound better.
But digital distortion is a different beast, and no one who has heard it would refer to it as
‘‘warm.’’ It is a very unpleasant sound that can be best likened to (sorry for the scatological reference) an electronic fart. Mixing at –6 dB makes this sound much more
unlikely.
Other Miscellaneous Stuff
In this section, you will also find tools such as your headphone level, two-track input,
and outputs, with level controls for each. You will also find a switch and level control
labeled Talkback. Some semi-pro boards actually have a mic built right into the console
next to this control, but usually you will need to plug a mic into the Talkback input to
use this feature. And it can go a long way in saving your sanity and your voice. The band
can always communicate with you by speaking into a mic. Talkback gives you the same
ability. Assign it to an aux send, engage the switch, and speak into the mic, and the act
should be able to hear you through the vocal monitors.
Anything that fosters communication between the act and those mixing the sound is a
good thing, and some acts make it a standard part of every show. Jimmy Buffett sings
through a normally routed vocal mic but wears a wireless lavalier mic in addition. That
mic is routed only to the monitor engineer and specified band members. It allows Buffett to step away from the vocal mic and communicate directly with the monitor engineer. Billy Szocska, who has been doing Buffett’s monitors for a few years, says that
when he first started, he made the mistake of turning off that lav mic’s output. ‘‘I did it
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The Master Section
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once,’’ he said in an interview and then repeated, ‘‘Once’’—with an emphasis and body
language that made it very clear that doing so had not made the boss happy. Billy will
tell you that in many ways, that lav mic is the most important thing onstage. At least it is
to the guy who signs the checks . . .
Splitting It Up
So far our talk of the master section has focused on bringing everything together, but
there is another set of controls that kind of lies between the channels and the mains.
These are either subgroups or VCAs (voltage controlled amps), and though both serve
similar purposes, they do so in very different ways. And in fact you may actually have
both on a console. And as long as we are talking about grouping channels—which is
what both subgroups and VCAs do—we should talk about something we left out in the
channel strip section. Remember we talked about the mute control? Some consoles will
allow you to assign multiple channels to mute groups. Why? Let’s say you are mixing an
act with three background singers and a horn section. However, those voices and instruments are not used in every tune, and sometimes they may actually leave the stage.
Assigning all of the backup vocal mics to a single mute group gives you the ability to
mute all of those mics at once, making for less chance of missing one or neglecting to
turn one back on. Okay, back to where we were before this little detour . . .
A subgroup is just what it sounds like: a feature that allows you to assign multiple inputs
to a single group. So you can, say, put all of your drums in a single group, and once you
have the relative levels right, you can raise or lower the level of the entire drum kit
without screwing up that relative balance. For most purposes subgroups will work just
fine in a live sound setting, but there are exceptions. Let’s look at those backup singers
again. Suppose you want to fade them out of the mix. Grab the subgroup fader and
bring it down slowly, and it all fades out, right? Not quite. Remember we talked about
post-fader aux sends used for effects? Well, unless you have multiple reverb units
available and are running a dedicated insert on the subgroup, then fading out the subgroup will fade the primary signal, but because it only controls the main outputs of the
channels in the group, the reverb on the vocals will continue to flow into the main mix.
This might be a cool effect, but it is likely not what you are looking for. Or, say you are
running your subs on a post-fader aux (an increasingly popular practice that allows you
to easily control the sub-bass level from the console); with a subgroup, changes made to
a group will not affect the amount of sonic info going to the subs.
VCAs are a whole different animal. In fact, they actually start back at the channel strip.
VCAs replace the potentiometers at the channel strip. Sort of... The fader is still a pot,
but no audio passes through it. Instead, the signal goes to an amplifier, the output of
which is determined by the voltage applied to the amp. So in a VCA console, the fader is
actually controlling the voltage sent to the amp, which raises or lowers the level of that
channel.
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In turn, you can assign multiple channels to VCA groups, which serve the same function
as subgroups but, again, do so without any audio signal actually passing through the
fader. Instead, the voltages of the channel VCA and any VCA groups are summed. So if
your channel is set at þ3 dB, and the VCA group is set at –5, then the actual output of
the channel is –2 dB from unity.
A little too geeky? Let’s look at the practical differences. We already looked at the example of a fadeout using a subgroup. In the same scenario using VCAs, as the voltage
from the VCA group is lowered, so is the voltage—or level—of the channels assigned to
that group. This means that any post-fader aux is affected as well. Think of it like this:
With subgroups, the channels assigned are combined into one signal, the level of which
can be controlled via the subgroup fader. But with a VCA, because the group control is
actually affecting the output voltage of the channels assigned to it, the result is the same
as if you grabbed all the faders of all the channels assigned to the VCA group and moved
them up or down by the same amount.
If it helps, visualize the VCA master actually moving all of the faders in the group. In
effect, that is what is happening here.
And remember that in most cases, a channel can be assigned to more than one sub or
VCA group. This can make your job a lot easier once you have things set up. For example, the drum inputs all go to the same group. But you can also route them to another
subgroup that includes all of the instruments and no vocals. That way, you can easily
raise or lower the level of the drums or turn the whole band down when they start to
overwhelm the vocals.
Finally, your groups will likely not have their own EQ or aux sends, but there is probably a group insert jack on the back of the console where you can add an effect or other
processor and have it appear only on that group.
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Gain Structure
B
efore we move away from the console and on to the drive rack, amps, and
speakers, I want to take one more look at proper gain structure. Steve LaCerra
has his own record label, is a very good studio engineer, and has been the FOH
engineer for classic rocker Blue Öyster Cult for more than a decade. I have been fortunate enough to have him write for me in several different forums, including the Live2Play Network, Front of House, and GIG. So here is his take—from someone who
does this every day and deals with a new PA every day. There is a ton to learn here. Some
of it will repeat ideas we have already touched on, but they are laid out here in a way
that is eminently usable. Thanks, Steve.
An extremely important aspect of sound system use, gain structure is the red-headed
stepchild of audio: It’s often misunderstood and typically neglected until there’s a
problem. Let’s try to demystify the idea of gain structure so that you can get on to
making music with the best sound quality possible.
What Is It?
Gain structure refers to the manner in which signal levels are set in (and between) the
various sections of an audio system. Sounds easy, right? Wrong. Turn a level control too
high, and you’ll have distorted audio. Set it too low, and you may find that your system
doesn’t play loud enough or that you can’t get sufficient level to tape. Symptoms of poor
gain structure include noise of the hissing type (as opposed to hum or buzzes), distortion, lack of headroom, and grossly mismatched readings between the meters on different devices used in your system. It may lead you to believe that a piece of gear is
malfunctioning or that you have a bad cable in the chain. When gain structure is set
correctly, you’ll get every last dB out of your PA, you’ll record cleaner tracks, and all of
a sudden your digital processors will have a better signal-to-noise ratio.
ABC’s of the Signal Chain
One of the most basic things you can do to ensure proper gain structure is to make sure
your sound sources are plugged into the right holes at the mixing console. There’s a
reason for separate mic and line inputs on your mixer: Line-level signals (such as those
from effects devices, tape machines, keyboards, and drum machines) are much stronger
than microphone signals. This is why manufacturers usually use different types of
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connectors for mic and line inputs. (Note that ‘‘tape’’ inputs essentially have the same
gain characteristics as line inputs.) A mic input incorporates an extra gain stage to boost
the microphone’s feeble signal up to something more usable. Line inputs are less sensitive, so when using an XLR-to-1/4-inch adapter cable to plug a mic into a line input,
you’ll have to crank the gain way high just to hear the mic. (This adds more noise.)
Conversely, if you plug a keyboard into a mic input, you’re probably going to hear
distortion because the signal from the keyboard is strong enough to overload the mic
preamp. Those are examples of poor gain structure.
Audio by Numbers
Once you’ve ascertained that the devices are plugged into the correct jacks, it’s a good
idea to check their operating levels. There are variations in line level, most notably those
referred to as þ4 and 10. Although the boundaries have become blurred in the past 10
years, most professional audio gear operates at þ4, while most semi-pro and consumer
gear operates at 10. (Technically speaking for you tweak heads, it’s þ4 dBm and 10
dBV, but we’re not gonna go there.) A general clue to operating level is the type of jacks
on the rear panel: If the jacks are RCA, you can be 99.44-percent sure the gear runs at
10. If the gear has XLR connectors for the line inputs and outputs, it’s almost certainly
þ4. If 1/4-inch jacks are used, you’ll have to get out the manual and read the fine print.
While you are at it, pay attention to whether the 1/4-inch jacks are TS unbalanced or
TRS balanced.
It’s important to understand how þ4 and 10 gear reacts when interfaced together, so
here’s an example. You patch a consumer-style CD player into a mixer. The CD player
has RCA jacks, and the mixer has 1/4-inch jacks, so you buy or make an adapter cable
to connect them. But when you listen to the CD player through the mixer, the level is
really low. To make it as loud as your drum machine, the faders have to be pushed way
up. This is because the CD player operates at –10 and the mixer operates at þ4. The
mixer is expecting to receive a stronger signal level. When it doesn’t, you have to crank
up the faders (which generally means more noise). What’s the solution? Look on the
mixer for a switch that changes the operating level from þ4 to 10. Doing so will make
the line input more sensitive, and you won’t have to open up the faders so much.
(Matching tape machine output levels to tape inputs in this manner is crucial to clean
mixes.) Some gear has two sets of input or output jacks for exactly this reason. If you
can’t adjust the operating levels of two pieces of gear to match, consider getting some
sort of level-matching interface. A good example is the Matchbox from Henry Engineering. It converts 10 audio on RCA jacks to þ4 audio on XLR jacks and vice
versa. Similar devices are available from Ebtech, Whirlwind, and other manufacturers.
Matching operating levels is particularly important when using compressors. Let’s say
you have a compressor patched between your mixer and your power amp in order to
prevent the power amp from being overloaded. The idea is that when the mixer starts
putting out excessive level, the compressor will compress, protecting the amp and
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speakers. Well, for a compressor operating at 10 (semi-pro), a þ4 signal from the
mixer looks like excessive signal, when in fact the mixer may not be putting out much
signal at all. This limits (HA!) the maximum drive to the power amp, putting a cap on
the amount of volume you can get in the room. The solution in this case is to look for a
10/þ4 switch on the compressor and set it to match the mixer.
The Microphone and Other Delicacies
Proper gain structure on a microphone is critical to clean sound because mics put out
such weak signals. Think of a mic signal as water flowing through plumbing. Much like
plumbing, audio consoles have a series of ‘‘valves’’ that influence the signal flow. If you
require water pressure sufficient to reach the fourth floor, you have to check several
valves. The most important one is the valve on the main water pipe entering the building. If the main valve is closed down, you can open up every valve feeding the various
hot and cold lines throughout the building, you can open up every faucet in every
bathroom and kitchen in the entire place—but as long as that main valve is closed,
water will not reach the fourth floor. The mixer channel’s mic trim knob is the equivalent of the water main. You must get the correct amount of level at the trim (or gain
control) in order to safely deliver the mic signal to the rest of the chain. You can boost
the fader up as high as you want, but if the trim is off, you’ll get nothing but noise.
Conversely, if you have the trim way up and the fader way down, chances for distortion
are much higher.
Depending upon the mixer, there are several ways to measure the mic signal. A popular
feature on many consoles is the PFL (Pre-Fader Listen) meter. Generally, pressing a
button labeled PFL on the channel switches the mixer’s main meter to show the level of
this one channel before that channel’s fader. In other words, it’s letting you measure the
water pressure right after the main valve but before the kitchen faucet. If you set the
level incorrectly here, you’re practically doomed to a career of distortion or noise. Adjust the trim knob while watching the meter. You can raise the trim until the meter reads
0, but remember this: Other microphone signals must make it into the audio ‘‘plumbing’’ during the mix, so leave a bit of headroom by PFLing the signal at roughly 7 to
5. When you start combining signals, you won’t overflow the main mix pipe. Since
adding EQ will likely change the PFL signal, allow a bit of room for that as well. If you
have the trim all the way down and the PFL signal is still way over 0, look for a pad
switch on the channel and use it; this will lower the sensitivity of the mic preamp by a
fixed amount, reducing the possibility of distorting the signal (sort of like narrowing the
water main).
Variations on this type of metering include ‘‘solo,’’ as implemented on most Mackie
analog consoles. The trick here is knowing that this type of solo does not show pre-fader
level, so the fader must be set at unity, or you will not get an accurate reading of signal
level at the input stage. On some consoles this spot is marked with a 0 or a small arrow.
This is the spot where the fader is putting out exactly what it is receiving, neither
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boosting nor cutting the signal. Other consoles might have a simple two-color LED with
green for signal present and red for overload. In this case, adjust the trim until the LED
barely shows red and then back it off by about 10 to 15 percent. Since some consoles
have more headroom than others, you’ll have to experiment to see how far you can push
the trim before distortion occurs.
Once the trim is set, you can bring up the channel fader to hear the signal. At least some
of the channel faders should be at or near the 0 mark; if all the faders are very low or
very high, something is wrong with the gain structure. Keep in mind that other ‘‘valves’’
affect the audio signal, such as the main mix fader(s), which should also be set at or near
0. If setting the master at 0 makes the volume in the room too loud, turn down the level
controls on the power amps. If you need to bring the master fader all the way up to get
adequate volume in the room, either the power amps are set too low or your system is
underpowered.
When sub-mixing channels (10 channels of drums to a stereo pair of subgroup faders,
for example), similar concepts apply. Think of a subgroup fader as a hot/cold mix valve.
For the mix valve to operate properly, you need correct pressure of hot and cold water
before mixing. Try using the kick drum channel as a reference, setting its fader to 0, and
then mixing the rest of the drum channels in to taste. If the mixer has a PFL switch at the
subgroup fader, use it to measure the flow right before the subgroup fader. A subgroup
PFL showing in the red will probably sound distorted no matter how loud or soft the
drums are in the mix.
Gain structure is equally important when using aux sends to route signal to effects such
as reverb or delay. Some mixers have a PFL on the aux send. Use it. Measure the level of
that snare drum send before it hits the reverb unit. Turn up the master knob for the send
and watch it hit the meter on the reverb. If the reverb has an input level control, turn it
up until the meter hits red and then back it down a bit. At this point it doesn’t matter
what the reverb sounds like; just get the level right. Then PFL the mixer’s effect return to
set the output level of the reverb as well as the trim on the mixer’s effect return (if there
is a trim). Once the levels have been set, bring up the effect return fader (or knob) to add
the sound of the effect into your mix. With digital effects, correct gain structure is extremely important because if you set the input too low, you won’t get the full benefit of
the unit’s A/D/A converters.
Once you get into good gain structure habits, you’ll find that you have more system
headroom, better signal-to-noise ratios, and cleaner mixes. Of course, the rules can be
broken, but first it’s a good idea to learn the game.
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Aux Sends and Returns
Y
ou should now be pretty well versed on the entire signal chain, from the point
on the console where the sound enters to where it exits. We still have amps and
speakers to deal with, but for now let’s take a little road trip into the Land of
Effects. Beware: It is easy to be seduced by the pretty things native to this part of the
audio world, and some of those pretty things have big, sharp, nasty teeth. So stay on the
marked trail and follow your guide at all times. Let’s go . . .
Hello? (hello . . . hello . . . ) Is There Anybody in There? (in
there . . . in there . . . )
First, let’s talk a bit about where effects get inserted in the signal chain. More than 90
percent of the time, effects units will be fed a signal from an aux send on the console. Pro
mixers will also offer ‘‘insert’’ points on each channel. So, if you have an effect or a
dynamic processor that you want on only one channel, there is no need to burn an aux
send. You just use the insert to put that effect on the channel where you want it and
nowhere else. If you have an insert out and an insert in (known as a dual point insert),
you use two cables and can keep everything balanced and pro. But most consoles you
will use feature single-point inserts, which share a single jack for both input and output.
To use this type of insert, you will need an insert cable. This is a Y-cable with a stereo 1/4inch connector on one and with each arm of the Y terminating in a mono unbalanced
1/4-inch connector (see Figure 14.1).
The signal flows out of the channel via one part of the connector (usually the tip), which
is wired to one of those mono connectors. It goes through the inserted processor and out
to the other mono 1/4-inch connector, which is wired to the second conductor on the
TRS (usually the ring portion). The last part of the TRS—the sleeve—is a common
ground for both of the mono 1/4-inch connectors.
Back to the aux sends . . . Remember, there are two kinds of aux send—pre-fader and
post-fader. Monitors use the pre-fader sends (you don’t want any adjustments you
make on the main fader to affect your monitor mix), while effects need to be a part of
the overall sound, so they are affected by main fader moves.
There are a couple of ways to get the sound from the effects box back into the main
signal path. One way is to use the returns that are dedicated to this purpose. The other
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Figure 14.1 A typical 1/4-inch insert cable.
way is to run the FX output into a pair of unused channels on the console. We’ll get into
why you might choose one way over the other in a moment. For now, let’s take a look at
the kinds of effects you might use in a live setting. They fall into two main types: timebased and modulation effects.
Time-based effects are generally understood to mean reverb and delay or echo. The
truth is that these are pretty much the same thing; it’s just a matter of degree. A delay or
echo replicates the sound of, say, your voice, bouncing off of a surface so you hear it
again. Reverb replicates the numerous very short echoes that make a room sound the
way it sounds. These short echoes are called early reflections, and the number, timbre,
and volume of these reflections make up the sonic signature of a space. Reverb is
sometimes used to make something sound bigger and more dramatic. It is best used to
make a sonically dry room sound livelier. It can also be used (or overused) as an effect—
such as on the vocal on a big ballad or the sound of a classic surf guitar song. Just so you
have your terms straight, reverb is often referred to as ’verb or grease.
Echo is a more distinct repeat of the original sound. Again, a little goes a long way, and
a good sound person who is familiar with the material being played will often ‘‘feather
in’’ a bit of delay at the end of a vocal phrase and then back off again as a new phrase
begins. Echo is sometimes called slap. So, if you are dialing in a mix and you get asked
for a short slap and a little grease, it means a quick echo and some reverb.
Getting Swishy
Modulation effects are the ones where something about the signal changes or modulates
at a certain speed. This can mean anything from volume (tremolo effects usually found
on guitar amps) to pitch (chorus) to time (flanging) to phase (phase shifting). All of these
effects result in some kind of moving and usually ‘‘swishy’’ or ‘‘swirly’’ sound. Chorus is
also used to fatten a thin sound and can be used to good effect on backing vocals.
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When old guys like me were coming up, most acts were lucky to have some kind of
spring-reverb effect—like those on guitar amps—built into the console or PA head. As
the world has gotten increasingly digital—and the cost of digital signal processing (DSP)
is now measured in pennies rather than hundreds of dollars—we have seen the birth and
growth of the multi-effect unit, which can generate two or three or a dozen effects all at
once from one box.
It has actually gotten even easier to add effects as the sound world becomes increasingly
digital. On digital consoles, those sends and returns are virtual, not physical, and the
box that houses the effects is a piece of software inserted in the signal path, called a
plug-in. Most plug-ins are made to emulate the sound of classic processors from the
past, and there is no quicker way to start a sound-guy fight than to say that a plug-in
sounds exactly like the gear it is trying to emulate. Although really good plug-ins sound
really good, there will always be those who really can hear the difference. I have seen
many major touring and installed rigs, and I often see an external unit patched across a
single vocal channel or across the whole mix. I know that there is likely a plug-in on the
console that emulates the hardware unit in the rack, but we use what we are comfortable with and what the client needs/wants.
The major advantage of plug-ins (in addition to the fact that being nonphysical means
they will not likely break down in the middle of sound check or even the show) is that
you can insert multiple instances of the same plug-in on different channels with separate
settings on each channel. That kind of power and flexibility can be a wonderful thing,
but it can also get you in boatloads of trouble very quickly. This is where the old adage
‘‘just because you can does not mean you should’’ takes on real meaning. Excessive
reverb, delay, chorus, or whatever will muddy up your sound and make it hard for the
listener to do things such as understand what the lead vocalist is singing.
Clap On, Clap Off
That brings us to some almost logistical ideas that are important to getting and keeping
a good overall sound.
The first is that whatever effect you are using may sound great when the band is singing,
but it probably sounds a lot less great when they are trying to talk to the crowd to set up
the next tune.
This brings us full circle to how the effect gets back into the console.
Any aux send intended to be used for FX will have a matching return with its own
volume knob. This is a kind of master volume for the FX—the send adjustment determines how much of the signal from a given channel goes to the FX unit, while the
return adjustment controls how much of the entire output of the FX device goes back to
the console. The aux send on the individual channel determines how much of that particular signal goes to the FX unit.
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You may opt not to use the returns and instead bring the FX output back to the board
via an unused channel. The advantages of this method include the fact that if the FX
output is coming back to a channel, it can easily be sent to the monitors so the performers can get a little grease without you having to buy a second FX unit. It is also an easy
thing to grab a channel fader and bring it down between songs—much easier than
finding the sometimes-buried return knob. Returning the wet signal through a channel
also allows you to EQ the effected signal without running a separate EQ unit specifically for the FX.
Dynamics—Compressors and Gates
When you hear a really great performer or band that understands dynamics (gasp!),
they pull you into the song by going from soft to loud and back to soft again. Well, a
compressor does just the opposite. It squeezes, or compresses, the dynamic range of an
audio signal. In simple terms, it does so by lowering the louder parts of a performance,
allowing you to bring up the overall level and make the softer parts more closely match
the loud parts. Think of it as a kind of automatic volume knob.
Most compressors have five controls—threshold, ratio, attack, release, and output or
make-up gain. Threshold determines at what level the compression kicks in. Signal
below the threshold passes unaffected. Ratio determines how much squeezing a signal
gets. A ratio of 3:1 means that for every 3 dB of signal over the threshold, 1 dB of sound
is allowed to pass. The higher the ratio, the more compression is being applied. Attack
refers to how fast the compression kicks in, and release refers to how quickly it lets go.
When you apply compression, you will usually lower the overall gain as you are bringing down the peaks, so make-up gain allows you to get the level back to where you
started.
In general, the lower the ratio, the more natural the sound. A ratio setting of less than
3:1 is often used to add punch or to tighten a performance or a recorded track. (These
are totally subjective terms, but they are impossible to really define. Your best bet is to
play with a compressor and hear it for yourself.) Higher ratios are used to get the
dynamics of a performance under control. Settings higher than 10:1 are called limiting
and are most often used as system protection to keep ‘‘Loud Larry’’ from blowing up
your speakers.
Sounds with hard transients, such as a snare drum, need a slower attack to avoid cutting
off the initial crack of the drum and making it sound unnatural. Fast attack and release—combined with a ratio over about 4:1—can result in breathing or pumping. This
is when the loud sound is quickly compressed and then released, and the subsequent
sound passes under the threshold and does not get compressed. You can really hear the
compressor working, which is something you never want to happen. As a rep from a
major high-end compressor manufacturer once told me, ‘‘If you can hear a compressor
working, it’s set wrong or it’s broken.’’
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When to Use It
I am one of those guys who believes that as little as possible is always best, but it really
depends on the act, the performance, and what you are going for sound-wise. Really
good players who really listen and adjust their playing to the others onstage with them
don’t usually require compression. I once asked the sound guy for Tower of Power, Ace
Baker, what kind of compression he used on the horn section, and he just wiggled his
fingers. Those guys are so dialed in that they ‘‘compress’’ themselves. Ditto a lot of oldschool singers who move toward and away from the mic as their performance gets
softer and louder. It can be overdone, but when it’s done right, we call it ‘‘good mic
technique.’’
A common use for a compressor in a live setting is to get a soft singer or one with a very
wide dynamic range up above a loud band. Another common use—one that sound guys
will argue passionately about—is to put a good stereo compressor with a very light
setting across the L-R outputs of the system to smooth things out a little bit. But this
means you use a good (read: expensive) compressor that brings something to the party
in terms of tone. Typically, we are talking about tube compressors here, and they are
not cheap.
A compressor can be used to bring up a weak kick drum (but be aware that using a
compressor to bring up a weak signal means you also bring up the noise floor) or to
tame an out-of-control snare drum. It is almost always used on the bass, and many bass
amps have a compressor built in. (Be aware that any tube amp will produce a bit of
‘‘natural’’ compression—a big part of the tube sound.)
Someone’s at the Gate
Compressors are part of a class of devices known as dynamics processors. The other
most common devices in this class are limiters, which we already touched on, and noise
gates.
A noise gate is used to clean up a ‘‘dirty’’ signal and keep the noise at bay during quiet
moments or, more commonly, to ‘‘turn off ’’ a mic when it is not being used so it does
not pick up signal from other sources around it and muddy up the whole sound.
