Gottlieb 2007 Shaping Sound in The Studio (Manual PDF

Shaping Sound
in the Studio
and Beyond
Audio Aesthetics and
Technology
Gary Gottlieb
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This book is dedicated to my three one and
only girls:
Kyla, my baby girl; Miranda, my big girl; and
Melanie, my best girl.
Foreword
If youre looking for a book thats going to tell you in ve easy lessons which
buttons to push to make a hit single in GarageBand, Shaping Sound isnt it. This is
a serious book: Its for people who take sound seriously and are serious about
learning about it. This book will take you to that very important place where
technology and art meet, where the technical decisions you make will have a
profound effect on someones art, and where the art will dene how you use
the technology.
Which is not to say that its ponderous or heavy. Far from it. Its thorough, but
its clear and logical and carries you along quite smoothly. Garys style is that of
a friendly, patient teacher, who is eager for you to understand what hes talking
about and wants to be sure that you do before you move on to the next topic.
Which all makes sense, since Gary is a friendly, patient teacher, as I learned when
we worked together in the Sound Recording Technology program at UMass
Lowell.
I met Gary when he was making the transition from practitioner to educator,
taking his years of experience at the craft of recording and guring out how to
pass on his knowledge and wisdom to students.
In the years since, he has learned a lot about teaching and how students learn. He
has built a terric program at Webster University, and this book is one happy
result of that achievement.
In Shaping Sound you will learn from Gary not only what recording engineers do,
but why they do it, and not only what the equipment does, but why we need it.
Youll learn how our tools can make things sound better, and also how they can
make things sound awful. Hes always providing context, making sure you have
the background to absorb what comes next. He starts at the beginning and takes
you efciently through the physics and acoustics that youll need to know to get
the most out of the rest of the book. When he talks about recording and editing,
he makes it clear that todays digital techniques didnt just pop up overnight:
They followed years of research, experimentation, and practice in the analog
world. He knows, and conveys to the reader, that understanding how our current
tools and practices evolved is critically important for anyone who wants to call
himself or herself an audio professional.
But like all good recording engineers, Garys primary concern isnt the theory,
the gear, or the techniquesits the music. Gary has recorded and produced a
huge variety of stuff, under a wealth of different conditions, and knows what
iv
works and what doesnt, what sounds good and what doesnt. And he knows
why. Now hes not only telling us, but also giving us ways, with the text, the
CD, and the exercises in each chapter, to learn what he knows through
experience.
Im listening, and you should be, too.
Paul D. Lehrman
Medford, Mass.
April 2007
Paul Lehrman is Director of Music Technology at Tufts University and is the
Insider Audio columnist for Mix magazine.
Foreword
v
Acknowledgments
Many people have helped me over the years. In my professional career in New
York, Barry Lazarowitz, Jerry Ragovoy, Gary Chester, Tommy Monst Civillo,
Ben Wish, and Leslie Mona-Mathus all taught me about the skills and psychol-
ogy needed to succeed in audio. Thanks also to Brooks Brown and the crew at
WEQX in Manchester, Vermont; working there reminded me how much fun one
could have at work.
As I began my career as an educator, those who helped me along the way include
my mentors Paul Nelsen and Will Moylan, both of whom taught me the value of
sharing my knowledge and the best ways to pass that along. My peers and friends
in education, such as Barry Hufker and Paul Lehrman, contributed to this book in
ways of which they were probably unaware. I also learned more from my stu-
dents than I should admit, at Marlboro College, Center for Media Arts, Plymouth
State College, Castleton State College, University of Lowell, Massachusetts, and
now at Webster University. Thank you all. And you probably thought you were
the students, when I was actually the one learning.
I would also like to thank those who helped me in the creation and production of
this book. The good people from Thomson who made this possible: acquisitions
editor, Orren Merton; my production editor, Cathleen Snyder; technical editor,
Barry Hufker; and the Man with the Handshake, Paul Lehrman.
Personally, I would like to thank my family and friends for their support
throughout the years. Irving, Shirley, and Lisa Gottlieb; all my cousins; and
my friends going all the way back to high school, too numerous to name. You
know who you are. I love you all, and thanks for all the help.
vi
About the Author
Longtime music business professional Gary Gottlieb refers to himself as a music
generalist. A professional musician since age 13, he worked in radio on and off
for 25 years, and was a music critic for 9 years. As a recording engineer and
music producer in New York, Gottliebs long and distinguished career includes
work with numerous Grammy Award winners and Rock & Roll Hall of Fame
inductees. His credits as a sound designer include numerous off-off-Broadway
productions, along with community and college theatre productions throughout
New England. Along with his history as a music critic and entertainment writer
for the Deerfield Valley News in West Dover, Vermont, and a disc jockey for
WEQX, a major modern rock station in Manchester, Vermont, Gottlieb
owned and operated a mobile DJ service. In 2002 he accepted a position as Pro-
fessor of Audio Production at Webster University in St. Louis, where he now runs
the Audio program.
vii
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Contents
Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xv
PART I
BASIC AUDIO THEORY 1
Chapter 1
Audio Aesthetics and Technology 3
In Search of Aesthetics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Audio and Aesthetics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Audio Technology Meets Aesthetics . . . . . . . . . . . . . . . . . . . . . . . . . 13
Chapter 2
The Properties and Characteristics of Sound 17
Sound as a Waveform. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Sound as a Form of Perception . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
The Speed of Sound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
The Human Perception of Sound . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
Loudness . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
Pitch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Timbre . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
Duration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
Location . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
Environmental Effects on Perception of Sound. . . . . . . . . . . . . . . . . . 38
Fletcher Munson Equal Loudness Contours. . . . . . . . . . . . . . . . . . . . 42
The Behavior of Sound Waves . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
Additional Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
ix
PART II
UNDERSTANDING AUDIO EQUIPMENT 51
Chapter 3
The Production Room 53
What Is a Production Room?. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
Key Components of a Production Room. . . . . . . . . . . . . . . . . . . . . . 60
The Performance Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
The Control Room . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
Consoles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Patch Bay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
Recording and Storage Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Tape Transports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Transport Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
Recorder Head Assembly. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Recorder Monitor Modes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
V.U. Meters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
Magnetic Tape . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Magnetic Tape Characteristics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Proper Tape Storage Considerations . . . . . . . . . . . . . . . . . . . . . . 94
Production Room Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Additional Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Chapter 4
Consoles 99
Versatility in Consoles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
Preamps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
Impedance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
Auxiliary Sends . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
Pan Pots . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
Equalizers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Summing Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
Using the I/O Module . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
Additional Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126
x
Shapi ng Sound i n the Studi o and Beyond
Chapter 5
Microphones 129
Dynamic Microphones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Condenser Microphones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Lavalieres . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
Boundary Microphones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
Shotgun Microphones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
Wireless Microphone Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
Polar Patterns . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
Critical Listening . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
Microphone Placement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
Additional Reading. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
Chapter 6
Speakers and Amps 161
Theory of Operation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Moving Coil Speakers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 165
Ribbon Speakers. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
Electrostatic Speakers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Woofers and Subwoofers. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Mid-Range Drivers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Tweeters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
Crossovers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
Studio Monitors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
Bookshelf Speakers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173
Sound Cubes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
Enclosures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 175
Line Arrays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 175
Efficiency, Frequency Response, and Distortion . . . . . . . . . . . . . . . . 177
Amplifiers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
Chapter 7
Digital Audio 187
Sampling Rates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189
Aliasing and Quantization Problems . . . . . . . . . . . . . . . . . . . . . . . . 192
Contents
xi
Error Correction, Compression, and Formats . . . . . . . . . . . . . . . . . 193
Bit Quantization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
MIDI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
SMPTE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
PART III
AUDIO METHODS AND OPERATIONS 203
Chapter 8
Editing 205
Objectives . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 207
Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208
Methodology of Simple Edits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 209
Methodology of Complex Edits . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
Possible Edit Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
Digital Editing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 222
Chapter 9
Signal Processing 225
Historical Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
Digital Signal Processing Technology . . . . . . . . . . . . . . . . . . . . . . . 233
Classification of Processors by Types of Perception . . . . . . . . . . . . . 236
Processors That Affect Loudness . . . . . . . . . . . . . . . . . . . . . . . . . . 237
Processors That Affect Pitch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
Processors That Affect Timbre . . . . . . . . . . . . . . . . . . . . . . . . . . . . 244
Processors That Affect Envelope. . . . . . . . . . . . . . . . . . . . . . . . . . . 254
Processors That Affect Location. . . . . . . . . . . . . . . . . . . . . . . . . . . 255
Processors That Are Fun . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
Signal Flow or Where to Use a Processor . . . . . . . . . . . . . . . . . . . . 259
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 262
xii
Chapter 10
Mixing 267
A Good Mix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268
I/O Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Master Section of the Console . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
Normaling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Using Monitors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Methodology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
Mixing in a Sphere . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 294
Chapter 11
Recording Studio Operations and Procedures 297
Production Facility Job Descriptions . . . . . . . . . . . . . . . . . . . . . . . . 298
Entry-Level Positions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
Studio Etiquette . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
Recording Studio Operations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Being the Best Assistant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 326
Chapter 12
Sound Design 329
Believability of Sound Cues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 331
Semiotics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333
Emotion, Anticipation, Subtlety, and Continuity . . . . . . . . . . . . . . . 336
Silence. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 338
Methodology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 339
Attention to Detail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
Incorporating Subtlety and Subliminal Cues . . . . . . . . . . . . . . . . . . 345
Incorporating Silence for Anticipation. . . . . . . . . . . . . . . . . . . . . . . 346
The Challenge of Ambience . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 346
Establishing Location Characteristics . . . . . . . . . . . . . . . . . . . . . . . 347
The Second Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 348
Selecting a Playback Format . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Incorporating Room Characteristics and Existing Equipment . . . . . . 351
Assessing Available Equipment. . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Contents
xiii
Speaker Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Choosing and Placing Microphones . . . . . . . . . . . . . . . . . . . . . . . . 353
Cue Sheets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 356
Appendix A 361
Bibliography . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 361
Appendix B 367
Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 367
Index 387
xiv
Introduction
After 10 or 15 years as an audio professional, I began to teach. I had done
bunches of albums and jingles, lm sound tracks, and Broadway cast albums; I
had sound designed plays and done sound reinforcement. My students were as
varied as my career. Some were aspiring recording engineers. Some were aspiring
sound designers. Some were musicians who wanted to learn the language and
communicate better with engineers and producers. There was no one book
that covered all these goals. So I wrote one.
I self-published that book for 10 years. It was used at a half-dozen colleges and
universities. Then something interesting started to happen. Digital technology
exploded and became affordable. Seemingly overnight, anyone who wanted to
could load some software on their computer and have a studio in their apart-
ment. Bedrooms became control rooms. Bathrooms became iso booths. And
the students changed. Many students now aspired to own successful home stu-
dios rather than work for large, commercial facilities. Some students just wanted
to learn how to use their software betteror learn why it sounded better when
they set something a certain way in their software. No book addressed these
issues in a clear and broad fashion. So I rewrote my book to cover this group
along with the other, more traditional students.
My goal in this book is not to single out any one form of audio. My goal is to
teach the basics of audio that are universal. While the day-to-day operations of a
sound designer, an assistant engineer in a commercial recording studio, and an
owner of a home studio are radically different, the principles that support their
work are identical. We all use technology and apply an aesthetic appropriate to
our goals. We are all grounded by the limitations of our systems, and we must
acknowledge those limitations. We all route and process our signal with the hope
that the signal will be clean and fulll our needs. Rather than limit this book by
directing it to one group, the book addresses the audio professionalthe com-
monality between us all.
It does this by following a workow model. First we will cover the basics. Why
do we do what we do? How do we route signal? What are the principles sounds
will follow, both as electricity in wires and as sound waves in air? How will we
be able to manipulate these waves to accomplish our goals?
Then we will discuss gear. Microphones, consoles, speakers, tape machines, and
hard drives. We will talk about how and where we process signals. As audio
professionals we need to understand our options regarding capturing, routing,
processing, storing, and reproducing sound. This section will familiarize us
with the gear, and we will learn how it hooks together.
After that we will talk about methods and operations. Now that we understand
how sound behaves and how we use gear to manipulate that behavior, how do
we put it all together? How does that audio signal actually become a CD or a
xv
sound cue in a play? How do we act in a studio or otherwise in front of a client to
increase the odds of keeping that client?
This book will not guarantee you success in the eld of audio, but it will help you
to understand how everything works, from gear to interpersonal relations. It will
give you an edge over your competition. What you do with that edge, whether
you use it at home or ride it to a brilliant career in audio, is up to you.
xvi
PART
I
Basic Audio Theory
1
1
Audio Aesthetics
and Technology
P
h
o
t
o
c
o
u
r
t
e
s
y
o
f
S
o
l
i
d
S
t
a
t
e
L
o
g
i
c
.
3
W
hen sound is used in entertainment and media, it bridges
the gap between aesthetics and technology. Whether on
stage, in a recording studio, on lm, on radio, or in a live
venue with a band or orchestra playing, the person who is engineering
or designing the sound should never be thought of as strictly a techni-
cian or strictly an artist. Indeed, if this person thinks of himself as
solely a technician or solely a designer, he is likely to fall short in pro-
viding the audience or end user with the best possible sound, regardless
of whether that sound is music, speech, sound effects, or any combi-
nation of the three.
The fundamental mandate of anyone working in audio has not
changed over the years. It still involves understanding and utilizing
signal ow to our greatest advantage. The role of the engineer or
sound designer, however, entails much more than simply passing
signal from source to speakers. It involves shaping that sound in a
way that makes it aesthetically appropriate for its particular use.
In music, if a guitarist plays an excellent riff that happens to be in a
different key than the rest of the musicians are playing, the guitar part
will sound ne when played alone, but will not work as part of the
musical whole. Much in the same way, any technician can capture
and amplify an audio signal, but that alone does not mean it will
work aesthetically as part of the artistic whole. From an aesthetic
viewpoint, a sound designer may create an incredibly beautiful sound
cue of a swamp at night, but if the play takes place entirely during the
day, this cue will be totally inappropriate. Audio professionals do not
operate in a vacuum.
In the history of recording, early engineers were little more than tech-
nicians in white lab coats who passed signal from a microphone
through a console to a recording device. Some of these engineers did
not bother to listen to the signal they were passing because their judg-
ments were primarily technical. Although most aspects of audio have
changed, the principle of signal ow has not. Signal ow requires
selection of a path for the electrons, and it is imperative that the
audio professional retain the maximum quality for that signal. In the
early days, engineers thought of the signal in its most primitive form,
4
as electricity passing through circuitry. Their simple goal was to keep
this electricity at an acceptable level while it was passing through their
equipment. Artists and producers would concern themselves with the
aesthetic elements of the sound that the engineer captured and routed
once it reached the speakers, where it was transformed once again into
acoustical energy, or sound.
In the last 40 to 50 years, pressure has increased on sound engineers
to be the rst line of defense toward an appropriate aesthetic. In lm
and video, closer attention has been paid by many directors to the
quality of audio in post-production. As broadcast technology has
improved, the quality of the signal of a TV or radio station has
come under greater scrutiny. With the advent of multi-track re-
cording and close miking (placing microphones close to instruments
for isolation), greater attention has been focused on the aesthetic
elements of music in recording studios. The modern engineer still
needs to be concerned about the technical aspects. The signal passing
successfully through the system without any signicant loss is still
necessary, but not enough. We are more concerned than our prede-
cessors about the aesthetic quality of the end result. As such, to sur-
vive in the modern world of audio, one needs both the ability to
technically support the signal and the aesthetic sensibility to judge
the resulting sound.
The technical approach to sound is fairly straightforward. Using a
microphone to capture live sound, or a source such as a CD sound
effects library, the sound is sent through a system following a specied
path (see Figure 1.1).
This path, or signal ow, is critically important from a technical stand-
point. A shorter path generally allows for greater signal quality, which
minimizes risk of distortion and circumvents other potential problems
before they occur. This path may terminate in a storage device such as
a hard drive or tape machine, a pair of speakers, a speaker array, or
any combination of these, and may pass through any number of de-
vices along the way. The common feature is that the engineer has
selected that path carefully to ensure the best possible quality at
every turn.
Chapter 1 Audi o Aestheti cs and Technol ogy
5
In Search of Aesthetics
Aesthetically, the approach is more complex. Choosing the proper
preamplication, for instance, becomes critical. In addition to choos-
ing a preamp that will give the signal a proper boost, the engineer
needs to choose a preamp that will introduce a minimum of noise.
Although the choice of a preamp that will introduce a minimum of
noise would appear to be purely technical, for many it becomes an
aesthetic decision. Some preamps have reputations for having a
warm sound, with some timboral coloration, while others are
thought of as transparent or cold. Even on the most basic level,
this is an example of how the technical and the aesthetic cannot be
separated.
Part of the audio professionals job is to expand that path as needed. If
the signal needs to be sent to outboard signal-processing gear, it becomes
Microphone Console Amplifier Speakers
Outboard
Equalizer
Outboard
Equalizer
Recording
Device
Figure 1.1 Routing can be as simple as sending a signal from a microphone through a
console, to an amplier and speakers. Every time we add a device, such as an outboard
equalizer or a recording device, the signal path becomes more complex.
6
the audio professionals job to perform these tasks with the shortest and
cleanest addition to the signal ow. Properly routing the signal to the
appropriate location and bringing it back without compromising that
signal is critical. While the decision to utilize equalization, compression,
or other signal processing is often aesthetic, determining the best route to
get there is a technical decision.
Routing can best be explained in everyday situations. Every time a
light switch is turned on, electricity is routed through that switch to
a light bulb. This is signal owelectricity following a specied path.
When you play a CD on your home stereo, you are actually routing
the signal as follows: The output of the CD player is routed to the
input of the receiver or the input of the preamp, which outputs to
the input of the amplier. The output of the amplier or receiver is
then routed to the input of the speakers, where the electrical energy
is converted into acoustic energy (see Figure 1.2).
Another example of signal ow in the home is a cable/VCR setup. The
cable or satellite TV signal coming into the home is routed into the
descrambling box, from the box to the VCR, and from the VCR to
the television (see Figures 1.3 and 1.4).
Although this may seem simple, its simplicity is its beauty. This type of
direct path ensures successful viewing with maximum signal quality. If
we were to attach the cable rst to the VCR, then to the cable
descrambler box, then to the TV, we would be unable to record the
incoming signals.
In the studio or production room, the path we choose is critical. The
basic signal path in the studio starts with the source (microphones or
recorded material we are processing), which is routed to the audio con-
sole, and from the console to a recording device. From the recording
device it is routed back to the console (for monitoring purposes), from
the console to an amplier, and from the amplier to the speakers (see
Figure 1.5).
In a sound reinforcement situation, the engineer may forgo the record-
ing device and send the signal directly from the console to a bank of
ampliers, then to the stacks of monitors or the speaker array in the
auditorium or stadium (see Figure 1.6).
7
In a radio station, the signal will be sent directly from the console to a
series of signal processors, which then send the signal to the transmit-
ter (see Figure 1.7).
In all of these situations, the signals path will rarely be as simple as the
ones described previously; the engineer will commonly route the signal
through a variety of processing gear to apply aesthetics to the sound.
Audio and Aesthetics
How can we effectively apply aesthetics? First we must attempt to dene
aesthetics and develop a deeper understanding of this perceptual and
To an AC Outlet
Compact Disc Player
Amplifier
Speakers
To an AC Outlet
Line Out
Speakers
CD
Figure 1.2 Every time you play a CD, you are actually routing signal along a specied
path. Based on original diagram courtesy of Sony.
elusive concept. In the most simple and classic denitions, aesthetics is
8
Wall
Cable Box
Television
In
Out
Figure 1.3 As you can see from this example, television signals also need to be routed.
Based on original diagram courtesy of Shure.
considered a branch of philosophy devoted to beauty and the arts. It
denes acceptable parameters of appearance and performance and
attempts to quantify quality. If that sounds difcult and relative,
thats because it is. You may like a certain style of music that your
best friend hates. Perhaps that style of music ts your personal aes-
thetic, your personal sense of beauty. To you, this particular style of
music is more than just acceptable. It sounds good to you. Perhaps it
even feeds your soul in some way. It ts your personal aesthetic. Your
friend, on the other hand, has a different set of criteria. This certain
style of music is beyond his parameters. It just doesnt work for him.
Your friend may in fact cringe at the same music that brings you joy
and fulllment. How can two people have different opinions regarding
the same piece of music? We have been told all our lives that tastes are
personal. We share some aspects of what we like and dislike with other
people, but it is never a clear distinction. It is amorphous and ethereal.
Sometimes it is difcult to articulate the reasons why a particular piece
of music or a particular piece of art will move you. That is because
9
Wall
Cable Box
VCR
Television
In
Out
In
VHF/UHF
Out
Figure 1.4 The addition of a VCR requires further proper routing. Based on original
diagram courtesy of Shure.
Recording
Device
Figure 1.5 A simple yet typical setup for a recording studio. This signal path will
change depending on the situation. Even the mood of the engineer will sometimes
dictate a change in the signals path.
Microphone Console
Stage
Amplifiers
Stage
Monitors
Figure 1.6 A simple yet typical setup for a sound stage.
10
aesthetics can be so personal that it is difcult to understand our own
preferences.
Have you ever walked into a museum with friends and noticed that
everyone gravitates to something different? You may look at a piece
of art and think to yourself that your three-year-old cousin could have
created it. The photo, painting, or sculpture next to it, on the other
hand, may move you deeply. Other patrons of the museum and friends
of yours may like the painting you disliked, and vice versa. It all comes
down to your personal aesthetic versus their personal aesthetics, and
no one is right or wrong.
There are those who consider themselves wine experts. They will tell you
which wines are best with different types of food. They will tell
you which vineyards and vintages are better. Suppose you buy a bottle
of wine that they recommend against, or serve a type of wine with a
type of food that is contraindicated? Suppose you like that wine in that
situation. Who is correct, you or the expert? According to your per-
sonal aesthetic, you are right if it works for you. We conform to cer-
tain parameters within our personal aesthetic. We would be unlikely to
prefer the avor of turpentine or gasoline; however, we are far more
subjective and judgmental in nuanced situations regarding what we
eat, drink, see, and hear.
In audio, the technical elements are fairly simple to understand and
uniform throughout the industry despite minor differences. Applying
an aesthetic sense to sound, however, becomes more subjective. Every
engineer will agree about how to read the level of a signal appearing on
a meter (although it can be open to interpretation); however, many
will have different opinions about the effect of minor changes in equal-
ization or reverberation. When assessing a sound from an aesthetic
standpoint, audio professionals are assessing a subjective quality
based on their personal experiences and history. Despite the subjective
Microphone Console Processors Transmitter
Figure 1.7 A simple yet typical setup for a radio station.
11
nature of this assessment, there are certain criteria we can use to break
it down.
Sounds need to be understandable within the context of their given
environment. Context is determined by the content of the material,
which comes with an assumption of the parameters regarding the
nal product. If it is supposed to be heavy metal, it should sound
like heavy metal. Context is also affected by the overall category
within the broad denition of the audio professionals eld, whether
it is a song that will be played on the radio or a sound cue for a play
that will be performed in a theatre. An engineers assessment of the
proper amount of reverberation on a guitar will be different in the con-
text of a recording studio than it will be in the context of a reverberant
theatre; the engineer will certainly want less reverb in the theatre cue
because the reective house will add quite a bit on its own.
In a recording or sound reinforcement situation, a guitar so blaringly
loud that it drowns out everything else is a bad aesthetic choice. Sim-
ilarly, if a singers voice is so quiet that it cannot be heard over the
band, the audio professional is not fullling his or her obligation. Ide-
ally, all elements need to be balanced, or in context with each other.
In lm and theatre situations, creating sounds in context is critical.
This can mean different things depending on the nature of the produc-
tion. In a farce, exaggerated sound effects can be very successful as
part of the overall production; however, in a drama they would be
inappropriate. In drama, the sound cues need to be as subtle and
believable as every other design element of the production, including
the direction, sets, costumes, and lighting.
To further complicate the theatre model, the sound cues need to sound
good in the space where they will be used. A sound designer might
create a sound effect that sounds perfect in the studio, but if this
sound is not as effective in the theatre, it is a failure. Sound cues for
the theatre must be created with the nal destination in mind, which
often necessitates a return trip to the studio to perfect a cue that
sounded good in the studio, but fell short in the theatre during tech
rehearsal.
12
Audio Technology Meets Aesthetics
Audio has taken many varied forms in todays world. The technology
used changes daily. Although this may seem intimidating at rst, the
basic skills carry over as each new piece of equipment, new method,
and new medium is introduced. New equipment requires devoting time
to master its applications, but understanding its use will build upon
knowledge of existing equipment and methods.
Audio has also attained a more prominent role in a variety of elds.
Many young engineers are seduced into the eld with dreams of
working in a recording studio or a Broadway theatre, but the reality
of the business is broader. There is a tremendous amount of work
available to the audio professional in other related elds, such as
sound for video games, websites, and multimedia presentations in
corporate and educational environments. The width and breadth
of possibility in audio is limitless and unpredictable; fortunately,
the building blocks of knowledge needed to succeed in all these elds
are shared.
This is where aesthetics and technology merge. When discussing aes-
thetics and technology, a distinction must be drawn between the phys-
ical reality of a sound and our perception of that sound. A sound wave
may appear a certain way on an oscilloscope, but a more important
feature of this wave to the audio professional is how it appears to the
listeners ears. One example of the difference between the physical
reality of a sound and our perception of it is described by the Fletcher
Munson Equal Loudness Contours, which we will explore more
deeply in Chapter 2, The Properties and Characteristics of Sound.
According to Fletcher Munson, sounds at different pitches of equal
loudness will seem to our ears to have different loudnesses. In other
words, our ears will perceive certain midrange frequencies to be louder
than the high and low frequencies in a sound, even if the amplitude of
the different frequencies is identical. A sound cue containing a variety
of frequencies will, therefore, stress different frequencies at different
playback levels, affecting the overall feel of the cue. As audio profes-
sionals, we are as concerned with the perceived sound as we are with
reality as portrayed on a meter (see Figure 1.8).
13
Much like a lighting designer in a theatre or an architect designing a
cutting-edge building, an audio engineer is constantly combining his or
her aesthetic approach with the technology necessary to make it work.
If an architect were to design and build the most beautiful and inno-
vative building of our agea building that collapsed within months
of its constructionthat building would be thought of as a failure
because aesthetics alone are not enough. The building also needs to
remain standing to fulll its function. The same architect could have
built a sturdy concrete bunker that would remain standing but be
unappealing to the eye. In this case, he has fullled the functional or
technical element but failed to create something aesthetically pleasing.
If this architect had constructed a building that was both appealing
and enduring, he would have accomplished that for which the audio
engineer strives. He would have bridged the gap between the aesthetic
and the technical.
Perceived Pitch
Frequency
A
m
p
l
i
t
u
d
e
Perceived Pitch
Frequency
A
m
p
l
i
t
u
d
e
Figure 1.8 When amplitude or volume is high, our perception is even throughout the
frequency spectrum. At low listening levels, we perceive pitches in midrange frequen-
cies as loudest and low frequencies as softest.
14
Neither aesthetics nor technology stand alone; each one requires the
other to be implemented. An audio engineer without any sense of aes-
thetics could successfully run current through a console, but would
be unable to judge whether the resulting sounds were effective for
the application. Similarly, someone with a well-developed aesthetic
sense who possesses no technical knowledge would fall short because
that person would have an idea in mind but be unable to execute it.
This book will endeavor to close the gap between the technical and the
aesthetic systems used in sound. First we will analyze the technical ele-
ments necessary to understand sound, produce and execute sound
cues, place microphones on instruments and record them, and nd
the shortcomings and advantages to a variety of sound environments.
We will then establish a set of aesthetic criteria that will apply to a
variety of applications, including theatres and recording studios. The
combination will allow form to follow function and give students a
solid basis to approach any real-life audio situation with both an aes-
thetic sense and the necessary technical knowledge to implement it. To
break conventions and play with perception, we rst need to under-
stand what is conventional. Lets begin by taking apart sound waves,
learning their components, and analyzing them enough so we can
manipulate them to conform to our growing personal aesthetic.
15
2
The Properties and
Characteristics of Sound
Time
A
m
p
l
i
t
u
d
e
17
O
ne must rst understand the physical properties of sound to
understand the various ways in which sound can be manipu-
lated to achieve believable audio environments. How that
sound is affected by the surrounding environment and the way in
which it is perceived by the human ear are also critical to the engineer.
Equally important is the idea that, while some of the technical ele-
ments cannot be modied, our perception of them may vary depending
on circumstances including the room in which we are listening, our
position in that room, and even the level at which we are listening.
These are perceptual factors, which we will delve into later in this
chapter. First, the reality of sound: How do sound waves behave?
Why? What can we predict about the performance of sound waves?
How can we relate the technical aspects of sound to the perceptual
aspects? How can we use this knowledge to create sound environments
and manipulate the listeners surroundings? When is it appropriate to
use this trickery and when is it inappropriate? What does all this
mean to the audio professional?
Sound as a Waveform
Sound exists as waves of compressed and rareed molecules moving
through a medium. When a force (such as a tree falling and striking the
ground) is applied to a medium (such as air), the molecules within that
medium are momentarily displaced and collide with neighboring mol-
ecules. This causes moving waves of varying pressure to propagate
spherically from the source, like the ripples in a pond when a pebble
is thrown in (see Figure 2.1).
When the pebble strikes the pond, the concentric circles grow larger as
they move away from the original force. The circles move in two
dimensions. Unlike the pebbles in the pond, sound waves move
away from the original force spherically, or in all directions at once.
When these molecules move away from each other as sound is being
propagated, they are in a state of rarefaction. As these molecules move
toward each other, they are in a state of compression. When waves
propagate in the same direction as the displacement of the molecules
in the medium, it is known as a longitudinal wave. Sound waves are
always longitudinal waves.
18
A Slinky, when snapped, is one representation of longitudinal wave.
The wave moves evenly and freely through the toy, compressing and
rarefying in a uniform direction and speed. Another example of a
longitudinal wave is contained in a row of billiard balls, which can
be thought of as the equivalent of large molecules that will be com-
pressed and rareed. The rst ball strikes the second, then recoils
toward its original position while the second ball rolls forward to the
third, sending that ball forward as it too returns toward its original posi-
tion. The force of the wave in this case is being pushed forward from
ball to ball, as each ball in turn attempts to return to its original posi-
tion after passing on its energy (see Figure 2.2).
Billiard balls simulate the actions of molecules in air, where they move
until they strike another molecule, then recoil in an unpredictable
direction. They may simply bounce back toward their original position
as the billiard balls do in the diagram. They may bounce in a different
direction, striking other molecules and propagating the sound wave in
other directions. If the sound wave is moving through another medium,
such as a liquid, molecules will move less freely, in a more minimal
range of directions. If the sound wave is moving through a solid, the
Figure 2.1 The concentric waves that form around a pebble striking a pond are an
example of a transverse wave.
Chapter 2 The Properti es and Characteri sti cs of Sound
19
molecules will be more likely to attempt to return to their original
position.
Sound as a Form of Perception
In its purest form, as described by physicists, sound is a mechanical
vibration of molecules in a medium as described earlier. As a form
of perception, however, the observer becomes an essential ingredient
in a sound event. As a form of perception, the mechanical vibrations of
molecules within a medium strike the ear and cause electrochemical
reactions in the brain that are interpreted as sound. This denition is
fundamentally different, as it requires an observer to interpret the
sound event.
The minimum requirements for a sound event as a form of perception
are therefore:
n
An applied force to initiate the displacement of molecules, such as a
hammer striking a wall or a nger plucking a string
n
A medium with molecules in sufcient quantities for the waves to
propagate, such as air or water
n
An observer to interpret the pressure waves as sound
1 2 3 4
1 2 3 4
1 2 3 4
1 2 3 4
Figure 2.2 When ball 1 is pushed toward ball 2, it strikes the ball, which moves for-
ward toward ball 3. Ball 1 will recoil, attempting to return to its original position, while
ball 2 strikes ball 3. Ball 2 will then attempt to return to its original position while ball
3 pushes on to strike ball 4. This is an example of a longitudinal compression wave,
where the wave is propagated in the same direction as the original disturbance.
20
We are not physicists; we are audio professionals. A physicist will only
require two elements for a sound event to occur. In audio we are
focused on more than the scientic reality, so as a form of perception,
the minimum requirements for a sound event to occur are force,
medium, and observer. If no medium exists (as in a vacuum), no sound
event will occur. Similarly, if no observer exists, there is no sound event
to manipulate. For the purposes of audio aesthetics, we will treat sound
as a perception rather than a mechanical wave. This gives us the free-
dom to incorporate perception and use subjective criteria when assess-
ing and inuencing the nal product.
Let us look again at the example of the tree falling in the forest, in
context of this denition. When the tree falls and strikes the ground,
the force displaces molecules that vibrate within the medium (air).
These molecules displace other molecules that eventually reach the
ear of the observer and are interpreted as sound. If the tree does not
strike the ground, there is no force and, therefore, no sound event. If
there are no molecules to vibrate and be displaced, as in a vacuum,
there is no sound event. If there is no observer to interpret these dis-
placed molecules as sound, there is no sound event. This is, of course, a
purely audio-based denition. A physicist will argue that the observer
is unnecessary, since a force and a medium are enough to create and
perpetuate a sound wave. For our purposes the observer is necessary
because from an aesthetic standpoint, we need to know not only that
the sound occurred, but also what it sounded like and the many ways
we can manipulate that sound to suit our purposes.
Let us look at another example. When a hammer hits an anvil, the
force of the hammer striking the anvil causes molecules to vibrate
and be displaced. When these vibrating molecules arrive at the ear
of the observer, they are interpreted as sound. If the anvil were struck
in a vacuum, the absence of molecules would preclude a sound event
because there is no medium. Without any of these three elements
force, medium and observerthere would not be a sound event. Since
we are dealing in audio, and the observers perception of the sound is
critical, we will simply accept the need for the observer in our deni-
tion. And besides, without an observer, who is going to sample the
21
hammer hitting the anvil and set it to trigger whenever the drummer
hits the snare drum? Well explore the concept of triggers in more detail
in Chapter 9, Signal Processing.
The Speed of Sound
As a longitudinal wave moves in the same direction as the original dis-
placement of molecules, it will travel at a uniform rate in a uniform
medium. The speed of sound will remain unchanged while the medium
remains consistent. In other words, as long as everything else stays the
same, the speed of our sound wave will also stay the same; in fact, it
becomes predictable. The aspects of a medium that are responsible for
the speed of a wave are density and elasticity.
Density is the mass per volume, or the number and mass of molecules
within a given physical space within the medium. Think of density as
how many molecules are crammed into a given spacethe more mol-
ecules we shove into that space, the higher the density will be. Elasticity is
the ability of molecules to return or spring back to their original location
after being displaced. Imagine molecules with high elasticity to be
attached to rubber bands, which return them to their original position
after they are pushed by a sound wave. In other words a typical solid,
such as aluminum, will contain more molecules within a cubic foot than
a typical gas, such as air. Aluminum, therefore, has a higher density than
air. Aluminum also has greater elasticity than air. When a block of alu-
minum is struck, it does not have much give and will therefore return to
its original state quickly.
Air, on the other hand, has a great deal of give and is relatively slow to
return to its original state. When we momentarily displace the mol-
ecules in a block of aluminum by striking it, we know that those mol-
ecules will quickly return to their original position. When we displace
air molecules we do not know where they will end up due to airs
extremely low elasticity. Greater elasticity within a medium causes
waves to travel through that medium more quickly because the mol-
ecules move as a result of the sound wave, push other molecules, and
then return to a state of rest rapidly. Waves will also travel faster
through a medium with low density, as long as there are enough mol-
ecules to strike one another.
22
One can hear distant sounds better on a summer night when the air
density is lower than it is during winter; however, if the air became too
thin and low in density, sound waves would not be able to travel at all.
At what point are there too few molecules? Sound will travel progres-
sively faster as the air thins (creating lower density); however, in outer
space the density is so low that a displaced molecule may not collide
with another molecule, making it impossible to propagate a sound
wave.
Assuming we have the necessary minimum number of molecules, in the
case of sound moving through solids, liquids, and gases, elasticity is a
greater factor than density. Sound will move fastest in a highly elastic
medium, provided the density is not of a magnitude that requires
excessive energy to displace molecules. Sound waves will therefore
travel fastest in mediums with low density and high elasticity. This
means that a sound wave will travel faster through water than air
due to the high elasticity, and it will travel even faster through metal.
Temperature is also a factor. Density increases as temperature
decreases, and molecules speed up as temperature increases. This ex-
plains why sound travels faster and further on a summer night than in
winter. Elevation alone has little effect on the speed of a sound wave
because elevation changes atmospheric pressure, which affects density
and elasticity proportionately. Since density and elasticity are both
reduced at higher altitudes, they essentially cancel each other out. Eleva-
tion does, however, affect the atmospheres ability to hold heat as a
result of lower atmospheric density. The speed of sound decreases
slightly at high altitudes due to the decreased temperature.
So what are we really talking about in terms of speed in various media?
The speed of sound in air at 32 degrees Fahrenheit is 1,088 feet per
second (ft/sec). The wave will increases in speed by 1.1 ft/sec for each
one-degree Fahrenheit increase. At 70 degrees Fahrenheit, the speed of
sound is 1,130 ft/sec. Sound travels faster through liquids and solids;
the speed of sound in salt water at 32 degrees Fahrenheit is 4,800 ft/sec,
while in aluminum the speed of sound is 17,000 ft/sec. While sound trav-
els faster in liquids and solids due to higher elasticity, it will not travel as
far due to the high density. While one might hear the sound of a freight
23
train from miles away through the summer night air, one would be
unlikely to hear that same freight train through a mile-wide block of
aluminum.
The Human Perception of Sound
We have established that sound waves may be viewed as a technical
phenomenon, waves of molecules displacing other molecules as a
sound wave propagates in all directions. We also know that in order
to manipulate these waves to our greatest benet, we need to view
them as a matter of perception and gain an understanding of how
our listener will perceive our handiwork. Since we favor dening
sound as a form of perception, we need to quantify the nuance of
sound. There are ve basic human perceptions of sound that allow
us to distinguish one sound from another:
n
Loudness
n
Pitch
n
Timbre
n
Location
n
Duration
These are the only ve means available to us while interpreting differ-
ences in sound. Obviously, few sounds will have only one difference;
more commonly two different sounds will have three, four, or even
ve perceptual differences. Sometime these variations in our perception
will be subtle, such as a slight difference in volume; other times they
will be signicant, such as the differences in sound produced by a trac-
tor compared to a lawnmower.
Loudness
Loudness is our perception of amplitude, which is measured in decibels
(dB). Loudness represents the perception and measurement of changes
in volume levels.
Amplitude is the size of a sound wave or the amount of displacement
of molecules within the medium. Increases in amplitude usually result
24
in the perception of increases in loudness. Normally, an increase in the
applied force causes an increase in amplitude and a subsequent increase
in loudness; however, the increase in loudness is not necessarily linear
with the increase in amplitude. While a logarithmic relationship exists
between amplitude and sound pressure level, the relationship between
amplitude and our perception of loudness is colored by other factors,
such as frequency, which will be discussed in the Fletcher Munson
Equal Loudness Contours section later in this chapter. Due to the dif-
ference between the objective, measurable increase in amplitude and the
subjective, perceptual increase in loudness, there is need for a separate
system of measurement for loudness.
The decibel is a unit of measurement used to calculate sound pressure or
sound intensity. Due to the enormous range of volumes that the human
ear is capable of hearing, a logarithmic scale is necessary to understand
and use this range. The vastness of this scale, which encompasses an
energy range of more than one trillion to one, would cause a linear sys-
tem to be unintuitive. Furthermore, our perceived difference in loudness
spans a different range at different listening levels. In the raw language
of phons as a measure of loudness, an increase of a few thousand phons
at a low listening level would be equivalent to an increase of a few
million phons at a louder listening level. Instead of asking your
roommate to raise the level 1,000,000 phons, the decibel scale was
created to make things more uniform and intuitive. The logarithmic
nature of the decibel scale allows it to cover large spans with small
numbers, making it easy to understand it in terms that the human ear
can understand. Simply put, a decibel value is derived from a ratio
comparing either a subjective value to a known value or comparing
two known measurable values (see Figure 2.3).
The logarithmic nature of the decibel scale has another advantage in
that it is similar to the way in which we hear. The logarithmic scale of
decibels allows us to quantify increases and decreases in amplitude in a
manner that is similar to the way our ears quantify loudness.
To illustrate the logarithmic nature of decibels in which a 3-dB increase
represents a 2:1 increase in power, while a doubling of sound pressure is
represented by a 6 dB-SPL boost, let us suppose that a 100-watt
25
amplier produces a sound pressure level of 80 dB. Doubling the power
by adding another 100-watt amplier will cause a 3-dB increase in
power and a 6-dB increase in sound pressure level. In other words, we
describe a doubling of amplitude by adding 3 dB. Similarly, if we double
the power of a 500-watt amplier by adding another 500-watt amplier,
we still increase the power by 3 dB. The intuitive mathematical relation-
ship remains the same (2:1) despite the difference in energy (100 watts
versus 500 watts). A 2:1 ratio in power will always result in a 3-dB
increase. Regarding our perception of these changes, while the doubling
of power causes a measurable increase of 3 dB, our perception demands
about a 10-dB increase for us to believe we hear a doubling of volume.
As we continue to explore perception, we will nd many more examples
0 Threshold of Hearing 20
100
1000
10000
100000
1000000
10000000
100000000
10
20
30
40
50
60
70
80
90
100
110
120
130
Noise
140 dB Threshhold of Pain
Jet Take-Off (100 m distance)
Jet Engine (25 m distance)
Pop Group
Heavy Truck
Conversational Speech
Library
Bedroom
Pneumatic Chipper
Average Street Traffic
Business Office
Living Room
Wood
Sound Pressure Level Sound Pressure
Pa
Figure 2.3 Some sound pressure levels expressed in dB-SPL, decibel values, relative to
loudness. Based on an original diagram by B&K Instruments.
26
in which a technical increase does not translate into precisely the same
perceptual increase.
The span of sounds that the human ear can perceive is known as the
dynamic range. The lowest point in dynamic range is the threshold of
hearing, which is the softest sound the ear can hear or the minimum
energy required for the average person to experience the sensation of
sound. The upper parameter of dynamic range is the threshold of pain,
the point at which sound intensity causes pain in the average listener.
A light breeze gently moving leaves in the trees or someone who is
whispering softly is producing sound at or near the threshold of hearing.
Standing next to a pounding jackhammer, in front of the stage at a heavy
metal show, or on a subway platform in New York as the train pulls into
the station represents sound events occurring at or near the threshold of
pain. In decibels we quantify this as ranging from 0 dB to somewhere
between 120 dB and 140 dB.
Dynamic range for musical instruments or electronic equipment is dif-
ferent. It can be described as the span between the noise oor (the point
at which desired sound becomes louder than the ambient or inherent
noise within that particular device) and the point of distortion, satura-
tion, or clipping. These differences will depend on the type of equip-
ment. Analog gear that has too much level tends to be referred to as
saturating or distorting, while digital gear tends to clip. As our signal
rises out of the noise oor and increases, our signal gets louder com-
pared to the noise, and therefore gives us a better signal to noise (S/N)
ratio. (More signal and less noise is always preferred.) As our signals
level increases we approach our optimal level, also known as standard
operating level (SOL), unity gain, or 0 dB V.U. Above SOL we have
headroom, which is the space between SOL and the point of clipping or
distortion. This area of headroom gives us the best S/N ratio since it is
so far from the noise oor, and operating in this area is the goal of the
audio professional (see Figure 2.4).
Pitch
The perception of pitch is the means by which we judge the frequency
of a sound. Frequency is the measurement of the speed at which a peri-
odic or repeating waveform oscillates. One oscillation or cycle of a
27
waveform is comprised of one complete compression and one com-
plete rarefaction. This is called one cycle of a wave. The frequency
of a sound is measured by the number of cycles a wave completes in
one second. A greater number of cycles per second results in a higher
frequency and a higher perceived pitch. Conversely, fewer cycles per
second represents a lower frequency, which is perceived as a lower
pitch. In common terms, when we refer to one sound as high and
another as low, we are discussing frequency.
The standard unit used to measure frequency is the Hertz (Hz). One
Hz equals one cycle per second, and 100 Hz equals 100 cycles per sec-
ond. Frequencies above 1,000 Hz are usually delineated in kilohertz
(kHz), where 1,000 Hz equals 1 kHz, 5,000 Hz equals 5 kHz, and
so on (see Figure 2.5).
The formula for calculating frequency is frequency cycles/second.
This is a simple formula that tells us that the frequency of a particular
sound is determined by the number of cycles completed in one second.
The physical distance required for one cycle of a sound wave to com-
plete is the wavelength of that frequency. Wavelength is frequency spe-
cic; if two sounds are of equal frequency in the same medium, their
wavelengths will be identical. Due to the greater number of cycles in
Saturation (+20 dbm)
0 VU (Standard Operating Level)
(+4 dbm)
Noise Floor (60 dbm)
Headroom
Signal/Noise Ratio
Improves as
Signal Increases
Dynamic Range
Figure 2.4 Dynamic range in equipment.
28
higher frequencies, a higher frequency will have a shorter wavelength
than a lower frequency.
The formula for calculating wavelength is wavelength velocity/
frequency. In this equation velocity is the speed of the sound being mea-
sured and frequency is the frequency, in Hz, of the sound being mea-
sured. The wavelength of a 100-Hz tone traveling at 1,130 ft/sec is
therefore 1,130/100, or 11.3 feet. Assuming the medium is consistent
and the frequency is unchanged, the wavelength will remain the same.
Period is the inverse of frequency. While frequency is cycles per sec-
ond, period is seconds per cycle. Period is the measurement of the
length of time one cycle of a wave takes to propagate. The formula
for calculating period is period seconds/cycle.
Wavelength and period are similar in that they both measure one cycle
of a wave; however, wavelength measures the amount of distance one
cycle travels, while period measures the amount of time one cycle
requires to compress and rarefy. Simply put, wavelength is a function
of physical distance while period is a function of time.
By combining the three formulas for frequency, wavelength, and
period, we nd that we can determine any of the variables if the others
Amp
Time - 1 second
1 Hz
Amp
Time - 1 second
2 Hz
Figure 2.5 Frequencies of 1 Hz and 2 Hz.
29
are known, since frequency, wavelength, and period are all specic to
each other.
n
frequency cycles/second
n
wavelength velocity/frequency
n
period seconds/cycle
If we know the frequency of a sound, we can determine its period simply
by inverting it (10 cycles/second 0.1 seconds/cycle). If we know
that the wavelength of a sound is 11.3 feet and the speed of that
sound is 1,130 feet per second, we can determine that the frequency
is 100 cycles per second (1,130/11.3 100) and that the period of
that same sound is 0.01 (the inverse of 100 cycles, or 11.3/1,130).
Similar to (but not to be confused with) the dynamic range for sound
pressure level, the range of human hearing in pitch is called the audible
bandwidth and it ranges from approximately 20 Hz to 20 kHz. The
highest frequency that the average person can hear is 20 kHz, while
the lowest frequency the average person can perceive is 20 Hz, although
this varies greatly from one person to the next based on genetics, expo-
sure to high sound pressure levels, health, and other factors. The ability
to hear higher frequencies wanes with age, particularly among males,
and few males older than 30 years of age can hear above 15 kHz.
Octaves, a musical interval that the ear perceives as having a common
quality, are a function of a 2:1 ratio in frequency. Sounds occurring at
100 Hz and 200 Hz are one octave apart, as are tones occurring at
5 kHz and 10 kHz. In music, sounds with different pitches that possess
a 2:1 ratio carry the same musical description, such as an A 440 (occur-
ring at 440 Hz) and an A 880 (occurring at 880 Hz).
As noted earlier, frequency is a measurable quantity, while pitch is our
perception of that technical reality. Although our frequency will not
easily change as a result of outside factors and inuences, our percep-
tion of that frequency, pitch, can readily change based on numerous
factors, such as volume, room construction materials, and position rel-
ative to the sound source. After we establish the other three ways in
which we perceive sound, we will explore some of these factors.
30
Timbre
Timbre is the subjective quality or feel of a sound. Adjectives such as
bright, dull, edgy, smooth, brassy, or tinny are descriptions of timbre.
The volumes of various frequency components of a sound create that
sounds timbre. All sounds other than a pure sine wave contain energy
at many different frequencies, and each unique combination of distrib-
uted component frequencies is responsible for a unique timbre. When
you raise the treble or bass settings on your home stereo or in your car,
you are changing the timbre. The fundamental frequencies of the music
are not changing, but the music is becoming more brittle or fatter. A
change that can be described in these or any subjective terms is usually
a change in timbre. Keep in mind that since we do not change the fun-
damental frequency when we adjust the timbre, we are not changing
the underlying frequency. The singer is still singing the right notes
even though we have made the vocal extremely bright or dark by
adjusting the timbre. These are purely perceptual changes.
There is no unit of measurement to quantify timbre; however, it can be
analyzed with a graph that plots amplitude versus frequency of the
component frequencies, called a Fourier analysis. Fourier analysis is
a concept in which complex waveforms are viewed as a combination
of many sine waves. These sine waves of varying frequencies and
amplitudes each represent one component, or harmonic, of the com-
plex waveform. When combined they form the composite, complex
wave. The relative volumes of the various frequencies contained in a
sound may be viewed as a visual indication of the quality of a sound.
The frequencies contained in any sound may be broken down into four
types: fundamental frequency, harmonics, overtones, and non-
harmonically related overtones. Any combination of these will give a
sound its unique character, and adjustment of any of these other than
the fundamental frequency will change that unique character.
The fundamental frequency, sometimes referred to as the fundamental, is
the frequency responsible for giving a sound its pitch. The fundamental
is usually the loudest in amplitude and lowest in frequency contained in
the composite. It is loudest because the pitch of the sound is louder than
the harmonics and overtones, and it is the lowest frequency because the
31
harmonics and overtones are higher in pitch than the fundamental.
When we discuss an A 440, 440 Hz is the fundamental, while the over-
tones and harmonics occur at higher frequencies and lower volumes
when compare to the fundamental at 440 Hz.
Harmonics are whole-number multiples of a fundamental and are
sometimes called partial tones or partials. The timbre of a sound is a
direct result of its harmonic content. Harmonics are tones that,
together with the fundamental, make up a sound. The rst harmonic
is the same as the fundamental (fundamental 1). The second har-
monic is one octave higher than the fundamental (fundamental 2).
The third harmonic is the fundamental 3. Think of a choir in which
the voices are in harmony (notes that are complementary to the lead
singer) with the harmony voices singing more softly to allow the lead
singers voice to come through. Just like voices singing in a choir, the
specic harmonics present in the composite sound and their relative
volumes are responsible for the quality of a sound (see Figure 2.6).
Overtones are harmonics. The overtone series begins with the second
harmonic, which is the rst overtone. The third harmonic is the sec-
ond overtone, the fourth harmonic is the third overtone, and so on. If
the overtones are whole-number multiples of a fundamental, they are
A 880 Hz
4th Harmonic/
3rd Overtone
660 Hz
3rd Harmonic/
2nd Overtone
A 440 Hz
2nd Harmonic/
1st Overtone
A 220 Hz
Fundamental/
1st Harmonic
Figure 2.6 As you can see, the rst harmonic is the same as the fundamental frequency
(220 1), the second harmonic is the fundamental times two (440 Hz), the third har-
monic is the fundamental times three (660 Hz), and so on. The overtone series follows
the same frequencies as the harmonic series, but is always one behind.
32
harmonics. In other words, if you can divide the frequency of an over-
tone by a whole number, such as two or three, and the product is the
fundamental, the overtone is a harmonic.
Non-harmonically related overtones are those overtones that are not
whole-number multiples of the fundamental. Non-harmonic overtones
are responsible for white noise, which is dened technically as any or
all frequencies occurring simultaneously and is dened perceptually as
any frequencies occurring randomly. For our purposes, we will use the
latter denition so we can identify sounds such as thunder or waves cra-
shing on a beach as having a white-noise component. Non-harmonically
related overtones and white noise are major components in wind, rain,
tape noise, crumbling paper, and drums.
Various waveforms tend to have characteristic harmonic content,
allowing us to group sounds by the waveform (see Figure 2.7).
Sine waves, or pure tones, have no overtones or harmonics. They are
the simplest of waveforms. While sine waves are convenient for dia-
grams, they do not exist in nature. A metal ute or a nger circling the
rim of crystal glass produces the closest sounds in nature to a sine
wave.
Triangle waves contain only odd harmonics at very low amplitude rel-
ative to the fundamental. This gives a triangle wave a warmer sound
than a sine wave. Sounds carried in triangle waves include a wooden
ute or a marimba. The formula for harmonics in a triangle wave
is 1/N
2
(N number of the harmonic), where the third harmonic
has 1/3
2
, or 1/9 amplitude of the fundamental; the fth harmonic
has 1/5
2
, or 1/25 amplitude of the fundamental; the seventh har-
monic has 1/7
2
, or 1/49 amplitude of the fundamental; and so on.