Drums—especially toms—are often gated so the mic is only feeding signal when the
drum is actually hit. In very loud settings, it not only cleans things up, but it also lowers
the probability of feedback in the drum mics. Again, be careful with the release settings
to keep it from sounding unnatural. Also, a gate on a lead vocal mic will help keep the
drums out of that channel when the singer is not at the mic.
There are a bunch of decent entry-level mixers—most notably from Yamaha—that have
built-in one-knob compressors that allow you to dial in how much you want to squeeze
the signal, and through the magic of digital signal processing, it makes the other adjustments for you. There are also some very good compressors with presets for different
kinds of inputs, including the PreSonus BlueMAX and the TC Electronic C300. Those
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are good places to start if you are just beginning to ‘‘get’’ compression. Outside of the
preset world, dbx is practically synonymous with compression, and some processors
with other brand names actually use a dbx chip in the compressor part of the circuit.
The Secret Is Out In our American Idol world, there is a use for noise gates that
many pro sound guys don’t know about but that is common on pop acts you think
might be lip-synching (and who probably are).
It works like this: Slight discrepancies between the movement of the singer’s mouth
and the track to which he or she is ‘‘singing’’ are not very noticeable to most people.
But when the singer stops and the vocal track continues, you have what we call in
sound-guy tech jargon ‘‘a problem.’’ If you want a great example, go to YouTube
and search for Ashlee Simpson and SNL. A classic crash and burn of a lip-synching
singer . . .
Since that incident, some folks have figured out a way to make sure that kind of thing
never happened to their clients. A noise gate is placed in the path of the recorded
vocal track and is controlled by the input of the ‘‘singer’s’’ mic. So as long as sound is
going into the mic, the gate stays open, and the recorded track plays. When the
singer stops, the gate closes, and the recorded sound stops. Pretty slick, huh? In other
words, the singer may actually be singing, but it is not always what the audience is
hearing.
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Monitoring
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s long as we are dealing with auxes, we’re going to take a big side trip into
Monitor World. Remember, I said there are two kinds of aux sends—pre- and
post-fader. This refers to where in the signal path the signal is diverted into the
aux. As discussed in the previous chapter, effects and processing should be affected by
any adjustments to the main channel fader, so the aux is post-fader. Auxes that get sent
before the fader (pre-fader) are generally used for monitoring, although they may be
used to send a mix to a place other than the stage, including auxiliary rooms in venues
such as churches and schools or backstage in dressing rooms.
Most consoles designed to be used as mixers for front of house will have between two
and four pre-fader auxes, and in many smaller gig situations, you will find yourself
running both the house sound and multiple onstage monitor mixes. (On some desks—
especially some made in the UK—monitors are referred to as foldback.) On these kinds
of gigs, if it is an analog console, you will likely deal with two or three mixes—probably
one for the lead vocalist (the ‘‘star’’ mix), a general band mix, and maybe a special for
backup singers, a second lead vocalist, or even a drum mix.
When we move into the world of digital, it gets more interesting. Even the smallest of
digital desks designed for live sound will usually have many more outputs than their
analog counterparts, and I know of many house engineers at venues in Las Vegas that
regularly mix the house plus eight or ten monitor mixes. With the growing adoption of
in-ear personal monitoring, a large number of monitor mixes has become more necessity than luxury for many artists. A ‘‘good enough’’ monitor mix is one thing when it
is coming from a wedge at the performer’s feet, but it quickly becomes not good enough
when it is being pumped directly into the performer’s ears.
In my experience, the job of the monitor engineer is much more difficult than the job of
the FOH mixer. Not only is the monitor engineer juggling multiple mixes, he is also
working much more directly with the artist, and the pressure can be intense.
The Gear
A specialized monitor desk (in the analog world) is generally known as a matrix
mixer—every input can go to any output in varying levels. For an example, take a look
at Figure 15.1, which shows a small monitor mixer that can be rack-mounted.
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Figure 15.1 Crest X20RM monitor mixer. Image courtesy of Crest Audio.
Starting at the top of a channel strip, you will see controls that are familiar—the trim
pots, a high-pass filter, and four channels of EQ with a sweepable mids. Next, we have
levels for each of the 12 sends. On the right-hand side of the console are 12 faders that
provide master output level for each of the sends. Each pair of outputs can be linked for
stereo operation if you are doing in-ear monitors. When a pair of outputs is linked, they
act as a single output. In other words, if you were to link all of the pairs, you would have
six outputs. So if, for example, you have three vocalists in the band all using personal
monitors, you could provide each of them with a stereo mix and have eight more left for
mono wedge mixes.
The inputs and outputs on this console are located on the rear panel. In addition to the
expected channel inputs and outputs for each of the 12 mixes, there is an output
matched to each input. On a dedicated monitor board, you will find an onboard split
like this. The idea is that sources come from the stage to the monitor console and then
via the split to the front-of-house console.
Each output will also—most likely—have an insert point that will probably feed a 1/3octave EQ. The biggest issue in monitors, aside from providing a mix that meets the
requirements of the artist onstage, is avoiding feedback from the stage wedges. The 31band EQ is used to lower the output of the specific frequencies causing the feedback—a
process known as ‘‘ringing out’’ the monitors. This is one area where ear training becomes crucial. If you are going to ‘‘call’’ the ringing frequencies, you have to be able to
identify them.
Do You Hear What I Hear?
To be anything like effective, the monitor engineer has to be able to hear as close to
exactly what the performer is hearing as possible. This is where the cue wedge comes
into play. The cue wedge is a stage monitor—the exact same kind as is on the stage—
that is set up for the monitor engineer. If you are stuck mixing monitors from the front
of house, you will still find a cue wedge very handy.
Very simply, the mix you are crafting for the performer is played for you through the
cue wedge so that the two of you have a common reference point. Few performers can
effectively communicate what they want in a monitor mix—but the ones who can are
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adamant about what they want—so being able to hear what they are hearing can make
the process much quicker and simpler.
The increasing use of personal monitors has made the job of mixing monitors harder in
some ways. (Side note: Personal monitors are also known as in-ear monitors or even just
ears, but those two terms are trademarked by one manufacturer—Future Sonics—so
although they are commonly used terms in the field, for publication we refer to them
generically as personal monitors or PMs.) Although there is less likelihood of feedback
than there is with stage wedges, the mix is more crucial. As someone who still performs
and who uses personal monitors, I can attest to the fact that a marginal monitor mix is
one thing when it is coming out of a wedge at your feet, and it is a whole other thing
when it is playing inside your head.
And because a good mix and a bad mix are totally subjective in Monitor World, that
means you get to do more mixes, because everyone hears differently, and no one wants
to share a PM mix with someone else. Another thing: If you mix a lot of monitors, do
yourself a favor and invest in at least a few sets of professional PMs. When you are
mixing PMs, you will almost always—if possible—want to use the same make and
model that your ‘‘star’’ is using so that you hear the same things he or she hears. Most
touring monitor mixers I know have a half-dozen or more sets of PMs, because every
time the client updates or changes, they need to do the same.
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The Drive Rack
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efore we get into this section, we need to note that we are talking about a generic
set of components that together handle some processing that happens postconsole and pre-power amp. For most of the history of the live sound biz, each
process was handled by a separate component; put together, most of us referred to that
group of components as a drive rack. Much changed as digital signal processing (DSP)
got more powerful and cheaper, and today all of those processes are usually handled by
a single component (and in some cases, that DSP is built right into the power amp).
Making it a bit confusing for some is the fact that one major audio manufacturer has a
line of processors called DriveRack. So just to be clear, we will be looking at each of
these processes as though each was handled by a separate piece of gear. Yes, that idea is
hopelessly outdated, but I find it the best way to make sure that inexperienced sound
techs really understand each process. The current crop of what we generically refer to as
speaker processors are really powerful and, if used incorrectly, can not only make your
show sound bad, but also can seriously damage the gear ‘‘downstream.’’ In other
words, you can blow up amps and destroy speaker drivers. We are going to take this
approach to make sure that does not happen on your watch.
The components that process the signal after the console are largely the same as ones
you would find on the console or inserted on an individual channel. However, they
serve very different purposes. We are going to start with the one you won’t find on the
console—the crossover.
You will find crossovers as passive devices inside a ‘‘full-range’’ speaker cabinet, as
stand-alone units, and as part of a speaker processor. All serve the same purpose—to
split source audio into multiple signals based on frequency. In a full-range cabinet, the
crossover really just serves the purpose of protecting the high-frequency driver. The
signal enters the cabinet and is split in two (or three in the case of a three-way cabinet),
and a network of old-school passive electrical components (you know, resistors,
capacitors, transformers?) keep the low-frequency parts of the signal from reaching the
high-frequency driver. Most passive crossovers send the full signal on to the low/mid
driver. This kind of crossover is known as a high-pass filter. It passes frequencies above
a set point (most passive crossovers are not adjustable) and blocks the rest. So what
happens to that part of the signal? Remember that the audio signal is energy, and—basic
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physics—energy can be neither created nor destroyed, but you can change its form. In
this case, the low-frequency energy is converted to heat.
The problem with this approach is simply the wasted energy—a full-range cabinet employing a passive crossover will require more amplifier power to achieve the same output as a cabinet in which the individual components are powered by separate amps.
Doing this requires a crossover that comes before the amp in the signal chain. There are
analog passive devices that can serve this purpose, but the chances of you coming across
one are not good. In fact, the chances of running into an analog crossover at all are slim.
But analog or digital, the idea is the same. A crossover basically acts as a traffic cop for
audio. It stands in the middle of the street and tells the stream of cars which way to go.
Order, Order, Order!
Let’s get some definitions of different kinds of crossovers out of the way. If you really
want to get into the math involved, there are some great online resources, including the
audio guide at rane.com (always a great reference for pretty much anything audio). For
our purposes, we are going to keep it a little less math-class and a little more real-world.
A crossover is a filter, and you will see filters referred to in terms of ‘‘order’’—first
order, second order, and so on. Again, there is a bunch of math here, but the easiest way
to think about them is in terms of how steeply they roll off frequencies above and below
the crossover point. A first-order filter is limited to a 6-dB-per-octave slope, which is not
adequate for most crossover purposes. Most crossovers you will find in an audio system
are second- and third-order filters, which roll off at the rate of 12 dB and 18 dB per
octave.
The other terms we need to understand are kinds of passes. A low-pass filter passes the
frequencies below a set point and filters those above. A high-pass filter does the opposite
and, as discussed earlier, is what you will find built into most full-range passive speaker
cabinets. A band-pass filter allows frequencies between two set points to pass and filters
those above and below. You can get a band-pass filter by combining a low-pass filter
with a high-pass filter, and that is the best way to describe most crossovers—the combination of a high-pass with a low-pass filter. (The area where the two filters converge is
often called the sum.) A very narrow band-pass is also known as a band-stop, bandrejection, or notch filter and is most often used to control feedback.
Now reverse some of those words, and you get into the guts of the filter. The pass band
is the part of the signal that is allowed to—you guessed it—pass through the filter, and
the stop band is the point where the filter kicks in. In theory, you would think that your
high-pass and low-pass filters would be set to the same stop band, but in practice those
filters usually overlap to some degree.
You will also see crossover filters referred to by the names of the people who defined
them. Linkwitz-Riley, Butterworth, and Bessel are the most common. A Linkwitz-Riley
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filter is popular because it is the flattest in the pass band, while a Butterworth filter will
give you about a 3-dB bump at the crossover point. The Bessel and Linkwitz-Riley are
the most similar in that they are both very flat, but the Butterworth has the sharpest
initial cutoff. The Linkwitz-Riley has moderate roll off and a flat sum. The Bessel has
the widest, most gradual crossover region and a gentle dip in the summed response. The
graph in Figure 16.1 gives a good picture of what the response of Butterworth and
Linkwitz-Riley filters looks like when the high-pass and low-pass functions are
combined.
Figure 16.1 The LR line represents the Linkwitz-Riley filter, and the Butterworth line is the
Butterworth. Note the area in the middle of the graph. This is called the crossover region, and the
two filter types give you two different results. The Linkwitz-Riley filter is flat in the crossover
region, while the Butterworth results in about a 3-dB ‘‘bump’’ in the same region. Illustration by
Erin Evans.
The graph in Figure 16.1 shows the response in a two-way system. But the truth is you
are unlikely to run into a lot of two-way systems. Even clubs and casual bands use threeway systems these days. As the processing gear and power amps have become less pricey
and more powerful, the use of subwoofers on even small gigs has become more the
norm than the exception.
System EQ
Earlier we talked about EQ at the console level, which is all about shaping tone in
individual ‘‘voices’’ in the overall sound. In the case of system EQ, it is more about
‘‘correcting’’ for anomalies in the speaker cabinets or in the room itself. When I started
doing sound, the system EQ was the first thing the signal hit after the console, with the
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crossovers coming just before the power amps. But the proliferation of cheap DSP
means that we can have EQ pretty much anywhere we want it, since all of the drive
components are really just different functions of DSP. It is possible to have EQ both
before and after the crossover. Ditto compression and delay. As a system-wide thing,
delay is used to time-align delay stacks in large venues. Now it can be used to align
individual drivers in a speaker cabinet. Compression (usually set at 10:1 or higher and
operating as a limiter) was used to protect amps and speakers from excessive signal
input in order to keep those amps and drivers from being damaged. Now you can apply
different settings to different frequency bands.
Most of the post-crossover processing falls into the area of ‘‘tuning’’ the speaker cabinets—EQ to make up for anomalies or deficiencies in the speakers themselves, delay to
align the output of individual components—and are beyond the scope of this book. For
our purposes, we have to assume you are working with a properly EQ’d and timealigned system at least at the individual cabinet level. A bit later we’ll talk about ways to
align a system once it is in the room.
Getting in Tune
System EQ is sometimes called tuning the room. It would be more accurate to say you
are tuning the system so it sounds good in the room. Depending on the gig, the amount
of setup time, and just the overall attitude and vibe of the client, you may be able to run
pink noise through the system (pink noise sounds like an enormous wall of static, but it
contains all possible frequencies in the spectrum at equal levels) and then use a device
called an RTA, or real-time analyzer, and a special measurement mic to give you a
visual representation of how the room is affecting the sound. Then, you use an EQ to
flatten what the mic is hearing by cutting or—if you must—boosting frequencies with
an EQ. You may have to do that by just listening to a familiar piece of music through the
system and adjusting using your ears alone. Regardless of the kind of gigs you do, this is
a skill that you must develop. Yes, there are measurement devices that will allow you to
make the system almost completely flat—as in no spikes in any part of the audio spectrum caused by the shape or construction of the room. But you won’t always have those
tools available, and even if you have them, you won’t be able to use them on every gig.
Being a competent member of the audio tribe is not about the gear. As I have often
heard, it’s not the car; it’s the driver. The same act on the same system in the same venue
can sound great or horrible in the hands of two different engineers.
Here’s an example. A few years ago I spent five days on a ship called the Rock Boat. It’s
a very cool idea that has grown over a period of years from a couple bands taking a
cruise with a couple hundred of their fans to 20-some-odd bands selling out the biggest
ship in the fleet, with shows all day and night and bands ranging from indie headliners
to ‘‘Hey, we just did our first CD’’ newbies. A ship at sea is not an ideal venue for that
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many shows, despite very high-end installed sound and lighting systems. Just the process of loading in and out requires the use of unusual gig gear, such as great big cranes.
The ‘‘big’’ room on the ship probably held 500 people, and there were at least a couple
of shows there every day. Toward the end of the cruise, a national headline act did a
show in that room with an opening act that was four or five rungs lower on the musicbiz ladder. The opening act was mixed by one of the lead sound techs for Atlanta Sound
and Lights, the production company that has been doing the Rock Boat shows since it
first set sail. The headline act brought in its own engineer. The opening act sounded
really good. The tech was very good. He knew his system, knew the room, and used his
ears—just the way it is supposed to be.
The headline act started, and I could not believe I was listening to the same system.
There was no punch in the low end and an overall dull and mushy sound that was at
least 8 dB lower in volume than the opener. (This is something that virtually never
happens. The politics of live music often dictate that the opening act has to run through
a limiter that keeps them substantially less loud than the headliner.) The system was
appropriate for the room and had sounded very good through several other shows I had
checked out there over several days. But at that point it just sounded bad, and I could see
that the visiting engineer and the soundco tech were having a ‘‘discussion’’ about the
situation. What happened?
In this case, the system EQ was provided by a digital unit with very high pedigree that is
used on lots of pro rigs in both the touring and installation worlds. This digital EQ
basically acts like a huge graphic EQ with the potential for 100 or more user-defined
‘‘bands’’ and is very useful for reducing or eliminating troublesome resonances in the
room. As the tech later explained, before the band even started playing, the visiting
engineer grabbed more than 20 ‘‘points’’ on this EQ and pulled them all down by at
least 6 dB—essentially cutting the amount of signal getting to the amps by about half.
And then he topped it all off by screaming at the tech that the system did not ‘‘have
enough gas.’’
We will get deeper into the roles of different players on the live audio team later in the
book, but for now let’s just say that, while the mix engineer will almost always be the
one ultimately held responsible if there are problems with the audio, every member of
the team has to be able to trust their teammates. A visiting engineer coming into a room
and wanting to make changes to make their act sound their best is common. Living and
working in Las Vegas, I get to know a lot of house sound teams, and I have yet to meet
one that does not have a horror story about a visiting engineer wanting them to basically re-engineer the entire system for their act. Sad to say, but the great majority of the
stories involve visiting engineers who don’t know anything at all about system design
and just want things the way they want them. But most good house crews will also be
able to tell stories about the guy who came into the room for the first time and made a
small suggestion that significantly improved the sound quality or system performance.
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HELLO, HELLo, HELlo, HEllo, Hello, hello . . .
If, like many of us, you are coming to the audio world via the music world, you are
probably used to using the terms delay and echo interchangeably. Echo is a commonly
used effect and is generated by a device called a delay—unless you are using real vintage
gear, and the hardware was still named using the word echo. But in a sound system,
echo is a bad thing, and the delay unit is used to get rid of it so that multiple sound
sources hit the listener at the same time. The difference in clarity between a properly
time-aligned system and one that is not aligned is huge. Same gear, same engineer, same
act, and a little delay can make things sound completely different. And this is one place
where you may be able to vastly improve the sound of the system with just a few small
tweaks.
Some explanation first . . . Delay in a system was first used in extensions to the system
called delay stacks or delay towers, which were used to extend the reach of a sound
system for very large gigs. For example, even a huge sound system is going to have a
hard time throwing enough sonic energy to the far reaches of audience space at a stadium gig. So at the point where the volume drops off, a second set of speakers is set up
for that extra energy needed to make it sound good in the cheap seats. The problem with
this second source is it is still within the sound field of the main speakers. The electrical
signal that flows to the speakers moves much faster than the sound coming from the
primary speakers. What you end up with is sound from the secondary speakers followed
a fraction of a second later by the sound coming from the main source. This makes
everything indistinct, muddy, and dull-sounding.
The reason the secondary speakers are called delays is because the signal is delayed so
the sound coming from the delays hits the listener at the same time as the sound from the
primaries.
On smaller gigs, the issue is more about the placement of speakers, and this is especially
true on club gigs. Some clubs have real systems that were installed by people who know
what they are doing and spent time designing the system to be right for the room. Some
are bars that book live music and pay little or nothing to bands so they can sell more
beer. The system—if they even have one—may be something they bought from a local
band or something they found on Craigslist and had installed by the kid up the street
who usually installs car stereos. The good news in the latter scenario is that you will be
making more than the band. The bad news is that you have to try to make a pile of
garbage sound decent. In other cases you may be working directly with a band that
carries its own PA. In any of these situations it is possible—even likely—that the system
will not be aligned properly.
The importance of a properly aligned system just can’t be overstated. There is a local
club here in Vegas that I played at while working on this book. I was looking forward to
the gig because I know the person who did the install, I know the gear involved, and I
was impressed by both. I advanced the gig and was shocked at how bad the system
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sounded in the room—that is, until I spoke to the person they had operating and
maintaining the system, and he told me that the line of subwoofers across the front of
the stage had originally been behind the stage and that by moving the subs forward, he
got more stage space because he put the monitor wedges on the subs. Two problems:
First, I know the installer well enough to know that if the subs were placed in an unusual
configuration, there was a reason for it. And, moving the monitors forward meant
moving the vocal mics forward, which put them well into the coverage area of the flown
mid-top boxes, turning the whole room into a feedback nightmare. When I spoke to the
installer, my suspicions were confirmed. The club had run out of money, and he had to
align the system physically rather than electronically because he had to do it without a
processor. And he had wanted to fly the mains two feet farther forward, but doing so
would have cut the coverage of the dance floor, so everything was a compromise, and
changing any part of it made the whole thing fall apart.
In fact, I was impressed enough with Brian Klijanowicz, who did the original install,
that I asked him to explain a couple of different approaches to time-aligning a club
system. Take it away, Brian . . .
Time-Aligning a Club System Time alignment is a very important, yet very often
overlooked aspect of system setup and tuning. A correctly time-aligned system has
many benefits, including more even coverage where two sound sources overlap and
a more even response across acoustical crossover points. It can give even the
cheapest of systems a couple decibels more in the area where engineers tend to like
them most: bass frequencies.
Two ways to quickly achieve this are using a sine wave and delaying the PA back to
the kick drum. Both approaches work, and the one you choose will depend on the
size of your gig as well as the time you want/have to work on it.
This is intended to be a minimalistic, quick way to time-align your system, so only a
few pieces of gear will be required. Using a sine wave will only need a sine wave
generator and the capability to delay an output signal and to reverse the phase of it.
To delay the PA back to the kick drum, all you need is a channel of delay and your
‘‘golden ears.’’
We will assume that the sound system we are working with is a standard frontloaded ‘‘stack’’ configuration. This means there is a subwoofer (producing subfrequencies from roughly 100 Hz and down) ground-stacked on the floor with a top
box (producing roughly 100 Hz and up) directly on top of it. We will also assume that
both subs and tops are currently producing the same polarity and facing the same
direction.
Sine Wave
Usually, an engineer would not want to use polarity to cancel out a signal. But that is
the whole concept behind this technique.
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First, it is important to find the acoustical crossover frequency between the subs and
tops. A measurement device is the most accurate way to do this. But if you don’t
have one, there is an easy way to rough it in. Assuming you don’t have a measurement device, flip through the pages of your crossover to see what the crossover frequency is and use that. (For this article, it will be 100 Hz.)
Let’s get back to the concept. We will take two similar sound sources, sub and top,
that are playing a 100-Hz sine wave, and we’ll flip one out of phase. The sound
sources will overlap and start to fade off from one another at the crossover point, but
more importantly, they will still reproduce 100 Hz, because that is the beginning of
the crossover filter on each source.
By physically looking at the speaker cabinets and knowing where the drivers are in
the boxes themselves, you will be able to determine which signal to add delay to.
Typically in the club world, top boxes are placed a little behind the front of the sub.
That way, if the top box falls, it will fall back and not onto the drunk audience. So,
assuming this is the case, delay should be added to the sub. While the 100 Hz is
playing through the sub and top, start increasing the delay to the subwoofer signal.
Eventually, the two signals will start to cancel out, and the total SPL will reduce
substantially. Find the point at which most cancellation occurs and leave it at
that. Change the signal that’s out of phase back into phase, and you should have
summation at the crossover point.
Some crossovers have an on and off function for the delay. A good way to check
whether summation is occurring is to flip the delay on and off to hear the difference.
This whole process can sometimes be done with music, preferably with driving, kick
drum–heavy music. However, it can be hard to distinguish what is cancelling and
summing in the crossover region with full-range music playing. A quick fix on a digital
console would be to throw a low-pass filter on the music channel.
Back to the Kick
Delaying the PA back to the kick drum is very simple. It works very well in the small
‘‘hole-in-the-wall’’ biker bar, but once you get into a bigger venue, it can become a
bit harder to hear the difference. Smaller venues that have lots of reflective surfaces
and 90-degree walls create standing waves. This makes it almost pointless to spend
the time to get out a measurement device—especially when the club owner wants
you to be done setting up before the first road case is even in the building.
Since this technique works better in small venues, we will assume your PA is in a small
venue and set up in front of a band. Even though it’s still surprising to club owners
and patrons, we all know how loud a drum set can be in a small room. That is why
typically, most engineers won’t even turn up all the drum mics, or they just won’t
mike the whole kit.
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So the drums are roughly 5 feet behind the PA. Think of the kick drum as another
speaker. If you are sitting in the audience, would you want to hear the main PA with
another set of PA 5 feet behind it? Probably not. So if you are sitting in the audience
listening to a band, you probably would not want to hear the kick drum once through
the PA and then again 5 feet later.