Sawtooth waves contain all harmonics at relatively high amplitudes
relative to the fundamental. The result is a rich and somewhat buzzy
sound, similar to string and brass instruments. The formula for the
harmonics contained in a sawtooth wave is 1/ N, where the second har-
monic has 1/2 amplitude compared to the fundamental, the third
harmonic has 1/3 amplitude of the fundamental, the fourth harmonic
has 1/4 amplitude of the fundamental, the fth harmonic has 1/5
amplitude of the fundamental, and so on.
33
Sine WaveNo Harmonics
F (Fundamental) = 1, H
2
(Second Harmonic) = 0, H
3
= 0, etc.
Triangle WaveOdd Harmonics Only, at Low Amplitude
F = 1, H
2
= 0, H
3
= 1/9, H
4
= 0, H
5
= 1/25, H
6
= 0, H
7
= 1/49, etc.
Sawtooth WaveAll Harmonics, at a High Level
F = 1, H
2
= 1/2, H
3
= 1/3, H
4
= 1/4, H
5
= 1/5, H
6
= 1/6, H
7
= 1/7, etc.
Pulse WaveAll Harmonics, at Level Equal to Fundamental
F = 1, H
2
= 1, H
3
= 1, H
4
= 1, H
5
= 1, H
6
= 1, H
7
= 1, etc.
Square WavesOdd Harmonics Only, at a Relatively High Amplitude
F = 1, H
2
= 0, H
3
= 1/3, H
4
= 0, H
5
= 1/5, H
6
= 0, H
7
= 1/7, etc.
Figure 2.7 A variety of waves and their formulas.
34
Pulse waves are very rich in harmonics, with all harmonics equal in
amplitude to the fundamental. With a buzzy, nasal sound, double
reed instruments, including oboes and bassoons, are represented by
pulse waves. The formula for harmonics comprising a pulse wave is
much simpler, since the amplitude of any harmonic is equal to the
amplitude of the fundamental. Here is the formula: Fundamental 1,
Harmonic 2 (H2) 1, H3 1, H4 1, H5 1, and so on.
Square waves contain only odd harmonics at a high amplitude and a
very hollow sound. Single reed instruments, such as clarinets and sax-
ophones, produce square waves. The formula for harmonics contained
in a square wave is the same as the formula for a sawtooth wave,
except with only odd harmonics: 1/N, where the third harmonic has
1/3 amplitude of the fundamental; the fth harmonic has 1/5 amplitude
compared to the fundamental; the seventh harmonic has 1/7 amplitude
compared to the fundamental; and so on.
Duration
Duration, or envelope, is the volume shape of a sound over time or the
lengths of time of the components of any soundhow much time
passes as the sound begins, continues, and ends. The envelope of a
sound gives it its unique characteristics as a function of the uctuations
of loudness over time. Stringed instruments have similar timbre
whether they are plucked or bowed; however, each of these sound
events possesses drastically different envelopes. As a string instrument
is bowed, the sound waves in the body of the instrument cause the
body to slowly begin to vibrate. As a result, a bowed instrument will
rise in volume slowly and remain at a constant level. A plucked instru-
ment, on the other hand, will rise in volume quickly and diminish more
quickly than the same instrument played with a bow. We perceive the
difference in duration between these two styles as a function of enve-
lope (see Figure 2.8).
The frequencies may also change with time; however, for the sake of
quantication, we will deal with volume over time.
Envelope can be charted graphically as amplitude versus time, with
points plotted for the attack, decay, sustain, and release (ADSR) of a
sound. The attack of a sound is how it begins, whether by plucking,
35
bowing, striking, or blowing; sustain is the continuation of that sound;
decay is the decrease in volume after the stimulus has been removed;
and release is the eventual cessation of that sound.
While we are discussing envelope, some mention should be made of
transients. Transients are instantaneous peaks in amplitude, commonly
found in drums, percussion, bass guitars, and vocals. Transients can be
insidious in that they will not show up on VU meters (only on peak
meters and peak indicators), and due to their speed they may elude
your eye if they do show on your meters. Transients require particu-
larly responsive microphones to pick them up and often require com-
pression to avoid speaker damage and tape distortion. This will be
discussed further in Chapter 8, Editing.
Location
The fth perception of sound is based on its location. The ear is able to
distinguish between sounds depending on distance and direction. This
ability is due to binaural hearing, or the existence of two ears. Two
ears allow us to perceive and localize sounds by hearing in three
dimensions. Binaural hearing has three components: interaural inten-
sity, interaural arrival time, and physiology.
Interaural intensity is a comparison between the loudness level of a
sound as it reaches each ear (see Figure 2.9).
Time
A A S S D D R R
A
m
p
l
i
t
u
d
e
Envelope of a plucked
string instrument.
Time
A
m
p
l
i
t
u
d
e
Envelope of a bowed
string instrument.
Figure 2.8 The same instrument with two very different envelopes.
36
Subconsciously comparing slight differences in the loudness of a sound
in each ear causes us to seek the sound source in the direction of the ear
in which it is louder. The difference in volume between the ears allows
us to determine the location of the sound in degrees to the left or right.
If a sound appears to be equal in both ears, we perceive it as being
directly in front of, behind, above, or below us.
Interaural arrival time, or binaural time of incidence, is a comparison
between the time taken for a sound to reach each ear. Similar to inter-
aural intensity, we seek the sound source in the direction of the ear in
which the sound rst arrives. The difference in time determines how
many degrees to the left or right the sound source must be. Sounds
arriving at both ears simultaneously are perceived as being directly
in front of, behind, above, or below us.
Physiology, or the shape of the outer ear, allows us to distinguish be-
tween sound sources above, below, in front of, and behind us. The
pinna, or outer ear, has two ridges that reect sound into the ear.
Direct sound travels directly into the ear canal. Some of the sound
that goes past the ear canal will be reected into the ear by the two
Sound Source
Figure 2.9 The difference between time of arrival (interaural arrival time) and loud-
ness (interaural intensity) at our two ears is among the clues we use to determine the
location of a sound. Based on original photo courtesy of Sennheiser Electronics
Corporation.
37
ridges of the pinna. The reected sound arrives after the direct sound, and
depending on the position of the sound source, a unique combination of
direct and reected sound will be produced. This unique combinationwill
determine the location of the sound.
Our perception of direction can also be impacted by frequency. We
will learn more about directionality of different waves in The Behav-
ior of Sound Waves section later in this chapter. For now, a simple
missive that lower frequencies bend more should explain the problem.
Since lower frequencies bend around things such as the listeners head,
they will arrive at the distant ear with more force, which will make it
difcult to determine the direction of a low-frequency sound source
compared to a high-frequency sound source.
Environmental Effects on Perception of Sound
The impact of the environment on our perception of sound should not
be underestimated. In the early days of recording, a group would often
stand or sit around a single mic to record. The producer or record label
would select a location for recording due to the rooms characteristics.
The band would be positioned just far enough away from the mic to
allow the rooms reections to sweeten the blend. When it was time for
the sax solo, the saxophonist would take a few steps forward, making
himself louder in the mix, then step back at the end of the solo. Con-
cert halls and ballrooms were popular choices for early recordings
because these rooms made everything sound better. In this day and
age of digital chambers and plug-ins, our best technology emulates
those early rooms. This is another area where the line between the
technical and the aesthetic blurs. We need to understand the basic
behaviors of reections and how they combine with the original signal,
but we also need to imagine how we will perceive these complex
combinations.
All surfaces affect sound. When a sound wave encounters a surface,
part of that wave will be absorbed, part will be refracted, and part
will be reected. The amount of the wave that will be absorbed,
refracted, or reected will depend on the nature of the surface and
the frequency of the wave. The nature of the surface, whether it is
38
hard or soft, painted or unnished, will determine the amount of that
sound which will be reected. The sound of a basketball player drib-
bling a basketball in a gymnasium with highly reective cement block
or masonry walls will be substantially different than the sound of a
basketball dribbled in a living room with drapes. Frequency is also a
factor because certain frequencies tend to be more reective than
others.
How much of a sound wave will be reected, refracted, or absorbed
when it encounters a surface? The amount that is absorbed will depend
on the absorption coefcient of that particular surface. Absorption
coefcients are ratios that compare the amount of energy absorbed
to the amount of energy that is reected. The amount of energy that
will be absorbed depends primarily upon the hardness of the substance
of which the surface is made. Although there are published absorption
coefcients for a variety of building materials, it is fairly intuitive that
hard surfaces, such as glass, cement, or ceramic tile, will be highly
reective, while soft materials, such as unpolished wood, drapes,
and carpet, will be highly absorptive.
The hardness and porousness of a given surface will also affect the
frequencies that will most readily be absorbed or reected. If a room
is designed that absorbs higher frequencies and reects lower frequen-
cies, then the result will be a muddy- or boomy-sounding room, since
only the low frequencies would be highlighted. Conversely, if only the
high frequencies were reected because the lower frequencies were
being absorbed, the room would sound edgy or brittle. Ideally, a neu-
tral room is the result of a good balance between materials that absorb
low and high frequencies.
Reections, also called early reections or direct reections, are the
rst sounds to travel to the observer in a straight unobstructed line
(see Figure 2.10).
A reection will tell the subconscious mind how close the nearest sur-
face is. A fast reection will indicate a surface nearby, while a long
reection will delineate a surface further away. The mind will translate
this information to inform us about the size of the room and, more
specically, the distance between the closest wall and the observer.
39
A short rst reection tells us we are in a small room, such as a garage,
a basement, or an intimate club, while a long rst reection tells us of
a concert hall or a large club. Room reections have another interest-
ing characteristic; they will combine with the direct sound from the
sound source and color, or change, the timbre and waveform of the
original sound, thereby adding to the perceptual puzzle we are creating
and interpreting.
An echo is a wave that has been reected with sufcient magnitude
and delay to be detected as a wave distinct from that which was
directly transmitted. In order to be perceived as an echo, it must reect
at least 80 milliseconds behind the original wave to distinguish itself.
Although an echo is a form of reection, the difference is our percep-
tual ability to distinguish it from the original sound, while a typical
reection will blend in with the original sound and not be distinct.
In other words, an echo is an individual, discrete repetition of a sound.
Reverberation is a series of random reections, rst increasing in
amplitude, then gradually decreasing in amplitude and spaced so
closely together that they are indistinguishable as separate reections
(see Figures 2.11 and 2.12).
In other words, reverberation, or reverb, occurs when reections
become too dense to distinguish from each other and the original,
and diminish in amplitude over time. Reverb is one of the cues we
use to judge the size of the room we are in. The greater the reverb
Figure 2.10 The rst sound to arrive at the listener after the source is the rst reec-
tion. Based on original diagram courtesy of Tannoy.
40
time is in relation to the sound source, the further we are from it.
When the reverb lasts a long time, we judge the room to be large.
Reverb can be added to any sound to give it depth and dimension
and to give the listener a sense of the size of the room in which the
sound was produced. Once we start getting creative with our mixes,
we will select specic reverbs to enhance perceptual rooms that we
create. If we want the artist to sound as if he or she is playing in Car-
negie Hall, we will apply a reverb with a long rst reection and a long
reverb time. If we imagine the artist is playing in a garage, we will use
shorter rst reections and reverb times. The combinations of the orig-
inal sound source, the early reections, and the reverb will evoke an
Figure 2.11 Following the rst reection, the sound waves bounce randomly off every
reective surface. This is reverberation. Based on original diagram courtesy of Tannoy.
Figure 2.12 An aerial view of random reections known as reverberation. Based on
original diagram courtesy of Tannoy.
41
image in the mind of the nal listener of a space in which the band is
playing. This space is entirely at the audio professionals discretion.
RT-60 is a method we use to assess the length of the reverb and there-
fore the size of the room. It stands for reverb time minus 60 decibels.
RT-60 measures the amount of time that passes while the sound pres-
sure level drops 60 dB after the reverb has begun. Larger rooms tend
to have longer RT-60s, while smaller rooms typically have shorter
RT-60s.
Fletcher Munson Equal Loudness Contours
We mentioned Fletcher Munson in the previous chapter. Since this is
an important concept in perception, lets take a closer look at the
details. The Fletcher Munson Equal Loudness Contours demonstrate
that there are differences between how we interpret a sound event and
the objective measurement of the energy in that event. Because our
interest lies with the perception of sound, these contours will affect
us dramatically as we place sounds in an environment.
Fletcher Munson states the following:
1. The human ear is not equally sensitive to all of the frequencies
within its range.
2. The degree to which the human ear favors some frequencies
over others changes when the listening level is altered.
3. The discrepancies in the ears sensitivity are most pronounced
at lower listening levels.
4. The discrepancies in the ears sensitivity are least pronounced
at higher listening levels.
5. Overall, the ear is most sensitive to mid-range frequencies and
least sensitive to low frequencies. The sensitivity to high fre-
quencies falls in between its sensitivity to low and mid-range
frequencies.
The implications of these facts are that if listening levels remain con-
stant and the perceived pitch is altered, the loudness will appear to
42
change, and that if the listening level is altered the perceived balance
between the levels of various frequencies will change. We can therefore
extrapolate that balances established for sound cues, for production,
or in music mixes must be monitored at a variety of levels to ensure
that they will be effective to a music consumer or audience member at
any loudness level.
Fletcher-Munson tells us that if a music mix is made while the engi-
neer monitors at high levels throughout, the mix will appear to be
midrange heavy and lack low frequencies when played back at lower
listening levels. Conversely, if a mix is done entirely at low listening
levels, it will be bass heavy when played back at high levels. With
theatre cues this is also a factor, because the director often chooses
to have cues played back at a different level than the sound designer
originally intended. To be sure of mixes and cues, it is sometimes
necessary to make concessions at certain listening levels. Keeping
our observers in mind (the audience or the purchaser of a CD), it
is better to produce a mix or a cue that will be effective though
imperfect at any level than one that is excellent at one level and ter-
rible at all others.
The Behavior of Sound Waves
While some of our perceptions of sound waves result from the physiol-
ogy of the body and the interplay between the different ways we per-
ceive sound, as mentioned earlier, other perceptions result from the
physical behavior of sound waves. It is critical to understand the man-
ner in which sound waves behave under any given set of circumstances.
The directionality of a sound wave depends on its frequency content.
Higher frequencies are more directional, while lower frequencies are
more dispersal. Simply put, high frequencies move in a straight line,
while low frequencies spread out. This is why AM radio signals, which
are broadcast at a lower frequency, seem to travel farther than FM
radio signals (see Figure 2.13).
The lower frequency AM signals are able to bend and hug the Earth
due to their longer wavelengths, while higher frequency FM signals
43
travel in a more direct path due to their shorter wavelengths, sending
them into the atmosphere and beyond our range more quickly.
Diffusion is the spreading out of a sound. Commonly, due to the phys-
ical properties of lower frequencies having larger wavelengths, low fre-
quencies diffuse while high frequencies are directional. Another
property of low-frequency sound waves is diffraction. This allows
low-frequency sounds to bend around corners more readily than
high frequencies.
These properties of low-frequency sounds have several implications. If
you walk into one end of an L-shaped bar with a band playing at the
other end, you will hear mostly bass. This is one of the many chal-
lenges faced by sound reinforcement professionals. Will you be able
to make the band sound as good to those listeners who are walking
in or standing at the bar as you can to the observers sitting at tables in
front of the stage? While this must be our goal, once again concessions
must be made on occasion to offer the best nal product to the greatest
number of observers.
FM Signal
AM Signal
Figure 2.13 The higher frequency and shorter wavelengths of FM radio waves causes
them to be more directional, while the larger wavelengths of the lower frequency
AM radio waves allows them to hug the Earth and travel much farther.
44
Also, the persistent spreading out of low frequencies partially explains
why, when your upstairs neighbor blasts his stereo, you only hear the
booming of the bass and drums. This is also the result of higher fre-
quencies losing energy more quickly as they change media, from the
air, to the wall or oor, then back into the air. Due to their larger
wavelengths, low frequencies will retain more of their energy as they
change media than the higher frequencies with their shorter
wavelengths.
When two or more sound waves combine, the result will be either con-
structive interference, destructive interference, or both. Simply put,
when sound waves meet they will alter each other in one of three
ways. Constructive interference occurs when two sound waves com-
bine and the result, referred to as the sum wave, is an increase in ampli-
tude of the sound waves (see Figures 2.14 and 2.15).
Constructive Interference
+ =
Figure 2.14 Two sine waves of the same frequency and amplitude, perfectly in phase,
will increase the amplitude.
Destructive Interference
+ =
Figure 2.15 Two sine waves of the same frequency and amplitude, 180 degrees out of
phase, will result in total cancellation. Destructive interference is commonly subtler,
resulting in a reduction of amplitude. Total cancellation is infrequent, but does occur
on occasion.
45
Destructive interference occurs when two sound waves combine and
the result is a decrease in the amplitude of the sound wave. The phase
relationship between the waves is crucial in determining whether they
will combine constructively or destructively. If both waves are in a
compression or rarefaction stage at the same time, then they will
tend toward constructive interference because they are pushing in
the same direction. If they are in different phases and are pushing
against each other, they will tend to lose energy and decrease the
amplitude (destructive interference). When two people push a car in
the same direction it will go faster. This is constructive interference.
If one person is pushing the front of the car and the other is pushing
the back, it either wont move or it wont move much. This is destruc-
tive interference. Another good way to envision and possibly experi-
ence destructive interference is with the help of a pair of stereo
speakers. When wired properly, both speakers will push sound at
the same timeparticularly those sounds that are located in the center
of the stereo eld. If one were to switch the wires on one of the
speakers (wired out of phase), one speaker would be pulling in
while the other was pushing out. Although this will have little effect
on sounds that are only on the right or the left of the stereo mix,
sound that is centered will be greatly reduced or entirely removed
from the mix. If the sound is completely removed, we are experiencing
total cancellation.
The phase relationship between two sound waves is determined by a
comparison between the point each waveform is at in its period when
they meet. Phase relationship can be viewed as a comparison between
the number of degrees each waveform has traveled in its cycle at the
time that they encounter each other. Each cycle of a sound wave, like a
circle, is divided into 360 degrees. The position of a waveform at any
given moment is measured by the number of degrees it has traveled
through one cycle (see Figure 2.16).
Another behavior of sound waves that can impact us greatly is masking,
which occurs when one sound covers up another. For example, it
becomes difcult to continue a conversation when a bus or a truck
passes by. This occurs because the louder sound will cover up, or
46
mask, the sound of your voice. Masking is at its most extreme when
frequency, timbre, and location are similar. It is easier, for instance, to
distinguish between the sound of a trumpet and a violin when they are
playing the same song than it is to distinguish between two trumpets or
two violins playing the same melody. If two of the same instruments
were playing the same melody in different octaves, it would be easier
to distinguish between them, since the frequencies are different. Also,
if we were to put the two violinists in different corners of the room, it
would become easier to distinguish between them. If we can change
any one of these factorsfrequency, timbre, or locationwe will
have a much simpler time hearing them as separate, and masking
will be reduced.
Masking comes into play in several ways. As an engineer it may be
optimal to have all instruments fully audible, but this will lead to
some instruments masking others. Part of the engineers task in the
mix will be to make all instruments audible without any covering up
another. As a sound designer, a cue that is full of white noise, such as
rain or wind, will mask the actors voices. This would be unacceptable
to the director and provide an inadequate experience to the audience.
There are a variety of techniques at our disposal to deal with masking,
all of which will be explored in Chapter 9.
The nal behavior pattern of sounds waves of which we must be aware
is the standing wave. When a sound wave traveling within an enclosed
space, such as a room or a speaker cabinet, encounters one of the sur-
faces of the enclosure, some of its energy will be absorbed and some
will be reected. If some of the energy travels back along the same path
as the direct energy, which happens between parallel surfaces, it will
interfere with the incoming sound wave. This causes increases and
0 90 180 270 360
Figure 2.16 The degrees of phase.
47
decreases in amplitude, depending on the phase relationship between
the two waves.
Frequencies with wavelengths that are whole-number multiples or
subdivisions of the distance between the surfaces off which they
are reecting will interfere with each other in such a way as to
cause increases and decreases in amplitude at specic locations
within the enclosure. These xed high-pressure (antinodes) and
low-pressure (nodes) locations within the enclosure, or room, are
the compressions and rarefactions that form a stationary waveform
called a standing wave.
Standing waves are important to the audio professional because they
affect the perception of loudness of select frequencies at specic loca-
tions within the control room, studio, or theatre. The acoustics of a
room can alter the perceived volume of a sound leaving the speaker,
as it interferes constructively or destructively with standing waves in
the room. The result is that two listeners, positioned in different loca-
tions within the room, may be hearing something completely different,
and for at least one of them the experience is sure to be unpleasant.
The best way to avoid standing waves is to avoid parallel surfaces in
any audio environment. If the parallel surfaces are already there, cur-
tains or sculpted ceiling tile or absorptive materials of any type may be
enough to break up the standing waves. To complicate things further,
standing waves can occur between the diaphragm of a mic and the
head of a drum, or the angled surface of a console and an identically
angled ceiling in the control room in which the console has been
installed. Always try to be aware of parallel surfaces in production
rooms; they will bring you grief if unattended. Now that we have
the building blocks of aesthetics and technology, lets apply this
knowledge to gear and the production rooms in which our equipment
will be housed.
Exercises
1. Walk around and listen to different environments. As you
move through each environment, observe the sounds you hear
as a combination of the ve perceptions of sound. Observe the
48
layers of sounds. Listen to them in terms of loudness. What is
the loudest sound you hear? What is the softest sound? What is
in between? Repeat for the other four perceptions of sound.
Try this in different environmentsindoors, outdoors, on a
road, in the woods, in a theatre, and so on.
2. Find or create a standing wave. Look for parallel surfaces, such
as a desktop and a ceiling or the top and bottom of a deep
window. Clap your hands in that spot. Listen for frequency
distortion. Adjust the frequency of your clap as you listen. Does
the standing wave change or disappear using different
frequencies?
3. Construct a wave resonator by attaching hacksaw blades of
varying lengths securely to the edge of a block of wood,
attaching a power drill to the block, and running it at varying
speeds. The drill will generate the resonant frequency of each of
the attached blades, showing that varying masses possess
varying resonant frequencies. Use a piece of foam pipe insula-
tion under the board both to increase the effect of the reso-
nance and to make sure the ailing blades do not hit the table
on which the unit sits.
4. Create an imitation Shive Wave Machine using about 5 to
6 feet of 1/4-inch audio tape or ribbon, suspended from the
ceiling, with straws attached horizontally at even intervals.
Transverse waves can be demonstrated by the motion of the
straws when twisted at the bottom, and longitudinal waves can
be observed by icking the bottom.
Additional Reading
Alten, Stanley R. Audio in Media, 7th ed. Belmont, CA: Wadsworth,
2004.
Backus, John. The Acoustical Foundations of Music, 2nd ed. New
York: W. W. Norton, 1977.
Campbell, Murray and Clive Greated. The Musicians Guide to
Acoustics. London: Oxford University Press, 2001.
49
Hutchins, Carleen Maley. The Physics of Music. San Francisco: W. H.
Freeman, 1978.
Katz, Bob. Mastering Audio: The Art and the Science. Burlington, MA:
Focal Press, 2002.
Olson, Harry. Music, Physics, and Engineering. New York: Dover,
1967.
Pierce, John. The Science of Musical Sound, Revised ed. New York:
W. H. Freeman & Company, 1992.
Rossing, Thomas. The Science of Sound, 3rd ed. Reading, MA:
Addison-Wesley, 2001.
Rumsey, Francis. Stereo Sound for Television. London: Focal Press,
1989.
Winckel, Fritz. Music, Sound, and Sensation: A Modern Exposition.
New York: Dover, 1967.
50
PART
II
Understanding Audio
Equipment
51
3
The Production Room
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t
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i
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i
c
.
53
N
ow that we have an idea of the theories underlying audio, we
can begin to apply that knowledge. We have already estab-
lished a connection between aesthetics and technology. We
have explored some of the technical realities of sound waves, and
we have looked at some of the relationships between those realities
and our perceptions of sound. Now lets begin to examine the types
of rooms where we can apply these ideas, and the equipment they con-
tain. This is where being a gear-head comes in handy, but well start
with the basics.
When discussing a commercial recording studio, a theatre booth, a post-
production facility, or a radio station, we generally have a picture in our
minds of the facility, what it looks like, and what happens there. All these
rooms, along with voice-over studios, foley rooms, edit rooms, home
studios, digital workstations, and broadcast production facilities, to
name a few, are forms of production rooms. As audio professionals, we
will spend a great deal of time in production rooms, so lets dene the
characteristics of a typical production room and explore the different
equipment used, how it all connects, and how we determine a path for
the signal.
What Is a Production Room?
A properly designed production room is a laboratory designed to accu-
rately capture, store, process, route, and reproduce audio information.
Not every production room will facilitate every one of these processes.
In the sound booth of a theatre, for instance, it may not be necessary to
store or record a performance. In the type of production room we call
a theatre, we will still capture information with a microphone and
route that signal through a console. We may even process the signal
from the microphone and add it to previously recorded sound cues, or
sound effects, and then we will reproduce that combination of signals
in the theatre for the audience (see Figure 3.1).
The signal may also be routed to the green room to cue actors or to
the conductor of the orchestra in an orchestra pit in a theatre situation.
In sound reinforcement, audio will also be routed to stage monitors so
54
Figure 3.1 Signal ow in live sound, whether sound reinforcement or stage sound, is
similar. Diagram courtesy of Mackie.
Chapter 3 The Producti on Room
55
the band can hear each other and possibly to a truck in the parking lot
for remote recording or to provide a satellite uplink. All of these live
situations share a primary goal of providing quality audio for the audi-
ence while serving the needs of actors and/or musicians to help them
provide the best performance possible.
In a multitrack situation in a recording studio, the criteria are different
(see Figure 3.2). Here the recording engineers primary goal is captur-
ing the musicians best performances with the highest possible level of
quality. To succeed at this, the musicians need to be comfortable and
at ease, they need to hear themselves and each other well, and they
need to have good communication between themselves and their pro-
ducer. While we consider our primary goal to be a clean, excellent
recording, it is just as important that the musicians have a good mix
in their headphones in order to provide us with the best possible
performance.
When we record live-to-two-track the routing changes, particularly of
our outputs. The criteria of excellent communications, good head-
phone mixes, and a clean recording of an excellent performance
remain the same (see Figure 3.3).