Most drummers do relatively the same sound-check routine. They will hit each drum
individually with quarter notes at a medium pace until you ask them to switch to the
next drum. When you get to the kick drum, get a rough sound in on the channel strip
and start tweaking the delay back. Add a little bit of delay at a time. You will start to
hear a change in tone and depth. Once you get to the sweet spot and are happy with
the sound, that’s it!
Take It to the Limit
The final step is to put some kind of ‘‘emergency brake’’ on the system—especially if it is
going to be driven by visiting engineers—in order to protect your amps and speakers.
Basically, this means using a compressor set to 10:1 or higher or a ‘‘real’’ brick-wall
limiter so that the amount of signal reaching the amps is below the level at which the
amp clips, which can damage both the amp and the speaker.
Now, let’s go back to the beginning here and remember that we used to have to deploy a
separate piece of gear for each of these processes. But the current crop of digital speaker
processors put all of these functions into one box. A typical processor will have two or
four inputs and six to eight outputs. Each input signal can be split, EQ’d, delayed, and
limited before moving on to the amps. Also, more and more of these functions are
moving directly into the power amp, and most major amp manufacturers sell at least
one model that has DSP processing in the amp. This leads to a conundrum when it
comes to buying gear. You only need one set of processors. What if you have outboard
processors and it is time to buy amps, and the amp already has DSP? Do you sell off
your processors and go with what is in the amp? Or do you keep the processors and
have the DSP in the amp as backup? To make things even more complicated, some
speaker companies insist you use specific processors that are preprogrammed with the
cabinet tunings for their speakers, and they will not release the data for those settings so
they can be transferred to the processors you may already own. (But note that the limiter settings are usually left open.)
Just an opinion here, but this kind of attempt to keep data proprietary feels an awful lot
like the battle that the record industry fought and lost a decade ago. Rest assured that
companies who make processors will figure out the proprietary settings and pass them
out freely to people and companies who buy their ‘‘non-approved’’ processors to use
with whatever speakers they want to. There are only two ways to keep this from
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happening. The first is to only sell your products as complete systems—amps, speakers,
and processors as a single unit. Several companies have taken that approach, but when
the pressure is on to make sales, those written-in-stone policies of selling only complete
systems tend to get a bit softer, and I know of only one company that consistently refuses—even on large systems—to sell individual components without the other parts of
the system (except for replacing something that has worn out or failed). The other way
to keep this from happening is to build all of the processing and amplification into the
speaker cabinet itself. And powered or active speaker systems are becoming increasingly
common. Which leads us to our next chapter . . .
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Active Speakers
efore we move to the end of the ‘‘traditional’’ signal chain, we need to take a
little side trip to a land where speakers, amps, and processing all live in the same
box. A magical place called the Land of Active Speakers . . .
If you have been paying attention, you will have noticed a thread running through much
of our content so far that involves a consistent move toward doing more with less. Well,
kind of. The power available has grown exponentially. When Bill Hanley did the sound
at Woodstock for somewhere north of half a million people, he had less total power
available than your average bar band has in their music-store-grade PA. But in general,
the trend has been more power—be it power or processing or whatever—in fewer
boxes.
It’s the Law
As we looked at in the previous chapter, most of that is in pretty direct relation to
Moore’s Law. (Gordon Moore is an engineer who, many years ago, while working with
chipmaker Intel, posited that the computational power of a given device would double
while the cost to produce it fell by half every 18 months. And that has been pretty
accurate for some three decades.) The ever-increasing speed at which technology advances has made it possible in a pretty short period of time to do with one box what
used to take four boxes to achieve.
This is both positive and negative. On the positive side, we get a ton of new toys and are
at the point where even regional acts in marginal venues can sound better than top
headliners did just 10 years ago. The negative side to all of this is that, as a former
touring sound guy who is now an exec with a major audio manufacturer told me, ‘‘We
have adopted the product cycles of the computer industry.’’ He is right, and the implications are huge. Two years before writing this book, I did a buyers’ guide for Front
of House magazine on digital consoles with a price tag that made it possible for the
average regional sound company to consider owning one. A few months before I wrote
this, we did another buyers’ guide, but this time we focused on the big, state-of-the art
units. In both cases we set the same price point—$75K. On the first go ’round, we had
companies screaming that the price was unrealistically low. Less than two years later,
we could find only a handful of consoles above the price that had been ‘‘unrealistically
low’’ not long before. At the time of the first guide, a digital console cost as much as a
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house in many places in the U.S. By the time the second guide rolled around, I knew of at
least a handful of touring sound guys who owned their own consoles and rented them
back to the band they were touring with because the price/performance ratio had fallen
to the point where buying a pro digital console was more like an investment in a good
car than in a house.
So what is the negative side? If you end up starting your own sound company, it is very
hard to know what to invest in when it comes to new gear. By the time you are working
in the field, it is very possible that most regional companies will have given up on trying
to stay current, opting instead to keep a few workhorse consoles in their inventory and
rent when they get an act or a venue that insists on something else. It is already at the
point where some system installers are telling performance venues that deal with touring acts to buy a really good analog console instead of the latest in digital technology.
The theory is that the lower-tech board will be perfectly adequate and acceptable for at
least half of the acts that come through and that when a larger headliner is booked,
there’s a good chance that the digital console they chose for the install would not be on
the rider (the list of acceptable gear for any gig), and they would end up renting one
from elsewhere anyway.
When you look at it objectively, it makes a lot of sense. But it also means that console
manufacturers are all competing for pieces of a smaller pie, which impacts profit margins, which impacts R&D budgets, which impacts how many new tools we get and how
often we get them.
Back on Track
Sorry, I got a little off track there . . . Important stuff, but what we started out with was
trying to make the point that putting the speaker, processing, and power all in one box
was less a function of implementing new technology than it was of making the currently
available technology easier to set up and run properly. Designing a sound system that
works to its full potential without damaging individual components means having a
firm grasp of the physics of sound, including the way sound waves propagate from a
source, how sound sources work together or interfere with one another, the behavior of
electro-mechanical devices such as speakers, and the power needs and potentials of the
devices feeding them. This is where classes on electrical engineering and acoustics come
into play. The concepts and knowledge those classes impart are well beyond the scope
of this book. (By the way, that is about the fourth time I have typed that phrase, and we
are not yet two-thirds finished. Are you getting the point that doing live audio right
means plenty of learning of both the book and real-world varieties?)
Matching the right amp with a given speaker requires more knowledge than even people
who have been around live sound for a long time may have a firm grasp on. Al Siniscal
founded the pioneering live audio company A-1 Audio and is one of the two people
most commonly cited as being the father of powered speakers for live event audio—the
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other being John Meyer, who deployed and integrated a system of amps, processing, and speakers for a 1971 tour. The disagreement is largely semantics, because
Meyer, then working with McCune Audio in the San Francisco Bay Area, designed and
integrated the system, but the speakers and amps/processing were in separate boxes.
Siniscal was mounting crossovers and BGW power amps right into the back of his
speaker cabinets and did his first international tour with those boxes in ’74. Both were
trying to overcome the same set of problems, mostly rooted in ignorance of the science
of sound. In a 2003 feature in Mix magazine, Siniscal recalled a symposium sponsored
by a major gear manufacturer at the 1992 Audio Engineering Society convention, where
engineers and dealers were asked how they powered a set of then-popular portable PA
speakers. There, he is quoted as saying, ‘‘For the five guys that can tune it, there’s another 95 that will screw it up. By the time we got around the room, there were all sorts
of things people used to power the speakers,’’ he says. ‘‘These would include cheap,
high-frequency amplifiers without overhead capabilities or dynamic range. They used
all kinds of off-brand crossovers and processors. They wouldn’t necessarily use the big
enough amplifier for the low frequencies, but they were out there competing. The preponderance of people didn’t power the speakers correctly, nor use the processors that
the factory recommended. These were music-store items, unbalanced, typically marginal units.’’
From the time of their introduction in the early ’70s until the mid ’90s, powered
speakers were largely confined to studio monitors (in fact, the first actual powered
loudspeaker Meyer made was meant as a studio monitor until the Grateful Dead used
them on tour) and Meyer Sound, plus guys like Siniscal who were making their own
boxes. In 1995, JBL introduced the first of what has become a ubiquitous class of gear:
Active systems mounted in a molded plastic cabinet that could be mounted on a tripod
stand. You will often see them referred to as speakers on a stick. It is usually a derisive
term, but I know of very few sound companies that don’t keep some in their inventory
for small gigs, extra monitors, or a million other uses. The market continued to grow,
but it really exploded with the introduction of the SRM450 by Mackie. Today, it is the
exception rather than the rule for most companies not to offer powered versions of their
speakers. And there is even an outboard speaker processor of the type we talked about
in the previous chapter, intended to be used specifically with powered speakers.
Pros and Cons
The advantages of powered speakers are many, including taking less truck space to
move because there are no amp racks to carry, as well as being secure in the knowledge
that amps are properly matched to speakers and that the processing (delay, crossover,
EQ—all of the things we discussed in the last chapter) is hardwired in so that the box
sounds as good as it is able to right out of the box. Another biggie is that the cable runs
from the amp to the speaker are short enough to be negligible in terms of impedance and
the power loss you can get when running long lines between your amp and speakers.
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But there are disadvantages as well. First, you have to get AC to each individual box
instead of to a single amp rack, and second, if you do lose an amp, you have lost that
box—probably for the duration of the show. Especially with flown systems, getting
access once the show is in progress is often impossible, and even if you could access the
box, you would have to either replace the amp module or have an external amp and
processor with all of the tunings already dialed in. Bottom line: It ain’t gonna happen.
Finally, while the cable runs between the amp and speaker are very short, the run to the
speaker will probably be as long as the initial run to the amp (and longer if amps are
located at the front of house—an undesirable but not unusual situation). While long
runs between the speaker and amp can have their own issues as outlined earlier, at least
the signal flowing through that line is strong enough that it is far less likely to pick up
extraneous noise. On the other hand, that long line-level run to a powered speaker is a
scenario that almost begs for interference coming from anything ranging from overhead
power lines to a radio station or any other wireless transmission.
For this reason, it is imperative that you run balanced lines between your console or
stage box and any powered speakers. I once did a small church gig that started outside
in the parking area, which just happened to back right up to a small radio station, before
moving inside for the second half of the event. I was worried enough about interference
that I used no wireless mics at all. I thought I had it covered, but when we fired up the
system, the radio station was coming through loud and clear. The problem ended up
being simple and was easily fixed, but . . . not to leave you hanging or anything, but I am
not going to get into it right now. We’ll save it for later in the book.
To close out this subject, I am going to tell you not to do something the whole industry
does and that I have in fact done throughout this chapter. Try not to confuse or interchange the terms powered and active. A powered speaker has an amp in it and maybe a
passive crossover. If you want the advantages of a speaker processor that we talked
about in the previous chapter, then you still need to run said processor between your
source (usually the console) and the speakers. On the other hand, an ‘‘active’’ speaker at
least implies that the processing is handled in the box. A real ‘‘active’’ box will basically
be pre-tuned, and many are made to be matched as specific sets of subs and top boxes.
A real-life example: About five years before writing this, I was doing a gig with an active
system that I had used for a few years. The crossover was built into the sub, so I just ran
signal to the sub and then looped it to the top box. This is a very typical setup for smaller
gigs, and you will come across it often, especially in small clubs. A friend had been
touting the services of a company that used a very high-end and precise audio analyzer
to measure a dozen different audio parameters and then make suggestions on how to
better tune the cabinets. This was a midsized sound company with an inventory that
included both powered and unpowered boxes and that used external processors on all
of them. The owner told me that the tweaks recommended after the analysis made his
speakers sound better and gave him a few dB of extra gain.
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So I had them come out to the gig to analyze this active system and see whether it would
benefit from an external processor. We set up the system using the external processor as
a crossover only set at an arbitrary 100 Hz. He ran his tests, and the system was a mess.
Drivers were out of phase, the response was so far from flat that the meter looked
something like a mogul run on a ski slope, and in order to be aligned without a delay,
the subwoofer would have had to be set 9 feet behind the mid-high top box.
It was hard to believe that the system was so out of whack, so we did the analysis again,
this time with the system set up as intended with the sub feeding the top boxes. The
result? While still not totally flat, the response smoothed out tremendously across the
spectrum, the out-of-phase drivers reversed, and the sub and top box were suddenly
aligned.
Could we have gotten the same or better results from a good external processor? Probably. In fact, almost certainly. But could we make it enough better to justify the expense
of the external processor? Probably not.
So what should you take away from this chapter? The fact that having the amp inside
the box has its advantages, especially for those who still have a lot to learn about sound
systems. But there are disadvantages as well, and those disadvantages make some pro
sound guys unwilling to give up the control they have with separate components. Also,
the fact that active and powered are not the same thing . . . High-end powered systems
are generally deployed with the assumption that they will be used with a processor.
Finally, although it is surely possible to improve on the response of an active system
through bypassing the onboard processing and tweaking it with a processor (assuming
you have the knowledge and the means to take the measurements in the first place),
doing so is probably not a cost-effective option.
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You Gotta Have Power . . .
W
e are getting near the end of the signal chain—approaching the transducer
that will convert the electrical energy (or data, in the case of digital mixing)
back into the acoustical energy that originated at the mic. But we’re going to
do it on a much greater scale (as in a lot louder). And making it loud takes power—a
bunch of it. Yep, you guessed it—it’s time to talk about power amps.
Let’s just make sure we have our terms straight first. A power amp is not the same as,
say, a guitar amp. It has no preamp or tone-shaping function. (Well, with the current
move toward including system DSP in the amp, that is not really true, but I am trying to
keep it simple here, so I am confining the discussion to traditional power amps.) A
power amp has one job, which is to make the incoming line-level signal powerful enough to drive the loudspeakers and ‘‘make’’ sound. You will see units out there with
digital readouts and all kinds of other controls and indicators, but the truth is that a real
power amp needs just four things accessible to the user—inputs, outputs, a gain control,
and an on/off switch.
Ins and Outs
Inputs and output connections on power amps come in a bunch of flavors, and once
upon a time you could make generally accurate estimations of how ‘‘pro’’ an amp was
by the kind of connectors it used. For example, if an amp had RCA inputs, it was consumer grade or at best something designed for the home or installed market. If it was out
on a live show, you probably had a problem. But today, one digital format uses RCAtype connecters as an option, so that benchmark is out the window.
Today you will find a couple of different analog input types and several possible digital
connections. On the analog side, what remains true is that pro gear takes balanced input
signals. It can be on a barrier strip, a 1/4-inch tip-ring-sleeve connection, an XLR, or a
combo jack that will take either one. RCA analog or unbalanced 1/4-inch tip-sleeve
connectors denote non-pro gear.
On the digital side, it can be an XLR, an RCA, or an RJ-45 (Ethernet) connector, depending on the amp and manufacturer and the kind of digital signal the amp can take.
As I noted back in Chapter 4 on cables and connectors, you really do need different
cables for the digital connections. In the case of an RCA or phono connection, those are
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used for the S/PDIF digital format and require the use of a higher-quality 75-ohm cable.
(Typical consumer RCA cables are more like 35 to 50 ohms.) The higher-rated cable
simply passes more data, and using cheaper cable can really degrade the quality of a
digital connection.
On the XLR side, it is a similar thing. The format is going to be AES/EBU (much more
common in pro use—especially in live sound—than the S/PDIF format), and a highquality 110-ohm cable is the spec for this kind of connection. At the time of this writing,
on the RJ-45 tip we were just starting to see really pro-quality Cat-5 or Cat-6 cable
making its way to market. While substantially more expensive than the cheap networking cable you get at RadioShack, it is worth the investment in a portable live sound
application. If you are installing a system and the cable is not to be moved, then any
decent networking cable should do the trick, but keep in mind that these cables and
connectors were never made with the intention that they be used for anything other than
a pretty permanent connection. Your standard RJ-45 connector is designed to withstand being plugged in and unplugged maybe 50 times, which is why everyone I know
who has gear with an RJ-45 connection either invests in better cable or carries multiple
backup cables to every gig, because those standard networking cables are going to
fail . . . and probably sooner rather than later. On the connector side, you are always
best off using gear that will take a Neutrik dataCON, which I discussed and showed a
picture of back in Chapter 4.
Outputs can take several forms. Even though few speaker systems use them anymore,
many amps still include banana-plug or binding-post outputs. These are made to take a
bare wire or simple plug that uses a separate plug for the hot and ground connections.
A straight 1/4-inch jack may be available, but again, this tends to be found on less-thanpro gear. The most common output connector is the Neutrik speakON (see Figure
18.1), variations of which are made by other manufacturers and called twist-lock.
Figure 18.1 Male and female Neutrik speakON connectors. Images courtesy of Neutrik.
speakONs come in flavors that include a straight two-pole connection, like a 1/4-inch
connector, to NL4 and NL8 connectors with four- and eight-pole designs that allow for
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bi-amp or tri-amp connections on a single cable. (Actually, with eight conductors, an
NL8 could handle a four-way system on one cable.)
What’s Cooking Inside?
We were talking about amps and got kind of sidetracked into cables and connections
again. (But keep in mind that more than 90 percent of the time you are looking for the
point of failure in any audio system, your search will eventually lead to a cable or connector.) Power-amp technology has made huge leaps in just a few years. When Bill
Hanley basically invented this business in the ’60s, one of his biggest competitive advantages was a refrigerated truck that held all of the Macintosh tube amplifiers and kept
them cool enough to power a large gig. The Woodstock festival was done with maybe
1,000 watts of total power. Today, that is one channel of a smaller pro power amp.
Amp design very quickly gets into electrical theory and formulas that are well beyond
the scope of this book. But we should at least take a look at the various amp classes
before getting into some of the more practical knowledge you will need for doing this
stuff on a day-to-day basis.
Class A
Those tube amps I referred to a moment ago were a Class A design. Class A amps sound
really good and are prized by audiophiles to this day. But they are inherently inefficient.
That is, the ratio of output power (what drives the speakers) to input power (what you
plug the amp into so it will turn on) is low, and much of the available energy is wasted
and dissipated in the form of heat. Remember basic physics: Energy can be neither created nor destroyed, but you can change its form. In this case the electrical energy that
powers the amp is sort of transferred, for lack of a better term (and amp designers
reading this are shaking their heads in disgust right now because that is not really accurate, but it is easier to understand without the math you need to really comprehend
the inner workings of an amplifier), to the input signal, making it many times more
powerful at the output than it was at the input. So Class A sounds good but does not
make efficient use of power.
Class B
Class B is much more efficient than Class A, but the design introduces changes to the
signal (in other words, intermodulation distortion) that make it sound less than great.
Class B amps were most often used in applications where the quality of the sound was
not a primary concern—things like portable tape machines and AM radios popular in
the pre-Walkman era. They are also found in things like bullhorns, walkie-talkies, and
emergency vehicle sirens.
Class AB
These were the first amps really designed with the high power and good sound needed
for live audio in mind. These are big, heavy beasts that do a very good job of amplifying
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sound in a pleasing manner, with much greater efficiency than a Class A design. But
they require a large output transformer as part of the design, which makes them less
than fun to carry around. As an example, a Crown Macro-Tech 2402 will put out about
500 watts into a ‘‘standard’’ 8-ohm load and weighs in at about 50 pounds. Given that
all but the smallest of backyard party gigs will require at least two amps, you now have
an amp rack that weighs well over 100 pounds (including the weight of the rack itself).
There are significant weight and size considerations—outside of the strain on your
back, big, heavy amps require you to have more truck space and burn more diesel to get
from gig to gig. A significant lowering of size and weight would make the financial
equation for touring sound a lot easier to handle. It is not uncommon for a large touring
show to carry somewhere in the neighborhood of 100 amps. That is 2 1/2 tons of amps
distributed among some 16 rack cases. Add those cases and all of the internal cabling,
and you are probably north of three tons. Reducing the space and weight by half would
lead to very significant cost savings for touring shows. Which leads us to . . .
Class D
Though first designed in the 1950s, the components needed to build a Class D were too
expensive to be used in mass production until around the end of the ’90s. Class D amps
operate on very different principles than more traditional power amps and are often
referred to with terms such as pulse-width modulation, switch mode, and switching
power supply. They may sometimes even be referred to as ‘‘digital’’ power amps, even
though the generation of power is an inherently analog process. The bottom line is that
they offer at least the efficiency of typical Class AB designs in a much smaller and lighter
package, and most of the touring amps you see now are Class D designs. Crown uses a
variation on the Class D pulse-width modulation design, which they have designated as
Class I. Their popular I-Tech series uses this topology. Another advantage of the smaller
and lighter vibe is that flown PAs can have the amp racks flown right next to the
speakers, keeping cable runs short. And when it comes to cable runs between the
speaker and the amp, the shorter the better.
In a perfect world, we would have perfect amplifiers that did nothing but increase the
energy of a signal without affecting the character of that signal in any way. But this is
not a perfect world, and there is no such thing as a perfect amp. All amps affect the input
signal in some way that introduces artifacts that are generally lumped under the common designation of distortion.
There are two main kinds of distortion to be concerned with. The first is harmonic
distortion, which basically introduces additional frequencies that are evenly related to
the original frequency. Small amounts of harmonic distortion can actually be pleasing
to the ear and are generally what is being touted when people speak of a piece of gear
sounding ‘‘warm.’’ The second type of distortion is intermodulation distortion, which is
similar to harmonic distortion except that the additional frequencies are not related to
the original in a way that any of us want to hear.
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There is a third kind of distortion that comes into play with any amplifier: clipping. If
you were to look at an audio frequency as seen on an oscilloscope, you would see the
tops and bottoms of the smoothly shaped waveform get lopped off, producing a flat
spot. This happens when the amp is pushed past the power it is able to supply and it runs
out of gas at the top and bottom of the signal, clipping those parts off of the waveform.
The overdrive and distortion used widely by guitarists is a kind of clipping and not
something you want coming out of your PA. In addition to sounding bad, clipping
makes an amp run hotter, which can cause it to overheat and shut down if it has protection circuitry and just burn out if it does not. Driving the input signal of the amp into
clipping can cause the same problem in the speakers to which it is attached. That’s right.
You can blow speakers not only by feeding them more power or power at frequencies
they were not made to handle, but also by feeding them with an amp driven into clipping. In fact, it is more common to blow speakers from using too little power than from
using too much. Think about that when you are thinking about skimping on power
amps.
One other function of an amp that can be affected by misuse is called the damping
factor. Again, too much math, but suffice it to say that changes in the damping factor
can result in fewer low frequencies being produced, making the audio sound thin
through the PA no matter how dense the original sound is. One big thing that can adversely affect the damping factor is using thin or low-quality cable between the amp and
speakers. When it comes to speaker cabling, use the best and beefiest cable you can
afford.
What a Load . . .
Power ratings on amps can be very misleading and are often more marketing than science. When you look at the power rating on an amp, there are a few things to take into
account and make sure you are comparing apples to apples.
To begin with, there are several ways to express output power. Until the marketing
departments and accounting suits took over the audio business and it became all
about making your gear appear to be better than the other guy’s by using whatever
irrelevant or inappropriate number it took to get there, every amp used the same spec to
express output power, called RMS or root mean squared. RMS is basically a kind of
average power output over time, and although none of us knew the math behind the
number, we didn’t need to because everyone was using the same objective basis for their
ratings.
But now you need to be careful, because even ratings with the same name can be arrived
at by different means. The other most common ratings are peak power and continuous
or program power, which sound the same but are usually arrived at by two different
routes. Peak power is the amount of power the amp can put out in short bursts. This
number is obviously much higher than the RMS. Both continuous and program are also
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generally higher than RMS as well. Continuous usually refers to the output power an
amp can put out continuously given a specific input frequency and power. But if two
amps have the same continuous rating but were arrived at using different frequencies,
then both numbers are meaningless. Program can be similarly meaningless because although it should be more ‘‘real world’’ than any other rating, no one is specifying what
kind of program material is being used. Is it a solo acoustic guitar or bass-pounding hiphop? Because of the inconsistency in the basis for power ratings between manufacturers
and even among models or series from the same manufacturer, it becomes even more
important to use your ears and input from hopefully objective third parties in making
your decisions.
Another thing that will have a huge impact on power ratings is the ohm load. A single
speaker cabinet will usually present a load of 8 or 4 ohms, and the output power of the
amp goes up as the load goes down. Given that, many manufacturers give their top
power rating based on a 2-ohm load. A couple potential problems here: First, while
many current power amps are designed with a 2-ohm load in mind and can handle it,
many (if not most) amps made before about 2005 may be able to run on a 2-ohm load,
but likely only for a limited amount of time. As that output power goes up, so do both
the current draw (the amount of current needed to drive the amp to that top rating) and
the amount of energy dissipated as heat. I once did a gig and had an old amp just die
during sound check, and in desperation I rewired the system, which meant the other
amp was running at less than 4 ohms. The result? I ended the night with two blown
amps instead of one.