When recording to hard disc our regard for the importance of quality
and communications remains the same; however, other criteria change
our signal routing drastically. When recording digitally, one of the
most important aspects of the signals path we are focused on is the
avoidance of repeated conversions between the analog elements and
the digital elements in our signal chain because repeated analog-to-
digital (AD) conversions and digital-to-analog (DA) conversions can
reduce our signal quality severely, preventing us from attaining our
goal of a quality recording (see Figure 3.4).
Similarly, a radio station may not need to record its signals, but it will
certainly capture and reproduce sounds and route them to a transmitter
all functions of a production room. It is also helpful for a production
room to be aesthetically pleasing simply because we will spend so much
time there, and a comfortable environment housing ourselves and our
gear will lend itself to creativity and efciency.
56
Figure 3.2 Typical signal ow in a recording studio. Diagram courtesy of Mackie.
57
Figure 3.3 We observe a different signal ow for live-to-two track recording. Diagram
courtesy of Mackie.
58
Figure 3.4 Hard disc recording has its own considerations, such as keeping the signal in
the digital domain for as long as possible to avoid excessive conversions between analog
and digital. Diagram courtesy of Mackie.
59
Key Components of a Production Room
It will be helpful for us to understand the various types of equipment
available to us as we work with audio. We will go into more detail
about all of these bits of gear as we move forward. First, an overview
of how these different pieces of equipment operate and interface is in
order. Energy is regularly converted from one form to another in a
production room in order to perform the operations we discussed in
the last section. These tasks are performed by microphones, tape
recorders, hard drives, and speakers, all of which contain transducers.
A transducer is a device that changes energy from one form to another,
the same way a light bulb changes electrical energy into light and heat,
or how a computer keyboard converts the kinetic energy in your n-
gers into electrical impulses that your computer can understand and
process. (See Figures 3.5 through 3.7.)
Changing energy fromone formto another is essential for performing the
various tasks requiredtocapture, store, andreproduce soundevents. Here
is another way of looking at this: We start with a sound wave, a form of
acoustic energy. We need that acoustic energy to become electricity in
order to manipulate it and send it to a tape machine or hard drive,
where it must become magnetismto be stored. We are converting acoustic
energy into electricity and then magnetic energy to record. Then, when we
play back, we convert the magnetic energy we have stored on tape or on a
Figure 3.5 Transducers convert energy from one form to another. A light bulb converts
electricity to heat and light.
60
hard drive to electricity, send that through an amplier to our speakers,
and convert it once again, this time into acoustic energy that we can hear.
Because the sound wave that hits our ear is converted in our brain into an
electro-chemical reaction, one might say that we are also transducers.
As noted in Table 3.1, examples of transducers in the production room
include microphones, which change the acoustic energy of the vibrating
air molecules into electrical energy; the record head of a storage device,
such as a tape recorder or hard drive, which converts the electrical energy
the microphone has generated into magnetic energy for storage; the play-
back head of a storage device, such as a hard drive or a tape machine,
which changes the magnetic energy stored into electrical energy; and the
Figure 3.6 A microphone is a transducer that converts acoustic energy to electrical
energy, or electricity, so we can send it wherever we wish. Photo courtesy of Shure.
Figure 3.7 A speaker is a transducer that returns electricity to acoustic energy so we
can hear it. Photo courtesy of Electro-Voice.
61
speakers, which convert electrical energy into acoustic energy that we
can hear. Without the help of transducers, our current methods of
recording and reproducing sound events would be impossible.
Lets begin with the oldest of our transducers, the microphone, invented
by Emile Berliner for use in Alexander Graham Bells telephone. While
they have come a long way from the old carbon microphones used in
early telephones, the primary function of the microphone has not
changed; it is still the tool we use to capture a sound event. As men-
tioned, microphones are transducers that convert acoustic energy into
electrical energy. The electrical energy output from a microphone may
be routed by a console to a recorder for storage, it may be routed to an
amplier and speakers to be reproduced, or it may be routed to both
places. The range of available microphones is wide, and each has its
own characteristics, so using different microphones to capture the
same sound will result in a noticeable difference. Choosing the micro-
phone that has the right characteristics for a particular application is
critical to the audio professional. Having a wide variety of microphones
to choose from increases the ability of the audio professional to capture
the sound event well and adds to the versatility of the facility and the
quality of the recording or production. We will discuss different micro-
phone types and applications in great detail in Chapter 5, which is
appropriately named Microphones.
The audio console is responsible for the processing and routing of most,
if not all, signals and is the center of any production room. The console
Table 3.1 Examples of Transducers
Common Transducers Energy Conversion
Microphones Acoustic - Electric
Speakers Electric - Acoustic
Record Heads Electric - Magnetic
Playback Heads Magnetic - Electric
Hard Drives (Writing) Electric - Magnetic
Hard Drives (Reading) Magnetic - Electric
62
is the link between all devices in the facility and the piece of equipment
where we determine our signals path (see Figures 3.8 and 3.9).
The microphones, recorders, monitoring system, and signal processing
equipment all connect to the console. The console will often simulta-
neously perform many tasks, and the exibility and versatility of this
device is often the hallmark of a quality facility. In non-studio settings,
such as lm shoots or theatres, the console should still provide the
recordist, sound designer, or board operator with the greatest possible
exibility and many options to achieve the same goal.
Storage devices such as multitrack analog, digital reel-to-reel, hard drives,
digital audio tape (DATs), and even consumer items such as cassette
Figure 3.8 Oneof the most sought-after consoles, the Neve 8068. Photocourtesy of Neve.
Figure 3.9 A Neve Kelso. This one was autographed by Rupert Neve. Photo by Gary
Gottlieb.
63
recorders convert an electrical signal froma console or microphone into a
magnetic eld, which is imprinted onto tape or disc as organized mag-
netic domains for storage and reproduction (see Figures 3.10 and 3.11).
Figure 3.10 A Studer A807 analog two-track tape machine. Photo by Gary Gottlieb.
Figure 3.11 ARevox PR99 MKIII analog two-track tape machine. Photo by Gary Gottlieb.
64
When reproducing stored information, the recorder converts the mag-
netic eld held by the tape or disc back into an electrical signal, which
is then routed to the console or directly to the amps and speakers.
Tape, discs, and hard drives are all forms of magnetic storage devices.
While both analog and digital tape recorders use tape, computers use
drives and discs to store and reproduce audio information in the form
of a magnetically imprinted algorithm. If tape can be thought of as
spaghetti, discs are pancakes, attened, circular, and enclosed. Com-
puters are popular for many audio applications, as control surfaces
and signal processors as well as recorders, and the audio information
resulting from the computing processboth manipulation of the signal
and storage of the signalis stored in a digital format on a magnetic
drive, available to be reproduced when needed.
The Performance Space
The studio or vocal booth is the performance area of an acoustically con-
trolledproductionfacility. Thepurposeof this roomis toprovideanacous-
ticallyisolatedandsonicallyneutral environment sothat the performances
within will be devoid of interference from outside sounds or unwanted
coloration from the rooms acoustic character. In addition to keeping
the room sounding as good as possible and keeping unwanted sound out
of the room, isolation keeps the desired sound in the room. This can be
particularly benecial in a home studio or other facility where annoying
your neighbors could be detrimental to your business continued success.
Consider isolation as offering a double bonus: You are improving your
sound quality while being kind to those around you (see Figure 3.12).
Sound is frequently recorded in non-controlled environments, such as
on a lm shoot, at a sporting event, during news interviews, or while
gathering sound effects. Each of these situations generates a different
set of problems. Due to the lack of isolation, noise, which we can dene
as the stuff we do not want to hear, can become a serious issue. Consider
the soundtrack of a lm. If the soundtrack had more camera noise than
actors voices, it would be unacceptable. The alternative for a lmmaker
faced with this situation (re-recording all dialogue on an isolated sound
stage) is expensive and time consuming, and it ensures that the director
will hire a different audio professional next time. The ideal situation is
65
that the recordist gets it right the rst time by selecting the best possible
microphones and learning the subtlety of angling them properly, making
the voices usable. Similarly, crowd noise at a sporting event or wind
noise while gathering sound effects detracts from the nal product,
potentially making the audio unacceptable and unusable. Lacking the
benet of isolation, a great deal of care and forethought regarding micro-
phone selection and positioning will positively impact the nal product.
Lets return to our isolated room. Ideally, when designing a studio, sev-
eral criteria are used to provide isolation. The walls, oors, and ceiling of
the roomare of proper design and sufcient mass to block out sound from
outside the room. Techniques used to achieve this include double walls,
where the dead air space between the walls causes the sound waves to
change media several times, losing energy in the process and reducing
the sound transference greatly. Another good technique involves sus-
pended or oating rooms, where the studio is uncoupled from the rest
of the building so that vibrations traveling through the building, such
as those produced by trucks passing by, will not affect the sound the
microphones are capturing. Along with isolation, other criteria used to
create the ideal studio include using no parallel surfaces, which avoids the
Figure 3.12 Clinton Recordings Studio A in New York, one of the best designed track-
ing rooms on the East coast, can accommodate up to 85 musicians. Photo courtesy of
Clinton Recording, New York.
66
creation of standing waves, and constructing all surfaces of a blend of
acoustically sound materials to ensure acoustically neutral reections of
sound waves. An ideal room will have a mixture of reective surfaces
(glass, polished wood) and absorptive materials (drapes, carpet) to pro-
vide an end result of a room that is neither too live nor too dead. Many
rooms will strive to have one part of the performance space a little more
live, or reective, while another part of the room will be more dead, or
absorptive. This gives the audio professional more options when using
the room to accent the original sound (see Figure 3.13).
When recording in non-controlled environments, care should be taken
whenever possible to fulll the same criteria: Avoid parallel surfaces
and try to record in an area that is sheltered from wind and other extra-
neous noise. If the situation allows, care should also be taken to record in
the most neutral environment possiblesometimes moving the subject(s)
a few feet one way or the other will greatly reduce reections, resulting
in a more natural and acceptable sound during a lm shoot, sound
effects creation or gathering, or an interview for broadcast.
Figure 3.13 Clinton Recordings oating tracking room, showing an iso booth on the
left. Photo courtesy of Clinton Recording, New York.
67
The Control Room
The control room is the heart of the production facility. Along with
housing the console, tape recorders, signal processing gear, and the
monitor system, it is the location where the audio professional deter-
mines the signal routing (see Figure 3.14).
Exactly like the studio, the acoustics of the control room are critical.
The audio professional must be able to believe his or her ears in the
control room. He or she must be condent in the accuracy of the sounds
that have been routed to the monitors. The acoustics of the room, there-
fore, should neither add to nor subtract from the sound leaving the
speakers. The same signal that is routed to the monitor speakers is
also frequently routed to a recorder for mixing. If this recording is
used to create CDs that consumers will buy, we need to be sure that
what we are hearing is an accurate reproduction of the sounds that
are recorded and an accurate forecast of what the consumer will hear.
Figure 3.14 Video Post & Transfer in Dallas is a ne example of a thoughtfully
designed control room, featuring a Solid State Logic OmniMix console. Photo courtesy
of Solid State Logic.
68
This will only be true if the acoustics of the control room are not color-
ing the sound leaving the speakers.
It is possible for the listening environment to color the sound leaving the
monitor to such a degree that the sound in the control room no longer
matches the character of the recording. This could result in nasty sur-
prises when the end user listens to the record or CD made from a master
created in the inadequate room we just described. The same basic rules
listed for the acoustics of the studio apply to the acoustics of the control
room. Double-wall construction and oating rooms are typical in high-
end control rooms. In addition to the criteria listed for creating a son-
ically neutral environment, the placement of the audio equipment in the
room will also play a major role in shaping the acoustic character of the
control room. It is just as easy to create a standing wave between a con-
sole and a ceiling that are parallel as it is to create one between two
parallel walls. In the non-controlled audio environment, it is rarely prac-
tical to set up a neutral control room in the eld. The same rules still
apply, however, and it is useful for the recordist to monitor on head-
phones with which he or she is familiar and trusts, in an environment
that neither adds to nor subtracts from the sound. This room should
also be aesthetically pleasing because, as an audio professional, you
hope to spend most of your waking hours (and possibly some of your
sleeping hours) here.
Consoles
When the average person thinks of an audio professional, he or she
thinks of someone operating a mixing console. The images from
music videos, lms about music, and magazines showing audio profes-
sionals usually position the engineer in a comfortable chair behind a
large console with lots of knobs, buttons, and faders. The mixing con-
sole is the center of the production room and the heart of the audio
process. At some point in the production process, all signals will pass
through the console. The consoles primary functions are the routing
and processing of input and output signal. The console is not a trans-
ducer; the signals remain electricity throughout. However, a good con-
sole can be your most useful tool and can accomplish amazing tasks on
your behalf.
69
A professional console has the ability to simultaneously route numerous
input signals to a wide variety of devices and locations. Multiple acous-
tic and electronic signal sources, such as instruments, reverb, and equal-
ization devices, may be simultaneously routed to signal processors,
while the sum of all the input signals are being sent to multitrack and
two-track recorders, satellite feeds, concert or theatre stages, and radio
transmitters. Audio consoles, although capable of performing complex
tasks, are relatively simple devices. The ability of a console to perform
complex tasks stems from redundancyin other words, there are many
modules containing the same simple devices, over and over again. If
you understand one module or strip in a console, you pretty much
understand them all, thanks to redundancy (see Figure 3.15).
Figure 3.15 The repetition or redundancy of modules makes it easy to understand
even the most complex consoles. Photo courtesy of Neve.
70
The versatility and exibility of a console, more than the number of
modules, determines its level of professionalism and usefulness. While
a simple DJ mixer will have very few options in terms of signal routing
within the console, ways to send out of the console, and other ways to
manipulate signal, a state-of-the-art console will give the audio profes-
sional many options for signal routing and signal ow, both within the
console and beyond. The option to shift the sequence of effects in the
console, as well as different ways to bring signal in and then send it
out, allow the engineer to be more creative and efcient. When the
audio professional runs out of ways to manipulate signal within the
console, it is time to step out beyond the console. This is when a
well-laid-out patch bay comes in handy.
Patch Bay
When an audio professional wishes to take a signal and send it beyond
the console to outboard signal processing gear, ampliers, or recording
devices that are not hardwired to the stereo bus, patch bays are com-
monly used (see Figure 3.16).
Patch bays are access points that can interrupt the normal signal ow
through a module and give access to the inputs and outputs of every
device in a production facility. A patch bay will also provide access to
signal ow between normally interconnected devices. The purpose of
the patch bay is to allow for the rerouting of the normal signal ow
and the insertion of additional devices. Patch bays are commonly used
for inserting compressors, expanders, noise gates, reverb, outboard
preamps, and outboard equalizers (see Figures 3.17 and 3.18).
Patch bays are also used to reroute the signal around defective gear
and interconnect rooms through tie lines. When making tape or disc
copies, the patch bay is used to create the shortest possible signal
path, thus limiting the possible introduction of additional noise
between two decks and maintaining quality. Patch bays are standard
devices in a production facility, and despite their simplicity they
add tremendous exibility by increasing the engineers signal ow
options.
71
Figure 3.16 While patch bays may seem complicated at rst glance, they are actually
simple to use and increase a production rooms exibility tremendously. Photo courtesy
of Neutrik.
72
Patch bays are a series of female jacks connected to wires that lead to
the inputs and outputs of various components in the console and the
production room. This allows a device to be inserted or the normal
signal ow to be rerouted. This is done by taking the wire that
Send
Master
Trim
FX
Return
or
Line Input
Send
Channel
Trim
Post
Fader
Signal
Fader
Master
Fader
Speaker
Volume
Amp
2 Track
Monitor
Speakers
Equalizer
1 2
3
Figure 3.17 Patch bays are commonly used to add effects to the signal path.
Line
Input
Mic
Input
Fader
to
Multi
Mon
Fader
Track
Select
Multitrack
Master
Fader
1 2
4
Trim/
Preamp
EQ
Compressor
Compressor
5
Figure 3.18 Patch bays can also be used to add compressors to the signal path.
73
leads between two devices, such as the multitrack recorder and the
consoles line inputs, cutting the wire in two, and connecting female
jacks to each end. One jack leads to the input of the console, and
the other leads to the output of the multitrack. They would be labeled
console line inputs, channel line inputs or line in and multitrack out-
puts, or tape outputs, respectively. The patch bay is made of a series of
these connections, allowing access to the signal ow between the devi-
ces in the studio. Applications for the illustrated patch points (channel
line in) include inserting noise gates or compressors and the direct
input of electronic instruments through the patch bay and into the
console.
Patch bays are normally output over input, where the output of a track
of the multitrack owing to its corresponding module in the console
would be interrupted, hence the output of Track 1 would be above the
input to the line preamp of Module 1. With signal processing devices
that are not normalled to anything on the console, typically the output
of a given device is directly below its input; however, when the output
of a device normally feeds the input of another device, the output will
be on top of that corresponding input, as noted with the example of
Track 1 feeding Module 1. As explained, when we create a patch bay,
the wire between two devices is cut and female jacks are connected to
each exposed end, which gives us access to the input and output of all
our devices. This also cuts off the normal signal ow between the devi-
ces. To correct this and allow signal to ow normally if we choose not
to insert a patch cord, the jacks are bridged with a connecting wire.
Two different types of jacks therefore must be used in patch bays. The
ones used for the output sides, known as half normals, allow the signal
to ow through the bridging wire whether or not a patch cord is
inserted. The other jacks, connected to the input sides and known as
normals or full normals, disconnect the bridging wire when a patch
cord is inserted. This is necessary since input signals cannot be com-
bined by simply jamming them together.
Although output signals can be split without signicant signal loss,
input signals need to be combined through an active summing net-
work, which we will discuss in more detail in Chapter 4, Consoles.
74
Breaking the normal on an input signal assures the engineer that input
signals will not be combined in the patch bay. To illustrate that input
signals cannot simply be passively combined, let us consider a plumb-
ing example. What would happen if everyone in an apartment building
were to ush his or her toilet simultaneously, the equivalent of too
much input? The output of the system would be too much for the
drain to handle, and the drain would back up into the apartments
on the lower oors. Conversely, if more drains were added (analogous
to splitting the signals), there would be no loss to the system. In other
words, if we had a mono signal and wanted to record it across two
stereo tracks, splitting that signal would not present a problem. On
the other hand, to take a stereo cassette (two tracks) and dub it to
mono (one track), we must bring it up on two modules of the console.
Once there, summing networks are available to combine the signals
effectively; otherwise, we would experience signal overload by trying
to just jam these signals together.
The term normals, or full normals, is used with two different mean-
ings. As explained earlier, a patch point where the normal signal ow
is interrupted by inserting a plug is called a full normal. The inputs to
all devices from the patch bay are full normalled. The term normal also
applies when one device is hard-wired to another, as is the case with
the output of the multitrack recorder and the line input of the console.
Because of the way they are wired, the normal signal ow is from the
output of the multitrack to the consoles line input; hence we say that
the multitracks output is normalled to the line inputs of the console.
Signal From
Multitrack One
Output Patch
Point One
Input Patch
Point One
Uninterrupted
Signal Flow
Console
Line In One
Signal From
Multitrack Two
Output Patch
Point Two
Input Patch
Point Two
Interrupted
Signal Flow
Console
Line In Two
Figure 3.19 Signal ow is different in half normals, or output patch points (top),
where the normal ow continues to Console Line In 1 and full normals, or input
patch points (below), where the normal signal ow from Multitrack Track 2 must be
broken to be replaced by the output of Track 1.
75
There are words we will abuse through overuse far worse than this as
we go along, like bus and monitor. This repetition of terms in audio
may seem difcult, but all of these terms can be recognized by their
context. When an engineer asks an assistant engineer whether a parti-
cular reverb unit is normalled to the console, the question is regarding
hard-wiring, not different patch point options.
The outputs of devices wired to a patch bay are wired half normal.
When a plug is inserted into a half normalled patch point, the signal
can be thought of as splitting into two directions. One path follows the
wire bridging the two female connectors (the normal). The other signal
path follows the patch cord, or wire that has been inserted into the
jack. In other words, in addition to sending the signal to another loca-
tion, an output signal also continues along its normal path. This can be
handy when an engineer wants to truly split a signal and have it con-
tinue along its normal route while also sending it elsewhere.
Another common feature of a patch bay is a tool called a mult, short for
multiple. A mult is a number of female jacks wired in parallel. When an
output signal, such as the output of one side of a tape deck or CD
player, is introduced into any of the jacks comprising the mult, it will
appear as an output at all of the other jacks, thereby dividing the signal
into multiple outputs, hence the name mult. A mult will allow a single
signal to be routed to more than one location for simultaneous process-
ing. Mults are commonly used when making multiple tape or CD copies
and when sending a single signal to numerous locations. Only one out-
put signal can be put into a mult at once to avoid excessive level, as
noted in our plumbing example. A stereo signal, for instance, cannot
be combined into mono in a mult.
A more modern form of patching frequently seen in digital studios is
called electronic patching. Electronic patching is sometimes a feature
on a digital console or workstation, and sometimes mounted in an out-
board rack. It allows equipment to be patched without physically
inserting a patch cord. This is accomplished by pressing buttons rec-
ognized by an inboard computer. The computer then routes the signal
to the desired device. The advantages of electronic patching are its sim-
plicity and the clean signal path, since the computer will seek the most
76
direct route to the selected equipment. The disadvantage is that clients
who are accustomed to seeing hundreds of patch cords in a patch bay by
the 20th hour of a mix session will miss the experienceand of course
assistant engineers who had to write up the patch bay at the end of the
session in the old analog days, listing every single one of those patch
cords from source to output, will miss the experience as well.
Recording and Storage Devices
While hard disc recording formats have gained tremendous popularity,
tape formats, both analog and digital, still account for a signicant
amount of use in the business, particularly in larger recording studios
and post-production facilities. Despite the differences inherent in dig-
ital and analog tapes, there are still many similarities. This is due to the
physical reality that magnetic tape of some type is still receiving the
information to be stored by being transported past record heads of
some type, which creates a magnetic imprint.
Tape Transports
Lets start with the tape transports. The function of the transport sys-
tem is to load and unload tape, as well as to move the tape across the
heads during record and playback. The tape transport system contains
the following components and is, with few exceptions, common to
both digital and analog (see Figure 3.20).
First, the basics. There are three motors on a tape machine: the supply
motor, located beneath the supply reel on the right side of the machine;
the take-up motor, located below the take-up reel on the right side of
the machine; and the capstan motor, near the machines heads.
The supply and take-up motors maintain the proper tension across the
heads by pulling gently in opposite directions while in play and record
modes, which prevents the tape from de-reeling. While in fast wind
modes (fast forward and rewind), the motor pulling in the direction
in which the tape is going pulls forcefullyfor instance, in fast forward,
the take-up motor pulls fast, while the supply motor pulls gently in the
opposite direction. This provides the tension necessary to prevent the
77
tape from breaking, spilling, or de-reeling. Connected to each motor is a
shaft to which a reel can be attached by any one of a number of popular
reel-locking devices.
The capstan and pinch roller initiate and maintain the motion of the
tape transport during play and record functions, controlling the tape
speed. The capstan shaft rotates whenever the tape recorder is on and
the tape is fully loaded. When play is pressed, the pinch roller presses
against the capstan, causing the tape, which is in between the capstan
and the pinch roller, to be pulled along. As noted, while this is hap-
pening, the supply and take-up motors are pulling in opposite direc-
tions to ensure that the tape maintains proper contact with the heads.
There are also numerous pinch-rollerless transport systems, which are
sometimes referred to as capstan-less transport systems. The philoso-
phy of these manufacturers, such as Otari, is that the pinch roller and
capstan may damage the tape or decrease its life span due to excessive
handling. As an engineer who appreciates both the quality of the Otari
Supply Reel Turntable
Tacho Roller
Guide Roller Guide Roller
Supply Swing Arm Takeup Swing Arm
Tacho Roller
Takeup Reel Turntable
Figure 3.20 A typical tape path of a pinch-rollerless tape transport system. Based on a
diagram by Otari Corporation.
78
MTR-90 (pinch rollerless) and the Studer A80 and A800 (with pinch
rollers), I have never found the presence or absence of pinch rollers to
determine my preference.
Its common to nd a reel size control on professional recorders. This
control determines the amount of tension necessary to hold the tape
against the heads, based on the size and presumed weight of the reels.
The amount of tension needed will vary with the size of the reel on the
recorder. Many modern recorders do this automatically. Some two-
track recorders have a separate control for each motor. This allows
different reel sizes to be used simultaneously, although under normal
circumstances most manufacturers recommend using the same size reel
on both sides of a tape machine.
Along with the tension provided by the supply and take-up motors, the
tape guides also ensure that the tape maintains proper contact with the
tape head. Tape guides have a horizontal slot in which the tape is held
during all operations. This not only ensures proper contact with the
heads, it also helps the tape to wind properly and neatly on the reel.
The tension idler is a multipurpose tape guide. In addition to the func-
tions listed earlier, the tension idler acts as an on/off switch. When we
load tape onto the machine, we engage the tension idler, moving it
from a passive position to an active position. If the tape breaks or
runs out, the lack of tension will cause this control to move from its
active position. When this happens, all motors will be switched off,
preventing the tape from being further damaged.
As the name would indicate, the tape speed controls determine which
speed the tape will move in play and record. Professional tape speeds
include 30 inches per second (30 IPS), 15 IPS, and 7 1/2 IPS. Thirty IPS
gives the best signal-to-noise ratio and high-frequency response
because more tape is passing the heads during any sound event.
Much in the way that more pixels create a better picture, more mag-
netic particles create a better sound. Thirty IPS is used for most multi-
track recording and mix-down sessions. At this tape speed satisfactory
recordings may be made without the use of noise reduction. Fifteen IPS
is also commonly used, but the signal-to-noise ratio is 3 dB less than at
30 IPS. Multitracking and mixing at 15 IPS is often done with noise
79
reduction due to poorer signal-to-noise ratio. Producers and artists will
sometimes choose to record at 15 IPS when the cost of the tape is a con-
sideration. Although professional multitrack recording is never done at
7 1/2 IPS, radio stations that have not switched to hard disc and satellite
downlinks commonly run at this speed, and dubs and promotional mate-
rial sent to radio stations are therefore often done at 7 1/2 IPS.
Transport Controls
The transport controls are the buttons used to control the movement
of the tape (see Figure 3.21).
Although this sounds very simple, it requires some skill, particularly on
tape decks without full logic. There is the possibility of damaging tape by
using the transports incorrectly. Most modern decks will slow down as
they approach the end of the tape, after a fast wind and before stopping,
but many older decks will not, and a stop or play command at the wrong
time will result in a pile of tape on the oor or a broken master tape. To
avoid this, with older or vintage decks in particular, the tape should be
shuttled, rocking back and forth between fast forward and rewind until
the tape is going relatively slowly before attempting to press play or stop.
All Safe
1
Ready
2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24
Safe
Repro
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23
Sel-Rep
Input
24
All Input
Vari
Fix
Ext
Speed
Display
Pitch Control Cue
Lo
Hi
%
IPS
All Sel-Rep
All Repro
Individual
Record Play Stop Rwd F.Fwd
Figure 3.21 This diagram of the Otari 24-track remote shows the tape transport con-
trols (bottom) and the ready/safe switches at the top, with the mode selection options
for each track in between. Based on a diagram by Otari Corporation.
80
While most transport functions are fairly intuitive and similar to the
controls on your home cassette or CD player, some mention should be
made of the various transport controls. Play will cause the tape to
move across the heads at the selected speed. Stop will cause the tape
to cease moving, regardless of direction or speed. Fast forward and
rewind modes will cause the tape to move forward or backward at
10 to 60 times the chosen playing speed, depending on the particular
deck and the tape speed selected. These controls are used to move the
tape quickly in one direction. Most analog tape players have head lifters
that engage during fast wind, moving the tape back off the heads. This
is important because when we double the tape speed, we increase the
signal in frequency by one octave. At 60 times normal tape speed, the
high frequencies and amplitude produced could easily destroy moni-
tors. To avoid this, lower the volume while rewinding or fast forward-
ing if there are no head lifters. Head lifters also prevent excessive head
wear in fast wind. The record button engages the record function. Pro-
fessional tape recorders also have a ready/safe switch to prevent acci-
dental erasure of the tape if the record button is pressed at wrong time.
Most professional decks offer edit mode, in which the take-up reel will
not move, allowing the tape to dump off the supply reel. This mode is
most often used to spool off unwanted tape and will be discussed fur-
ther in Chapter 8, Editing.
Recorder Head Assembly
The head assembly of a recorder contains the components that are
responsible for imprinting and reading magnetic information to and
from the tape (see Figure 3.22).
Figure 3.22 Tape heads of a Studer 24-track recorder.
81
These functions are performed by electromagnets placed in blocks along
the tape path. There is an individual electromagnet on each block for
each track of information the recorder is capable of recording. Two-
track recorders have two electromagnets on each head block, four-
track recorders have four magnets, eight-track recorders have eight
magnets, 16-track recorders have 16 magnets, and 24-track recorders
have 24 magnets. Professional tape recorders have a three-head stack:
n
Erase
n
Record
n
Playback
The modern erase head, or bias head, was a major breakthrough in
multitrack recording. By activating the magnetic particles on the tape
into an excited state, they can be recorded on with superior signal-to-
noise ratio. This relates to a theory in physics about a body in motion
preferring to stay in motion, but another way of looking at this is to
consider an inelder in a baseball game. When the pitcher throws the
ball to the plate, the inelders do not merely stand there and wait to see
what happens next; they rock back and forth from one foot to the
other. They put themselves in motion. Even if they are rocking left
while the ball is hit to their right, they get a better jump on the ball
because they were already moving. Similarly, if we get those particles
spinning and jumping on tape before we smack it with a signal to
imprint, we will be able to get a better, more accurate imprintin
other words, more signal and less noise. The erase head does this
through use of a very high-frequency signal (the bias tone), many
times higher than anything we can hear. This signal is converted into
a magnetic eld, which is applied to the tape just like any other signal,
except that due to the extremely high signal, those molecules vibrate
for a moment. If we then print information on the tape, we can do so at
a higher level; if we do not, the tape is simply erased. The erase head
only operates when the recorder is in the record mode.
The record head, like the erase head, converts an electronic signal into
magnetism and applies it to the tape. Unlike the bias tone, this signal
represents an acoustic wave within the range of human hearing. The
82
signal that is sent to the record head is the analog of the original sound
wave in an analog recorder and an algorithm in a digital recorder.
The playback head converts the magnetic information stored on tape,
either an analog of the original wave or an algorithm representing that
wave, into an electronic signal. This electronic signal is routed to the
recorders outputs so that the stored recording can be monitored and/or
processed.
Recorder Monitor Modes
The monitor modes of a tape recorder determine the source of the sig-
nal that will be routed to the outputs of the machine. Professional
recorders have three monitor modes:
n
Input, source
n
Repro, playback
n
Sync, sel-sync, or sel-rep
In input mode (also called source), the signal entering the recorder is
split and routed directly to the meters and the recorders output, as
well as to the record head. When we listen to the output or look at the
meters in this mode, we see and/or hear the same signal that is being
routed to the record head. This is used in multitracking in the initial
recording session and is also used to check the level going to the two-
track during a mix prior to recording. Leaving a machine in input is
also a great way to create a feedback, since the output of the console
feeds the inputs of the machine, which feeds back to the console, back
to the machine, and so on. Never place a recorder in input without a good
reason.
In the reproduce mode (repro), the output of the playback head is routed
to the meters and the output of the recorder. When we listen to the
recorders output or look at the meters, we are seeing and/or hearing
what was imprinted on tape. This is used on a multitrack during mix-
down and on a two-track or cassette while recording to ensure that the
signal sent to the deck is being recorded. During a mixdown, if you left
your recorder in input and for any reason you accidentally failed to put
the machine into record, you might not realize it if you left the machine in
83
input, since the meters still show activity. Conversely, if you set the
machine in repro and failed to go into record, the meters would be at
because there is no signal being recorded on tape. Switching to repro on
your mixdown machine before you roll tape is an excellent double-check
that you have set everything properly and that your signal is arriving and
being recorded in the desired location.
Sync mode is a special monitor mode used for overdubbing on analog
machines when a new performance is added to an existing performance
on tape, such as when a singer is added the day after the band laid down
their tracks. The reason this mode is needed is that there would be a
delay, due to the physical distance on the tape between the record
and playback head, if the repro mode was used for monitoring. This
happens because the record head comes before the playback head. A
performer playing or singing along with the recording would hear
the information from the playback head and stay in sync with that sig-
nal. The new performance would be applied to the tape by the record
head, which is at a different location. The old and new performances
would therefore be at different locations on the tape, meaning that they
would not sound simultaneously, or be in sync, when the tape was
played back.
To overcome this, sync mode was created. The record head acts as a play-
back head for all previously recorded tracks. Using sync mode means that
the signal will be applied to the tape at the same location from which the
performer is monitoring. There are controls to select the monitor mode
for each individual trackso that any combination of tracks may be used in
an overdubbing session. One advantage of using digital storage media
while overdubbing is that sync mode is not necessary. The physical lim-
itations of tape do not exist in the digital domain; with virtual tracks and
instant recording, the audio information is simply stored in sync to the
other audio information without the need for a special sync mode.
V.U. Meters
Since tape will only function properly within a limited energy range, and
our equipment will only tolerate a certain amount of voltage, it becomes
necessary to monitor the amounts of energy we are routing and recording.
84
To do this, specially calibrated voltmeters are inserted in the signal ow
within the recorder and the console (see Figure 3.23).
The engineer uses these meters to ensure that the electricity in the con-
sole and the magnetic energy being applied to the tape remain within
operating tolerances. Two types of meters are used for this purpose:
the V.U. meter and the peak meter.
V.U. meters are the most common type of meters found on professional
recording equipment. V.U. stands for volume unit, which is a unit of
measure by which these meters are calibrated. The V.U. meter is cali-
brated to have a response that is similar to human hearing, displaying
the average volume of the program material. Like the human ear, V.U.
meters do not respond quickly to transients. Due to this, some training is
required to properly use these meters. An engineer using V.U. meters
Figure 3.23 V.U. meters: Top-left, low level; top-right and bottom-left, good operating
level; bottom-right, too much level. It is important to aim for the proper level to avoid
distortion or clipping on one hand and to ensure maximum signal-to-noise ratio on the
other. Photo by Gary Gottlieb.
85
must have an awareness of the transient content of the material being
recorded. If the meter does not respond quickly enough to show the
actual level of transients, the overall level must be lowered or other mea-
sures must be taken to ensure that the peaks in level do not overload the
tape.
Peak meters are not calibrated to respond as the human ear does. The
response of a peak meter is much faster than that of a V.U. meter.
Because of this, the actual level of any signal can be seen at any
time, showing us the transients in our signal. This means the engineer
will always be sure of the amount of energy being applied to the tape.
Some peak meters will be embedded in V.U. meters, giving the audio
professional the best of both worlds: a meter that acts in a similar fash-
ion to the ear and one that will expose transients.
One form of peak meters, peak-hold meters, are particularly useful,
since peak meters respond so quickly that they can sometimes be dif-
cult for the eye to follow (see Figure 3.24).
A signal may rise to its peak and fall so quickly that the engineer fails to
notice that there is a problem. The peak-hold meter leaves a trace or ghost
marker behind at the position of the highest peak for a few seconds,
allowing the engineer to see it before it disappears (see Figure 3.25).
One more thing needs to be mentioned about meters in general. They are
calibrated to our own arbitrary criteria, so they are only as reliable as the
last personwho alignedyour equipment. Furthermore, 0 dBcan represent
different things on different pieces of gear. Professional equipment is nor-
mally rated as 0 dBVU4 dBm, where dBVUis an arbitrary value that
we can adjust with a screwdriver and dBm is a specic, measurable
amount of current. Non-professional and even some semi-professional
Figure 3.24 Peak-hold meters operate far faster than V.U. meters or human hearing,
allowing us to see transients on our meters. Photo courtesy of ADT Audio.
86
(or prosumer) gear is typically rated as 0 dBVU 10dBm; in other
words, line level on some gear will be 14 dB lower than other gear, mak-
ing it noisier. Professional gear and semi-professional gear will occasion-
ally offer a switch on the back between 4 and 10. Be sure to select the
setting that matches the level of the rest of your gear if you are presented
with the opportunity to do so.
Magnetic Tape
It is useful to understand the nature of magnetic tape, the tape we use for
both analog recording and recording on digital machines (rather than
hard drives). Magnetic tape is composed of a thin mylar strip, a form
of plastic, coated on the back with a substance used to increase traction,
reduce static electricity, and improve adhesion, so the tape loads easily
and packs well during winding. The front is coated with a binder solu-
tion, in which tiny metal particles are suspended (see Figure 3.26).
Figure 3.25 The red trace or ghost markers on the center two tracks will hold for a
moment on a peak-hold meter to give the engineer a better chance of seeing the exces-
sive level. Photo courtesy of RME.
Carrier
Carrier Adhesive
Mylar Surface
Working Adhesive
Figure 3.26 The components of magnetic tape. Based on a diagram by Editall.
87
The binding solution must hold the metallic particles in place solidly
enough so they do not fall off, yet they must be elastic enough to allow
them to vibrate, move, and realign when they need to conform to an
applied magnetic eld.
Tape is where we keep our tracks. A track is a memory location on
tape, a strip across the magnetic particles. A track consists of a single,
linear, horizontal magnetic band through the particles on tape, repre-
senting one channel of audio information (see Figure 3.27).
Each band lines up with an individual head on the erase, record, and
playback head stacks. In between each track is a small area of blank
tape, called a guard band, to help prevent crosstalk, which is information
from a track playing back on an adjacent head. Much in the way that
increasing our tape speed in analog increases our signal-to-noise ratio,
the greater the width of the magnetic band forming the track, the more
magnetic particles it encompasses, and therefore the greater the reso-
lution of the information and the better the signal-to-noise ratio. This
is once again similar to the number of pixels on a computer screen or
Full Track
Two Track (Half Track)
Recorded Information
Guard Bands
Four Track
Figure 3.27 Full-track, two-track, and four-track congurations. The dark gray areas
represent the horizontal bands created by the electromagnets located in the record
head, while the white space represents the guard bands.
88
the number of dots per inch resolution of a newspaper or magazine
picture. The faster the tape moves and the wider the track, the greater
the number of magnetic particles passing by the heads per second will
be. The two easiest ways to achieve higher sound quality on tape are
therefore higher speed and wider tracks. Professional track formats
use numerous tape sizes and track congurations. In digital, of course,
track width and tape speed are unimportant to achieve a quality,
noise-free recording.
Full track uses almost the entire tape for one track. There are small
guard bands on either edge of the tape to help prevent edge damage
to the program material during handling. Full track is a mono format,
and once it is recorded on, the tape can only be used in one direction
unless all recorded material is erased. The full-track format is found in
1/2-inch and 1/4-inch congurations.
Half track, or two track, is a two-track format in which the entire tape
consists of two tracks. Each track uses approximately one-third of the
tape width. The remaining third of the tape is used for a guard band
between the tracks to prevent crosstalkone tracks signal bleeding
into the otherand at the edges of the tape to prevent edge damage.
Half track can be either a mono or a stereo format and can only be
played in one direction once it is recorded on.
Four track is a professional 1/2-inch tape format that can be used
either as a mixdown format for commercials or as a multitrack format
in an antiquated setup. Four track should not be confused with
1/4-track stereo, a nonprofessional 1/4-inch format in which one can
record in stereo on one side, then ip the tape over to record in stereo
on the other side (which is really the bottom of the same side, not the
other side of the tape). (See Figure 3.28.)
As a rule of thumb, if you can ip it over and play the other side, its not
a professional format. The professional four-track format uses the
entire tape (except for guard bands) in one direction. Once recorded
on, it can only be used in that same direction unless all recorded mate-
rial is erased.
Professional eight track, an antiquated format that is extremely cheap
on online auction sites, uses 1-inch tape, and the eight tracks use the
89
entire tape, which moves in only one direction. There are also some
semiprofessional 1/2-inch eight-track formats.
The professional 16-track format uses 2-inch tape only, and as is the
case with professional four and eight track, it moves in one direction
once it has been recorded upon. As with the eight track, there are
1-inch 16-track machines that are considered semiprofessional. The
professional 24-track format also uses 2-inch tape and is still holding
out as the industry-standard analog multitrack format.
On the digital side, digital audio tape (DAT) was a functional mix for-
mat in the industry briey and is still used in some places for archiving
and backups (see Figure 3.29).
Four TrackProfessional Format with All Tracks in One Direction
Quarter TrackConsumer Format with Two Tracks in Each Direction
Figure 3.28 Professional four-track conguration versus consumer 1/4-track format.
Figure 3.29 The HHB Communications Portadat. Photo courtesy of Mark Trew and Trew
Audio.
90
Using the same mechanics as a VCR, with tape moving slowly across a
rapidly rotating drum, DATs are an excellent and inexpensive digital
two-track storage format, although DATs have mostly given way to
CDs for archiving and mixing in home setups due to the CDs attrac-
tive price and convenience.
Another common digital tape format is Digital Audio Stationary Head,
or DASH, a multitrack format. DASH-format machines have enjoyed
tremendous popularity since their introduction. DASH recorders rec-
ord from 2 to 48 tracks on 1/4-inch or 1/2-inch tape that is specially
formulated for digital use, and because the tape is not pulled across a
rapidly rotating head, as with the DAT, the tape life is increased.
ADATs were also tremendously popular on the digital multitracking
front and were among the earliest inexpensive digital recording for-
mats (see Figures 3.30 through 3.33).
ADATs are eight-track digital recorders that record on videocassettes and
can be easily linked or synced up. The advantage to this, besides digital
Figure 3.30 A popular ADAT recorder/player, the Tascam DA-38. Photo courtesy of
Michael Conn.
Figure 3.31 Another popular ADAT recorder/player is the Alesis ADAT. Photo courtesy
of Alesis.
91
quality at a reasonable price, is that if a studio owns three ADATs, they
have 24-track capability. If a client only needs eight tracks, they can use
one of the ADATs, saving the client some of the tape expense. Of course,
most ADAT enthusiasts have given up this now-outdated technology in
favor of hard drive recording, but many of these devices are available at
bargain-basement prices through online auction sites.
Magnetic Tape Characteristics
Professional magnetic recording tape is tough enough to stand up to
day-to-day handling; however, there are some guidelines to follow to
ensure maximum life. Tape does not require a white room, a com-
pletely dust-free environment, or gloved hands. It is recommended,
Figure 3.33 ADAT recorder/player Roland DM-800 multitrack disk recorder. Photo
courtesy of Roland.
Figure 3.32 Another ADAT recorder/player is the Sony PCM-800. Photo courtesy of
Sony.
92
though, that the tape be handled as little as possible. Touching the
emulsion side of tape can cause sweat or oil from your skin to coat
the tape or break down the chemical composition of the binder. If
there is oil or sweat on the tape, dirt and dust will stick to it more
easily. This will make it more difcult for the tapes magnetic elds
to be read by the recorders playback head. When the binder breaks
down, which can result from contact with sweat and body oils, the
magnetic particles will no longer be held in place. The result is known
as drop out.
Drop out occurs when the magnetic particles held into place by the
binder fall or drop off of the tape, which leaves an area that cannot
hold any audio information. When a tape with drop out is played,
moments where the sound disappears or drops out will occur. Drop
out is usually caused by a breakdown in the chemical compound of
the binder. As mentioned, excessive or improper handling of the
tape may cause this to happen more frequently, so tape should be
handled minimally and held by the edges, much in the way you
would hold a photo if you did not wish to get a ngerprint on it.
The other common problem that may be caused by improper tape han-
dling is tape stretching. Although professional-quality tape is difcult
to break, it can be stretched somewhat if not handled properly. If the
tape becomes stretched, a clearly audible and sudden dip in the fre-
quency of the recorded material, called wow, will occur.
The amount of time available on a reel of tape depends on both the length
of the tape and the speed at which the tape is moving. A2,500-foot reel of
tape, which is the studio standard, will offer 15 minutes of recording time
at 30 IPS, 30 minutes at 15 IPS, and 60 minutes at 7 1/2 IPS.
Stored properly, magnetic tape can last for decades. It is not unusual to
pull out a 20-year-old tape that is in virtually original condition, as long
as it has been stored properly. Tapes should be stored in a dark, cool,
relatively dry location, within a temperature range of 40 to 70 degrees
Fahrenheit and out of direct sunlight. Excessive moisture, excessive heat,
or direct sunlight will destroy tape. Another good way to ruin a tape is
with a magnet. Since the information contained on a tape is stored in a
magnetic eld, running a magnet through a tape library can erase
93
portions of masters that are stored within. Keep in mind that speakers,
microphones and headphones all contain magnets and should never be
stored near tapes.
Proper Tape Storage Considerations
Here are some things to keep in mind when we consider storage areas
for tape:
n
Temperature
n
Humidity
n
Light
n
Magnet-free environment
n
Locked and secure environment
Tapes should always be stored tails out with a smooth wind. When the
work is done, simply let the tape play until it spools off the supply reel.
Letting it play off the reel, rather than using fast forward, creates a
smoother wind. Storing tapes tails out has two advantages: It protects
the recorded material, which is usually toward the head of the tape,
and it avoids noticeable print through. Print through occurs when
some of the magnetic energy stored on the tape passes through a
layer of tape and is stored on an adjacent layer, which is more likely
to happen when the recording levels are high. Tape is stored tails out
so that if print through occurs, it will occur after the original sound.
This will sound like an echo, which is a more natural sound than pre-
echo, which can result from storing the tape heads out.
Production Room Procedure
Although every production room is unique, there are some procedures
that are common to all production rooms, including considerations
when doing any of the following:
n
Powering up
n
Powering down
n
Normalling
94
The number-one consideration when powering up a production room is
to protect the speakers. Many devices will output a surge when they are
rst turned on. There are two simple steps to follow when powering up
to prevent this surge from reaching the speakers and potentially damag-
ing them. As any audiophile will tell you, the last thing you do when
powering everything up is to raise the volume. When the unit is off, the
volume should be down. Do not raise it until everything is turned on.
Also, the power amplier should be the last piece of equipment to be
turned on when powering up. So the sequence for powering up is as fol-
lows: Check the volume to make sure it is all the way down, power up
everything except the amp, power up the amp, raise the volume. Con-
versely, because equipment also surges when it is turned off, when power-
ing down rst lower the volume, then turn off the amp, then power
everything else down.
Normalling equipment or a room is standard procedure and prepares
the room for the next session. It is a huge distraction to walk into a
studio or pick up a remote recorder for a session and nd someone
elses leftover equalization settings from the last session. Both out of
consideration to the next engineer and to ensure that the next session
goes as smoothly as possible, always return the room and every piece
of equipment to its normal when your session is done. All equipment
has a normalled position. When a device is in its normalled position,
all of the controls are set to a standard starting position. While this will
vary from one production room to another and from one piece of
equipment to another, some basic guidelines are as follows:
n
All faders and monitor pots should be all the way off.
n
All pan pots should be set to center.
n
All aux sends should be all the way off.
n
Equalizer boost/cut knobs should be set to center.
n
All bus send switches should be disengaged.
And now that we know what our console should look like when it is
normalled, lets take a closer look at how far out of normal we can
take it on a day-to-day basis.
95
Exercises
1. Keeping the console volume down to avoid feedback, plug a
microphone into the console. Route the signal from the input
module, through a multitrack bus, to a recording device. Route
the signal back to a channel for monitoring. Send the output
of that channel to the stereo bus and make sure you see level in
the meters.
2. Use the patch bay to send signal from a CD player to an input
module, then route the signal as earlier until it appears in the
meters of the stereo bus. Nowturn up the console volume. (Make
sure that microphone is off or disconnected before you do!)
3. Create a voice-over commercial. Record music from a CD onto
a multitrack machine or multitrack computer program. Add
your voice as an announcer on another track. Balance the levels
between the music bed and your voice, and mix down to a CD.
4. Plug in four microphones in front of four sound sources, which
could be instruments, vocals, or a combination. Bus them to
two tracks, practicing both combining inputs to tracks and
balancing live-to-two track levels.
Additional Reading
Aldred, John. Manual of Sound Recording, 3rd ed. Kent, England:
Dickson Price, 1988.
Aldridge, Henry and Lucy Liggett. Audio/Video Production: Theory
and Practice. Englewood Cliffs, NJ: Prentice-Hall, 1990.
2004.
Bartlett, Bruce and Jenny Bartlett. Practical Recording Techniques:
The Step-by-Step Approach to Professional Audio Recording, 4th ed.
Boston: Focal Press, 2005.
Baskerville, David. Music Business Handbook and Career Guide,
8th ed. Thousand Oaks, CA: Sage Publications, Inc., 2005.
96
Benson, K. Blair ed. Audio Engineering Handbook. New York:
McGraw-Hill, 1988.
Camras, Marvin. Magnetic Recording Handbook. New York:
Springer, 2001.
Clifford, Martin. Modern Audio Technology: A Handbook for Tech-
nicians and Engineers. Englewood Cliffs, NJ: Prentice-Hall, 1992.
Davis, Don and Eugene Patronis Jr. Sound System Engineering,
3rd ed. Boston: Focal Press, 2006.
Davis, Gary and Ralph Jones. The Sound Reinforcement Handbook,
2nd ed. Yamaha, 1988.
Eargle, John. Handbook of Recording Engineering, 4th ed. New York:
Springer, 2005.
Ford, Ty. Advanced Audio Production Techniques. Boston: Focal Press,
1993.
Hausman, Carl, Philip Benoit, Frank Messere, and Lewis B.
ODonnell. Modern Radio Production: Production, Programming,
and Performance, 6th ed. Belmont, CA: Wadsworth, 2003.
Horn, Delton. DAT: The Complete Guide to Digital Audio Tape. Blue
Ridge Summit, PA: Tab, 1991.
Huber, David Miles and Robert E. Runstein. Modern Recording
Techniques, 6th ed. Boston: Focal Press, 2005.
Hurtig, Brent. Multi-Track Recording for Musicians. Sherman Oaks,
CA: Alfred, 1988.
Jones, Steve. Rock Formation: Music, Technology, and Mass Commu-
nication. Newbury Park, CA: Sage, 1992.
Jorgensen, Finn. The Complete Handbook of Magnetic Recording,
4th ed. Blue Ridge Summit, PA: Tab, 1995.
Lockhart, Ron and Dick Weissman. Audio in Advertising: A Practical
Guide to Producing and Recording Music, Voiceovers, and Sound
Effects. New York: Frederick Ungar, 1982.
97
Nardantonio, Dennis. Sound Studio: Production Techniques. Blue
Oringel, Robert. Audio Control Handbook: For Radio and Television
Broadcasting, 6th ed. Boston: Focal Press, 1989.
Reese, David, Lynne Gross, and Brian Gross. Radio Production
Worktext: Studio and Equipment, 5th ed. Boston: Focal Press, 2005.
Shea, Mike. How to Build a Small Budget Recording Studio from
Scratch, 3rd ed. Blue Ridge Summit, PA: Tab, 2002.
Siegel, Bruce. Creative Radio Production. Boston: Focal Press, 1992.
Utz, Peter. Making Great Audio. Mendocino, CA: Quantum, 1989.
Wadhams, Wayne. Dictionary of Music Production and Engineering
Technology. New York: Schirmer, 1988.
Watkinson, John. The Art of Digital Audio, 3rd ed. Boston: Focal
Press, 2000.
White, Glenn. The Audio Dictionary, 3rd ed. Seattle: University of
Washington Press, 2005.
Woram, John. Sound Recording Handbook. Indianapolis: H. W.
Sams, 1989.
Zaza, Tony. Audio Design: Sound Recording Techniques for Film and
Video. Englewood Cliffs, NJ: Prentice-Hall, 1991.
98
4
Consoles
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99
A
s mentioned, when the average person thinks of an audio
professional, he or she envisions someone operating a mixing
console. Those romantic images showing audio professionals
sitting behind a large console with a veritable ocean of knobs, but-
tons, and faders are what pulled some of us into this business in the
rst place. In the production room, the mixing console is the center
of everything. All signals will pass through the console. The consoles
primary functions remain as routing and processing of input and out-
put signals. A professional console has the ability to simultaneously
route numerous input signals to a wide variety of devices and loca-
tions. Signals arrive at the console, where the audio professional
sends them wherever they are needed. These amazing devices are
simple to understand. It is important to recall redundancy, the seem-
ingly endless repetition of modules, which makes them easy to under-
stand and use. If you take the time to understand one module or strip
in a console, you will understand them all, thanks to redundancy (see
Figure 4.1).
As stated earlier, versatility and exibility of a console, more than the
number of modules, are the hallmarks of a quality console. A state-
of-the-art console will consistently give the audio professional many
options for signal routing. Shifting the sequence of effects in the con-
sole and choosing different ways to bring signal in and then send it
out allow the engineer to maximize quality and fulll his or her
aesthetic.