And that 2-ohm rating is meaningless if you can’t get enough power out of the wall to
drive it that hard. As a good friend of mine at a major manufacturer once said, some of
those 2-ohm ratings are only good if you are powering the amp with ‘‘solid copper bar
connected directly to the Hoover Dam.’’ Again, something to keep in mind when
shopping . . .
Oh, and another marketing ploy: Is that rating power per channel or is it with the amp
in bridged mode? This is important because in bridged mode, the two channels combine
into a single output at close to double the power of either channel on its own.
You Can’t Do That with an Amplifier
If things continue the way they have been going lately, then by the time you read this,
you may be hard-pressed to find an amp that does not have onboard DSP to perform
most (if not all) of the functions of an outboard speaker controller. Over the course of
just a few years, DSP included in the amp has gone from the province of only the most
expensive, high-end pro amps to being almost ubiquitous to the point where even amps
designed directly for the musician market include some kind of limited system control
functions.
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Okay, let’s wrap this up with a few rules of thumb.
1.
Class D (and Class I) amps are generally more expensive, but they are smaller
and lighter than Class AB amps, which could have a long-term effect on the
bottom line.
2.
Use the shortest, heaviest cables you can between the amp and speakers.
3.
Use an amp appropriately matched to the speaker. An overpowered amp can
blow the speaker through sheer power, and an underpowered amp can damage
the speaker if it is driven into clipping in search of more volume.
4.
A lower ohm load can mean more power, but don’t go lower than four unless
you are sure the amp is designed for it.
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Loudspeakers
I
t’s been a long journey, but we have finally arrived at the end of the signal
chain. Remember that at the beginning we converted acoustic energy via a microphone; sent it on through the console where it was tweaked via EQ, dynamic
processing, and maybe some additional effects; mixed it together; and sent it along to be
amplified. Now we are going to convert that energy back into acoustic energy via
another transducer—the loudspeaker.
Speakers can get very complex, and this is the area where acoustics and the physics of
sound come into play more than any other. Getting really deep into that is outside the
scope of this book, but I will give you the basics. This will end up being the largest part
of this book, and I will approach it from the inside working out. First, we will examine
the actual loudspeaker transducer units or drivers, and then we’ll move on to how the
design and construction of the actual box affects the sound, how you use the device,
and finally, some common ways speakers are actually deployed in typical live sound
situations.
The Drivers
Any kind of loudspeaker is an electromechanical device. In other words, it has moving
parts. And anything with moving parts is prone—to some degree or another—to
wearing out and breaking down. This can actually be an opportunity and job security
for newbie sound techs. Learning how to troubleshoot and repair/re-cone speakers will
put you in much greater demand than someone whose only skill is mixing. Remember,
there are seven days in a week, and for the majority of sound companies, only two or
three of those are actual gig days. The rest of the time is all about prep and making sure
the rig is working right. Taking a class on speaker repair may be one of the best moves
you can make when it comes to staying employed even when things are slow.
We are going to limit our discussion of driver types to the most common ones you
will see—cone-based speakers and compression drivers. Other driver types, including
ribbons and piezo, are out there (and ribbons are becoming increasingly popular,
especially in line-array deployments), but the vast majority of boxes you will actually
use will have these two component types inside.
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We’ll start with a typical woofer. The diagram in Figure 19.1 shows a speaker made
by JBL.
Figure 19.1 Cutaway drawing of a typical JBL low-frequency transducer. Diagram provided
courtesy of JBL/Harman.
At its most basic, a cone speaker consists of a frame or basket that houses and supports
the cone, a surround (flexible surface—rubber or polypropylene or a similar material)
that connects the cone to the frame, a permanent magnet and metal parts forming a
magnetic gap, and a voice coil. The voice coil creates an electromagnet. The incoming
signal causes the electromagnetic voice coil to change polarity hundreds of times every
second. Suspended in a magnetic field formed by the permanent magnet, the coil moves
toward and away from the magnetic gap as the polarity changes. (Assuming a positive
charge on the permanent magnet, when the coil is negative it is attracted to the magnetic
gap and moves toward it, and when the charge of the coil is positive it is repelled and
moves away from the magnetic gap.) The voice coil is attached to the cone and causes
the cone to move in sympathy with it. The resulting motion of the cone moves the
surrounding air, which reaches the ear and is perceived as sound.
Those are the basics. Let’s take a look at a real speaker. Figure 19.2 shows pictures of
actual speakers, and we will be using the diagram in Figure 19.1 as a reference point.
Starting at the top-left and working in a counterclockwise direction, let’s look at the
individual parts of the speaker.
■
The frame. This is the metal housing that supports, protects, and houses the actual
cone.
■
Input terminals. These two input points (one positive and one negative) are where
the amplified signal enters the speaker mechanism. The input tinsel leads carry the
signal from the inputs to the voice coil.
■
Voice coil. This creates the electromagnetic field that interacts with the permanent
field from the magnet in the gap.
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■
Magnet, protective tire, back plate, and top plate. The magnet is the source of the
permanent magnetic field, the tire surrounds and protects the magnet, and the back
plate and top plate form the magnetic path to the gap in which the voice coil is
suspended.
■
Center pole piece. This is the piece that the voice coil is ‘‘wrapped’’ around and
completes the magnetic gap circuit path.
■
Vent. This allows air to escape the structure as the coil moves toward the magnet.
This air movement is all about keeping the voice coil as cool as possible. Some
speakers also use a metallic fluid called ferrofluid in this space to absorb heat and
keep the voice coil cool.
■
Shorting ring. This reduces distortion induced by the action of the magnetic fields.
■
Former. This keeps the voice coil in shape (not as in going to the gym in shape—as in
round in shape).
■
Spider. This keeps the coil centered.
Figure 19.2 Back, side, and front views of a cone driver, courtesy of Eminence.
Note that there is a structure both inside and outside the voice coil that forms the
magnetic gap. The amount of clearance between these structures and the coil is tiny, and
the coil can expand with the tremendous amount of heat generated by the power
through the coil. As a result the tiniest amount of misalignment can result in scraping
and damage as the coil moves. Yes, this means that a poorly designed, badly built, or
improperly used speaker can catch fire. Something to keep in mind . . .
Figure 19.3 presents a diagram of a voice coil off center and tilted within the air gap.
Figure 19.3 Off-center and tilted coils. Diagram courtesy of Eminence Loudspeakers.
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The long wavelengths of low frequencies mean that the cone has to really move in order
to re-create the bass tones. A high-frequency driver reproduces sounds with much
shorter wavelengths, which therefore need to move much shorter distances. A compression driver works on the same principle, but instead of a large cone that is inverted
toward the magnet, the surface is a dome-shaped diaphragm that protrudes away from
the magnet.
Figure 19.4 Diagram of a typical compression driver provided courtesy of JBL.
While a cone loudspeaker produces a sound field that is omnidirectional—going off in
all directions—at low frequencies, a compression driver is usually attached to a horn,
and the size and shape of the horn determine the coverage of the higher frequencies.
This is an important distinction, as you will see when we start to look at box designs and
deployment. Low frequencies tend to be omnidirectional. High frequencies are very
directional, which means that the coverage of a typical box is defined mostly by the
dispersion of higher frequencies.
Having said that, it is easy to fall into the trap of seeing the horn as being nothing more
than a way to guide sound in a specific direction, which is not the case. Compression
drivers are also often referred to as horn drivers. As the vibrations move from the
narrow throat of the horn through its flare and to the exit, larger and larger ‘‘slices’’ of
air are excited, which results in a much louder sound than was present at the driver
itself—which is why we are dealing with horns in the driver section rather than in the
‘‘Box Design’’ section, where they would seem to be more appropriate.
Horns are most often used with mid and high frequencies because their shorter
wavelengths lend themselves to this format. When you get down into bass frequencies,
the amount of space needed to make a horn that properly transmits longer wavelengths
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is pretty prohibitive. One way around this is the folded horn, which we will cover in the
‘‘Box Design’’ section later in this chapter.
It is not only the size of the horn, but also its shape that contributes to the overall
frequency response and to the dispersion pattern. And many horn designs have been
employed over the years in search of one that puts out the most accurate sound and
allows for pattern control. When it comes to pattern control, the idea is to keep the
horizontal dispersion wide and limit the vertical dispersion for two reasons. First, most
of the energy on the top part of the vertical dispersion shoots over the heads of the
audience and is wasted. Worse, in an enclosed environment, those sound waves bounce
off the ceiling and walls and back into the intended sound field, which results in muddy,
incoherent sound. Second, because horns are usually in the same box or stacked on top
of a cone driver, the two sound sources can interfere with each other, something we will
look at in the ‘‘Box Design’’ section.
Most of the horns you run into will be either radial or constant directivity designs.
A radial horn has two surfaces based on an exponential flare rate and two straight
walls. While this works great for overall pattern control, the dispersion pattern narrows
as the frequency increases. The overall effect is of a ‘‘beam’’ of high-frequency sound on
axis, with those frequencies falling off rapidly as the listener moves off axis, resulting in
a dull or muddy sound.
In May 1975, some of the beaming problems were addressed by D. Broadus ‘‘Don’’
Keele, Jr. of Electro-Voice. Instead of the typical practice of making a horn that flared at
a constant rate from the driver to the exit point, by having a conical section in the
middle part of the horn and then going into a rapid flare at the end, he was able to
minimize the ‘‘beaming’’ of high frequencies, resulting in a much smoother response
over a wider area. This design is referred to as a constant directivity, or CD, horn, and
variations on it are still in use today. See Figure 19.5.
Figure 19.5 Diagrams from the original patent application for the constant directivity horn design, provided courtesy of Electro-Voice.
Most horns you will see are going to be of the radial or CD design or some variation on
those designs, including Bi-Radial horns pioneered by JBL.
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The last horn to look at is the multi-entry horn first patented by Ralph Heinz of RenkusHeinz and also used extensively by Tom Danley in work at Sound Physics Labs,
Yorkville, and Danley Sound Labs.
A multi-entry horn has multiple midrange and high-frequency drivers mounted to the
same horn, as shown in Figure 19.6. The result is that more linear transient response
and smoother polar patterns are possible, and greater power output can be achieved
from a smaller box.
Figure 19.6 Multi-entry horn.
Impedance
Everyone repeat after me—oooooommmmmmmm. Relax your mind and take a chill
pill—time to get into a little math. . . .
Impedance is just what it sounds like. It is the degree to which the flow of electricity in
any wire or circuit is blocked or impeded. Imagine this: You are holding a garden
hose that has water flowing through it. Now imagine putting your thumb over the hose
opening. If you cover the entire opening, you can actually still feel the pressure of the
water, but nothing comes out of the hose. As you move your thumb, water is able to
leave the hose, and the less you cover the opening, the lower the pressure behind your
thumb. Think of your thumb as a kind of resistor and the degree to which you cover the
hose exit as impedance.
Okay, here’s the math part. Impedance is measured in ohms. (Get it? Ohms? Like the
meditation sound I referred to above? Gosh, I’m witty. . . . ) Way back in 1827, a
German physicist by the name of Georg Ohm put forth the idea that in electrical
circuits, the current through a conductor between two points is directly proportional to
the voltage across the two points and inversely proportional to the resistance between
them, provided that the temperature remains constant. This is known as Ohm’s Law,
and it is expressed as voltage/impedance ¼ current or voltage/current ¼ impedance.
For example, if a speaker rated at 8 ohms receives a 10-volt signal, then the current is
1.25 amperes. If you lower the impedance of the speaker to 4 ohms, you double the
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current flowing to 2.5 amps. So, on the surface it would appear that the lower the
impedance of the speaker, the more current (or power) is running through the speaker.
And that is a good thing, right?
Not exactly, because as the impedance approaches zero, you are asking the power
amp to produce more and more current, and current produces heat. More current
means more heat, and you can easily toast an amp by asking it to produce too much
current.
A couple of other things to wrap your brain around . . . (And yes, we will have a nap
break after this chapter so you can recover.) First, the impedance of a speaker is
dynamic. In other words, it changes constantly in relation to the frequency of the signal
it is receiving. This is why speaker specs are listed as nominal impedance. It is kind of an
average or best guess of the impedance at any given time.
Here is the other thing that is going to sound backwards: As you add speakers to a
circuit, the impedance goes down, not up. How is that possible? Remember that the
voltage output of an amplifier is constant. Adding one 8-ohm speaker to the chain
requires a 10-volt amplifier to produce 1.25 amps of current. Adding another speaker
that requires another 1.25 amps means that the same amplifier now has to produce
2.5 amps. 10/2.5 ¼ 4. So now your total impedance is 4 ohms. Adding a third 8-ohm
speaker that requires 1.25 amps brings the power draw on the amplifier to 3.75 amps.
10/3.75 ¼ 2.6 amps. Remember, Ohm’s Law also says that impedance is equal to voltage/current, so on that last example, 10/2.66 ¼ 3.75 ohms. The bottom line for a
guesstimate on total load is that every time you double the number of drivers all at the
same impedance, the total load as measured in ohms is cut by half.
Just to make sure you were paying attention in math class, what if you have two speaker
cabinets, one with an 8-ohm impedance and one with a 4-ohm impedance? Assuming
that 10-volt amplifier again, the 8-ohm cabinet will require 1.25 amps of current, and
the 4-ohm cabinet will need 2.5 amps. So the total power draw for the two cabinets is
3.75 amps. Voltage/current ¼ impedance, so that means our two cabinets are presenting
a load of 2.66 ohms (10 volts/3.75 amps ¼ 2.66 ohms).
Power Handling and Efficiency
A speaker will have a power rating. This will usually be expressed as an average amount
of power that the speaker needs to be driven properly. The unit of measurement is
watts. (Yes, it’s named after another physicist. James Watt was a Scottish engineer who
lived in the 18th and early 19th centuries. He had nothing to do with electricity, but
the unit of measurement was named for him to acknowledge his contributions to the
development of the steam engine.)
Wattage is figured with—yea!—another mathematical formula. In this case it is the
output voltage squared divided by the impedance. So, going back to our 10-volt
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amplifier and 8-ohm speaker, the power handling of the speaker would be 100 divided
by 8 (100/8), or 12.5 watts. Modern amps put out much higher voltage, so using
60 volts and that 8-ohm speaker (602/8) gives you a power rating of 450 watts.
So a speaker rated at 450 watts should not be hooked up to an amplifier that puts out
more than 450 watts of power, right? Again, it makes sense on the surface, but that
450 watts is an average, so a 450-watt amp hooked to a 450-watt speaker means the
amp is being asked to put out its maximum power all the time. This is a good way to
blow an amp. A good rule of thumb is that the amp output should be about double the
power rating of the speaker.
Not only will pushing the amp to its maximum put the amp in jeopardy, it is also bad for
the speaker.
Speakers fail in one of two ways: thermal failure or mechanical failure. Mechanical
failure is always related to the movement of the voice coil. Too much movement in
either direction results in what is called over excursion. If this happens as the coil is
moving away from the magnet, the coil can ‘‘jump the gap’’ and either short out or
cause the coil wiring to break, which results in an ‘‘open’’ coil. The bottom line in either
case is that the coil no longer functions as an electromagnet, and the speaker stops
working. If it happens as the coil is moving toward the magnet, the coil can bottom out,
causing it to deform and be unable to move within the air gap. Again, no movement of
the coil means no sound.
Thermal failure has more possible causes, the most common of which are too much
input power, signal power outside the frequency range that the speaker can produce, or
amplifier clipping. The first two of these make sense right away. Too much input
or input frequencies that the speaker can’t handle have to go somewhere. (Remember,
energy cannot be created or destroyed, but its form can change.) In these cases the
excess or unusable input gets converted to heat, which can deform the coil or actually
make the speaker catch fire.
The last one is not so obvious. A clipped signal from the amp means that the amp is
being pushed harder than it was made to. Since it does not have enough power to
reproduce the entire signal, the top and bottom of each wave are cut off or clipped.
Trying to reproduce a clipped signal produces heat, which can toast your speaker.
Bottom line: Power matching is crucial, as an amp that is either too small or too large
can result in a blown speaker.
One last piece of math before we move away from drivers and on to actual cabinets . . .
Sound output is measured as sound pressure level, or SPL, which is expressed in decibels
and is arrived at using a standard of 1 watt of input measured at 1 meter of distance.
Without getting really deep into the math, just remember that the amount of amp power
needed to produce a specific amount of SPL doubles every time the speaker efficiency
drops by 3 dB. So if a speaker with an efficiency rating of 93 dB puts out 120 dB SPL at
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1 watt/1 meter, then a speaker with a 90-dB efficiency will require twice the power to
achieve the same output level.
The rub is that as speaker efficiency goes up, the accuracy or fidelity of the reproduction
often goes down, so achieving really good sound requires a lot more power. In the early
(or even just earlier, as in prior to a decade ago) days of live sound, power was a
precious commodity, so there was a constant balancing act between making speakers
that were efficient enough to put out sufficient SPL and making ones that still sounded
good. In the past decade, the output power of typical power amps has increased massively, which makes the efficiency of the speaker less crucial. It is, however, something
you should still be aware of.
Time to move on . . .
Box Design
As you have probably already figured out, box design is not just putting together some
wood and throwing a couple of drivers into it. The truth is that there is at least as much
math and physics involved in good box design as there is in driver design. And it is not just
design; material used in construction also has an impact on the final sound you hear.
If you really want to get into the nuts and bolts and calculus of speaker boxes, there are
plenty of online resources to turn to. I am going to stay away from the math and go with
an overview of the most common types of speaker design.
Sealed Box
This is not a design you will see often in PA systems, with the exception of some midrange drivers, but it is the most logical place to start. Think of the typical 410-inch
half-stack cabinets used by guitar players. Now visualize a cone speaker in action. The
movement of the cone moves or excites the air particles in front of it, resulting in audible
sound. But what about the air in the cabinet? As the speaker cone moves, that air is
moved as well, so what happens to it?
In a totally sealed cabinet design, the air vibrates the actual walls of the box, which is
where that energy goes. But it takes a lot more energy to move a 1/2-inch-thick piece of
wood than it does to move a paper cone, so the vibrations produced are not enough to
really contribute to overall volume. And even though most sealed cabinets are not really
sealed (they have a small air leak built in to allow for the equalizing of air pressure
inside and outside the box), the box itself still vibrates, which can have a big effect on
the timbre or sound quality.
One way to use the energy from the back of the speaker and keep the box sealed is to use
what is called a passive radiator. This is basically a speaker with no signal routed to it.
Instead of the cone moving in response to the signal being sent to the voice coil, it is
moved by the air in the cabinet. This is not a common design in PA cabinets, but I have
seen it in some studio monitors.
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Bass Reflex
This is probably the most common design you will see. In a bass reflex design, there are
holes, or ports, usually located on the same surface on which the speaker is mounted. It
is one thing to just drill holes into a speaker cabinet, but it is quite another to properly
design and build a port calculated to increase the speaker output at a specific frequency
range. A bass reflex cabinet can have some internal structure attached to the port, which
is calculated or tuned to maximize the output of the desired frequencies.
Figure 19.7 Typical bass-reflex enclosure design.
Because the ‘‘inside’’ energy is actually making it out into the airspace that the listener
hears, bass reflex cabinets loaded with the same drivers and getting the same signal at
the same power level as a sealed cabinet will generate more bass output.
Horn-Loaded
This term refers to any speaker box that uses some form of horn to combine the signals
of a single or multiple drivers to increase the efficiency and control the dispersion
pattern of the entire box. This can mean anything from drivers attached to a horn and
the whole thing mounted to the box, to a design in which the box itself acts as a horn.
This last one is not common anymore (although it was very popular at one point in
the development of live audio reproduction) and can take the form of a folded horn.
This can mean an enclosure in which the driver fires out the front, and the back energy
expands through some twists and turns to replicate the length of a horn needed to
reproduce low frequencies, and those frequencies exit via a horn mouth, strengthening
the output of the actual driver.
In a folded horn design, it is not uncommon to see only the mouth and not the actual
driver from the outside of the enclosure. This is because the driver is mounted inside the
box, expands through the internal horn structure, and sound exits from the horn
mouth only.
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Deployment
In all but small gigs, you will likely use multiple boxes of the same type, and how you place
the boxes can radically affect the sound. Here comes the whole physics thing again. . . .
Each box can be termed a point source, which is the point at which a radiating pattern
begins. This could be anything from a speaker to a light bulb to a rock thrown into a
pond. In reality, each of the components in a box is its own point source, but we’ll get
into that in a minute.
Look at the diagram in Figure 19.8 and imagine the rock-into-the-pond thing. The areas
where the ‘‘ripples’’ from each rock intersect are called an interference pattern. And it
does not matter whether we are talking sound or light or ripples, the principle is the
same. The point in the development of the waveform determines what will happen
when the two waves meet. If they meet at the peak or trough of the wave, the waves
will combine and become stronger. If they meet at some point in the middle, they will
destroy each other and basically cancel out that part of the wave at that spot.
Figure 19.8 Diagram of two waves colliding. Illustration by Erin Evans.
In sound, the overall effect is called comb filtering, and it looks like what you see in
Figure 19.9.
The audible result is a carved-out or hollow sound and much less overall output level.
With multiple boxes, we need to place them in a way that minimizes this interference.
Many full-range boxes made since the ’80s use a trapezoidal design that is calculated to
minimize interference between boxes placed side by side. If you are using older boxes
where all of the sides are at right angles, then you need to space them to avoid this
kind of filtering of the sound. A friend of mine, Paul Overson, who has a small sound
company in Utah, goes out on gigs with a homemade ‘‘template’’ that, when placed
between his subs and top boxes, shows him the ideal angle at which to place his top
boxes in order to minimize comb filtering.
One approach to multiple boxes that has become very widely used (although poorly
understood by many sound personnel) is the vertical line array. The idea is that the
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Figure 19.9 Comb filtering. Illustration by Erin Evans.
waveforms produced by a vertical stack or array of drivers of the same size will couple
as long as the individual drivers are no farther apart from center point to center point
than the size of the wavelength of sound at the frequencies being reproduced. The result
is that the entire array acts like one big source with wide horizontal coverage and
narrow vertical coverage, resulting in much clearer sound.
I am not going to get into the ins and outs of line array theory except to note that one of
the advantages of a properly designed and deployed line array is that instead of output
dropping by 6 dB with every doubling of distance, it drops only 3 dB. The bottom line is
that you can get more sound farther from the source with fewer boxes.
As if just getting speaker boxes placed right on the ground were not hard enough, since
the early ’70s, the demands to minimize blocked sightlines have resulted in the practice
of flying, or hanging, speakers. (Actually, the first flying systems were platforms that
held speakers arrayed just like they would have been on the ground, but then the entire
platform was lifted into the air. Bruce Jackson working with Elvis Presley and Stan
Miller working with Neil Diamond were two pioneers of this practice.)
When flying speakers, good sound is complicated by serious safety issues. This is not a
tome on the art of rigging, but just keep in mind that you need to really know what you
are doing before you start hanging heavy speaker enclosures above an audience. Never
put eye bolts or anything like that into a cabinet not designed to be flown. Speakers
designed with flying in mind will have ‘‘flyware’’ built into them for this purpose.
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Getting Your Hands Dirty
ow we have been through the entire signal chain, and you should have at least
a basic understanding of the gear, how it works, and how each thing gets
hooked up. So now it is time to start talking about putting together a mix.
Well, hang on there, little campers. There is still much to be done before we start messing with all the knobs and buttons. First, let’s get real. The average touring or venue
sound crew has from three to ten people, and only one or two of those is actually mixing. So what is everyone else doing? This is important, because if you are just coming
into the business from school or some other training program, you will start out as one
of those ‘‘other guys.’’ The only way you walk out of school and into a mixing position
is by starting your own soundco, by having a rich uncle in the business, or by going to
work for an act with whom you have a preexisting relationship.
The second two options are pretty rare, so my first advice is to get yourself a job with an
established company or venue doing the ‘‘other guy’’ stuff for a paycheck. This is still a
business in which experience counts for a lot and time has to be spent paying dues. In
the meantime, hook up with a local artist, promoter, or venue, do mix gigs at the usually
pitiful prevailing rate, and look at it as experience and a stepping stone. A combination
of those two things will keep your mixing chops up, keep the rent paid, and give you
solid, real-world education about how things work on real-live gigs.
Kicked by the Wind, Robbed by the Sleet . . .