Versatility in Consoles
Imagine if your home stereo only had one input, dedicated to a radio
signal, and only one output, dedicated to one speaker or one pair of
speakers, like a clock radio. You would be very limited in your choices.
Now imagine you had many inputsone for a CD player, one for
DVD/ TV, one for a turntable, and one for auxiliary, in addition to
the radio input. Imagine you have three outputs instead of one, each
to a different pair of speakers. Now you can select what you wish to
listen toCD, DVD, radio, and so on (source)and you can select
which set or sets of speakers you wish to listen to (output). Now
you are versatile. You have options. Now imagine that you could
100
access dozens, or perhaps hundreds, of sources and dozens of outputs.
You can take signal from many different sources, combine them and
effect them however you see t, blend them and re-blend them, before
you send them to different locations to t your clients needs and your
own aesthetic criteria. This describes signal routing and signal ow,
the basic building blocks of audio. To gain a better understanding of
the basics of routing signal ow, we need to understand the various
components of the console.
Figure 4.1 The redundancy of modules makes it easy to understand even the most
complex consoles, such as this SSL XL 9000 K in Studio Davout, France. Photo courtesy
of SSL.
Chapter 4 Consol es
101
There are many different types of consoles available, from different
manufacturers and with different basic design elements. Some offer
dedicated input modules and dedicated output (monitor) modules.
Others offer modules that handle either input or output signals, but
only one or the other at one time. Then there are modules that can
simultaneously route an input signal and an output signal. If you
understand the last of these, known as the input/output module (I/O
module), you will be able to understand all types of modules, so this is
the console design we will examine (see Figure 4.2).
The I/O module is the most common part of the console. It contains
several inputs and outputs, each having its own specic purposes,
as well as an equalizer to adjust timbre and other features. By under-
standing how signal ows through one I/O module on a particular con-
sole, all I/O modules become easily understandable on that console.
Also, if you understand modules on a handful of different consoles,
then any console you sit behind will make a certain amount of sense.
Figure 4.2 Consoles like the Neve 88R in Skywalker Sound utilize I/O modules. Photo
courtesy of AMS Neve.
102
Preamps
Although all consoles will not contain every feature described here
(and some high-end consoles will contain far more features), consoles
are generally similar, so we can cover the basic features common to
most modules (see Figure 4.3).
Figure 4.3 The features of this module, from a Mackie CR series console, are typical of
many modules. Photo courtesy of Mackie.
Chapter 4 Consol es
103
The rst point at which the signal is manipulated by the console is
determined by the type of signal present and selected. Consoles accept
primarily two types of signals, line level and mic level. Although DJ
consoles will also accept signal at phono level, as will older home ste-
reos, we will remain focused on the more common line and mic levels.
There is a need for two separate types of inputs because there is such a
vast difference between the voltage levels of each of these signals.
Microphones output tiny amounts of voltage, somewhere in the
range of 60 to 70 dB below the consoles optimal operating level,
while a line levels signal is already roughly in the range of the con-
soles standard operating level. As a result these signals need to be
dealt with differently when they arrive at our console in order to
ensure maximum quality and proper pre-amplication of each signal.
Due to its extremely low voltage, between 60 and 70 dB, a mic
level signal requires a tremendous boost so that it will be sufcient
to be processed by the consoles components while retaining reason-
able signal-to-noise characteristics. These signals will need as much as
a 70-dB boost in order to reach our consoles standard operating level.
That is a lot of amplication, and if one were to treat it cheaply, a
great deal of noise would be introduced to the signal, which would
be unacceptable because it would go against our goal of quality audio.
This signal therefore requires a large amount of clean amplication. The
microphone input of a console contains a special amplication circuit,
a preamp or mic pre, which performs this function. The microphone
preamp is typically the single most expensive component and is consid-
ered by some to be the most important component on any console since
it must have excellent signal-to-noise characteristics while amplifying the
signal tremendously.
The microphone preamp is also one of the components that is chiey
responsible for the character of the sound, or the way a console will
color a sound. The characteristics of a consolesuch as a Neve, which
is known to have warm mic preamps, as opposed to a Solid State Logic
console, which has a reputation for transparent mic preampswill
often determine an engineers console preference. When customizing
a console, a popular x involves removing the mic preamps and adding
replacements that suit the engineers taste more closely. This type of
104
modication can make an inexpensive console sound like a top-grade
model; it could also give an engineer the versatility or cleanliness
of one console combined with the mic pre characteristics of another.
Several manufacturers sell rack-mountable or stand-alone boxes with
high-quality preamps that may be used as outboard signal processing
gear to improve the sound quality of the average console, and of course
plug-ins are available that emulate all the most popular preamps. The
aesthetic choice of a particular preamp can greatly affect the nal
product and can become part of an engineers aesthetic as well as his
or her signature sound.
Home stereo ampliers have special inputs with preamps for turnta-
bles, line inputs, and possibly even microphones as needed, just as
modules on a console have line inputs and mic inputs. All preamps,
from your favorite DJs phono preamps to your stereos line preamps
to your consoles mic preamps, provide the rst stage of amplication
to an incoming signal. Like microphones, turntables output a small
amount of voltage (though not as low as a microphone) and require
an extra boost to reach a usable range. If you have ever plugged a turn-
table into a line level input on a home stereo, such as the input for a
CD player or cassette, you have experienced the problem of inap-
propriately matched level as you turned your volume knob higher
and higher, hearing mostly noise. Conversely, if you have ever plugged
a CD player into a phono input, your level was crashingly loud and
distorted. Both at home and in the studio, we need to select the correct
input for our incoming signal to ensure the best possible quality and, in
some cases, in order to hear the signal at all.
While quality is always benecial, it is less critical in a consoles line
input preamp. Since consoles operate at or near line level, line level
signals do not require much boost. An audio consoles line inputs are
typically used for the output of electronic musical instruments, such as
electric guitars and basses, synthesizers, samplers, and drum machines.
Another primary use for a consoles line input is to receive the output
signal from recorders and signal processors. The separate inputs on an
audio console can be compared to the separate inputs on a home stereo
amplier. The home stereo has separate inputs for turntable, CD,
DVD/TV, and auxiliary sources. Due to the varying output levels
Chapter 4 Consol es
105
of these devices, it is necessary for devices that are being connected
electrically to be well matched, as mentioned previously. If an improper
signal is routed to an input with either too much or too little level, the
result will be either damaged components due to excess of current or
level, or poor signal-to-noise ratio, since the engineer will be unable to
bring the sound up to an adequate level without also bringing up the
noise as mentioned above.
Impedance
The difference between mic level signals and line level signals is not
described as simply a difference in level, it is described as a difference
in impedance, and that has other extremely important implications. A
mic level signal, at somewhere around 65 dB, is known as a low-
impedance signal and can travel long distances through cable without
signicant loss because such small amounts of electricity encounter lit-
tle resistance. Conversely, a line level signal is at or about optimal con-
sole level, somewhere between 30 dB and 0 dB, and is also known as
high impedance. Due to its higher level of current, a line level signal
sent through a long cable will experience resistance resulting in dete-
rioration, or signal loss. Some consumer microphones are high imped-
ance. These can sometimes be identied by their low price, their
location on a shelf at a local discount big box retailer, or by the fact
that they terminate in a 1/4-inch jack, also known as a guitar plug.
There are several different types of 1/4-inch jacks. There is the type
just described, which will only have one ring on the jack (these
will typically carry a high-impedance, unbalanced signal). There is
another with two rings, known as tip-ring-sleeve, which is balanced
but still high impedance and typical of 1/4-inch patch cords. There is
also a stereo 1/4-inch jack, frequently seen terminating a pair of head-
phones, which is also high impedance. Low-impedance signals typi-
cally terminate in a barrel-shaped three-pin connector known as an
XLR, or Cannon plug (see Figure 4.4).
On a regular basis, engineers will nd themselves in situations in
which it is necessary to turn a line level signal into a mic level signal.
This is common in recording studios while recording an electric bass
or a synthesizer. The musician does not want to play in the control
106
room (where he or she could have a short cable run to your input);
instead, he or she wishes to remain in the studio to play with band-
mates. Basses and synthesizers output at line level, but it is a long
way from the studio to the control room. It would be messy to have
a 50-foot wire running to the control room and would result in tre-
mendous signal loss due to the high-impedance output of a bass or
synthesizer. When the engineer wants to take the line level signal
direct into a mic input, a direct box, or DI is used. A DI is actually
a step-down transformer that splits the signal, sends an unaffected
high-impedance signal to a 1/4-inch jack (if the musician wishes to
monitor himself through an amp in the studio), and lowers the level
of the other signal to low impedance, mic level signal, routing it to an
XLR output that allows it to be plugged into a microphone input and
make the long trip to the console without any signicant signal loss.
While this method is effective on some instruments, other instruments
such as guitars that benet from the coloration of passing through an
amplier sound better if the musician plugs into his or her amp, upon
which the engineer then places a microphone.
Once signals reach the console, we may need to adjust the rst level of
amplication. Mic pres and line pres are controlled by mic trims or
line trims. On some consoles there is only one trim pot (potentiometer)
per module for both of these functions, while others offer separate trim
Figure 4.4 Various connectors (left to right), male XLR, female XLR, RCA plug, 1/4-inch
phone plug, 3.5mm mini plug. Photo courtesy of Neutrik.
Chapter 4 Consol es
107
pots for mic and line. Sometimes there will be a switch to determine
which signal the engineer seeks to amplify and use as an input; other
times the console will sense whether an input is plugged into a mod-
ules 1/4-inch input or XLR input and assume that it is a line or mic
signal based on the jack. In this case the trim will adjust whichever pre
is associated with the signal that is plugged into that module, line, or
mic. If you have plugged in a microphone, look for a switch that says
mic if there is only one trim pot. If there isnt a switch and there is
only one trim pot, and if you have a microphone and no line input
plugged into that module, the module will probably know which signal
you desire to amplify, and it will know which preamp will have the
proper impedance to give you the proper range of amplication.
Trims are passive attenuators. They are simple resistors controlled by a
rotary pot. A trim is used to vary the level of a signal before it reaches
the amplication phase of a circuit. When set for maximum level, gen-
erally turned all the way to the right, the trim control is adding no resis-
tance to a circuit. When turned to the left the trim control is reducing the
level of the signal by adding resistance. This allows a signals level to be
varied before it reaches the amplier, which is a xed gain stage and can
therefore be easily overloaded if the input signal is too large. As such, it
is a good idea when plugging in a microphone or other source to begin
with the trim set to full attenuation, all the way to the left, and turn it up
gradually until standard operating level is reached (see Figure 4.5).
This is not necessary with line inputs because their levels tend to be
more consistent. While there will be some variation between the out-
puts of tape machines, computers, synthesizers, signal processing gear,
CD players, and other line level sources, these differences are smaller
than the differences between the outputs of different microphones. Fre-
quently a line trim will provide a detent, or click-stop, at the twelve
oclock position as a normal setting for most line inputs.
While we are discussing preamps, lets talk about the other device we
can use to increase the level of a signal in our module, the fader. Faders
are the linear sliders located at the bottom of each module, closest to
the engineer. Most faders are linked to Voltage Controlled Ampliers
(VCAs) (see Figure 4.6).
108
Figure 4.5 The mic and line trims on the SSL 6000, with the multitrack bussing and bus
pan above it. Photo courtesy of SSL.
Chapter 4 Consol es
109
Unlike potentiometers, which are passive and reduce resistance to
increase level, pushing up a fader attached to a VCA actually increases
the amplication of the signal and potentially adds a certain amount of
noise. The assumption that faders control VCAs is not universal. Some
consoles have main faders or channel faders (other names for the big
faders on each module) that are actually passive attenuators. Some
consoles have a channel fader attached to a VCA and a potentiometer
above it known as a monitor pot that is a passive attenuator. Still other
consoles will have two faders, a large channel fader at the bottom
linked to a VCA and a smaller fader above it that is a passive attenu-
ator and operates as a monitor pot (see Figure 4.7).
There are many combinations possible, and an audio professional
needs to learn the combinations on all consoles he or she regularly
operates in order to be aware of all the options for amplifying signal
with a minimum of noise.
The reason for the separate fader and monitor pot mentioned in the
previous paragraph is that our I/O module is designed to support and
route both an input signal and a return from tape, or output signal,
simultaneously and discretely. While recording basic tracks in a
recording studio, one path leads from microphones and instruments
Figure 4.6 The large fader on an SSL console is linked to a Voltage Controlled Ampli-
er. Photo courtesy of SSL.
110
Figure 4.7 The small fader on an SSL 6000 is a passive attenuator, not a VCA. Photo
courtesy of SSL.
Chapter 4 Consol es
111
through the preamp and fader to the multitrack recorder, while the
other returns from the multitrack through the line preamp in the mod-
ule, then through the monitor pot to the ampliers, then to the monitor
speakers. The path is actually a little more complicated, as we will
soon see. Each of these signal paths has separate volume controls,
which are usually the channel faders going to the recording device
and monitor pots or monitor faders going to our amps. In more sophis-
ticated consoles these paths can be ipped, sending the monitor faders
to the house or multitrack and sending the channel faders to the moni-
tors. In other situations an audio professional will route the signals
differently, as is appropriate for the operation he or she is trying to
accomplish. In a live situation, with a band or in a theatre, the channel
faders will typically be sent through the stage ampliers to the speak-
ers in the house while the engineer may use the monitor pots or mon-
itor faders to create a cue mix for the musicians or actors. The
possibilities are endless and will be dictated by the needs of the project,
the demands of your client (the producer or director), the capabilities
of the equipment, and your personal audio aesthetic.
Auxiliary Sends
In addition to our primary mixes, the audio professional has the option
of creating sub-mixes. Depending upon how the audio professional
chooses to set it up, a console is capable of creating many simultaneous
and discrete sub-mixes that can be used for any purpose. A sub-mix of
instruments can be sent to a reverb unit, to headphones or stage mon-
itors for cuing, to a satellite uplink for broadcast or netcast, to a remote
recording truck out in the parking lot, to the dressing room in a theatre
to cue the actors, or to the orchestra pit for the musicians. These func-
tions are usually performed by the auxiliary sends (aux sends), some-
times referred to as sends, while the channel faders feed the multitrack
or the house and the monitor faders feed the monitors (see Figure 4.8).
Any desired conguration is possible with a well-designed console and
a knowledgeable engineer. In terms of signal ow, the signal feeding
our aux send can be either pre-fader or post-fader. Most consoles
default to post-fader, where the signal is sent to the aux sends after
the fader, which means that our fader movements affect the level
112
sent out of the aux sends. When we choose a pre-fader setting, our
fader movements do not affect the level of the signal going to or com-
ing out of the aux send. We will discuss applications for these two
settings in Chapter 9, Signal Processing.
To send the signal from the output of a channel fader, we use a channel
assignment switching matrix, also known as multitrack busses, or bus-
sing. Put most simply, a bus is a send. The auxsends mentionedpreviously
Figure 4.8 The SSL 6000 features six aux sends, including the stereo pair at the top.
Also shown are the pre/post switches. Photo courtesy of SSL.
Chapter 4 Consol es
113
and the stereo bus, which sends the stereo mix to the ampliers and
speakers for monitoring as well as to the mixing machines, are also
busses. A bus is anything that sends and potentially blends a signal;
this will merit more discussion in a moment. In the multitrack busses,
a combination of buttons and pan pots (refer to Figure 4.5) will deter-
mine where a signal in the channel fader will be routed, usually onto a
track of the multitrack storage device to be recorded as an individual
element, or into the stereo bus, where it will combine with other sig-
nals to create our stereo mix. Along with the ability to bus a signal to a
single track on the multi, a signal can also be split between two tracks
using the pan pot to create a stereo pair on your multitrack. This tech-
nique is valuable when combining ve or six inputs from a drum set
(tom-toms and overheads) into a single stereo pair or when recording a
large orchestra, for combining a section into a stereo pair. When work-
ing with the virtually innite number of tracks offered by many com-
puter programs this technique is unnecessary, but when the number of
tracks available limits the audio professional, combining inputs in this
manner becomes second nature.
Pan Pots
Pan pots, or panoramic potentiometers, will continuously vary a single
signal between two or more output busses. This is useful when creating
a stereo mix or when creating a stereo pair of tracks on a multitrack
(see Figure 4.9).
Figure 4.9 Pan pots will move the signal in the channel across the stereo eld. Photo
courtesy of SSL.
114
When creating a stereo pair on the multitrack, by bussing a signal to
both Tracks 1 and 2, it will appear to be in the center. Bussing a signal
to Track 1 will make it appear to be only on the left side; bussing it to
Track 2 places it only on the right side. By bussing it to both tracks and
panning it part of the way toward Track 1, it will appear to be mostly
toward the left, but not hard left. Pan pots are also used on the mon-
itor fader to determine location in the stereo eld while monitoring.
Much like the balance control on your home or car stereo, a signal can
be moved from left to right within the stereo eld by using the pan pots
on the monitor signal, or on the channel fader signal in a mix situation.
Since the pan pots can affect either the stereo bus or the multitrack
bussing, many modules will feature two pan potsone near the
fader and monitor pot to adjust the signal on the way into the stereo
bus and another in the multitrack bussing dedicated to panning the
signal between tracks on the multitrack.
Equalizers
Another feature common to I/O modules in most consoles is equaliza-
tion. Equalizers (EQ) are frequency-selective ampliers. They will
increase or decrease the volume of a user-selected range of frequencies
within the audible band. The chief function of an equalizer is to alter
the timbre, or the subjective feel of a sound (see Figure 4.10).
Similar to the bass and treble controls on your home or car stereo,
basses may be made deeper-sounding, and the attack of drums and
percussion instruments may be emphasized or de-emphasized by
increasing or decreasing the amount of energy at the proper frequen-
cies. We will explore different types of equalization later, in Chapter 9.
For now we need to understand where the equalizer is placed in the
signal ow of our module. As with many aspects of signal ow the
answer can be as simple as, EQ is placed wherever we want it in
the signals path, or as complicated as, It is not where I want it,
but my console will not let me change that.
Most sophisticated consoles incorporate an EQ IN switch, which pla-
ces the EQ in the signals path. When EQing an input signal it will
typically be after the mic-pre and before the fader and aux sends.
Chapter 4 Consol es
115
Consoles that boast greater exibility offer many more options, such
as placing the EQ after the aux sends on any module you wish or giv-
ing the audio professional the option of placing the EQ before or after
a lter section or dynamics section (which we will also learn more
about in Chapter 9). With EQ, as with every component in our signals
Figure 4.10 The equalizer section on the SSL 6000 offers many options for rerouting
signal, depending on which switches are pressed. Photo courtesy of SSL.
116
path as it winds its way through our module, more options and versa-
tility translate into greater opportunities for creativity for the audio
professional and the ability to shape the sound to conform to the engi-
neers personal aesthetic.
In addition to choosing the sequence for module components, we can
frequently choose which signal within the module will be affected by
EQ. You may recall that two signals can coexist discretely in our I/O
module. A quality console will allow the engineer to select which of
these signals will be EQed in any given module. Often, there will be a
switch that will assign the EQ circuitry to the monitor path (such as
the MON switch in Figure 4.10). This is called monitor EQ, and
unlike input EQ, it will not affect the signal as it is recorded on the
multitrack. It is highly recommended that basic-level recordists avoid
input EQ, because EQ cannot be removed once it has been recorded on
a track.
Summing Networks
You may be wondering by now how the console manages to combine
all these signals. Simply jamming them together in an ever-increasing
manner would result in a tremendous amount of level, which would
soon exceed the tolerances of the equipment, causing distortion. Con-
soles use summing networks, or combining amps, to combine separate
signals into one composite signal. There are summing networks in
every bus on the console. Every time two signals need to combine in
a bus, each individual signal is reduced by 3 dB, and then combined.
Because 3 dB represents an approximate halving of the signal, when
they are combined the end result is an output level consistent with each
of the two input levels. Consoles have numerous summing networks,
since they need to be present wherever two signals may combine.
Faders, monitor pots, multitrack bussing, the stereo bus, and aux
sends all provide functions that require combining amps, one for
each possible bus on each module. It may help to return to the example
of the plumbing system that we used in the last chapter, where the
output of a variety of plumbing xtures combines. If everyone in a
dormitory or apartment building were to run their sinks and showers
at the same time while ushing toilets, the resulting output might
Chapter 4 Consol es
117
overload the system and cause it to back up, much in the way our con-
sole will distort if we jam too much signal through it at once. Also, like
a plumbing system, each individual signal source has its own volume
controls, similar to a valve, which modulates its ow into the network,
or drain pipe. The entire network also has a volume control, a master
valve, which will modulate the ow of the entire network or drain pipe
to some output source, or back into the stereo bus, where it is summed
once again. All summing networks must have a master controlling the
output. Lets look more closely at the master valves.
The master fader, or any master trim, will control the output level
of the summing network that it modulates. The master fader controls
the output to the monitors and the mix machines in the recording
studio, and the house speakers in a live sound situation. Similarly,
the bus faders modulate the level of the sum of the multitrack busses,
and the send masters control the overall output of the aux sends. The
need to raise or lower the stereo bus while mixing should be obvious;
it is one more tool at our disposal to ensure that we send the proper
level to our mix machine. Similarly, if we create a composite signal
and assign it to a track on our multitrack, we may need to lower the
overall level. If we are happy with the internal balance we have cre-
ated, it is more efcient and accurate to reduce the level at a bus
master than it would be to attempt to lower all faders feeding that
bus by an equal amount. Our various masters are helpful in achiev-
ing correct levels throughout our processes, but we must remember
to raise our master whenever we wish to send signal out through any
bus (see Figure 4.11).
By now we have all the pieces of the puzzle. Lets t them together. If
we want to send a signal from a module through an aux send to a reverb
unit, we must have signal in the module. The module must be turned on
and the fader pushed up (assuming our aux send is set in post-fader).
We must turn up the aux send on the module and have the correspond-
ing aux master turned up, and the reverb unit must be turned on. If the
reverb unit is not normaled to one of our aux sends, we must place it in
the signals path through the patch bay, as discussed in Chapter 3. If we
are actually trying to hear a signal come out of the console (imagine
that!), we must route that signal accordingly as well. To listen to a
118
monitor mix, we must have signal in our monitor faders. The monitor
faders must be turned on and turned up. These signals must be assigned
to feed the stereo bus. The master fader must be up. The monitor source
selector switch, which determines which signal is being sent to the mon-
itors, must have the stereo bus selected. The volume control must be up,
determining how much of the signal is being routed to the ampliers,
which must be on, and then to the speakers.
This may sound complicated, but how different is it from your home
stereo? If you want to listen to a CD, everything must be powered up.
You need to put a CD in the player. You must have your CD player
hooked up to the proper input. You must select the CD input on your
amp. You must send the signal to the correct speakers. And you must
have your volume up. Its not all that different, except that with a pro-
fessional console you have many more options.
Figure 4.11 The master section contains the master fader plus all send masters. Photo
courtesy of SSL.
Chapter 4 Consol es
119
Using the I/O Module
We have talked about the manner in which the I/O module handles two
different signals discretely, but we have not discussed the ramications of
this, nor have we discussed the way the audio professional uses this tech-
nology to his or her advantage. Lets talk methodology for a moment.
Suppose we are recording a rhythm section, drums, electric bass, elec-
tric guitar, and piano. I will start by placing microphones on the drum
kit, one each on the kick drum (K), snare drum (SN), hi-hat (HH), each
of the three tom-toms (TOMs), and two on the overheads (OHs). I will
plug the bass (BS) into a DI, which will be treated as a microphone
input, and then place a microphone on the electric guitar (EGT) ampli-
er. Two microphones will be placed in the piano (PNO), one for the
low end (left hand) and one for the high end (right hand). Table 4.1 is
the beginning of my input list.
Now let us suppose that we are handed this input list and we only have
eight tracks available. We cannot spread our instruments out during
recording. We need to combine some elements as we record in order to
Table 4.1 Input List
Input Instrument
1 K
2 SN
3 HH
4 TOM1
5 TOM2
6 TOM3
7 OHL
8 OHR
9 BS
10 EGT
11 PNOL
12 PNOR
120
t this rhythm section on eight tracks. To do this, I will add my multi-
track bussing assignments, or track assignments, to this list as we see in
Table 4.2.
I have now t all twelve inputs on the eight tracks available to me. Of
course I cannot add anything else to these tracks, such as a vocal or a
guitar overdub, since I did not leave any tracks open. In a real-life sit-
uation I would have to combine further to keep tracks available for
other instruments to be added. Notice I combined the drum tracks
into a stereo pair on Tracks 3 and 4, as we discussed earlier. This is
a good start, but I am not nished yet. We are discussing two discrete
signals living side by side in each module, and so far we only have an
input signal. In other words, the band is playingwe have turned up
our mic pres and channel faders on the rst 12 modules and assigned
each module (Input in Table 4.2) to the appropriate and corre-
sponding track (Track in Table 4.2) using the multitrack bussing.
We have sent these signals to our recording device where we see the
meters moving, indicating that they have arrived as planned, but we
have not heard anything yet. Heres the next step.
Table 4.2 Input List with Track Assignments
Input Instrument Track
1 K 1
2 SN 2
3 HH 4
4 TOM1 3
5 TOM2 3, 4
6 TOM3 4
7 OHL 3
8 OHR 4
9 BS 5
10 EGT 6
11 PNOL 7
12 PNOR 8
Chapter 4 Consol es
121
Monitoring will take place through the monitor pot of the module that
corresponds to the track assignment. If I wish to hear the kick drum I
need to turn up the monitor pot or monitor fader on Module 1, assign
the monitor pot to the stereo bus (which may happen automatically,
depending on the console), turn up the master fader, and turn up the
gain control. In other words, I need to route the input signal to the
recording device and route the return from the recording device to
the monitors. These two signalsthe input signal from the microphone
in the kick drumthat we are sending to be recorded and the return from
the recorder of the kick drum that we are routing to the monitorsare
discrete and simultaneously passing through Channel 1. Channel 2 is
identical, as the signal from the snare drum microphone is recorded
and monitored discretely and simultaneously through the same
module.
When we get to Module 3, things become a bit more interesting. The
signal from the microphone on the hi-hat is routed to Track 4 (the
right side of our stereo drums) to be recorded. In order to hear the
hi-hat, we must raise the monitor pot on 4. Similarly, the signal from
the microphone on our rst tom comes into our console on Module 4
and is assigned to Track 3. To hear it we must raise the monitor pot
on Module 3. Now that we have turned that up, Module 3 contains
the input signal from the microphone on the hi-hat along with the
return from the recording device of the rst tom on Module 4. We
will pan the monitor pot on Module 3 to the left side of our stereo bus
and the monitor pot on Module 4 to the right side of our stereo bus
since this will eventually be our stereo drum mix.