So, back to the question of what the ‘‘other guys’’ do. The world of live audio is not
standardized by any stretch of the imagination. So you may have to adjust some of the
terms I use here for what is the norm in your part of the country. I live in Las Vegas, and
the terminology for audio personnel here is very different than what it is in L.A.—at
least among local companies. The terminology we are using is something that comes
from the theatre world and is commonly used by the union that represents audio people—IATSE, the International Alliance of Theatrical Stage Employees.
Audio guys are generally divided into three categories and are often referred to as A1,
A2, and A3. And, yes, the A means ‘‘audio,’’ and the numbers denote the place in the
pecking order.
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An A1 is usually a mix engineer for the house but can also be a system tech who tunes
and maintains the system on a tour or in a venue. An A2 is often a monitor engineer and
may also be a system tech for either house or monitors, depending on the venue and gig.
As you can see, there is some nebulousness and crossover between A1s and A2s.
An A3 is an audio stagehand, and if you are the new guy on a tour, in a venue, or
working for a soundco or even a union gig, this is where you will likely start out.
Duties of an A3 can include everything from pinning the stage (running connections
from individual mics and DIs to the split or stage box), to wiring speakers under the
direction of the system tech, to pushing cases and coiling cable.
The Union This is a touchy subject, and one I am going to get a lot of grief for
addressing, but here goes. IATSE is not held in very high esteem by many in the audio
tribe. The attitude stems from the insularity of many locals (I know plenty of experienced guys in Vegas who can’t get a union card because the local is more concerned
about protecting longtime members than it is about bringing in new, better-educated blood), as well as the reputation of many locals for sending out people based
more on seniority than on qualifications. On the other hand, many gigs require that
you are an IATSE member before you can even be considered. The balancing act is to
be in the union but not of the union. In other words, you need a card to work in many
places but a union attitude will get you fired very quickly from many gigs.
Don’t despair that you went through all this training only to do grunt work. First, there
is a lot to learn as an A3 if you can keep your mouth shut and your eyes and ears open.
Second, it is all about experience. I have been running sound since the 1970s, but because I don’t make a living at it anymore or even do it every day, when I go out and
work for a soundco in Vegas (which I still do), I go out as an A3. In other words, I know
how to do everything on the stage or dealing with the system, but I would have to work
my way into an A2 or A1 position. It’s not that far from the situation you will be in
when you first start working. I’m just a lot older.
The need to understand the basics of implementing and deploying sound systems, from
connecting mics and speakers to knowing enough about how things work to choose the
right tool for the job, can’t be overstated. And most of that knowledge is not something
you can get from a book.
Had My Head Stove In . . .
There are some programs out there that take a real-world approach to live audio and go
way beyond just teaching students how to run a digital console and how to mix. One
that stands out is Belmont University’s program, which operates as an actual sound
company doing events both on and off campus. Students learn everything about doing
real gigs, from prepping and loading a system to getting paid and getting the gear back
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to the shop. There are others out there that do similar stuff, and if you are going to work
in the live audio field, it would behoove you to seek out that kind of program and not a
glorified recording program that takes a few weeks to cover live sound.
As I said before, it is virtually impossible to take this stuff too seriously. In my day gig as
the editor of Front of House magazine, I talk to mix engineers and system techs for the
biggest tours and shows in the world and speak regularly to the owners of sound companies, from the small local guy to the enormous international touring firm with dozens
of clients on the road at any given time. Not long ago, as I was putting together the
outline for this book, I started asking these people what they wished green sound guys
coming out of school knew when they got to the gig. Not a single answer concerned
anything technical, with the exception of a few people who expressed the wish that new
guys had a better handle on the proper mic to use for a given application. The rest of the
answers were all about work ethic, teachability, and attitude. Some of the answers to
the question ‘‘What do you wish they knew’’ appear near the end of the book. But the
bottom line is very simple: Soundcos, whether they are the touring, one-off, or in-house
venue variety, want people who are willing to work hard, not complain about it, and be
open to suggestions and able to follow directions.
One of the biggest issues with new sound guys is a tendency to think they know everything there is to know when they arrive at their first gig. Worse, many will ignore or
even actively contradict the instruction or advice of seasoned audio guys. Part of it is the
arrogance of youth. Put it aside. The guy with the gray hair who gets a little lost navigating the latest digital console or system controller has a wealth of information and
experience. And most of them are quite willing to share if given the chance.
But I’m Still on My Feet . . .
This is not just a question of the best and fastest way to get a job done—although that is
crucial information when you are working a show that has three hours from the time
the trucks arrive at the venue until things have to be ready for sound check. Many times
this could be information that saves your life.
Working in a live production environment can be inherently dangerous. Think about it.
We are moving and deploying big, heavy pieces of gear, large portions of which ‘‘fly’’
above the stage or audience. We work with electricity, and a large show may have
things such as hydraulics and pyro to worry about as well. Every year we lose at least a
handful of usually young techs because they were not following basic safety requirements. Experienced crewmembers only become such by keeping from getting injured or
killed on the job. If someone tells you to do something that you know is cutting corners
in the area of safety, you should refuse to do it and talk to a crew chief or production
manager. But the instances of someone doing that will be pretty rare. This is a place
where you really want to watch and listen to crewmembers who have some time in the
field. This is a business where the best training is of the on-the-job variety.
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Now it’s time to get down to business and look at working a typical gig. For the purposes of this book, I will use a smaller gig with a crew of two or three people as an
example. If you are on a big show, you will only do a fraction of this. But just as I used
an old-school analog mixer for the examples in the gear part of the book because the
principles are the same as you get higher tech and more digital, so it is with small-gig
activities having a comparable function on bigger shows and in bigger venues.
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he gig does not start when the band starts to play or even when you start setting
up gear. The truth is, it starts weeks earlier in most cases. As we address this
subject area, I’m assuming that you are going out as the audio provider. Either
you are working for a small company or you are working for yourself. On a tour or
working directly for a venue, these are not things you will be concerned with. But I find
that even guys who work for a larger sound company often have their own smaller
company as well, if only because that is the way they get to actually mix and do more
than just what the crew chief tells them to do. And virtually every sound company I
know started like this—from the smallest local shop to the biggest international touring
company. Not everyone reading this will opt to run his own sound company. But I can
virtually guarantee that most of you will at least try it.
As we move through the day on a typical gig, I am using as a point of reference a festival
gig that I did for many years. This was a parish festival at a local Catholic church. The
two-day event included everything from acoustic duos, to three-piece punk bands, to
student big bands, to children’s choirs and ‘‘folklorico’’ dance troupes, to a headlining
dance band each day. Between acts we played canned music that I provided. Over the
years I went from basic sound, to sound and lights, to sound and lights and my band as
one of the headliners. The person at the parish I worked with went on to run a bigger
fundraiser at the large private high school in town and hired me again. That was big
enough that I had to hook up with another sound provider to have enough gear, and I
got to mix one of my favorite singers from the ’70s, Brenton Wood. (As a side note, one
of the festival organizers ended up in my band and has been singing with us for almost a
decade now.)
Here is the key thing: I got the gig when a sax player I worked with who had done the
gig the year before called and asked me if I could take the gig because he just didn’t want
it. He told me that it was a giant pain and that the people were difficult and yada yada
yada . . . . I ended up doing the gig eight out of the next ten years, and the two years I
didn’t do it were one when it was cancelled because of construction at the church and
another when a new group of organizers decided to do it with all volunteer providers.
They called me back for the gig the next year even though the gig was near L.A. and by
that time I had moved to Las Vegas.
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What I am getting at is that a gig that someone else decided was not worth the time led
to lots of other work, and by the time we were done, I was making more than double
what they were charged the first year. As time went on, my gear got better as the festival
got bigger, but I did not keep the gig because of my gear. I got asked back over and over
because I was easy to work with, did a good job, and went the extra mile to ensure that
the festival was without hiccups. But most important was that I understood who I was
working for.
Too often on gigs like this, local sound providers make the mistake of trying to act like a
touring provider who is working for the band. Always remember that the person who is
signing the check is the person who you are ultimately working for.
Example: One year the festival moved from its normal location to a pavilion outside the
Rose Bowl. If you have ever seen the area around that world-famous stadium, you know
that it is surrounded by multimillion-dollar homes. These people bought big, expensive
houses up the hill from a stadium that holds about 100,000 people and expected that
the stadium would never be used. And because they have money, they have power, and
so the volume restrictions at the Rose Bowl are flat-out crazy. (Trust me, in another
professional lifetime I was a newspaper editor in Pasadena, California, and the fight
between the residents and people using the stadium for anything other than occasional
football games was legendary and something I became very well acquainted with.)
I knew from years past that we were in for a ‘‘situation’’ because the crowd at this gig
wanted to party, and the gig often went an hour past the scheduled time—which was
fine with me, because I got paid extra. But the Rose Bowl rules said that any measurable
amplified sound after 10 p.m. resulted in fines of something like $1,000 a minute. So I
reminded the band of that when they set up. I reminded them again as they were starting
their last set and again at 15 minutes out. And again at 10 minutes. And again at five,
four, three, two, one . . . and at 10 p.m. I shut down the PA mid-song. The band was
angry and made sure I knew about it. But the city sound cop went away happy, as did
the organizer, and I got asked back again the next year.
Okay, lecture over. Back to the gig . . .
The first thing any sound person running a gig of any size should do is find out just
what he is in for. This is known as advancing the gig, and it includes everything from
communicating with the acts or their reps to determine what they need and making
sure you can provide it, to actually going to the gig site to make sure you have the lay of
the land.
Advancing the Gig
In a perfect world, you would be able to go to the venue a couple weeks before the gig,
but this is not always practical, especially in a touring situation or even if it is a local act
playing a one-off out of town. Regardless, there are certain things you need to know.
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If you can’t go and look with your own eyes, then make sure to ask all of these questions
and get photos if you can.
■
Find out the venue location. This needs to be totally specific, as in a real address
and ZIP code. That last one is crucial. Why? I’ll use a local example. There is a
freeway called the 215 Beltway that circles the entire Las Vegas valley. So if the
venue manager tells you the room is at ‘‘Durango and the 215,’’ it could be in one of
two very different places.
Because the 215 is a beltway, it crosses Durango at both the south and north ends of
the valley. And if you go to the wrong one, you will add a good hour to your travel
time once you figure out you are in the wrong place, get directions to the right place,
and make your way the nearly 20 miles between the two points. Or worse, suppose
the venue verbally told you that it’s at the 15 and the 215. Same deal: The two
highways cross at two points again, and if you go to the wrong one, you may have
more serious problems. Not only does the resort corridor known as the Strip lie
between the two points, but so does the crazy interchange in downtown Las Vegas
called the Spaghetti Bowl and the big construction north of that. An hour is getting
off easy. Not only that, but the two areas may as well be on different planets. At the
south intersection, you are a couple miles south of the Strip in a very nice, upscale
area, and the clubs here are the kinds of places where the sound crew will be expected to do the actual gig wearing something nice, just like the folks in the club.
Farther north, out by Nellis Air Force Base, the area is a lot rougher, and your main
need will be a cell phone and perhaps some personal protection. And no one cares
what you wear.
■
Get a list of what gear the venue has in house. I am an advocate of the ‘‘I would
rather have it and not need it than need it and not have it’’ school of thought, so I
tend to bring more gear than I need to. So even if I am told that the house supplies an
installed house PA and all I need to do is plug my console into their system, I will
probably bring speakers and amps anyway, just as a precaution. But when the truck
is packed, the speakers will be the first thing on it so that if I get there and find out I
don’t need them, they can stay on the truck and save time and labor on load in and
load out.
■
Get an accurate description—with a diagram or pictures—of the load in/out situation. The difficulty of the load in will have a huge impact on how much pre-gig
time you need. Is there a loading dock? Is there a ramp? Remember that a typical
loading dock is not helpful if you are carrying gear in a van. A ramp is much more
helpful in that situation.
How about the distance and terrain from the gear drop point to the stage? I remember one gig I did for years where the load in was down a fairly narrow corridor
that was lined with nice wood wainscoting, and we had to be very careful to get the
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gear in and out without marring that wood. It added a good 20 minutes to the
load in.
Are there stairs? My tech editor reminded me of a gig he used to do—by
himself—where the stage was 10 feet and a flight of stairs above the main floor.
Amps, speakers, everything up the stairs with no help. Fun, eh?
■
Find out where the FOH position is in relation to the stage. On smaller club gigs, do
not be surprised if there is no FOH position and you are expected to mix from the
side of the stage. If that is the case, find out whether there is an area that you can use
as a mix position and make sure you are carrying cable covers—pulling 20 feet of
gaff tape off a snake at the end of the night is no fun at all.
■
Find out about the electrical situation. If this is a small gig, chances are that the
person you are asking for that information will have no clue. This makes it doubly
important that you know what you need. A little more math: If you know the voltage, you can calculate the power needed to drive your gear. The formula is
watt/volts=amps (current). So an 1,100-watt power amp in a 110-volt circuit needs
10 amps. Amplifiers and lights pull the most current of anything on the stage. For a
typical audio-only gig, you are generally safe asking for four 20-amp services.
Again, it is likely that the person you are talking with will not have that information,
so—either in a trip to the venue to scope things out or before load in—there are a
couple of things you can do to figure out your power. The first one is cheap (in the
$50 range at Home Depot)—it is called a circuit sniffer. You plug one piece into an
outlet and use the other piece to pass over the circuit breakers in the service panel.
When you are above the breaker that controls the outlet you are plugged into, the
sniffer will light up. A little bit of moving things between plugs and some trial and
error, and you will find out just how much current you have available and where.
Unfortunately, few small venues were wired with the needs of audio and lighting in
mind, and it is not unusual to find that all of the outlets in the stage area are on
the same circuit, which could spell disaster. When faced with that kind of situation,
you have a couple of choices. The first is plenty of good, heavy-duty extension cords
(not the orange ones from Home Depot). The other option is a power distributor.
For a long time, this was something you needed a licensed electrician to hook up,
but lately Peavey has been making a great piece called the Distro that plugs into a
220-volt appliance outlet and provides 16 outlets on eight 20-amp services.
See Figure 21.1.
Figure 21.1 Back view of a Peavey Distro. Image courtesy of Peavey Electronics.
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As long as you can get to a 220 outlet, this will solve most of your power issues.
(Hint: You will usually find an outlet like this in the kitchen, so you will still need
those good extension cords.)
■
Know who to call. Make sure to get at least two names and cell phone numbers of
people you can talk to if you get to the venue and something major has changed, the
doors are locked, or whatever.
All of this is just advancing the physical venue—it is not the hardest and most detailed
thing about advancing the gig. For that, we need to enter the magical world of tech
riders and stage plots.
You may have heard or read about the outrageous demands made by some artists for
how their dressing room is to be stocked. But this is not about bowls of M&Ms with the
brown ones removed or fresh flowers on every table-like surface or a case of Jack
Daniels in the fridge. There is a document called a technical rider that outlines the gear
that an act expects. When you accept the gig, you are accepting the rider unless you have
negotiated something different beforehand.
Riders can be really over the top in their requests. This seems to be especially true of upand-coming artists. They may have been doing clubs on a 16-channel mixer and blownout speakers six months ago, but now that they are really touring, they want the best.
The list that follows is from an actual rider for a not-yet-famous touring act. Take a
look at it, and we’ll meet on the other side to talk about what it all means.
Front of House Audio System
(A) Front of house sound system shall be an active four-way stereo system, capable
of producing an unequalized frequency response of +/3 dB 50 Hz–18KHz at
an undistorted signal of 120 dB SPL at the front of house console in any venue.
For outdoor events, delay stacks should be made available.
(B) The FOH (Line Array Only) enclosures will be EAW, ADAMSON, V-DOSC,
McCAULEY MLA-6, MEYER M3D, or JBL VERTEC. Any proprietary enclosures
must be approved by Producer/Artist’s Production Manager.
(C) The FOH speaker enclosures must be properly positioned and capable of producing a flat response for all sold seating areas. This includes front filled position
(in front of stage) driven by a matrix or auxiliary send.
(D) The FOH speaker enclosures are to be powered adequately and accordingly to
speaker and driver requirements.
(E) Power amps are to be CROWN, LAB GRUPPEN, or QSC. Ex: Crown IT-8000 to
power subs. Crown IT-6000 to power mids. Crown IT-6000 to power mid-high
and highs.
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(F) Cross Overs/Systems Processors accepted are LAKE CONTOUR, XTA, KT, or
BSS. (Located at FOH)
Front of House Console and Processing (Fly Dates Only)
(A) FOH console is to consist of fifty-six (56) channels. Ex: Consoles accepted:
A. T. I. PARAGON II, YAMAHA PM 5000 or 4000, MIDAS XL-4, MIDAS
HERITAGE 3000. NO EXCEPTIONS!
(B) FOH processing equipment is to consist of: 1. Lake Contour EQ, KT-Helix,
KT-3600, KT-DN360, TC Electronics EQ Station, TC Electronics 1128, or BSS
1/3 Octave E.Q.’s.
(C) One (1) EVENTIDE H/3000, One (1) LEXICON 480, One (1) LEXICON 200,
Two (2) YAMAHA SPX-2000, One (1) T.C. ELECTRONICS D-TWO or
ROLAND SDE-330.
(D) Twenty (20) channels of compression, KT, BSS, or DRAWMER.
(E) Eight (8) channels of gates, KT, BSS, or DRAWMER.
(F) One (1) KT-DN60 or A.C.I. SA-3051 spectrum analyzer.
(G) One (1) pro compact disc player.
(H) Three (3) Clear-Comm stations with beacons and handheld sets. This is to be
separate from lighting communications.
Monitor Console and Processing (Fly Dates Only)
(A) Monitor console is to consist of fifty-six (56) channels. Ex: Consoles accepted:
YAMAHA PM-5D, YAMAHA PM 5000, MIDAS XL-4, MIDAS HERITAGE
3000. NO EXCEPTIONS!
(B) Monitor processing equipment is to consist of: 1. Sixteen (16) channels of E.Q.
E.Q.’s accepted: T.C. Electronics EQ Station, T.C. ELECTRONICS 1128’s
(w/remote fader controller), KT-Helix, KT-3600, KT-DN360 or BSS 1/3 Octave
E.Q.’s.
(C) Fourteen (14) channels of compression, KT, BSS, or DRAWMER.
(D) Eight (8) channels of gates, KT, BSS, or DRAWMER.
(E) Seven (7) YAMAHA SPX-2000, REV 500, or PRO R3 Reverbs.
(F) Five (5) Shure PSM-600 Hardwired IEM units.
(G) Eight (8) Shure PSM-700 IEM systems with beltpacks and antenna combiner.
(H) All necessary cabling for IEM systems and spare beltpacks.
Rider current as of February 2, 2010
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Microphone and Mic-Stand Package (Fly Dates Only)
(A) Twelve (12) Radial J-48 direct boxes
(B) Four (4) Shure SM-58
(C) Four (4) Shure KSM-32
(D) Three (3) Shure KSM-27
(E) Five (5) Shure SM 57
(F) Four (4) Shure SM 98 w/ mic mounting hardware
(G) One (1) Shure Beta 52
(H) One (1) Shure SM 91
(I) Four (4) Shure KSM-137
(J) Twenty-four (24) round base mic stands with booms. Twelve (12) tall and
twelve (12) short.
(K) Six (6) Z Bars.
Okay, let’s start from the top.
■
They want it loud.
■
There are a lot of venues where a line array is a bad call, but they are spec’ing a line
array ‘‘only.’’ That along with the fact that one of the few speaker choices they are
giving came out a decade before this rider was written and that the company in
question has released at least three highly regarded line arrays since that time says
one of a few things: 1) What they really care about is brand name. If you have one of
those brands, you are probably going to be able to get the act’s PM to sign off on it.
2) The rider was written by management and not the sound guys. 3) This young act
took the rider from another act and just copied it.
Let’s skip ahead a bit to the consoles. Note that the choices are pretty narrow and that
the FOH models are all analog. The monitor choices include one industry-standard
digital desk. Now go back to the list of possibilities of the origin of the rider and what
that means. Given the console choices in conjunction with this, I can tell you virtually
for certain that the act is traveling with their own FOH and MON engineers, and they
either wrote or had a large hand in writing the rider. Either that or it is copied from
another artist and those are the consoles their sound guys like.
So if you don’t have all this stuff, does that mean you don’t get the gig? No, it does not.
In my experience the real make-or-break items are console, amps, and speakers. If I
have at least two of the three of those, I am probably okay as long as one of those is the
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console. If you have all of the big three, then everything else can likely be negotiated.
(Also note that this is a very strict rider. Most of the time you will see the words ‘‘or
equivalent’’ next to every piece of requested gear.) The exceptions in this case are the
mics and IEMs. Note that all of them are spec’d as being from the same manufacturer.
What that usually means is that the artist or engineer is endorsed by said brand, which
can throw a whole other layer of complexity into the mix.
The important thing is to balance the request against what you have available, and if the
scale is even or tilting just a bit in your favor, then get on the phone and find out what is
negotiable and what is not. You may have to put your sales hat on here—there are
plenty of brands that work just as well as the ones listed, but you are going to have to
make a case for this.
The Plot Thickens
The next items you will deal with are the input list and stage plot. These will vary
radically depending on the kind of music the artist plays. For example, most rock and
country acts will ask for a fully miked drum kit—that is, a mic on every drum (often two
each for the kick and snare) and the hi-hat plus overheads. A jazz act may ask for just
overheads and maybe a kick mic.
For our purposes, in Figure 21.2 I am using the plot I provide for gigs I play with my
own band. Some terms to make sure you have straight: Stage left and stage right are
always defined as being from the perspective of a person standing on the stage and
facing the audience. In other words, if you are at the FOH position, stage left is always
on your right. Got it? Downstage means closer to the audience.
First note that it is oriented as a bird’s-eye view, and the text assumes that the person
reading it is on the stage and looking toward the audience. Everywhere you see the word
‘‘vox’’ is a vocal mic. The drum miking is not specified because most of my gigs are in
places where the kit is provided and pre-miked. Also, everyone uses in-ear personal
monitors, so there are no wedges on this plot.
The input list may be an actual separate document with a list of the instruments and
vocal mics needed, or it may be a series of notations on the plot itself.
The final piece of paper you need before the gig is the most important one—a contract.
We are not going to get deeply into the contract except for a few notable items.
■
List all of the gear and personnel you are providing.
■
Note any substitutions from what is listed on the rider along with the name of the
person who approved the change. When possible, make these changes via email so
you have a paper trail in case of a problem later.
■
Note the date and time of both the show and the load in/load out.
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Diva 2
(Laura)
Vox
wireless
Diva 3 (Linda A)
Vox wireless +
2nd Vox headset
Guitar 2 wireless
Acoustic guitar DI
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EVERYONE IS ON EARS!
Bill
Vox wired
Guitar 1 wireless
te
Flu ic
M
Music stand
Music stand
Music stand
Sax
(wireless?)
Trumpet
(wired)
Trombone
(wired)
Percussion
Table &
Diva 1 (Linda
E) Vox wired
Music stand
Bass Vox
DRUM CAGE 8' wide x 6' deep (??)
Guitar Amp 2
Fender
Princeton
miked
Guitar Amp 1
Line 6
Bogner/
Radial DI or
miked
KEYS
Kurzweil PC88 or Motif
Korg CX3
On two-teir stand
4 DIs
Bass Amp/DI
6 FEET
10 FEET
24 FEET
REV BILL and the SOUL BELIEVERS
JANUARY 9 & 10 2009
CANNERY EASTSIDE--MARILYN'S LOUNGE
Figure 21.2 A Rev. Bill stage plot for a casino lounge gig.
■
Note the amount you are to be paid, including a reasonable deposit, and the timing
and form of payment. Unless this is someone you deal with regularly, it is pretty
standard to ask for a deposit of up to 50 percent and to have the balance due and
payable by credit card, cash, or cashier’s check prior to the start of the gig.
The reason for the last part is that the sad truth is that local production companies are
the first ones thrown under the bus if the gig goes badly. If the show sells poorly and the
promoter and/or venue lose money, it is not your fault. But rest assured that someone
will try to make it your fault, and you will find yourself chasing payment. Many sound
companies make a policy of having the client/promoter pay after the system is producing sound but before the gig starts. Spell everything out and get paid in advance. After
the gig, your only recourse is to sue. Before the gig, you can pull the plug and leave with
the gear. The latter provides a lot more leverage.
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On the Gig
ongratulations! You got the gig. Now what?
Again, we are looking at this with the assumption that you are working for a small
company where everyone does everything, or you are working shows with your own
gear.