Moving our way down, we will turn up the mic pre and fader on
Module 5 (the middle tom) and bus it to both Tracks 3 and 4 on our
multitrack so it will appear in the center when the signal returns from
our multitrack to the monitor pots on both Modules 3 and 4. The signal
on Tom 3 arrives in the input section of Module 6 and is assigned to
Track 4, so when it returns to the monitor side of Module 4 it will
appear on the right side of our stereo drum pair. Our overhead micro-
phones on the drum kit, on Inputs 7 and 8, will be assigned to Tracks 3
and 4, respectively, and will return on Modules 3 and 4, blending with
the toms and hi-hat and completing our stereo drum pair.
122
We plugged the DI on the bass guitar into Microphone Input 9 and
recorded it on Track 5, so the mic pre and fader boost the bass input
signal on Module 9, where it is assigned (or bussed) to Track 5. The
signal returns on Module 5, where it passes through the line pre on the
way to the monitor pot, which sends the signal into the stereo bus,
which allows us to hear it. Similarly, the electric guitars microphone
that is plugged into Input 10 is assigned to Track 6. We can manipulate
the guitars level to the recording device on the mic pre or channel fader
on Module 10 and adjust the level in the mix on the monitor pot on
Module 6.
We nally arrive at the piano, which is plugged into Inputs 11 and 12
and assigned to Tracks 7 and 8. When we raise the monitor pots on 7
and 8 to hear the piano, lets not forget to split them in the stereo bus,
with 7 toward the left and 8 toward the right. Make an aesthetic
choice whether you like these tracks hard-panned left and right or
panned to a lesser degree. Trust your ears to tell you what sounds
best.
In consoles without I/O modules, modules will be dedicated to input or
monitor. The input modules will generally be to the left of the con-
soles master section, while the returns from the recording device
will typically be on the right. The only thing that changes in this
type of setup is that you adjust your input levels on the faders on
your left and adjust your monitor levels on the faders on the right.
The numbers still line up as listed earlier, with the bass on Module 9
on the left (input) and on Module 5 on the right (monitor). Some engi-
neers prefer these consoles, with dedicated monitor sections, while
others prefer a console with I/O modules. Like so many facets of
this industry, this is a personal choice.
Now that we think we understand the split in the module, lets con-
sider some other practical implications and applications. If our level to
our recording device is low, where do we adjust it? Will that affect
anything else in our signals ow? What if the producer wants to
hear more guitar in the monitor mix in the control room, or if the gui-
tar player wants to hear more of himself in the headphones? Every
time we adjust level, it affects everything downstream. Thinking
Chapter 4 Consol es
123
of our signals path in a linear fashion may help to clarify this (see
Figures 4.12 and 4.13).
Lets consider the input signal rst. Suppose our bass guitars signal to
tape is too low. Based on the earlier example, we would adjust the level
of the signal at either the preamp or the channel fader on Module 9,
regardless of whether we were recording on tape or disc. By raising
the level to tape, we would also be raising the level of the return signal
of the bass, because any change along the signals path affects every-
thing downstream. If the level to the recording device is ne but we
wish to hear more bass in the control room, we would raise the mon-
itor pot on Module 5, affecting only our monitor mix. Since there is
little else downstream of our monitor pots, little else is affected. In the
case of aux sends, they will be affected by changes in level at the mon-
itor pots only if the aux sends are set in post-fader, as previously
discussed.
This concept can be difcult at rst. We can make it easier. Looking
back to Table 4.2, there are two columns of numbers, one listed as
input and the other as track. Anytime an adjustment needs to be
made to level for recording, that is input. The number will correspond
to the number in the Input column (such as 9 for the bass guitar), and
Mic
Signal
Mic
Preamp
Channel
Fader
Signal Flow
Multitrack
Busing
Bus
Master
Track On
Multitrack
Figure 4.12 Typical ow of an input signal from a microphone to a recording device.
Any change made along this path will impact everything farther down the path,
including on the monitor side.
Recorded
Signal
Line
Preamp
Mon. Pot
or Fader
Stereo
Bus
Master
Fader
Volume
Control
Amps and
Monitors
Signal Flow
Figure 4.13 Typical ow of a monitor signal as it returns from our recording device,
through the console and to the speakers. Moving the monitor pots will affect the signal
coming out of the aux sends only if the aux sends are set in post.
124
the adjustment will be made either in the preamp or channel fader.
Anytime an adjustment needs to be made regarding monitoring, hear-
ing, or listening (such as the producer saying, I want to hear more
bass), look for the track number5 in the case of the bassand
adjust the monitor pot. If the guitar player wants to hear more of him-
self in the headphones, the operative word is hear, so we would go
to Module 6 and turn up the aux sends from which we are feeding the
headphones. Easy as pie (after a few years of practice).
Now that we have dissected the heart of our control room, the console,
it is time to take closer examination of the extensions of our ears into
the studio: microphones.
Exercises
1. Look at the input list and track assignments in Table 4.2.
Obtain 12 microphones and cables, plug them in, and set up
your bussing according to the table. Check every microphone
input to see that you are passing signal to the correct track.
2. Building off the last exercise, treat this like a real session. Have
a friend or fellow student call out, Make the bass louder in
here, or, The guitar player wants more of himself in the
headphones. If you are alone, make a list of producer
demands beforehand. Once you have set up the console, go to
your list and randomly pick out tasks to accomplish. Do this
until you are adept at distinguishing between the input side and
the monitor side.
3. Continuing with the previous exercise, add EQ to the equation.
Try to imagine brightening up the bass going to tape. Try
monitor EQ on the kick drum to give it more oomph.
4. Continuing further with the previous example, go to the patch
bay and add a reverb on Aux Send 1. Return it to any available
faders (patch from the outputs of the reverb to the channel line
in on any module that is unused). If you have signal in your
channels, send it through Aux 1 to the reverb (make sure your
aux master is up) and monitor it through the returns.
Chapter 4 Consol es
125
Additional Reading
2004.
Benson, Blair ed, Audio Engineering Handbook. New York:
McGraw-Hill, 1988.
Springer, 2001.
Clifford, Martin. Modern Audio Technology. Englewood Cliffs, NJ:
Prentice-Hall, 1992.
Davis, Don and Eugene Patronis, Jr. Sound System Engineering, 3rd
ed. Boston: Focal Press, 2006.
Springer, 2005.
Ford, Ty. Advanced Audio Production Techniques. Boston: Focal
Press, 1993.
Hausman, Carl, Philip Benoit, Frank Messere, and Lewis B. ODonnell.
Modern Radio Production: Production, Programming, and Perfor-
mance, 6th ed. Belmont, CA: Wadsworth, 2003.
126
Hurtig, Brent. Multitrack Recording for Musicians. Sherman Oaks,
CA: Alfred, 1988.
Jones, Steve. Rock Formation: Music, Technology, and Mass Com-
munication. Newbury Park, CA: Sage, 1992.
Jorgensen, Finn. The Complete Handbook of Magnetic Recording, 4th
ed. Blue Ridge Summit, PA: Tab, 1995.
Oringel, Robert. Audio Control Handbook, 6th ed. Boston: Focal
Press, 1989.
Press, 2000.
Sams, 1989.
Chapter 4 Consol es
127
5
Microphones
P
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o
t
o
c
o
u
r
t
e
s
y
o
f
N
e
u
m
a
n
n
.
129
M
uch like our ears, the primary job of the microphone is to
capture a sound event. Microphones are transducers that
convert acoustic energy into electrical energy. As discussed
in Chapter 3, a transducer is a device that converts energy from one
form to another. The electrical energy output from a microphone may
be routed by a console to a recording device for storage, to a transmit-
ter for broadcast, or to an amplifier and monitors for reproduction. A
variety of microphones are available, each with its own characteristics.
Sounds picked up by various microphones will exhibit these character-
istics and therefore sound different. These characteristics may be
judged technically, aesthetically, or both. As a result, an audio profes-
sionals decision regarding a particular microphone can be a very per-
sonal choice. Choosing the microphone with the right characteristics
for a particular application is crucial. An engineers personal aesthetic
should always be a factor. In a studio situation, the criteria for micro-
phone selection may include frequency response and accuracy; in a live
music setting, durability of a microphone may become important;
and in an outdoor film shoot or live broadcast situation, rejection of
unwanted sound becomes critical. Access to a wide variety of micro-
phones increases the sophistication and versatility of a sound profes-
sional and allows the engineer to select the ideal microphone in any
situation to address both the technical needs of the project and the
audio professionals personal aesthetic.
Much has changed since the earliest days of microphones (see Figure 5.1).
While the carbon microphones used in telephones were perfectly
adequate for the task, and in fact were technological marvels at the
time, the modern engineers demands upon microphones have become
much more exacting. An audio professional begins shaping the mix
through microphone selection and placement. A good engineer is
familiar with many of the microphones available and learns which
microphones will produce the best results in any given situation.
Converting acoustic energy to electrical energy in a microphone requires
a diaphragm, which is a surface made of a thin, flexible substance under
tension, similar to a drum skin or the paper in a kazoo, which vibrates
in response to the changes in atmospheric pressure caused by the
compression and rarefaction of molecules of a sound wave. The physical
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motion of this diaphragm, analogous to the original sound wave, is
converted into an electrical signal, which is also analogous to the orig-
inal sound wave. There are several methods of accomplishing this. A
general microphone type represents each one. Each of these types of
microphones has assets and liabilities, and each will affect both the tech-
nical and aesthetic aspects of a particular audio situation.
Dynamic Microphones
Dynamic microphones work on the principle of inductance, in which
electric current is created, or induced, by a wire or any conductor as it
moves within a magnetic field (see Figure 5.2).
There are two types of dynamic microphones, moving coil and ribbon,
although when someone refers to a dynamic microphone her or she is
usually discussing the more common moving coil variety, which we
will explore first.
In a moving coil microphone, a wire in a magnetic field is attached to
the diaphragm, which causes it to move in concert with the diaphragm.
The wire is surrounded by a magnetic field, and the free electrons
Figure 5.1 The classic Shure 55 series microphone. Photo courtesy of Shure.
Chapter 5 Mi crophones
131
within the wire begin to flow as a result of this movement, producing
electricity. To understand the relationship between electricity and
magnetism, consider an experiment frequently done in high schools
that involves spiraling a wire around a nail with electricity from a bat-
tery flowing through it, causing the nail to become an electromagnet.
In the case of the high school experiment, the nail is converting elec-
trical energy into magnetic energy, while a moving coil microphone
converts acoustic energy into electrical energy, but the principles are
similar. Since the movement of the attached wire causes the flow of
electricity in a moving coil microphone, and the diaphragm moves in
response to the sound waves in the air, the electrical signal will be an
analog of the original sound waves. A moving coil microphone, which
most audio professionals describe as a dynamic microphone, has one
end of a spring-like wire coil attached to the diaphragm while the other
end of the wire coil feeds the microphone output. Either the wire coil is
surrounded by a permanent magnet or the coil itself surrounds the magnet.
Historically, moving coil microphones do not have the best frequency
response; however, they are inexpensive and durable, making them an
ideal microphone in many instances.
Durability is not only a factor when a microphone is dropped or
thrown into a flight case to be transported; it is also related to high
sound pressure level (SPL) situations. Moving coil microphones are not
only difficult to break, they are difficult to overload with amplitudein
other words, they can be used in situations in which the sound source is
Figure 5.2 In this diagram, showing inductance in a dynamic microphone, the magnet,
which is attached to the diaphragm, moves through a wire coil, which creates an elec-
trical signal analogous to the original sound wave. Image courtesy of Shure.
132
very loud. This makes them useful for instruments with tremendous
transients, such as drums, percussion, and guitar amps, as well as for
instruments with high SPL. Their toughness also makes them ideal for
live sound, film shoots, and other work outdoors or in distant facilities
since these situations involve travel, and dynamic microphones are less
likely to be damaged in transit. Moving coil microphones, such as the
Shure SM57 (see Figure 5.3) and SM58 (see Figure 5.4) and the Sennheiser
MD421 (see Figure 5.5) and MD441, perform well on guitar amps and
drums, particularly snare drums and tom-toms, due to their high resis-
tance to overloading.
Figure 5.4 Like its counterpart, the Shure SM58 is a studio
standard dynamic microphone. Photo courtesy of Shure.
Figure 5.3 The Shure SM57, a studio and live-use workhorse of a dynamic
microphone. Photo courtesy of Shure.
Figure 5.5 The Sennheiser MD421 is a dynamic microphone offering a different
texture than the Shure dynamic microphones. Photo courtesy of Sennheiser.
133
Bass drums require even more resistance to overloading due to the
extremely high SPLs. The old-school recommendations for bass drum
microphones include the Electro-Voice RE20 (see Figure 5.6) and the
AKG D12 (see Figure 5.7) or D112.
A 421 can also be used successfully in a bass drum. The Shure SM58 is
also an ideal vocal microphone in live situations, whether music,
broadcast, film, or video, where durability is a primary factor.
Although the microphone choices outlined here are considered by
many to be studio standardsmeaning many audio professionals will
use these without thinking about it too muchthere are always new
microphones worth considering and fresh ears (yours!) that should be
making the final determination and microphone selection.
All of the dynamic microphones we have discussed thus far offer rela-
tively small diaphragms. There is a new generation of large-diaphragm
dynamic microphones that is gaining tremendous and well-deserved
respect. This new wave of microphones is led by Heil Sound, with its
Figure 5.6 An EV RE20 is a classic choice for a bass drum and is also a favorite of many
radio announcers. Photo courtesy of Stan Coutant.
134
PR20 (see Figure 5.8), PR30 (see Figure 5.9), and PR40 (see Figure 5.10)
gaining great success throughout the industry. The strong rear rejection
of all of Heil Sounds products makes them excellent in broadcast and
interview applications.
Figure 5.7 The AKG D12E attained status as the studio standard bass drum micro-
phone. Photo courtesy of Stan Coutant.
Figure 5.8 The large-diaphragm dynamic Heil Sound PR 20 is excellent on vocals, snare
drums, and drum overheads. Photo courtesy of Heil Sound.
135
Ribbon microphones also work on the principle of inductance and are
therefore another type of dynamic microphone; however, engineers will
never refer to ribbon microphones as dynamic microphonesthey are
always referred to as ribbon microphones. In the ribbon microphone,
the diaphragm is a thin, metallic ribbon, which is extremely fragile.
Figure 5.10 The PR 40, also a large-diaphragm dynamic microphone from Heil Sound,
is considered by some engineers to be positioned to replace the D12 as the studio-
standard kick drum microphone. Photo courtesy of Heil Sound.
Figure 5.9 The Heil Sound PR 30, also a large-diaphragm dynamic microphone, excels
on guitars and toms. Photo courtesy of Heil Sound.
136
This metallic ribbon is thin enough to be responsive to the vibrations in
the air. As with the moving coil microphone, the ribbon microphones
moving conductorin this case the diaphragmis suspended in a mag-
netic field, created by permanent magnets built into the microphone.
Despite their excellent frequency response, ribbon microphones are
much more delicate than moving coil microphones and perform poorly
in outdoor conditions when gusts of wind are present. They also do not
respond well to transients. A new generation of ribbon microphones has
emerged that boasts a more rugged design, but many in the engineering
community remain unimpressed by the quality of the sound captured
compared to the classic ribbon microphones. To many audio profes-
sionals, there is still nothing as good as the sound of an RCA 77DX (see
Figure 5.11) on a cello.
The most widely used ribbon microphones are older models, such as
the RCA 77DX and RCA 44BX (see Figure 5.12). They are popular
with voice-over announcers and, as with many types of classic or vin-
tage equipment, they have become sought after and expensive. Ribbon
Figure 5.11 The RCA 77DX is a classic ribbon microphone, delicate and sweet. Photo
courtesy of Stan Coutant.
137
microphones are also used on string sections and brass sections; how-
ever, close-miking is not recommended in this case because ribbon
microphones are overly sensitive to the wind produced from the
bells of some brass instruments. Ribbon microphones are commonly
used for plucked-gut or nylon-stringed instruments as well.
Condenser Microphones
Condenser microphones work on a completely different principle than
dynamic microphonesthe principle of capacitance. A capacitor is a
device that, like a battery, is capable of storing and discharging an elec-
trical charge. The turn signals or intermittent speed of windshield
wipers in a car use capacitors in their circuits, which store a charge
for a user-selected or predetermined period of time and then discharge.
Figure 5.12 The RCA 44-BX is another classic ribbon microphone, popular with
announcers and singers, and delicious on string sections. Photo courtesy of Stan Coutant.
138
Condenser microphones work on the same principle, where the stored
charge is released in a fashion that is analogous to the original acoustic
wave (see Figure 5.13).
Think of two buckets of water, each with a hole in the bottom. If these
holes are of different sizes, the water will trickle out of each at a differ-
ent rate, just as electricity will trickle out of a capacitor at a specific
rate, unique to each capacitor. The diaphragm of a condenser micro-
phone is a capacitor. This capacitor has a minimum of two opposing
platesone fixed in the rear, called the base plate, and one moving plate
that sits in front. The stored voltage is discharged depending upon the
distance between these two plates. Sound pressure waves entering
the microphone cause the front plate (diaphragm) to vibrate. As the
front plate vibrates, its relative distance to the rear plate changes.
This is the means by which the output voltage is modulated. The
Figure 5.13 The inner workings of a Neumann U 67, a popular condenser microphone.
Photo courtesy of Neumann.
139
name condenser microphone derives from old terminology, in which
capacitors were called condensers. Although some audio professionals
refer to these microphones as capacitor microphones, the vast majority
in the engineering community continue to refer to them as condenser
microphones.
The electrical charge held by the capacitor within the condenser micro-
phones diaphragm is supplied by an external source. Older models of
condenser microphones have a separate power supply unit, which is
sometimes tube-powered, about the size of a lunch pail. This plugs
into an electrical outlet to provide power to the microphones capsule
(see Figure 5.15).
Newer models receive their power directly from the audio console.
Known as phantom power (48 volts D.C.), it derives its name from
Figure 5.14 The Neumann U 67 is a popular condenser microphone, although it is no
longer in production. Photo courtesy of Neumann.
140
the fact that the power supply is no longer visible (and taking up loads
of valuable floor space). More expensive modern consoles have switch-
able phantom power on each input/output module. Moderately priced
and inexpensive consoles will have one switch to enable all modules for
phantom power. Phantom power should always be turned off while
microphones are being plugged in or unplugged, since the resulting
pop can damage monitors and other equipment. Phantom power is sup-
plied through the microphone cables in DC as noted above; it is blocked
by capacitors from entering the consoles microphone preamps, result-
ing in no effect on the audio signal passing through.
Condenser microphones are very sensitive and can break or overload
easily; however, they offer superior frequency response. Due to their
extreme sensitivity, they are considered by many to be the best vocal
microphones and are capable of picking up the slightest nuance of a
Figure 5.15 The Neumann M 149 is a tube/solid state hybrid condenser microphone,
still in production today. Photo courtesy of Stan Coutant.
141
performers voice. Condenser microphones are exceptional all-around
microphones and are commonly used on vocals, brass, woodwinds,
strings, pianos, drum overheads, and any acoustic instrument. General-
purpose condenser microphones may be inappropriate for live, broad-
cast, or film work due to their extreme delicacy and sensitivity; however,
there are specialty condenser microphones used for live and film use,
such as shotguns, which will be discussed in more detail later in this
chapter, in the Shotgun Microphones section. Neumann U 87 micro-
phones are the most widely accepted and commonly used condenser
microphones, featuring fine frequency response. Other Neumanns
include the U 89, a little brighter than the U 87; the vintage FET47
and U 47 (a tube microphone), which are both heavily sought after for
their richness of tone; and the KM series, including the KM 84, KM 100,
KM 130, and KM 140fine microphones for drum overheads and pi-
anos. The KM 130 is particularly well suited for live use (picking up the
ambience of the concert hall) because it is omnidirectional. We will dis-
cuss directionality in great detail in the Polar Patterns section later in
this chapter.
The AKG 414 is also a widely accepted condenser microphone, useful
in all situations except close-miking, as is the AKG 451, which is excel-
lent in most situations, particularly for instruments that need a strong
edge at higher frequencies, including drum overheads and toms, and
low strings, such as cello and double bass (see Figure 5.17).
Figure 5.16 Not all con-
denser microphones are
made by Neumann, as
you might think by look-
ing at the previous pho-
tos. Audio-Technica makes
this condenser micro-
phone. Photo courtesy of
Audio-Technica.
142
Lavalieres
Lavalieres are clip-on microphones typically used where an invisible or
unobtrusive microphone is needed (see Figure 5.18).
Common applications for lavalieres include use by television news-
casters and as body microphones on theatrical performers. While lava-
lieres were originally hung around the neck and frequently referred to
as lapel microphones, these microphones are now mounted on
lapels or ties, or buried in a performers costume or even his or her
wig. The term lavaliere is now used for any small microphone con-
cealed on a performer. Lavalieres also generally have a high-end boost,
used to compensate for the directionality of higher frequencies. A per-
former will never speak or sing directly into a lavaliere, because it is
mounted somewhere in his or her costume or clothing, so the higher,
more directional frequencies need to be boosted. As a result, the
Figure 5.17 A popular AKG condenser microphone, the 414. Photo courtesy of Stan
Coutant.
Figure 5.18 A lavaliere can be hidden in an actors wig
or costume. Photo courtesy of Stan Coutant.
143
microphone would be hissy and overly bright if it was used directly in
front of the mouth. There are two types of lavalieres availablemoving
coil and electret capacitor. The capacitor versions are generally of
higher quality.
Lavalieres have become accepted into standard use in the theatre, par-
ticularly in musicals. Using these microphones allows actors on stage
to have intimate moments boom out over the loudspeakers in the
house and to be heard clearly even if his or her voice does not project
well, or if he or she is not facing the audience. There has been some
discussion regarding the aesthetics of lavalieres in the theatre, specifi-
cally whether it is more pleasing to have the actor project well or it is
more efficient to have the actor amplified regardless of the strength of
his voice or the direction in which he is speaking or singing. Nonethe-
less, most directors on large productions have become accustomed to
the use of lavalieres compensating for other deficiencies in a produc-
tion. In musicals, the sound designer is expected to provide the clear,
direct sound that only lavalieres can provide. One word of caution
regarding the use of lavalieres: When an actor exits the stage, a lava-
liere goes off with him or her, rather than staying on stage the way a
stationary microphone does. To avoid extraneous noise from offstage,
make sure the lavaliere is turned down when an actor exits.
Contact microphones are another type of small, clip-on microphone.
Somewhere in between a lavaliere and a guitar pickup, contact micro-
phones are convenient in live situations when a musician moves
around or dances while performing. By using a contact microphone
in these situations, the musician will never go off microphone.
The Sennheiser MKE system of lavalieres is a generally accepted stan-
dard in theatre situations, providing consistently good results. Other
notable lavalieres include the Sony ECM 66 and 77, the Electro-Voice
CO94, and the Crown GLM-100.
Boundary Microphones
Boundary microphones are usually attached to stiff, sound-reflecting
surfaces, such as walls, floors, or desktops. Since sound is picked up
by such a large boundary, these microphones are relatively free of the
144
phase problems usually associated with highly reflective surfaces.
Common uses of boundary microphones include floor microphones
on stage, room microphones to pick up surrounding ambience, confer-
ence rooms, and courtrooms. Using boundary microphones in theatre
situations is an excellent method of obtaining a balanced sound from
all the actors and avoiding the expense and limitations of lavalieres. By
mounting boundary microphones on the edge of the stage (downstage)
at 8- to 10-foot intervals, all voices within 10 to 12 feet of the front of
the stage will be captured well. In these situations the sound designer is
hoping for cooperation from the director and the actors, hoping that
no one steps on any of the microphones.
The industry-standard boundary microphones for many years were the
Crown PZMs (see Figure 5.19). Their replacement, the Crown PCCs,
offer a superior frequency response and polar pattern.
Another notable boundary microphone is the Crown SASS, a stereo
microphone that is usually mounted on a stand rather than on a
hard surface. Although boundary microphones are not particularly
musical, making them less attractive in the studio, they are extremely
valuable when recording corporate or civil events, or when used in
theatre applications.
Shotgun Microphones
Shotgun microphones have extremely narrow polar patterns and are
usually mounted and focused on a particular location in situations in
which it is undesirable to have a microphone in view or impractical to
get close to a sound source. By eliminating ambient sound around the
Figure 5.19 The Crown PZM boundary
microphone. Photo courtesy of Crown.
145
microphone and focusing on a distant sound, shotguns can pick up
sounds at a greater distance and with greater clarity than conventional
microphones. The one drawback to shotguns is the bass frequency
response. Due to the filtering process that reduces or eliminates ambi-
ent noise, the larger wavelengths of lower frequencies are not picked
up well by shotguns, and the bottom, or low frequencies, are some-
times either muddy or absent.
Applications for shotguns include theatre productions, film produc-
tions, sporting events, sound effects gathering, and clandestine surveil-
lance. In theatre situations, the use of boundary microphones as
described here will only cover the first 10 to 12 feet of the stage. Shot-
guns hung down from the pipes above the stage can cover the rear of
the stage, or the upstage area. Due to the noise caused by light
dimmers, these microphones cannot be mounted on the same pipes
as stage lights; however, empty pipes or pipes with scenery can be
used for shotguns, which will effectively blanket the upstage area.
Shotguns can also be mounted offstage in the wings, pointed at a par-
ticular problem area on stage.
In film productions, the advantages of shotguns are twofold: Not only
are they unobtrusive and invisible when kept out of the cameras
frame, they also eliminate camera noise by focusing on the sounds of
the actors. Care needs to be taken in film situations to hold the micro-
phone steady because jostling the microphone adds noise, making the
recording unusable. At professional sporting events, look for a guy on
the sidelines running back and forth with what looks like an umbrella.
That umbrella is actually a parabolic microphone, similar in concept
to a high-quality shotgun microphone. These particular microphones
are so discerning that the operator can point the microphone (carefully
and with a steady hand) into the huddle at a football game from the
sidelines, with tens of thousands of fans screaming their heads off, and
still capture the conversation going on 50 or 100 feet away. This is also
why these shotgun microphones are excellent for clandestine surveil-
lance, or spying, and for sound effects gathering. In both of these sit-
uations the audio professional does not want the subject to be aware of
his or her presence, and the long-range quality and capabilities of these
microphones allows the recordist to remain anonymous.