Welcome to the Working Week
Most of the companies I know run on a Tuesday through Sunday workweek. The majority of gigs are on the weekends, so Monday tends to be everyone’s day off, and a lot
of small shops are just plain closed on Mondays. On Tuesday morning, the agenda often
calls for breaking down and cleaning up gear from the gigs of the past week. Cables that
went bad are repaired, rack cases get a fresh coat of paint, work boxes get restocked. In
other words, everything that is independent of any specific gig gets taken care of.
Next up is looking at the gigs for the coming weekend and making decisions regarding
gear and crew. Even if you are a one-man shop and there is one gig, this is a needed step.
Time constraints, location, and even just the limitations of your body will often require
hiring additional help at least for load in and load out. If the gig requires extra hands,
then book them early—if you wait until late in the week, the good ones are already
taken.
Side trip—I did a gig a few years ago at a county fair in rural Utah. The crew consisted of
me and the sound company owner, who was mixing FOH. (I was mixing monitors.) The
promoter was to supply three hands for load in and load out and to handle extra stuff
during the gig. What we did not know at the time was that the labor pool consisted of
inmates from the county jail. A good hand at least knows which end of an XLR cable to
plug into a mic. Let’s just say these guys were less than helpful. . . .
Okay, back to the shop, where it is Wednesday. Time to pull gear for the gig. If you
are working for a small company, this may be a moot point because you will be using
everything in the shop on every gig. But even if that is the case, this is when gear gets
taken out of cases, inspected, and tested. Especially in cases where the gig is not near
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the shop, this is a crucial step. One thing you can have happen is that you get to the
gig and have an equipment failure that could have been avoided with a bit of prep
time in the shop.
If space and time allow, I like to take this time to set up the rig exactly like I will at the
actual gig and test each part to make sure that not only are all of the individual parts
working, but that they are all working together as well. If you have the means and the
space, you want to test everything at as close to show volume as possible. There are
problems that may not be apparent at lower volumes but that are very obvious at show
levels. An intermittent connection on a piece of gear may not crop up in the shop at low
volume, but under the assault of multiple subwoofers . . . well, you get the picture.
Once the rig has been assembled and tested, then it is time to pack it all back up and get
it ready to go. Crucial: Always use a written list! The list will be your bible at several
points—when packing up, when loading the truck, when loading into the venue, when
loading out and repacking the truck, and finally when unloading the rig back at the
shop. Use a checklist at each of these junctures, and you will have moved a long way
toward a more successful gig.
Showtime—No Sleeping In for You
Show day has arrived, and it is going to start early and end late. Some people like to load
the truck the night before the gig, but unless you have 24-hour armed security or the
ability to park the truck inside a building and lock the building down, I do not suggest
this approach. I almost never hear about gear being stolen from a shop, but I hear about
loaded trucks and trailers being stolen on a pretty regular basis. Remember, theft is
usually a crime of opportunity, and a loaded and unattended truck or trailer just
screams ‘‘opportunity.’’
So how early do we start? Probably the most common reason for a gig going badly is
rushing setup. Rushing leads to forgetting and shortcuts that come back far too often to
bite you in the butt. The Rev.’s Rule of Thumb is take the amount of time you think it
takes to unload the truck and load into the venue (of course, you already advanced the
venue and are accounting for things such as the loading area being located farther away
from the venue, needing to use freight elevators, stairs, and so on, right?) and double it.
So if I think it takes two hours for me to load in, set up, and line check, then I shoot for
being at the venue four hours before the band. The same rule goes for travel. If the venue
says they are an hour from your shop, and you have checked Google Maps and they
appear to be right, allow for two hours anyway.
So using that math, if the band is supposed to arrive at 5 p.m. for a 6 p.m. sound check
with doors at 7 and the show at 8, and you have a two-hour setup and a one-hour drive,
then I would suggest being at the shop by 9 a.m. Remember, you still have to load the
truck.
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Now we magically jump ahead in time. You are at the gig, the truck is unloaded, and the
cases (or actual gear for things like speakers if you are not at the ‘‘case mandatory’’
point yet) are more or less where they need to be. Try to make sure that you have a place
to store empty, or ‘‘dead,’’ cases. If there is no place to do that (here comes the whole
advancing the venue thing again . . . ), and you have to put them back on the truck, then
that time has to be figured into both load in and load out. The order in which setup
occurs really depends on the type of gig, the number of people in the crew, and the way a
particular group does things. If you are new, it’s best to find out how things are done
and then do them that way. No one wants to hear the new guy saying, ‘‘Well, when I
worked for Rock Star X, we did it this way.’’ No one cares how you used to do it. One
of the cardinal rules of the gig (more of which you will find as we approach the end of
this tome) is that there is the right way, the wrong way, and our way.
The big exception here is when the issue is safety-related. If you are being asked to do
something that puts you or someone else in physical danger, then bring it up to your
direct report. If they can’t or won’t address it, then take it up the ladder. If the answer
you get is, ‘‘That’s just the way we do it,’’ then be prepared to bail and lose the gig and
probably any future work with that company. But do it, because it is better to be gig-less
than dead. There have been far too many instances of new crewmembers doing things in
an unsafe manner either because they were told to do it that way or because they just did
not ask and get told the right way. And too many young crewmembers are killed or
crippled every year. Don’t become a statistic. Safety is paramount—always.
How We Roll
My approach is to get the main part of the system set first: FOH console, speaker
processors/crossovers, amp racks, and house speakers placed, wired, powered up, and
tested. When I know that part of the system is up and working, then I can deal with
everything else.
You may notice that I am not giving specific instructions on how to connect parts of the
system. That is because although certain principles apply to every system, there are too
many variables even with legacy analog systems to do an accurate overview, and the
onset of digital makes it even more open to interpretation. But there are some inviolate
rules, mostly having to do with power.
■
Whenever possible before connecting two pieces of gear, make sure both are powered off.
■
If you have to plug into something live—say, a line from the stage into a console
channel—make sure the receiving end is powered down. In the case of a console,
mute the channel.
■
Mute the console before powering up or powering down any gear hooked into it.
Many pieces of gear will produce a very loud transient on power up or down, and
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that transient can damage amps and speakers farther down the line. Muted or not,
never turn the console on or off when the amps are still on.
■
When applying power, start at the ‘‘small’’ end of the signal chain and move toward
where the signal is the loudest. On power down, go in the opposite direction. Stage
elements (not including monitors) first, processing gear next, console (muted and
main faders down), speaker processors, and finally power amps. Then unmute the
console with the main faders still all the way down and slowly bring up the faders to
avoid any nasty surprises.
When working with digital components, the cardinal rule is that once the signal is in the
digital domain, leave it there as long as you can. Every A/D-D/A conversion degrades
the quality of the audio. If you are using a digital split snake, and your console has the
appropriate inputs, use them. The same goes for every step in the chain. Ideally, the
switch from analog to digital will take place as close to the source as possible, and it will
stay digital until as close to the speakers as you can get.
Getting Pinned
Once the main system is up, do the same thing for the monitor world. Once you know
that everything is up and on and that signal is getting from the console to every wedge or
personal monitor receiver and making sound, then you can move on to setting, or pinning, the stage.
Refer to your stage plot. If backline (amps, drums, and so on) are already there or you
are providing them, then get everything miked and direct inputs plugged into the system. If backline is coming with the band, then set mics and DI where the plot indicates
they should be. Line check every input at both mains and monitors. Have someone on
the stage tap each mic as you confirm signal is getting to the console.
Once you know everything is working, it is time to dress the stage. This means making
sure that cables are out of sight and do not pose a tripping hazard. Gaff tape (never duct
tape), plastic cable tunnels, and stage draping are all your friends here. And although
this step may have nothing to do with you providing good audio, it may have everything
to do with you getting the gig next time.
When you interview for a job or meet with a potential client, you don’t show up in
shorts and a T-shirt, even if that is what you wear every day at the shop. If you’re smart,
you don’t do the gig dressed like that, either. Think of the stage as an extension of you
and your services. Given similar production experiences and prices between two companies, with one providing a stage that looks tight and clean and the other a stage with
cables strung across it and scratched and dented gear, which one will get the call for the
next gig?
With the stage set, pinned, and dressed and audio being produced by the system, you are
off the clock until the artist or their rep arrives. This is a good time to get with the client
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and get paid. Unless this is a client with whom you have a regular, ongoing relationship,
you need to do everything possible to get paid prior to the gig. If you are working for an
entity like a corporation with invoicing and billing policies that prevent getting paid
before the show begins, then make sure you have a hefty deposit. My policy is that I
always look at any money that comes to me after the gig as gravy. I have to cover all
costs, including labor and fuel, with the deposit. After the gig—unless it is a regular
customer—the only way I can go after monies owed and not paid is through the court
system, which can take a long time and can be less than effective. I once had a contractor working for me on some home repairs. Things were going poorly, and I finally
lost it with him and told him that if I did not get what I had paid for, I would sue. He
told me to go ahead, that he would just go out of business and reopen under another
name, and that there was nothing I could do about it. The point here is that if someone is
planning on cheating you, they’ll find a way to cheat you. The only way out is to get
paid in advance. A good contract can help, but given the choice of a good contract or
cash in hand, I’ll take the dinero every time.
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T
Hello (Hello . . . Hello . . . Hello)—
and Welcome to the World of
Delay
he terms delay and echo are often interchanged. In the world of FX that is fine,
but we are now talking about time-aligning a system, and the idea is to get rid of
echoes, not to create them.
Without getting into scads of physics, there are a couple of reasons why this is a necessary step. Let’s start with a situation where in order to achieve even coverage, we need
to place speakers in places other than the main hang. Now let’s say an audience member
is standing next to one of those auxiliary speaker stacks. If you don’t align the system,
what that audience member gets is the sound from the speaker closest to him followed—often louder—by the sound from the main PA. Why does the sound from the
speakers farthest from the stage hit before the sound from the speakers right at the
stage? Because light moves much faster than sound, and the electrical signals in the
system are moving at the speed of light. So, the signal to that second speaker placement
gets to that speaker at virtually the same time as it does to the main speaker hang. The
sounds from each source are produced at virtually the same time, and it takes the sound
from the main stack longer to hit your ears than it does the sound from the speaker right
next to you. Depending on the distance between the two, the result will be anything
from a muddy, indistinct overall tone to actually hearing the same sound as two distinct
events. And that is why those aux speakers are usually called delay stacks. To clean up
the sound, you need to apply delay to the speakers farthest from the stage so that the
sound emanating from them is formed at the same time the acoustic energy from the
main stage is arriving at that location.
And it is not just big gigs with multiple speaker locations where this is an issue. In fact,
small gigs often sound horrible because the PA has not been delayed to the backline.
Brian Klijanowicz, who writes for me at FOH magazine did a great piece on this subject
that he has allowed us to use here. Take it away, Brian . . . .
A Realistic Approach to Subwoofer Time Alignment Time alignment is a very important, yet very often overlooked aspect of system setup and tuning. A correctly
time aligned system has many benefits, including more even coverage where two
sound sources overlap and a more even response across acoustical crossover points.
It can give even the cheapest of systems a couple decibels more in the area where
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engineers tend to like them most: bass frequencies. This leads to the topic for this
article—subwoofer time alignment.
Two ways to quickly achieve this are by using a sine wave and by delaying the PA
back to the kick drum. Either one or both will work better in different situations depending on the size of your gig, the time you want/have to work on it, and just how
much you care.
With the more frequent use of digital crossovers, system controllers, amps, and
consoles, it has become easier than ever to add delay to multiple signals, whereas
years ago you would have to eat up an entire rack space and insert cabling for just
one channel of delay.
This is intended to be a minimalistic, quick way to time align your system, so only a
few pieces of gear will be needed. Using a sine wave will only require a sine wave
generator and the capability to delay an output signal and to reverse the phase of it.
To delay the PA back to the kick drum, all you need is a channel of delay and your
‘‘golden ears.’’
For this piece, we will assume that the sound system we are working with is a standard front-loaded stack configuration. This means there is a subwoofer (producing
subfrequencies from roughly 100 Hz and down) ground-stacked on the floor with a
top box (producing roughly 100 Hz and up) directly on top of it. We will also assume
that both subs and tops are currently producing the same polarity and facing the
same direction (azimuth/splay angle).
Sine Wave Usually an engineer would not want to use polarity to cancel out a
signal. But that is the whole concept behind this technique.
First, it is important to find the acoustical crossover frequency between the subs and
tops. A measurement device is the most accurate way to do this. But if you don’t
have one, there is an easy way to rough it in. Assuming you don’t have a measurement device, flip through the pages of your crossover to see what the crossover frequency is and use that. (For this piece, it will be 100 Hz.)
Let’s get back to the concept. We will take two similar sound sources, sub and top,
that are playing a 100-Hz sine wave and flip one out of phase. The crossover frequency is the point at which both sound sources overlap and start to fade off from
one another. But more importantly, they will still reproduce 100 Hz, because that is
the beginning of the crossover filter on each source.
By physically looking at the speaker cabinets and knowing where the drivers are in
the boxes themselves, you will be able to determine which signal to add delay to.
Typically in the club world, top boxes are placed a little behind the front of the sub.
That way if the top box falls, it will fall back and not onto the drunken audience. So,
assuming this is the case, delay should be added to the sub. While the 100 Hz is
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playing through the sub and top, start increasing the delay to the subwoofer signal.
Eventually, the two signals will start to cancel out, and the total SPL will reduce substantially. Find the point at which most cancellation occurs and leave it at that.
Change the signal that’s out of phase back into phase, and you should have summation at the crossover point.
Some crossovers have an on/off function for the delay. A good way to check whether summation is occurring is to flip the delay on and off to hear the difference.
This whole process can sometimes be done with music—preferably with driving, kick
drum–heavy music. However, it can be hard to distinguish what is cancelling and
summing in the crossover region with full-range music playing. A quick fix on a digital
console would be to throw a low-pass filter on the music channel.
Back to the Kick Delaying the PA back to the kick drum is very simple. It works very
well in the small ‘‘hole-in-the-wall’’ biker bar, but once you get into a bigger venue, it
can become a bit harder to hear the difference. Smaller venues that have lots of
reflective surfaces and 90-degree walls create standing waves. This makes it almost
pointless to spend the time to get out a measurement device—especially when the
club owner wants you to be done setting up before the first road case is even in the
building.
Since this technique works better in small venues, we will assume your PA is in a small
venue and set up in front of a band. Even though it’s still surprising to club owners
and patrons, we all know how loud a drum set can be in a small room. That is why,
typically, most engineers won’t even turn up all the drum mics, or they just won’t
mike the whole kit.
So the drums are roughly five feet behind the PA. Think of the kick drum as another
speaker. If you are sitting in the audience, would you want to hear the main PA with
another set of PA five feet behind it? Probably not. So if you are sitting in the audience listening to a band, you probably would not want to hear the kick drum once
through the PA and then again five feet later.
Most drummers do relatively the same sound check routine. They will hit each drum
individually with quarter notes at a medium pace until you ask them to switch to the
next drum. When you get to the kick drum, get a rough sound in on the channel strip
and start tweaking the delay back. Add a little bit of delay at a time. You will start to
hear a change in tone and depth. Once you get to the sweet spot and are happy with
the sound, that’s it!
Thanks, Brian. Before we leave the whole subject of delay, there is a reason to mess
around with delay besides cleaning up sound and getting even coverage. I am just going
to touch on this because it gets into the whole idea of psychoacoustics, or the way
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humans perceive sound. This is done in the theatre world all the time. Think of the
coverage area—especially the source point of the PA—as a kind of stage. Great care is
taken to ‘‘place’’ individual voices and other sound sources in a way that results in the
reinforced sound coming from the same direction as the original source. Then, the
whole PA is delayed back so that it is just a few milliseconds behind the actors onstage.
The conceit of theater is that real actors don’t need a PA. That may have been true in the
small acoustically designed theaters of times past. But with huge traveling productions
of Broadway shows becoming the norm, and as more contemporary and rockin’ music
has made its way to the stage, those days have gone the way of $35 seats for a Broadway
show. Really good theatrical sound designers know how to use placement and delay to
fool the audience into believing that the sound they are hearing is coming from the stage
and not from a PA system. Here is the psychoacoustic part: If a sound starts in one place
and then continues from another source, the brain will think that the entire sound is
coming from the initial source.
Get it? By making sure the initial attack is heard right from the stage and that the delay
to the PA is an imperceptible few milliseconds, the brain will tell its master that the
sound is coming from the stage. Neat trick.
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Backline Basics
W
e are going to step away from the sound system for a minute and into
another area that can be a great job for those into live event audio. You see
those things on the stage that are making all the noise? The amps and drums
and stuff? Well, here’s something that may come as a bit of a revelation: Except on
really small and really big gigs, that stuff is usually not carried by the band.
It comes down to a case of straight economics. At the low end, local bands and
the proverbial ‘‘band in a van’’ are carrying everything they own, which may actually
include basic PA. At the high end with big tours, they are carrying all backline and
production (audio, lights, video, and so on). And there is business and money to be
made everywhere in between. Some acts carry backline and board groups (consoles and
processing) and rent ‘‘racks and stacks’’ (amps and speakers) locally. And plenty carry
only things that they can get into the overhead bin on an airplane. The rest of it gets
rented.
Behind the PA
This is the wonderful world of backline—pretty much everything behind the PA. It can
include guitar and bass amps, drums, percussion rigs, keyboards, and drum kits. It can
occasionally mean instruments such as guitars and basses (although they are usually
there as backup, with the player bringing one main instrument on him and renting a
backup).
So, two questions. First, why would you want to deal with this stuff? The answer to that
is twofold and pretty easy: There is a lot of money to be made if you do it right. (One
sound company owner I know got a nice Gretsch guitar as a Christmas present from the
backline company he uses, which gives you an idea how much business he was sending
to them.) The second part of the answer is that although there is generally less overall
business in backline in any given city, the ratio of provider to amount of business works
much more in the favor of the provider than the ‘‘real’’ world of audio does. The bottom
line is that you make a lot less per gig than the company providing the big stage system is
billing, but to use my adopted hometown as an example, on a given weekend there may
be a dozen sound companies doing gigs in 20 venues. But 10 of the 12 companies are all
using the same backline provider, and no one is angry about it.
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The second question is whether you can do full audio and backline as well and just bill
for the whole thing. The short answer is yes, but the truth is a lot more complicated. The
two services require some very different investments and skill sets among those working
the gig. It takes every bit as much knowledge to maintain and set up backline as it does
to do the same with a PA. It is just a different kind of knowledge.
Getting the Gear Right
If you are thinking about doing the backline thing in addition to sound reinforcement,
there are ways to do it and make it work. The biggest challenges are getting the right
gear in and getting someone to care for it and set it up. Drums are pretty easy. Many—if
not most—drummers carry their own cymbals as a matter of course or can be persuaded
to carry them, so if you have good-quality ‘‘concert’’ kits with at least two mounted
toms, two floor toms, and double kicks or a double kick pedal, and they say Yamaha,
Pearl, Ludwig, or DW on them, you are probably okay. When setting rental prices,
make sure to keep in mind that even though drummers use the same heads on their own
drums for a significant amount of time, they will generally expect the heads on a rental
kit to be new.
You or your backline person need to know how to change heads, tune, and set up a kit
at minimum. But drums are pretty simple overall.
Percussion rigs are likewise pretty standard. An LP percussion table and a set of congas
that includes a conga, a quinto, and a tumbao (three sizes of conga drums) with a Toca
or Latin Percussion logo on them in a nice basic black will usually be sufficient. Percussionists will usually bring a suitcase full of hand percussion toys.
Guitar and bass rigs are simple. No stompboxes—most guys bring their own. For guitar, a good Fender Twin, a Marshall head and half-stack cabinet, and maybe a VOX
AC-30, and you’re in business. You or your tech needs to know how to do basic maintenance and repairs, plus how to change tubes and bias the amp correctly afterward.
For bass, an Ampeg SVT and maybe a Gallien-Krueger RB800 plus a single or dual 15-inch
cabinet and a 410-inch, and you’re set. You have the same maintenance and repair issues
as with guitar rigs.
This hardly covers every request, but it will cover the vast majority of them. I once got a
call from a local venue to rent my Line 6 Vetta guitar amp because the rider called for
one and they could not find one at a rental house in town, but they knew I had one. The
reason the rental houses did not have one is that people who want an amp like that will
almost always carry one with them.
This brings us to the biggie: keyboards. This is the tough one, because remember that
most synths are just big computers with piano-style keyboards on the front end. And
your computer is obsolete by the time you get it from the store to your car, so what do
you think happens with keyboards? Same deal. It becomes a case of making sure you
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have certain meat-and-potatoes stuff—Kurzweil PC88 piano, Yamaha Motif, Korg
Triton, and a good Hammond organ (very expensive and notoriously hard to maintain)—and you will be able to cover most requests.
A word about organs: There are very good tone-wheel organ emulators out there made
by Korg, Nord, Roland, and even Hammond but . . . I did the keyboard rental thing for
a couple of years. I had a Motif, a PC88, and a really nice Korg CX-3 organ. The Motif
and PC88 paid for themselves several times over. The CX-3 never did, despite the fact
that lots of guys play them. The bottom line is that if the request is for a Hammond B3,
then only a B3 will do. As for the rest of it, keeping up with the latest flavor and feature
set is really hard and very expensive, and each one you buy is really just a bet on how the
market will go over the next year. Stick to the basics.
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Hands on the Knobs
e are finally going to put together a mix. And you would think this would be
the longest, most detailed chapter in the book, but it is probably one of the
shortest and most general.
Mixing is very subjective, which makes it vital that you know who you are mixing for.
I can hear the collective sighs now: ‘‘The audience.’’ That’s the obvious answer. Except
the audience does not sign the checks. If you are to succeed, one of the hardest skills you
will learn is to identify who the person signing the checks is listening to when it comes to
production matters, including the quality of your mix, and make that person an ally.
When it comes to the actual mix, I have found some general principles to be helpful.
Fix It at the Source
The old joke about ‘‘fixing it in the mix’’ is only funny because those who really know
will tell you that there is only so much you can fix in the mix. Garbage in equals garbage
out. A lousy guitar sound coming off the stage can only be fixed to a certain degree in
the system.
It is always best to get the source sound as close to what you want to hear in the systemas you possibly can. Once again, here is where your interpersonal skills come into
play. You can’t go to a lead guitar player and start ripping apart his tone. A better
approach is to say, ‘‘It sounds great onstage [regardless of whether it does], but I am
having a really hard time getting it to sound like that in the system. If we could roll back
the low end a little, then I can get more of your sound in the system, and we’ll get the
low end back to you onstage through the monitors.’’
That still may not work, but telling the guitarist his tone stinks is a guaranteed nonstarter.
The other big stick in your arsenal is mic selection and placement, which is a subject that
could fill a book all by itself. For our purposes, let’s just say that you should experiment
with different mics on different sources with different placement anytime you get the
chance (but not on the gig).
But don’t get so set in how you do things that you can’t try something new or take
something someone else does and apply it to your gig. Mic selection and placement is
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one of the places where there is no real substitute for experience. Listen. Keep your eyes
open. Ask questions. And experiment—on your own time.
Find Your Foundation and Flow
How you set up a mix should be determined by the kind of gig, the kind of audience, the
kind of music, and the gear you have to work with.
When I first started with live sound as a musician, the PA was just the way we got the
vocals up above the band. Instruments and amps did not get miked unless they were
horns or acoustic guitars. And we were using small boards, so I always put the lead
vocal in Channel 1. And I kept doing things that way for a very long time.
I have since migrated to the typical rock setup of the kick in Channel 1 and the first
group of channels dedicated to drums. But this only happened as I started to do gigs
where everything was going through the system, because then it made sense. The kick,
snare, and bass were the foundation upon which the music was built, so once I was past
the stage of using the PA to add vocals to a mix of instruments coming off the stage, it
made sense to start there.
The foundation of the music will change by genres. For example, a straight-ahead jazz
drummer will almost never even have a mic on the kick. A couple of good overhead mics
to pick up the entire kit is a common setup. The hi-hat becomes the rhythmic foundation, so on a jazz gig I might put that in Channel 1. Note that if you do all rock gigs with
the kick in Channel 1, you make any assignment changes at your own peril. When an
adjustment needs to be made in the heat of battle, instinct takes over, and if you reach
for what you think is a kick mic and it’s a hi-hat, it can be a problem.
A couple of weeks before writing this, I was out on a gig in an arena that was using a
totally computer-driven mixing system with no physical control surface. Because the
control surface was virtual, it could be adjusted to the mixing styles of individual engineers. And the two guys on the gig had very different styles and were able to fit the
surface to their style, and neither had the inputs in anywhere close to the order that the
other one had them in.
Finally, don’t force a sound that is inappropriate to a musical style just because it is
what you are used to. Folk music needs to be mixed very differently from a rock show.
And failing to make that adjustment to honor the music is a huge mistake. I once heard
Howard Paige (whose list of A-list clients takes up several pages) ask the immortal
question, ‘‘Since when did the kick drum become the lead vocal?’’
Listen to every kind of music you can find and have a solid foundation of what it is
supposed to sound like before you take a gig mixing that genre. Then go with the flow
and let the music do the talking. It is not about us or our huge systems and egos. It is
about letting the artist onstage communicate his artistic vision to as many people as
possible with finesse and clarity. (Unless it is a punk gig . . . just kidding!)
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Triage
In medical terms, this is the practice of surveying the victims of some kind of accident
with a lot of injuries and deciding who to treat first based on the severity of the injuries
and the odds for survival.
In many situations today, we have the advantage of recorded shows and the ability to
mix shows before the curtain ever rises. It is not unusual for an engineer to spend a week
or more building a mix in a rehearsal studio using a recording of the act. But in one-off
situations, you may be mixing an act you have never heard.
Spend some time listening before you start adjusting anything. Get your main elements
under control. Drums and vocals are a great place to start. Then bring in what is needed
to build on the drums and support the vocal. If there is something you can’t fix, try to kill
it in the mix if you can and move on. Remember, it is not just an order-of-importance
thing; it is also a ‘‘Can I make it better?’’ thing.
And here is where knowing the music really comes into play again. Mixing horns is a
great example. People who do not understand horn-driven music will often get the mix
priorities all wrong. For example, in a three-piece section of trombone, sax, and trumpet, what gets the short end of the mixing stick? The trombone, almost all the time . . .
But an engineer who has really listened to the music is likely to tell you that the ‘bone is
the most important voice in the section—the foundation, as it were—and that he builds
the sax and trumpet on top of a really solid trombone sound.
Know the music. Know what’s important. Know where you can make the biggest
difference.
Less Is More—Softer Is Louder
This is especially true when mixing monitors, but it applies to FOH mixing as well. You
can often make a bigger difference and a better mix by taking things out rather than
continually boosting things. Here is that whole ‘‘know the music’’ thing again. Knowing the basic frequency range of various instruments is one of the most helpful pieces of
knowledge you can ever possess.
Let’s use monitors as an example. When a vocalist complains that he can’t hear himself,
don’t give in to the immediate reflex of boosting that voice in the mix. Instead, take a
good listen to the mix and identify things you can take out in order to carve out more
sonic space for that vocal. Maybe it is not about boosting the vocal, but rather bringing
down the guitar and giving the singer just kick drum instead of the entire drum kit in the
mix.
At FOH, this really applies to EQing a specific input. Make it a goal to always cut and
never boost EQ. No one can do that 100 percent of the time, but make that your goal.
For example, consider a hi-hat that is not cutting through. Maybe dumping all of the
frequencies below the high-mids is a better answer than boosting the highest
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frequencies. Or consider a guitar and vocal that are fighting each other. Maybe scooping some of the sonic range that the guitar and vocal share out of the guitar sound is a
better approach than just pushing the vocal higher in the mix.
Keeping your mix subtractive rather than additive will keep your sound cleaner, will
make it feel louder at a lower total volume level, and will lessen the possibility of
feedback.
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T
here are many places where you may end up working in the sound biz—
churches, schools, performing arts centers, theatres, resorts, and so on. But no
matter where we end up, most of us at least toy with the idea of going out on a
tour. There is a certain romance to the whole idea of being part of a big tour, and it can
be a very cool thing. Plus, a major tour always looks good on a resume, no matter where
you end up.
Ironically, living in the Live Entertainment Capital of the World (a.k.a. Las Vegas),
most of the guys I see on big shows got there because they were looking for a way to get
off the road. Once you are a touring guy, it can be hard to settle down and find something that pays as well and is as much fun as being on the road with a rock show. But the
job itself is a lot harder than most people think.
Before you get on the bus, it would behoove you to have a good idea of what a typical
touring day is like. At the time of this writing, one of the biggest shows out was Bon
Jovi’s ‘‘The Circle’’ tour. Mike Allison is a 30-year road veteran who works for Clair—
one of the biggest sound companies in the world—and is the FOH system tech and audio
crew chief on the tour. You might even say that touring is the family business, as Mike’s
son recently joined the tour and is part of the core audio crew that travels with the band.
Mike was kind enough to take time from his very busy gig to make notes on a typical
day. I get tired just reading it. Want to go out on tour? Here is an idea of what your day
might look like.
My Day—by Mike Allison
■
4:30 a.m.: Bus call (rigging call started at 2:30 a.m.). I take my bags and head to Bus
3. That is the lighting bus, but it leaves earlier, so I will take it to the venue and
transfer my bags to my own bus when I get a chance after it arrives at the venue at
8:30 a.m. I try to get the cobwebs out and start thinking about the gig.
■
5:00 a.m.: We arrive at the gig. I head in with my two other audio guys. As I go in, I
look at the load in. Does it have docks? Is it just dumping on the parking lot? How
much space is there as I walk in? If I have been here before, I tell my guys my plan.
This gig—the Bell Centre Montreal—I have been here many times. So as I walk in, I
see Michell, the head sound guy, shake his hand, and say hello to the other guys I
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recognize. The stage manager is in the process of dumping the stage trucks, so we
have some time. My two guys head to catering. I head to the floor to have a look.
■
5:15 a.m.: I get with our tour electrician and find out where we will get our
connection for power. As this tour is on the verge of needing generators, I need to
check every day to make sure where we connect.
■
5:25 a.m.: I see the head rigger and ask whether he has any problems with my points.
If not, then I look to see what kind of space we will be working with. We have so
much gear behind the stage that space is at a premium. Today, the rear-hang PA cabs
are flying right next to the back wall. We will need to get at them ASAP, or we will
not be able to get them up until the lighting truss goes up, which could be a long
time. I look for space to place the amps—not good today.
■
6:00 a.m.: I head to catering and grab some breakfast. It’s hard some days to eat
heathy; it’s all greasy food. But I try.
■
6:20 a.m.: I talk with the lighting crew chief about having some of the space that is
marked out for lighting repair world. (It’s very large out here . . . lots of lights need
work all the time.) I am glad we don’t have to repair the PA as much!
■
7:00 a.m.: It looks like we may get a truck soon. I call my guys on the radio and tell
them soon. Maybe . . . Take a look at space and let me know whether you want me
to send it out on the floor or around the back, through the vom. [ Note: Vom is short
for ‘‘vomitorium.’’ Really. An entrance into a theatre through a banked tier of seats.
It comes from the Roman term for an entrance into an amphitheatre.] This place has
lots of entrances. My guys think that the PA cabs should go on the floor. The amps
stage right have the long push around the stage, along with the cable caddies and
truss. Now the rear fill and side fill PA need to go another way.
■
7:30 a.m.: We get the PA truck and start unloading. It takes a few tries of telling guys
(local stage hands) that it’s not all going out on the floor. If they would listen, we
could get it done without so much confusion.
■
8:00 a.m.: First truck done. Now starting on the next truck. This one is just monitors
and FOH gear. But again, it’s not all just dumping out on the floor. Some goes down
the hall to FOH, some out where the stage is being built, some to stage left . . . We
get it all in place.
■
8:30 a.m.: Truck is unloaded, and now it’s time to get the sound hands and start
working. We get six guys. I take two and split the other four between my guys.
Dustin heads to the stage to get started on the cabling he needs to get done before
everyone else gets there, and he has to fight to get into the space under the stage.
Chris takes his guys and starts stage right PA. It’s the harder side, with all the dimmers and cabling for the ton of lights on the floor. The sooner he gets in to get space,
the better. I take my guys and start running the 2/0 feeder for power. It takes about
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15 minutes. Then it’s time to get on the rear fills. Stage left is a mess. All the lighting
and motion cables are on the floor, right where I need to fly the rear fills. I take a
look. If we can slide the cables out a little, I may be able to just get the speaker cabs
in. It’s a job, but we get space. It takes about 30 minutes to get them up. It’s three
stacks of four i3 cabs—not big. Thank god, because the space is so tight. We get
them cabled and flown.
■
9:00 a.m.: The other sound guys arrive (an FOH engineer and two monitor
engineers). Chris has a small problem and calls on the radio. I tell my hands to
wait for me by the cable caddies and see what’s up with the PA. Nothing serious—just
a question about pinning for the cabs. I take a look at the cable truss and everything
he has done so far just to make sure it all looks good and safe.
■
Right about now I lose track of time.
■
I get my hands and head out to monitor world. With the unions here, we need to use
our guys to tip the two monitor boards and get them in place. After we get them in
place, we head to FOH. And do the same thing . . . [ Note what ‘‘tipping the board’’
means. This tour is on all big, heavy Midas analog consoles. Each weighs about
1,100 pounds in the case. ‘‘Tipping’’ means getting them from the ground onto their
stands.]
■
By now, Dustin is over at stage left getting ready to fly the PA.
■
I take one guy and give him to Dustin, and the other I take to Chris.
■
During this time I am listening to the room. I have been here many times, so I know
how it will sound. But if it’s a place that I can’t remember or that’s new to me, I listen
to the room. When the stage guys drop something, does it echo a ton or does it taper
off? As the motion guys raise the video truss, does the sound of the motors slap
around where the stage will be? You get the idea—I listen to the room. Later, when I
am tuning, it will help me.
■
I head out to the stage left amps and wire them up with power, signal cables, and
network cables. They do the same for stage right. Now I have to figure out where to
run the cross-stage cable (signal and Cat-5 network cables). If I run them on the
ground, they will get buried with all the lighting, video, motion, and who-knowswhat cables. Not good for audio cables . . . So I run the cables around the first level
of seats. I have 250 feet of cable, so it’s not usually a problem to reach from stage left
to stage right. But today it just makes it.
■
I go to check on Chris. Everything is looking good. I head to stage left and tell Dustin
he can go to the stage and finish what he needs to do there. I will finish flying the PA.
■
With both sides of the PA floating just above the floor, I head up to the upper balcony to get a better look at the 11 motors that comprise the main PA, the downstage
side-hang and rear side-hang PA, and supporting truss. From below, if one motor
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stops, it is really hard to see; but from above, I can see it really easily. We have tape
measures on the main, so as we go up I can get a reference. We fly the PA at 32 feet to
the bottom. I get all the speakers and truss up to height. I look at level, do the pullbacks, and head to the rear speakers. We fly them independently, because they are so
surrounded that this is the only way to do it.
Next to stage right. During this time we are rotating out the stagehands to take a
break. Cabling the PA is always a job, with 220 feet of cables and long runs.
■
About now, the lighting and video trusses are up, and the stage is getting ready to
roll into place.
■
Chris heads to stage right to cable up the amp racks. I head to stage left amps. I have
two stagehands. We try to keep the cables falling straight down, and then we route
them to the amps. Sometimes, depending on where I get the amps, we have to add
the extension (between 30 feet and 60 feet). All the cables are Clair Hi-D, which is a
very large-gauge cable, so it’s like wrestling a snake to get it where you want it to go.
Very good for low loss to the speakers, but not fun to run.
■
The stage rolls into place, so it’s time to put the transformer split/patch rack into
place. This is where all the stage boxes (with all the mics) converge to get split to the
four different places (FOH, two monitor boards, and recording mix). This is a large
caddy that looks like a cable bomb when all the connections are made. It just fits
under the stage with a little room to get in to make all the connections.
■
As I am doing this, Chris takes the stagehands and starts running the cables for the
subs that are under the stage along with the front fills. You got it right: more Hi-D.
We have eight subs, 12 FF-2 (small front fill cabs), and two S-4s, the big old fourway Clair PA under the stage. So we need six of the Hi-D cables. They run under
stage left and out to the amps, all 100 to 120 feet in length. More fun with the cables,
as we need to run them in a very neat order and put them into large cable ramps. The
Fire Marshal really likes it that way, and we need to keep anyone walking from
tripping and keep gear from rolling.
■
After I make the connections, I take the hands to FOH to run the real snakes. We
have two runs, Bon Jovi and then the supporting act. They both are 100 meters, or
about 330 feet long. They have to connect to both the transformer split and the
amps.
■
Finally, it’s time for lunch. Most days it’s a little after 1:00 p.m. It’s always interesting in catering now, as we have the stagehands in eating. Some of the conversations are very interesting.
■
After I finish, I head to FOH to power up the system. I turn all the FOH gear on and
get all the computers running. All of the consoles are old-school analog—Midas
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XL-4 at FOH and a pair of Midas Heritage 3000s at monitors. But the drive system
is a very new Lab.gruppen amp Lake I/O system. I run three computers—one tablet
has Smaart on it, and there are two for controlling the drive. I start the I/O software
and wait as it talks with all the amps. (Each amp has a Lake digital controller in it,
which is on anytime the amp has power to it, even if the amp is off.) It takes about
five minutes (on a good day) for all the modules to check in and see whether they are
in the same state as the controller thinks they should be. If they are not communicating, or if they are showing offline, then I have to figure out what’s up and how
to fix it.
■
I get them all talking: 60 amps, 208 amps channels. It’s time to turn them on. With
the new amps, the real way to do this is with the software. I go to the PLM page and
hit All On. Wait 10 seconds, and they all start turning on. The software reports all is
good. Now it is time to check the system. Again, new amps—the only way to really
do this is with the software. So I start to mute the different outputs that go to the
Low, Mid, and Hi, Sun S-4, and FF-2. With all the channels, this takes a few minutes. I turn on the pink noise and start unmuting the outputs one at a time to make
sure every component is working and all connections are working. With a large
system, you can see this takes a while.
■
All systems go. Dave (Eisenhower—FOH engineer for Bon Jovi) comes out, and we
start tuning the system. We have a good system, and we know what we want, so it’s
a relatively easy time with the initial tune. Dave and I have the same feeling about
the PA. If you have good sounds coming in, and you know the PA sounds good, then
the tune should be easy. We do minor tweaks at FOH. Get the main PA delayed to
the subs. Then I fire up the second tablet and start the VNC program, which is
basically a server on the main tablet and a client on the second tablet. We run a
remote desk on the wireless to walk the room. It runs Wi-Fi at 2.4 GHz and 5 GHz.
As we are running a 360-degree system, we have to check from top to bottom, front
to rear. So we drive all the other guys crazy with the same song repeating for a while.
The goal is to reproduce the sound at FOH all over. Most days I believe we do a
good job. The system is so separated out that it makes this easy.
■
System tuned. It’s time for line check. I relax for a few and start planning the load
out. It has been in the back of my mind as we load in. When I am placing anything,
flying the PA and loading in, I am thinking about how the load out will go. If I do
that or place that there, how will that affect the load out?
■
Bon Jovi line check done. It’s time for support check. This is always one of the
hardest times for me. Depending on the act, most have not ever had any time in a big
arena, so they are a little out of their experience level. How do I keep them happy,
keep the gear safe, and keep the main act and production happy? Some of the mixers
think that LOUD is good, but a big arena with loud in it never works. The politics
for this are way too long to go into here.
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■
We get finished and wait for Bon Jovi sound check. Most days it’s a few songs. We
work on anything new they are doing tonight.
■
5:00 p.m.: Bon Jovi sound check on stage.
■
6:30 p.m.: Doors. Start the walk-in music. Time for dinner.
■
7:15 p.m.: Run the house safety announcement.
■
7:30 p.m.: Support act on. Keep their engineer ‘‘in the program.’’ Help if I can;
otherwise, just count the time.
■
8:15 p.m.: Set change. Quick line check. Get ready for the show.
■
CD in the recorder.
■
Radio announces four minutes to band. Start the intro.
■
The show starts. I wait for one song to let the band settle and Dave to get into the
groove. Then I take the secondary tablet and walk the arena. Usually it’s just small
changes—gain up here or down there.
■
End of show. Load out. The best way to describe it is controlled chaos.
Note a few things: His day started at 4 a.m., and lunch was at about 1 p.m. That’s a
nine-hour day, and he is not even half finished yet. On the show I saw, the band finished
around 11 p.m. A typical load out is two hours or so. This is a really big tour, so it is
likely closer to three . It is a good 22-hour day.
And there’s another show tomorrow . . . .
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T
Just Because You Can Doesn’t
Mean You Should
he title of this chapter may be the best advice anyone can give you, and it doesn’t
matter whether the advice is technical or just about the way you comport
yourself on the gig. For example:
■
The audio systems in use today are very powerful, and it is possible to run volume
levels exceeding 110 dB. Don’t.
■
The digital systems that are increasingly the norm allow an engineer to put processing on every channel. Compression, reverb, and any other effect you can think of
are as close as the nearest computer plug-in. And you can stack five or six inserted
processing units on any channel. Don’t.
■
Given the right combination of stimulants, you may be able to party like a rock star
and still be able to get through the next day’s gig—for a while. Don’t.
This next part is something that has been around for years but never before published.
Titled ‘‘The Road Bible,’’ it was written by a touring sound guy when—after a 12-hourplus pre-gig call—he was unwinding on the bus and a guest came aboard, looked at him,
and said, ‘‘I want your job . . . .’’
So You Want My Job?
By Roadie ‘‘M’’
Don’t anyone take this offensively. I’m taking time from figuring out my float to give
some insight to anyone who wants to work on the road. Read on . . .
Do you have a family, a pet, a wife, kids, a home, a disgusting habit, a desire for lots
of sleep, a diet that consists of water and tofu, motion sickness, the inability to lift
more than 75 pounds (just to name a few)?
If you answered yes to any of these, stay at home, because you won’t make it out
here, and what you had at home will probably be gone when you get back.
While you’re thinking of your mate getting ‘‘entertained’’ by your neighbor or Fido
running into traffic or your kids telling you about your ‘‘long-lost brothers,’’ who
sleep over while you’re gone, all after whining to us about your $1,000 cell phone
bill, I am worrying about you dropping a shackle from the grid, tipping over a PA
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stack, electrocuting a lighting guy, or puking in someone’s bunk, which may lead to
you getting yourself in deep trouble.
There’s nothing worse than having to pick up a slacker’s slack (especially when it’s
preventable).
Think you’re a good candidate? Hardcore? Ready for battle? Read on . . .
Schools
Schools are a good foundation, but no school can simulate an 18,000-seat building
or teach you how to lift a console, pull cable, stack, power up, and repair gear without hurting yourself or someone else, all while making it sound good and look good
all while getting it there in one piece. If you have the money, go for it. But don’t think
for a minute that a sound or lighting company is going to make you a trade-magworthy engineer overnight. After you spend 30 grand and get your first gig, guess
what? You’ll still be stacking PA and trusses, pulling cable, loading trucks, and driving
trucks, buses, and so on. There is no substitute for experience, but in a school you will
learn some basics. Knowing what to do when something goes wrong will come with
experience—unless you’re ignorant or lazy.
The Big Green Book of Rock and Roll
Page 1: The gig comes first.
Page 2: You will never see anything in this book that tells you this job is fair.
Page 3: Don’t be late. If you are, bring a copy of your obituary.
Page 4: Treat people well on the way up. If you don’t, those you stepped on will step
on you as you come back down.
Page 5: The band will always win.
Page 6: Know what to do when something goes wrong. Or else.
Page 7: Be nice. Being a jerk will only get you a window or aisle seat—one way.
Page 8: No one cares how you did it with XYZ band. There’s a right way, a wrong
way, and our way.
Page 9: Don’t fix it ’til it’s broke.
Page 10: The wheel has already been invented; sometimes it needs new tires or a
little bit of air. Have a workbox and an air hose ready and know how to use them.
Trying to reinvent the wheel may cause you to get locked in a workbox.
Page 11: The rumor you started will end. See Page 7.
Page 12: Everyone gets paid to do his job. Want more money? Get another job or be
the best at what you do. Learn your boss’s job.
Page 13: Trust and respect are earned.
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Page 14: Thou shalt not lie. Thou shalt shut up. Eventually the truth will come out.
Earn respect or get a choice of seats (see Page 7—again.).
Page 15: Don’t be stupid. Check and double-check, or someone may not live to see
their next paycheck.
Page 16: Tour managers/production managers do exactly what their title says. They
arrange travel as well. And always have cash and credit cards. Anger them, and you’ll
find out what else they can do.
Page 17: Nothing is secret or sacred on the road. We will find out eventually.
Page 18: Ask if you don’t know. Just don’t ask the wrong person.
Page 19: What happens on the road stays on the road.
Page 20: Understand politics but don’t get involved with them.
Get the picture? Eventually you will . . . maybe. Keep reading . . .
If you really want to work in the touring industry, find a touring-related job.
Go to a sound company, a lighting company, a trucking company, a pyro company, or a
bus company and apply for a job. Go to a local show and get on the ‘‘call,’’ get hired, and
pay your dues, and you will be partially on the road to success. Keep quiet. Listen and
learn and have no fear of manual labor. No one will hire you from job corps, the newspaper, or the lottery. If you want this job, you have to get it. It rarely gets handed down.
The stuff you put up with in the beginning will educate you for later—school or no school.
We don’t want snivelers, whiners, or crybabies. Ex-military folks do well on the road and
adapt quickly. Want to know why? Discipline, familiarity with lousy conditions, knowledge of teamwork and organization, knowing how to go through a chain of command,
and above all, learning to be an expert in their field and understanding unit camaraderie.
There is no grievance board, company nurse, sick days, or foo-foo baskets on birthdays. The tour is the nation, with different ranks, rules, and policies. Many are unwritten. This isn’t just a job; it’s a different way of life. Be good at something. If that
something is desirable, you will always be working. Don’t sell yourself short. If you
can’t live on what someone wants to pay you, find another job.
You want my job? Get up and do something. Don’t just sit there; do something.
Open a phonebook, look in a newspaper, in Pollstar, or at a tour schedule online and
do something!
On another note, the rumors are true. ‘‘Touring can be like boy scout camp with girls,
liquor, and [fill in the blank].’’
After the work is done.
Now go!
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When I was asked to do this book, I put a question to the members of ProAudioSpace.com.
I asked them what they wished the ‘‘new guy’’ who just came out of school knew before
he got on the gig. I got dozens of responses, and not one of them had anything to do with
gear or technology.
One of the most common requests was that new folks learn the right way to coil or wrap
cable. It’s called ‘‘over-under,’’ and it is really hard to illustrate in a book. Just remember that you never wrap cable in a loop between your hand and elbow. More
quickly than anything else, it will tag you as knowing nothing. If you don’t know how to
‘‘over-under’’ a cable, ask. And then practice at home.
Another common issue was the inability of many newbies to troubleshoot. We have
become a ‘‘throw it away’’ society. Most of us in the audio biz grew up when there were
still things like ‘‘shop’’ classes in high school, and we learned the basics of an electrical
circuit and how to build and repair one. Today it often costs more to repair an electronic
product than it does to just replace it. The result is a serious lack of ability in the troubleshooting department.
First, many problems can be avoided with proper preparation. For example, check all
snakes with a condenser mic and make sure there is signal on every channel before you
even leave the shop. Do it again before pinning the stage, and you will avoid problems
that will take longer to find later in the process.
Here is how I troubleshoot. It comes down to a process of elimination to find the
problem. You have to find it before you can fix it. Say there is a mic onstage that is
not sending signal to the board. In a simple setup, it could be the mic, the cable, the
snake channel, or the console channel. Or it could be something as simple as the phantom power not being engaged on a channel using a condenser mic. Here are the steps I
take.
1.
Check the board to make sure the head amp is up, the channel is not muted,
another channel is not soloed, and the phantom power is on if it is a condenser
mic. Also check where the snake outs connect to the board. A connector may not
be inserted all the way.
2.
Replace the mic with a simple dynamic that you know is working. Make sure to
mute the channel or bring the gain all the way down before taking this step.
3.
Still using the mic you know is working, replace the cable.
4.
If it is still not working, then it is either the snake or the console. (Remember, this
is a simple setup.) The easiest thing to do is to plug that same working mic directly into the console. If it still does not work, replace the cable, just to be sure.
If it is still not working, figure that the console channel is bad. Tape it off and
worry about it after the gig. Repatch that input to another channel.
Large-Music Software
Chapter 27
5.
Just Because You Can Doesn’t Mean You Should
169
If the mic works plugged directly into the channel, then it is a problem with the
snake. Repatch it into another channel.
The same basic process should work no matter what the problem. If one side of the PA is
humming, switch the outputs (if that is possible). If the other side starts to hum, then the
problem is at the console. If the same side hums, then it is somewhere between the
console and the speakers. Just keep moving down the line and isolating individual links
in the chain until you find the one that is bad. Always have spares. If the faulty piece
goes on the gig, the chances of having the time to fix it are not good. That gets done at
home or at the shop after the gig is done. Refer to the signal flow chart at the end of
Chapter 19 if you are confused.
Next request from the tribe: Know the lingo. Do you know what a spanner, a torch,
and a stinger are? In order, that would be a wrench, a flashlight, and an electrical
extension cord. But no need to reinvent the wheel here; there is a great Lingo section on
Roadie.net. Learn it.
The following sections provide some more of the assembled wisdom of people who have
been doing this for a long time.
Dave Shirley
My first day in a Sound Reinforcement 101 class was in September, 1981, at Belmont
College (now University) in Nashville. The first day of audio class, they taught us that
the most important rule to remember is, ‘‘No matter what happens, it’s always the
soundman’s fault!’’ If you can’t remember anything else, remember that rule, and it will
make those ‘‘learning experiences’’ at first gigs make more sense. If the singer loses his
voice, it’s the soundman’s fault. If the PA died, it’s the soundman’s fault. If the audience
couldn’t hear the lyrics in the cheap seats, it’s the soundman’s fault. If they hated the
band, it’s the soundman’s fault. I’m sure you follow the pattern. It’s somewhat funny,
yes, but also completely true. All kinds of stuff will get blamed on you, deserved or not.
Deal with it, learn from it, and move on.
The second thing they taught us the first day of class was the six questions every sound guy
sound should ask about a gig—and this works in almost any business interaction, especially
if you are self-employed, as most of us are. The five Ws and an H: Who? What? When?
Where? Why? And how much? If you can’t answer those six questions, you didn’t get all
the info you needed to book that gig! Get all the pertinent details in the first contact with the
client. It will make you appear professional and cut down on playing phone tag and sending
lots of emails back and forth. Not that there aren’t follow-ups with additional info and
variables, but if you know the basics, you can add it to your calendar with confidence.
Oh, the extra-credit question on the end of semester final: ‘‘Whose fault is it?’’
Large-Music Software
170
Live Sound Fundamentals
Jordan Wolf
1.
Learn how to effectively communicate with people.
2.
Learn how people perceive your communication toward them.
3.
Learn the various ways to wrap cables. (Different companies have different
preferences.)
4.
Be willing to work long hours for little pay, respect, or acknowledgment.
5.
You’re never too important for sweeping floors or cleaning fader caps.
Ian Silvia
Someone should kindly explain to them that if they work for a sound company, they
might not get to mix bands very often. It seems like most bands (including openers) have
their own ‘‘engineers’’ these days. I think it might have something to do with the serious
number of kids these schools churn out. They get out of sound college and go on tour
with their buddy’s band for $50 a day and a ham sandwich.
If you like knobs, faders, nice clothes, and catering, find a gig working for a band. If you
like feeders, motors, gig butt, and gas-station egg salad sandwiches, get a gig with a
sound company.
I prefer the latter.
Mike Reeves
I notice a lot of new guys have a habit of patching the cable from the microphone first,
making a whole bundle of cable by the snake. Obviously, we know to start at the snake
and neatly drop the cable underneath the microphone stand.
Considering when techs get out of school, chances are they aren’t going to be behind a
console anytime soon. Since they are learning about how to run the consoles, they need
to know what they are getting into once they arrive. So load-in/load-out etiquette . . .
Learn how to properly lift gear. Learn spatial relations on how to load trucks correctly.
Suggesting they get a chauffeur’s license might be good, because the new guys are
usually the ones who have to make the deliveries in the box trucks. Learn that you work
as a team, and never leave your man hanging. Nothing angers a tech more than the new
guy vanishing during a load in/load out.
D.V. Hakes II
In my neck of the woods—Hawaii—there are many decent A-1s. Basically, someone has
to die or retire before these kids even get a shot at touching the board. We get an onslaught of them every now and then, but none of them stays for long (which is a shame
Large-Music Software
Chapter 27
Just Because You Can Doesn’t Mean You Should
171
in some cases), because they never get the chance to mix, and they don’t know how to
do anything else (or they don’t want to). They want to bypass the five to 10 years of
being an A-2 and go directly to being an A-1.
They need to teach these guys/gals (or they need to learn as quickly as possible) mic
placement; speaker placement; speaker rigging; mic type for different applications; how
to take amps apart and put them back together again; how to mark the cable ends and
the snake heads (for everything); how to set up, run, and fix different COMM systems
and how to get them to talk to each other; and so on.
Having worked with 30 or so of these kids over the last 15þ years, I’ve learned to let
them know straight up that they aren’t going to touch a board until they know how to
set up and fix everything else (as well as show their work ethic, and so on). I know it’s
harsh, but that’s the way it works around here. We have more experienced A-1s than we
have work. We also have a list of A-2s who have put in the time and have the knowledge, who are waiting (and very ready) to step up.
Jeanne Knotts
1.
Calm down and pay attention.
2.
Know your frequencies. If you didn’t learn this in school, then you can teach
yourself with an audio training CD or an EQ in line with your stereo.
3.
Don’t lie your way into jobs. It generally backfires, and you stop getting booked.
4.
Be good to your cables and tape them down.
5.
Learn to solder and learn basic repair skills, because you will usually need them
in a high-stress situation, on the fly, and with no light. Keep your soldering iron
on during the show, just in case.
6.
Have backups for all your tools and keep them locked up! Unless I specifically
need something, I generally only carry what I can fit in my pockets and on my
belt.
7.
If you think you will fail, you’re right. See #1, keep a positive attitude, and stay
busy or ask what you can do. Sure, there’s a lot of hurry up and wait, but if
people think you’re lazy or an idiot, they’ll book someone else. There is a difference between not knowing something and being an idiot. Learn it.
8.
Smoking and swearing are hot-button issues now, and unless you’re good friends
with the talent, you’re better off keeping the relationship professional. Also, you
never know where or how you will run into someone, so try not to step on people
if you can help it.
Large-Music Software
172
Live Sound Fundamentals
Steve McCarthy
It is easy to spot the newbies when they try to wrap cable. They always wrap it around
their palm and elbow. They lift with their back bent over instead of with their legs.
‘‘Keep your hands out of your pockets, kid.’’
‘‘Don’t stand there gawking with your jaw hanging down.’’
‘‘Ask questions. No one is going to preach audio to you.’’
‘‘Put your gloves on and stop crying about that cut on your hand.’’
‘‘You want to patch the snake head? Try counting from 1 to 10 and 10 to 1 at the same
time, for starters.’’
‘‘Stay in your department. Don’t let me catch you drifting off with your friends in
props.’’
‘‘You want to reinvent the wheel, kid? Save it for the day you become a department
head.’’
Final Words
A final thought: We have so much control at our fingertips that it is easy to forget that
sometimes too much choice is a bad thing. It is not a coincidence that great live records
were released on a seemingly monthly basis in the ’70s and ’80s, and there are very few
of them anymore. But most shows have the ability to record and even do full multitrack
sessions on every gig.
Those great live records were made on gear that baby bands won’t put on a rider. The
Beatles played Shea Stadium with gear that no self-respecting bar band would use
today.
The thing is that music was the primary concern when those records were made—not
technology. Whatever you do, remember that you are there in the service of the music or
whatever the performance is. The most successful gig is the one where no one knows
you are there. It’s not about you. It’s not about the gear. It’s about the performance.
And never forget that extra credit question: Whose fault is it?
Now get up. Doors are in 30, and there’s still work to be done.
Large-Music Software
Index
Special Characters
1/4-inch connector, 23, 53, 78
3-to-1 rule, 18
A
A1 category, 128
A2 category, 128
A3 category, 128
acoustic energy, 7
active DI, 66
active speakers
Moore’s Law, 97–98
overview, 9–10
pros and cons, 99–101
address, microphone, 12–13
advancing gig
overview, 132–135
stage plot, 138–139
technical rider, 135–138
AES/EBU format, 26, 104
After Fader Listen (AFL) button, 66
Alesis ADAT digital recorder, 26
Allen & Heath ZED R16, 54
Allison, Mike, 159–164
amps (amplifiers)
Class A, 105
Class AB, 105–106
Class B, 105
Class D, 106–107
matching to speakers, 98
modeling, 19
VCAs, 9, 59, 71–72
analog
connections, 103
distortion, 70
antennae
parallel, 36
positioning, 43
attack control, compressor, 80
audio
balanced connection, 24–25
doubling volume, 6
loudness, 5–6
operating levels, 74–75
Audio Reference Companding, 42
Audio-Technica AE2500, 17
auditioning, wireless, 40–43
aux (auxiliary) returns, 69–70, 77–82
dynamics, 80–82
effects, 78–79
send/return adjustments, 79–80
signal chain, 77–78
aux (auxiliary) sends
channel insert, 58–59
channel strip, 9
dynamics, 80–82
effects, 78–79
gain structure, 57–58
send/return adjustments, 79–80
signal chain, 77–78
subgroups, 60
two-track jacks, 59–60
B
back plate, speaker, 113
backline, 151–153
Baker, Ace, 81
balanced audio connection, 24–25
bandrejection, 88
bands, EQ, 61–62
band-stop, 88
bandwidth, 42
bass
positioning microphones, 19
rig, 152
bass reflex design, 120
Bell, Graham, 5
Bessel filter, 89
big green book of rock and roll, 166–169
BNC connector, 32
body, mic, 12
boosting, 62–64
boundary mic, 18
box design, speaker, 119–120
brick on a rope power
connection, 27
Butterworth filter, 89
Large-Music173Software
174
Live Sound Fundamentals
C
cables
1/4-inch insert, 78
Cat-5, 30
insert, 56
insulators, 23
Musical Instrument Digital Interface
(MIDI), 28–29
power, 26–32
shielded versus unshielded, 21–25
tip-ring-sleeve (TRS), 23
tip-sleeve (TS), 23–24
TRS send and receive, 58–59
capsule, mic, 12
cardioid pickup pattern, mic, 13
Cat-5 cable, 30
CD (constant directivity) design, 115
center pole piece, speaker, 113
channel insert, 58–59
channel strip, 9, 53–56
channels, 65–67
mute, 66–67
phantom power, 65–66
solo, 66–67
circuit sniffer, 134
Class A amps, 105
Class AB amps, 105–106
Class B amps, 105
Class D amps, 106–107
clipping, 107
comb filtering, 121–122
companding, 36–37, 42
compression, wireless, 34–35
compression driver, 114
compressors, 80–82
condenser mics, 11–12, 39
conductors, 21–22
cone speaker, 112
connectors
1/4-inch, 23
BNC, 32
data, 28
female three-pin XLR, 24
five-pin MIDI, 29
IEC electrical, 26
male three-pin XLR, 24
Neutrik etherCON, 31
Neutrik speakON, 104
overview, 25–26
power, 26–32
RCA (phono), 25
RJ-45, 30, 104
seven-pin MIDI, 29
three-pin XLR, 24
twist-lock (powerCon), 28, 104
XLR, 20
console, 49–52
auxiliary sends, 9, 57–60, 77–82
education, 51–52
hostility, 50–51
incompetence, 51
overview, 8–9
progression, 49
constant directivity (CD) design, 115
continuous power, 107–108
Crest X20RM monitor mixer, 84
crossover region, 89
crossovers, 87–88
cue wedge, 84–85
cutting, 62–64
cymbals, miking, 18
D
damping factor, 107
Daniels, Drew, 50
Danley, Tom, 116
data connectors, 28
Davis, Rich, 45
decibels (dB), 5
delay
back to kick drum, 94–95, 149–150
secondary speakers, 92
sine wave, 148–149
subwoofer time alignment, 147–148
time-based effect, 78
delay stacks, 92, 147
delay towers, 92
deployment
loudspeakers, 121–123
microphones, 15–18
desk, 49
DI, 65–66
digital connections, 103
digital distortion, 70
digital signal processing (DSP)
cost, 79
drive racks and, 9
power, 108–109
digital snake, 48
direct input, 19
distortion
diversity, 36, 43
Dog Whistler app, 64
doubling volume, 6
drive rack
filters, 88–89
limiters, 95–96
overview, 9
system EQ, 89–90
time-aligning club system, 92–95
tuning, 90–91
Large-Music Software
Index
drivers, loudspeaker
cone speakers, 112–113
high-frequency, 114
horn, 114–116
overview, 111
drums, miking, 16
DSP (digital signal processing). See digital signal
processing (DSP)
dual point insert, 77
dynamic mics, 11, 39
dynamics
compressors, 80–82
noise gate, 81–82
processors, 81
E
ear training, 64
early reflections, 78
ears, 85
echo, 78, 92, 147
education, 51–52
effects
inserting, 77–78
modulation, 78
proximity, 11–12
time-based, 78–79
energy, acoustic, 7
EQ (equalization)
bands, 61–62
cutting, 62–64
parametric, 62
semi-parametric, 62
system, 89–90
using preamplifier, 9
expansion, wireless, 34–35
F
Federal Communications Commission (FCC), 35
female three-pin XLR connector, 24
ferrofluid, 113
figure-8 pickup pattern, mic, 13
filters
Bessel, 89
Butterworth, 89
first-order, 88
high-pass, 62, 87–88
Linkwitz-Riley, 89–90
low-pass, 88
notch, 88
second-order, 88
sweepable, 62
third-order, 88
true cut-and-boost, 61–62
first-order filter, 88
five-pin MIDI connector, 29
flying speakers, 122
foldback, 83
folded horn, 115, 120
formats
AES/EBU, 26, 104
S/PDIF, 25–26, 104
wireless, 40
former, speaker, 113
foundation, adapting, 156
frame, speaker, 112
frequency
bands, instrument, 63
translating to notes, 4
frequency agility, 38
full-range speaker cabinet, 87
Future Sonics manufacturer, 85
FX unit, 79–80
G
gain structure
audio operating levels, 74–75
defined, 73
importance of, 57–58
microphone, 75–76
signal chain, 73–74
gate, noise, 81–82
gig
advancing, 132–139
pinning stage, 144–145
set up, 143–144
showtime, 142–143
work week, 141–142
grease, 78
Ground Lift switch, 46
guitar
positioning microphones, 19
rig, 152
H
Hakes II, D.V., 170–171
Hanley, Bill, 105
harmonic distortion, 106
harmonics, defined, 4
head amp, 8
Heinz, Ralph, 116
hertz, defined, 4
Hertz, Heinrich, 4
high-frequency driver, 114
high-pass filter, 62, 87–88
hi-hat mic, 16
horn drivers, 114–116
horn-loaded design, 120
horns, miking, 19
hostility, between crews, 50–51
Hughes, Mick, 18
hyper-cardioid pickup pattern, mic, 14
Large-Music Software
175
176
Live Sound Fundamentals
I
IATSE (International Alliance of Theatrical Stage
Employees), 127–128
IEC electrical connector, 26
impedance, 116–117
incompetence, 51
in-ear monitors, 85
input connections, power, 103–105
input section, console, 8
input terminals, speaker, 112
insert cable, 56
inserts
channel, 58–59
defined, 55
dual point, 77
insulators, cable, 23
interference pattern, 121
intermodulation distortion, 106
International Alliance of Theatrical Stage Employees
(IATSE), 127–128
interpersonal skills, 155
Inverse Square Law, 6
I/O panel, Allen & Heath ZED R16, 54
I/O panel, Mackie Onyx 1640i, 53
Iso Outputs, 46
J
Jackson, Bruce, 122
JBL low-frequency transducer, 112
Johnston, Tom, 51–52
K
Keele, Don Jr., 115
keyboards, 19–20, 152–153
kick drum
delaying PA back to, 94–95, 149–150
positioning mics, 17
Klijanowicz, Brian, 93, 147
Knotts, Jeanne, 171
L
LaCerra, Steve, 73
lavaliers, 39
LED-based meters, 70
limiters, drive rack, 95–96
limiting settings, 80
line inputs, 74
line-level signals, 73
Linkwitz-Riley filter, 89–90
lip-synching, 82
listening, 61–62
live production environment, 129
load, power, 107–108
load in/out situation, 133–134
loudness, 5–6
loudspeakers, 7–8, 111–123
See also speakers
box design, 119–120
deployment, 121–123
drivers, 111–116
impedance, 116–117
power handling and efficiency, 117–119
low-pass filter, 88
M
Mackie Onyx 1640i, I/O panel, 53
magnet, speaker, 113
male three-pin XLR connector, 24
master section, 69–72
additional tools, 70–71
aux returns, 69–70
LED-based meters, 70
voltage controlled amps (VCAs), 71–72
master section, console, 9
Matchbox level-matching interface, 74
matrix mixer, 83
McCarthy, Steve, 172
McDonald, Michael, 18
mechanical failure, speaker, 118
Meyer, John, 99
MI (musical instrument) mixer, 59, 61
mic (microphone)
address, 12–13
condenser, 11–12, 39
dynamic, 11, 39
gain structure, 75–76
input, 74
pickup patterns, 13–15
positioning, 15–18
mic pre, 8
MIDI (Musical Instrument Digital Interface) cable,
28–29
Miller, Stan, 122
mixers
auxiliary sends, 9, 57–60, 77–82
education, 51–52
hostility, 50–51
incompetence, 51
matrix, 83
musical instrument (MI), 59, 61
overview, 8–9
mixing, 155–158
adapting foundation and flow, 156
fixing it at source, 155–156
softer is louder, 157–158
triage, 157
mobility, wireless, 41–42
modeling amps, 19
modulation effects, 78
Large-Music Software
Index
monitor placement, 14
monitoring, 83–85
cue wedge, 84–85
gear, 83–84
Moore, Gordon, 97
Moore’s Law, 97–98
Morgan, David, 20
mult, 47
multi-entry horn, 116
Musical Instrument Digital Interface (MIDI) cable,
28–29
musical instrument (MI) mixer, 59, 61
mute, 66–67
N
Neutrik etherCON connectors, 31
Neutrik speakON connectors, 104
noise gate, 81–82
nominal impedance, 117
notch filter, 88
O
off-axis placement, mic, 19
Ohm, George, 116
ohm load, 108
Ohm’s Law, 116
omni-directional pickup pattern, mic, 13, 15
orders, filter, 88
OSHA, sound safety standards, 6
output connections, power, 104–105
output control, compressor, 80
over excursion, 118
Overson, Paul, 121
P
PA, delaying back to kick drum, 94–95
Page, Howard, 49
pan control, 66–67
parallel antennae, 36
parametric EQ, 62
passes, 88
passive DI, 66
passive radiator, 119
patterns
interference, 121
pickup, 13–15
peak power, 107
Peavey Distro, 134
percussion rigs, 152
personal monitors (PMs), 85
PFL (pre-fader listen) button, 55, 66, 75
phantom power, 11, 65–66
phono (RCA) connector, 25
phono (RCA)–style inputs, 53
pickup patterns, mic, 13–15
pinning stage, 144–145
pitch, 4
plug-ins, 79
PMs (personal monitors ), 85
point source, 121
positioning
antennae, 43
loudspeakers, 121–123
microphones, 15–18
post-fader aux send, 58
power
cables, 26–32
Class A amps, 105
Class AB amps, 105–106
Class B amps, 105
Class D amps, 106–107
connectors, 26–32
DSP, 108–109
input connections, 103–105
load, 107–108
output connections, 104–105
signal chain, 10
switching supply, 106
wireless, 35–38
powerCon connector, 28, 104
powered speakers
Moore’s Law, 97–98
overview, 9–10
pros and cons, 99–101
preamplifier, 8
pre-aux sends, 58
pre-fader listen (PFL) button, 55, 66, 75
pre-fader sends, 77
prep. See advancing gig
pressure zone (PZM) mic, 18
processed signal, 10
protective tire, speaker, 113
proximity effect, 11–12
pulse-width modulation, 106
PZM (pressure zone) mic, 18
R
radial horn, 115
radio frequency (RF), 41
Rapco LTI 100, 20
Rasmussen, Jeff, 18
Rat, Tommy, 63
ratio control, compressor, 80
RCA (phono) connector, 25
RCA (phono)–style inputs, 53
RCA two-track jack, 59
real-time analyzer (RTA), 90
receiver, wireless, 41
Reeves, Mike, 170
release control, compressor, 80
return adjustment, 79–80
Large-Music Software
177
178
Live Sound Fundamentals
reverb, 78
RF (radio frequency), 41
rider, technical
front of house audio system, 135–136
front of house console and processing (fly dates
only), 136
microphone and mic-stand package (fly dates only),
137–138
monitor console and processing (fly dates only), 136
RJ-45 connector, 30, 104
RMS (root mean squared), 107
roll off, 61–62
root mean squared (RMS), 107
RTA (real-time analyzer), 90
S
schools, 166
sealed box design, 119
second-order filter, 88
semi-parametric EQ, 62
send adjustment, 79–80
seven-pin MIDI connector, 29
shielded cables, 21–25
Shirley, Dave, 169
shorting ring, speaker, 113
side-address, mic, 12
signal chain
gain structure, 73–74
inserting effects, 77–78
overview, 7–10
powering, 10
signal flow, 123
Silvia, Ian, 170
Simmons, Patrick, 51–52
sine waves
cancelling each other out, 17
delay, 148–149
overview, 3
time-aligning club system, 93–94
Siniscal, Al, 98–99
slap, 78, 92, 147
snakes, 45–48
snare, miking, 16
solo, 66–67
solo in place, 66
sonic signature, 78
Sony/Philips Digital Interface format (S/PDIF), 25–26,
104
sound
balanced audio connection, 24–25
doubling volume, 6
loudness, 5–6
operating levels, 74–75
speed, 5
splitting, 46–47
vibrations, 3–5
sound pressure level (SPL), 5, 118–119
spaced-pair technique, 18
S/PDIF (Sony/Philips Digital Interface format), 25–26,
104
speaker processors, 9, 87
speakers
active, 9–10, 97–101
back plate, 113
box design, 119–120
center pole piece, 113
cone, 112
deployment, 121–123
drivers, 111–116
flying, 122
former, 113
frame, 112
impedance, 116–117
input terminals, 112
power handling and efficiency, 117–119
protective tire, 113
shorting ring, 113
spider, 113
thermal failure, 118
top plate, 113
vent, 113
voice coil, 112
speakers on a stick, 99
speed of sound, 5
spider, speaker, 113
SPL (sound pressure level), 5, 118–119
splits, 8, 45–48, 55
splitter, 46
stage, pinning, 144–145
stage box, 8, 45–48, 55
subgroups, 60, 71
subsnake, 8, 48
subwoofer time aligning, 147–148
sum, 88
super-cardioid pickup pattern, mic, 14
sweepable filter, 62
switch mode, 106
switching power supply, 106
system EQ, drive rack, 89–90
Szocska, Billy, 45
T
Talkback control, 70
technical rider
front of house audio system, 135–136
front of house console and processing, 136
microphone and mic-stand package, 137–138
monitor console and processing, 136
thermal failure, speaker, 118
third-order filter, 88
three-pin XLR connector, 24
threshold control, compressor, 80
Large-Music Software
Index
time-aligning
club system, 92–95
subwoofer, 147–148
time-based effects, 78
tip-ring-sleeve (TRS) cable, 23
tip-sleeve (TS) cable, 23–24
tom-toms, miking, 16–17
top plate, speaker, 113
top/front-address, mic, 12
touring, 159–164
transducer, 7
transmitter, wireless, 41
trim, 75–76
TRS (tip-ring-sleeve) cable, 23
TRS send and receive cable, 58–59
true cut-and- boost filter, 61–62
true diversity, 36
TS (tip-sleeve) cable, 23–24
tuning, drive rack, 90–91
tuning the room, 90
twist-lock (powerCon) connector, 28, 104
two-track jacks, 59–60
two-way split, 46
VHF (very high frequency), 34
vibrations, 3–5
vocal mic, 12
voice coil, speaker, 112
voltage-controlled amplifiers (VCAs)
defined, 9
faders, 59
master section, 71–72
volume, doubling, 6
VU meters, 70
W
UHF (ultra high frequency), 34, 35
underhead technique, 18
unidirectional mic, 13
unshielded cables, 21–25
wall wart power connection, 26–27
Watt, James, 117
wind screen, mic, 12
wireless, 33–43
compression/expansion, 34–35
dynamic or condenser mics, 39
format, 40
frequency, 38–39
history, 34
maximizing performance, 43
mobility, 41–42
power limitations, 35–38
radio frequency (RF), 41
reception impediments, 43
unplugging, 33–34
versatility, 40–41
Wolf, Jordan, 170
V
X
Van Druten, Ken (‘‘Pooch’’), 8
VCAs (voltage-controlled amplifiers). See voltagecontrolled amplifiers (VCAs)
vent, speaker, 113
venue location, 133
verb, 78
versatility, wireless, 40–41
XLR connectors, 20, 23–24, 46, 53, 74
X-Wire wireless system, 42
X-Y technique, 18
U
Y
Young, Tom, 20
Large-Music Software
179
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