146
The Sennheiser 816 is an industry standard among long-reach shot-
guns. It features excellent rejection of incidental sound and a sharp
boost from 5 to 15 kHz. The Sennheiser 416 is, similarly, the standard
for short-reach situations, where the sound source is closer to the
microphone but not close enough for conventional miking techniques.
The Neumann KMR 82, also a long-reach microphone, is superior in
music recording from a distance. The AKG 451 CK 9 is also an excel-
lent long-reach shotgun, while the AKG 451 CK 8 works well in short-
reach situations.
Wireless Microphone Systems
Wireless microphone systems are used anytime the use of a micro-
phone cable would be confining or unsightly. Along with theatre appli-
cations, they have become very popular in sound reinforcement,
particularly when the performer wants the freedom to dance or
move around the stage (see Figures 5.20 to 5.22).
Figure 5.20 A Telex FMR-1000 wireless microphone system. Photo courtesy of Telex.
147
Wireless microphones broadcast on specific radio frequencies and
require a transmitter, antenna, and a receiver to broadcast and pick
up the signal for routing to an audio console (see Figure 5.23).
These microphones must be in the line of sight of the antenna attached
to their receivers, and the batteries must be changed regularly. When
used with a lavaliere, a wire runs from the microphone to a body trans-
mitter with antenna, which broadcasts to the receiver. For handheld
Figure 5.21 The EV REV wireless microphone system. Photo courtesy of Electro-Voice.
Figure 5.22 A wireless microphone system by Shure. Photo courtesy of Shure.
148
wireless microphones and wireless headset microphones, the transmit-
ter and antenna are usually incorporated into the microphone itself.
Because the signal broadcasts in the UHF and VHF bands, there is a
risk of interference by ham radio operators or CB operators.
Polar Patterns
A polar pattern is a map of a microphones directional sensitivity,
graphically depicting the way a microphone will respond depending
upon the position of the sound source relative to the microphones
position. Also known as pickup patterns, there are three basic types:
omnidirectional, or nondirectional, which is equally sensitive in all
directions; bidirectional, which is sensitive only to the front and
back; and unidirectional, or directional, which is sensitive only in
the front. Cardioid is a commonly used name for a directional micro-
phone with a heart-shaped pickup pattern, as are super-cardioid,
hyper-cardioid, and ultra-cardioid for shotgun microphones with
tighter heart-shaped patterns.
A microphones directionality is determined by the microphone type,
since certain patterns are inherent to certain types of microphones, and
affected by openings on the sides or rear of the microphone, which
allow phase cancellation of sounds from certain directions. Openings
Mixer
Wireless
Receiver
Wireless
Microphone
Transmitter
Figure 5.23 This is the basic theory of wireless microphones. Based on an original
diagram courtesy of Sennheiser.
149
close to the diaphragm cancel high frequencies, while openings farther
from the diaphragm cancel lower frequencies.
Moving coil microphones are inherently omnidirectional (see Figure 5.24).
Adding ports to the microphone casing and adding an acoustic phase
shifting network creates a cardioid pattern (see Figure 5.25). This causes
sound waves approaching the microphone from the rear to impact both
sides of the diaphragm. These sound waves will have their phase
reversed on either side of the diaphragm and therefore will be greatly
reduced in intensity. This increasingly reduces the microphones sensi-
tivity as the sound source moves off axis, with its minimum sensitivity
located at 180 degrees off axis (the rear of the microphone).
Ribbon microphones are inherently bidirectional, making them most
sensitive to sounds entering the microphone at 0 degrees and 180 degrees
and least sensitive to sounds entering at 90 degrees and 270 degrees
(see Figure 5.26). This pattern can be valuable when recording an inter-
view or two singers singing a duet.
0
Figure 5.24 The omnidirectional pattern shown means that the microphone is equally
sensitive in all directions. This pattern is inherent to both dynamic and condenser
microphones, but other patterns are easy to create. Based on an original diagram cour-
tesy of Sennheiser.
150
Condenser microphones are inherently omnidirectional; however, they
can easily be made multidirectional, offering a variety of patterns,
based on which plate is charged. Many condenser microphones have
a third plate, and by variously reversing, reducing, or eliminating the
0
Figure 5.26 The bidirectional pattern is most sensitive to the front and rear of the
microphone and least sensitive to the sides. Based on an original diagram courtesy of
Sennheiser.
0
Figure 5.25 The cardioid pattern is most sensitive to the front and least sensitive to the
rear. It is a common pattern for vocals and instruments when separation between
sound sources is desirable. Based on an original diagram courtesy of Sennheiser.
151
charge to selected plates of the three plates, bidirectional, omnidirec-
tional, and cardioid can all be made available.
Critical Listening
An important concept in the entire field of audio, which becomes par-
ticularly significant when discussing microphones, is how to listen,
understand what you are hearing, and trust your ears. Learning critical
listening takes time, but it begins by simply listening to the sounds
around you. Next time you walk up a concrete stairway, listen to the
reflections of your footsteps bouncing off the walls. Listen to the com-
bination of the reflections and the footsteps themselves. Next time you
walk through the woods or down a city street, listen to the layers of
sounds. If you are in the woods, what do you hear close to youyour
footsteps as they crush leaves and twigs, perhaps the sound of a nearby
stream? What do you hear that is a little more distantthe chirping of a
bird, the croaking of a frog? How about in the distance, perhaps an ani-
mal moving through the woods or the wind quietly rustling the leaves
atop the trees? If you are in a city, stand on the street and listen. What
sounds are closetraffic noise and perhaps peoples voices? A little fur-
ther away do you hear the sound of a bus or a car without a muffler a
block or two away? Perhaps a siren in the distance? And what about
that underlying ambient rumble that most cities seem to have; if you
listen at a moment when there are no voices and no traffic, does the
city have a certain basic sound, perhaps a combination of distant sounds
that combine to become indistinguishable? Identifying and analyzing
these layers is the beginning of ear training, or learning how to use
your ears most effectively.
Much in the same way that you can tear apart the layers of sound
described a moment ago, you can dissect what you hear in an audio
situation. Instead of simply accepting the complex, aggregate sound,
an audio professional is constantly listening to the layers that comprise
that sound. In addition to tearing apart the layers of sound on the basis
of loudness as described a moment ago, we analyze the frequencies and
locations of the various sounds that arrive at our ears. What is the
highest sound in pitch of everything we are hearing? Perhaps its the
chirping of birds, the wail of a siren, or the squeal of faulty brakes.
152
We also ask where sounds are coming from. Is the full-frequency,
white noiserich sound of the wind moving from left to right? Is it
coming from behind us? If we were able to change the timbre of the
wind, would it be easier to carry on a conversation on our cell phones?
These are the same questions we will ask in a studio or a theatre
throughout the audio process. When we compare two microphones on
a single instrument, the criteria that determine which microphone we
will ultimately use are the same. What frequencies are we hearing? Are
we hearing enough of the high and low ends of this particular instru-
ment? Are we hearing too much of any particular range of frequencies?
Does it sound as good through the microphone as it does when we
stand in the room next to the instrument? How do we know that?
These questions are not always simple to answer; often it is a matter
of comparing sounds between different microphones until we deter-
mine which one, or which combination of microphones, sounds best
to our ears. And the answer to the final questionhow we know that
is simple. We answer all these questions by keeping our ears wide open
and using them in every situation, discerning and dissecting all the
information we are given.
This is not simple to learn, but the way to start is by using your ears
critically every day. Use your ears to listen to and analyze everyday
sounds. Every sound event can be analyzed; do not pass up any oppor-
tunity to pick apart a sound, especially complex sounds that offer
interesting entertainment. This will enable you to listen more critically
in the studio, on the set, or in a live situation. There is little right and
wrong in critical listening; it is subjective. The ultimate goal is simply
to understand what you are hearing; increase your understanding of
what you hear and increase your trust in your ears. Trusting your
ears is the ultimate aesthetic goal.
Microphone Placement
Based on our ideas of critical listening, we know how crucial micro-
phone placement can be. Where a microphone is placed on an instru-
ment is as critical a decision as which microphone the audio
professional has chosen. Musical instruments do not always produce
153
sound in the way one would expect, so it is important to understand
the advantages and disadvantages of various locations on various
instruments. The most important criterion in choosing and placing a
microphone is listening. If one microphone does not produce the
desired sound, try another. If you are unsure of the best location to
place a microphone, have the musician play while you circle the instru-
ment, listening closely to select the best location. Odds are that the
microphone will sound best in the same location where it sounds
best to you. That being said, here are some ideas for various
instruments.
There are numerous ways to place a microphone on a drum set. The
simplest form, placing two microphones at a slight distance, will give a
nice airy sound, but the bass drum and snare will lose much of their
power. With jazz, bluegrass, or folk this may be adequate, but with
rock or dance the snare and bass drum, or kick, are critical because
they drive the tune. As such, most modern recording professionals
favor a minimum of four microphones on a drum kit, one each in
the kick and snare and two at a distance. If you have enough microphones
Figure 5.27 The Shure Model VP88 stereo middle-side microphone. Photo courtesy of
Stan Coutant.
154
and a console that is capable of enough inputs, the ultimate setup would
include one microphone each on the kick and snare, one for the hi-hat,
one on each tom-tom, and two overheads.
When placing microphones on skinned drums (everything except the
overheads), always use a dynamic microphone so close to the drum
head that it is almost touching. Aim the microphone at an angle to
prevent standing waves from occurring between the head of the micro-
phone or the diaphragm and the head of the drum. For overheads, bet-
ter results are achieved by thinking of each microphone as picking up
half of the drum kit, rather than simply miking the cymbals. For the
left overhead (the engineers perspective, not the drummers), think of
the microphone as the center of a triangle described by the toms and
the cymbal. Similarly, think of the right overhead as the center of
a triangle described by the snare, hi-hat, and cymbal. Also be aware
of the potential for phasing between microphones. Left and right over-
heads can cause phasing unless they are tilted away from each other,
and the right overhead can have a poor phase relationship with a hi-hat
microphone if it is placed too high. Generally overhead microphones
placed at a height of about six feet are pretty safe.
For a kick drum, an AKG D 12, a Heil PR-40, a Sennheiser 421, or an
Electro-Voice RE20 is an ideal choice. Any of these microphones will
avoid the problems of overloading. Place the mic deep into the drum
and remember to angle it against the skin. For a snare, try a Shure SM58,
a Heil PR-20 or a Sennheiser 421. On the hi-hat a Shure SM57 is a
good choice; it is identical to the SM58, except it is more directional,
giving better separation between the snare and the hi-hat. Any of the
Neumann KM series would also be a good choice for the hi-hat. Try a
Sennheiser 421 on the toms, or, for a brighter, crisper sound, use an
AKG 451 or a Heil PR-30. On the overheads, a Neumann KM 84 or
an AKG 451 is always a good choice, as is a Neumann U 87 or an
AKG 414.
An electric bass is often best miked without a microphone. Using a
direct box on a bass, bypassing an amp entirely, not only gives a supe-
rior sound, with a rich, full bottom and a lot of pop on top, it also
avoids the potential problem of the bass amp bleeding into all the
155
other microphones in the room. Many engineers prefer to split the bass
signal in a DI, sending the clean signal to a microphone input and
sending the high-impedance output to a bass amp so it can be miked.
Sometimes in a mix the direct signal and the miked amp can be
blended successfully.
With an electric guitar, use of an amplifier is necessary since taking a
guitar direct results in a thin sound. When miking a guitar amp, use a
Shure SM58, a Heil PR-30, or a Sennheiser 421, less than an inch from
the speaker and on an angle. This will maximize the depth of the sound
of the amp, while the angle will prevent a standing wave from forming
between the diaphragm of the microphone and the speaker cone in the
amplifier.
With an acoustic guitar, the sound will vary depending on the guitar
itself. The base of the neck, just above the sound hole, is often a sweet
spot, as is the bottom of the guitar. Avoid the sound hole itself because
this is often too reverberant and muddy to record well. As with most
instruments, a good audio professional will walk around the instru-
ment listening closely to decide where it sounds best. Try a Neumann
U 87, a Heil PR-30, or an AKG 451. In a pinch, a Sennheiser 421 will
work well.
Pianos offer several interesting problems. If there are no other musi-
cians playing in the room, open the piano to full stick and record it
both close and from a distance. If there are other musicians, close the
piano to half stick, microphone it close, and throw a few blankets over
it to avoid leakage into the piano track from other instruments.
Remember where the sound comes from. The vibrations of the strings
may create the sound, but the richness of the sound board gives a piano
its fullness. When you need to place microphones inside a piano on
half stick, you may be better served by placing them close to the
sound board rather than pointing them at the strings. Engineers use
a variety of microphones on pianos, although virtually any decent con-
denser microphones will do the job nicely. Try a pair of Neumann
U 87s or AKG 414s, one over the high end (where the short strings
meet the sound board) and one over the low end (where the long
strings meet the sound board). Compare the sounds of miking the
156
sound board or miking the strings themselves: Each has its advantages,
and what sounds best will often be determined by the type of piano
and the style of music.
Brass instruments provide us with a bit of a paradox. On one hand they
can easily overblow condenser and ribbon microphones; on the other
hand, condenser and ribbon microphones accentuate their sound.
A condenser microphone, such as a Neumann U 87 or FET47 (see
Figure 5.28), or a ribbon, such as an RCA 77, placed two to three
feet from the horn of the instrument solves this problem.
As long as the microphone isnt too close, the problem is solved. Brass
instruments produce sound from the horn, making microphone place-
ment obvious and easy.
Woodwinds also sound best with ribbon or condenser microphones,
such as a Neumann U 87. However, the sound does not emanate
from the horn; it comes from the finger holes. As such, miking the
bell of a clarinet will give an inferior sound to miking the finger
holes toward the top of the neck. This is also true for saxes, oboes,
bassoons, flutes, and piccolos.
With high strings, such as violins and violas, the best sound is captured
from about two to three feet above the instrument, using ribbon or
condenser microphones. Neumann U 87s or U 89s, RCA 77s, and
AKG 414s are all good choices. Cellos tend to sound best with a
Neumann FET 47 or other good condenser microphones with a strong,
round low end set about one foot or less in front of the instrument.
Figure 5.28 A Neumann FET 47,
as pretty to look at as it is to lis-
ten to. Photo courtesy of Stan
Coutant.
157
For a double bass, try a Neumann KM 84 or an AKG 451, which will
both accentuate the high end of the instrument and reinforce the strong
bottom. Like a cello, a double bass should be miked from in front of the
instrument, from a distance of one foot or less.
Due to the wide scope of percussion instruments, it would be impos-
sible to include them all with specific instructions. They can, however,
be divided into three groups: mallets, including marimbas, vibes, and
xylophones; skins, including tympanis and congas; and toys, including
tambourines and shakers. For all percussion instruments, condenser or
ribbon microphones can be used, as long as care is taken to avoid plac-
ing them too close to the skins. A pair of Neumann U 87s placed four
or five feet above the percussionists setup will work nicely as he moves
back and forth through a piece, playing tympani sometimes, marimba
other times, and triangle still other times.
The ultimate rule, as stated earlier, is to use your ears. Each situation,
whether miking to record or miking an orchestra pit for a live perfor-
mance, is unique. Never allow yourself to be so confident with your
setup that you stop listening and start relying on what you think you
know. That sometimes happens to old engineers shortly before they
are taken out to pasture. Never stop listening. Always take the time
to listen critically.
Exercises
1. Have a friend who is a musician play an instrument. Listen
carefully. Move your head around the instrument to see where
the sound is. Is the sound fuller from the front, back, or side of
the instrument? Is the sound deeper or brighter in one location
as opposed to another? Where does the instrument sound
best to you? Where would you place a microphone as a result
of this information?
2. Place a microphone on the instrument and listen through the
microphone. Does it sound different through the microphone?
Move the microphone to different locations. Does the sound
change?
158
3. Place a different microphone on the same instrument. Does
the sound change? Try different locations. Is the location
determined to be best by your ears also the best location
with a microphone?
Additional Reading
2004.
Borwick, John. Microphones: Technology and Technique. London:
Focal Press, 1990. Excellent guide to microphone usage.
Burroughs, Lou. Microphones: Design and Application. Plainview
NY: Sagamore, 1974.
Clifford, Martin. Microphones, 3rd ed. Blue Ridge Summit, PA: Tab,
1986.
Springer, 2005.
Press, 1993.
CA: Alfred, 1988.
Press, 1989.
159
Press, 2000.
Sams, 1989.
160
6
Speakers and Amps
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W
e now have a good idea of how and why microphones
work. We also know the importance of listening when we
determine the subtle differences between these different
microphones. To make these judgments, the devices we use to repro-
duce audio information are at least of equal importance to the devices
we use to capture sound in the rst place.
The production room monitor speakers are considered by some to be the
most important components in the studio system. Most consumer audio-
philes will be happy to tell you that they spent more on their speakers
than on any other single component in their systemsometimes
more than they spent on every other component combined. And for
good reasonthe monitor speakers are the only components that output
any sound and tell us what we need to know. The audio professional is at
the mercy of the sound output by the speakers. Without the monitor
speakers, the engineer has no practical insight into the character of the
signal that is being recorded or reproduced and cannot judge either the
aesthetic aspects or the technical accuracy of the signal. It is difcult to
imagine that there is any single change that will alter the character of a
studios sound more than a change of monitor speakers. The choice of
Figure 6.1 The musikelectronic geithain (MEG) RL 901 Studio Reference Monitor offers
accurate sound in a wide range of frequencies. Photo courtesy of MEG.
162
monitor speakers is therefore one of the most important decisions made
in the design of a studio, and it can be a very personal decision. It is of
primary importance that the monitor speakers give an accurate picture of
all audio signals as they are recorded, reproduced, and manipulated. If
the other components in the system are not of the highest quality, an
accurate monitoring system will show the signal as it is recorded or
reproduced, demonstrating any deciencies introduced by lower-quality
gear in the signal chain. If the monitor system does not give an accurate
picture, the engineer will be operating at a disadvantage and may be
laboring under false impressions. As such, it is critical that we, as
audio professionals, not only know what we like in a monitor, but
also that we understand the nature and theory of monitor speakers.
Theory of Operation
We can think of speakers as reversing the process undertaken by
microphones, since speakers perform the opposite function of micro-
phones. Microphones are transducers that convert acoustic energy into
electrical energy, while speakers are transducers that convert electrical
energy into acoustic energy. A simplistic way of demonstrating and
understanding microphone and speaker system design is to look at
the original design for the telephone. There are two basic components
to the telephone: a microphone and a speaker. The original micro-
phone used was composed of a diaphragm ush against a capsule lled
with carbon granules, which acted as a variable resistor between a bat-
tery and an electromagnet. When someone spoke into the mouthpiece,
the diaphragm began to vibrate. The vibrations caused compressions
and rarefactions of the loosely packed carbon granules. As the density
of the carbon granules changed, so did the amount of voltage supplied
by the battery passing through the capsule. The modulations in the size
of the signal passing through the capsule were a direct analog of the
sound-driven vibrations of the diaphragm. When the size of the voltage
uctuated, so did the intensity of the electromagnet. In other words,
the sound wave hit the diaphragm, the diaphragm moved in a way that
was analogous to the wave, and that movement was reected in the
carbon granules. With the help of a battery and an electromagnet, this
movement was converted into electricity.
Chapter 6 Speakers and Amps
163
Next to the electromagnet was another diaphragm, the speaker. This
diaphragm was pulled by the electromagnet when it became strong and
released when it became weak. The uctuations in voltage caused uc-
tuation in the strength of the magnetic eld, which in turn caused the
diaphragm attached to the magnet to vibrate. These movements repli-
cated the movement within the carbon granules, so these vibrations
were also an analog of the original sound waves.
Obviously technology has come a long way since the days when
Alexander Graham Bell invented the telephone. The telephone was bril-
liant in its time, but the understanding of speaker technology it offers is
simplistic. Although a speaker is still essentially a magnetic force of
some type modulating the movement of a diaphragm of some type,
there are now so many ways in which a speaker can accomplish this,
much as there are different ways that different types of microphones
attain the same goal. Still, the underlying technology to remember is
that a magnet is incorporated to convert electrical energy into physical
movement, which causes the mechanical motion of the diaphragm that
imitates the sound wave. The movement of the diaphragm causes com-
pressions and rarefactions of the molecules of the surrounding medium,
which is acoustical energy that we perceive as sound.
Another way of looking at this is that a speaker contains a voice coil,
which is a coil of wire, attached to a cone-shaped diaphragm made of
materials such as paper, mylar, or polypropylene. When electrical
energy moves through the coil it becomes an electromagnet, which
Figure 6.2 The Focal SM8 is a two-way near-eld monitor featuring an 8-inch bass/mid
driver with passive radiator and a 1-inch Beryllium tweeter. Photo courtesy of Focal.
164
pulls and pushes against the large magnet mounted toward the rear of
the diaphragm. This movement in the voice coil pulls and pushes the
diaphragm in a manner that is analogous to the original wave repre-
sented by the electrical current. The resultant movement of the dia-
phragm causes compression and rarefaction in the air molecules,
which creates the acoustic sound wave.
Moving Coil Speakers
Like microphones, there are three basic speaker design types: moving
coil, ribbon, and condenser, or electrostatic. Moving coil speaker design
is by far the most common among consumer products. Virtually all of
the speakers that the average person will ever encounter are moving coil
speakers. This design is so popular because of its physical toughness.
They are, generally speaking, difcult to break and, like their micro-
phone counterparts, they can handle high SPL efciently. Moving coil
speakers consist of a wire coil suspended in a magnetic eld attached to
a diaphragm, very much like a moving coil microphone. In this case the
diaphragm pushes the air to initiate the sound waves. When current is
run through the wire coil it becomes an electromagnet of varying inten-
sity and polarity. This causes it to move as it attracts or repels the per-
manent magnet. Amplitude is duplicated in the speaker by the amount
Figure 6.3 Several Meyer Sound monitors. Photo courtesy of Meyer Sound.
165
of current present in the voice coil, because greater current results in
greater movement by the coil, which in turn results in greater movement
of the diaphragm, which translates into stronger compression and
rarefaction.
The quality of moving coil speakers should not be underestimated. They
are still considered by most audio professionals to be the ideal design for
a woofer, where the larger wavelengths create grosser movements within
the cone. Their toughness becomes a key attribute when dealing with the
low frequencies that we direct toward the woofer.
Ribbon Speakers
Ribbon-style drivers are found primarily in tweeters, speakers that
handle high frequencies, and occasionally in mid-range drivers. Some
consider them more accurate in higher frequencies, but among con-
sumers they are less common than moving coil speakers because
they are not as versatile. Although they are very sensitive, they cannot
tolerate the high sound pressure levels required to reproduce low
frequencies, which is why they are relegated only to speakers that
exclusively handle higher frequencies. They are most often found in
high-frequency drivers where extra sensitivity is needed, including stu-
dio, theatre, and audiophile applications. These elements work well in
these situations where they receive only the highest of frequencies since
the amount of voltage applied to high-frequency drivers is usually
relatively small. These speakers are well suited for the ner movements
and lower sound pressure levels required by higher frequencies.
Ribbon speakers consist of a thin metal strip that acts as a diaphragm
surrounded by the magnetic eld created by a permanent magnet. Oth-
erwise, they are similar in theory to the moving coil speaker. As tech-
nology marches on, there are many new ribbon tweeters entering the
market every yearsome to great acclaim. As with microphones, the
audio professional needs to get comfortable with a variety of speaker
types, both to establish his or her own preference among the huge
selection available and to be adept at any of the popular models an
engineer may encounter in any studio.
166
Electrostatic Speakers
Electrostatic, or condenser, speakers are a high-end consumer format
and are almost as rare as ribbon speakers. The operating principles
require a knowledge of electronics beyond the scope of this book; how-
ever, in principle they are similar in design to condenser microphones,
where the varying distance between charged plates releases an electrical
charge that is analogous to the original sound wave. Just like condenser
microphones, they require a charge; in this case they receive their cur-
rent from an A.C. outlet. The variations in distance between the
charged plates cause the diaphragm, which is far larger in electrostatic
speakers than in other types, to push and pull, thereby simulating the
sound wave. In other words, the electrical energy is sent to a moveable
plate. The varying distances between the moveable plate and a xed
plate cause capacitance, or the storing and discharging of current, to
occur. This in turn translates into physical movement of the diaphragm,
which moves in a fashion that compresses and rarees the surrounding
molecules in air in a manner that is analogous to the original electric
signal. It is only a matter of time until these appear more commonly in
professional settings, but as of now they are rarely used professionally.
Woofers and Subwoofers
One speaker alone will not simultaneously reproduce the entire
audible spectrum accurately, because low-frequency vibrations in
the speaker will distort the smaller high-frequency vibrations if a
Figure 6.4 The Superior Line Source (SLS) 8290 with a planar ribbon high-frequency
transducer can double as a compact PA and as a stage monitor. Photo courtesy of SLS.
167
single speaker is used to reproduce them at the same time. Due to
this, high-end monitoring systems consist of cabinets containing
numerous speakers. As a general rule of thumb, the greater the number
of speakers in a cabinet, the smaller the range of frequencies that
each speaker is required to produce, and the more accurately it will
do so. It is not unusual to nd speaker cabinets that contain up to
four different speakers, each reproducing a specic part of the audi-
ble frequency spectrum.
While three-way systems are common for home use, many studios pre-
fer the greater accuracy of two-way systems for their near-eld mon-
itors, nding the additional mid-range push unnecessary. Then they
use huge studio monitors containing four or even ve speakers in
order to remain condent that they can hear the entire audio spectrum.
The woofer (and subwoofer) is responsible for reproducing the lowest
frequencies in the spectrum. The range of frequencies output by the
subwoofers begins as low as 16 to 30 Hz and can reach as high as
200 to 750 Hz. Subwoofers are used in more expensive speaker sys-
tems and in theatre, home theatre, surround sound, and sound rein-
forcement applications. They are the speakers that give that deep,
bass push and the oomph of the explosion onscreen in a movie, or
the thump you feel in your chest on the dance oor. If your upstairs
neighbor has a subwoofer, it could be the reason why your glasses keep
falling off the shelf. The highest frequency reproduced by the sub-
woofer will be in the range of 100 Hz.
Figure 6.5 The SLS LS8800 is a full-range bi-amped true line source array module. The
LS8800 high frequency section features a high performance planar ribbon transducer.
Photo courtesy of SLS.
168
Mid-Range Drivers
Mid-range drivers reproduce the frequencies that fall in between those
produced by the woofers and the tweeters. These frequencies range
from about 200 Hz to 750 Hz on the low end and reach as high as 1 kHz
to 5 kHz. This means that the mid-range drivers will not only give a little
Figure 6.6 The SLS LS8695AX is a bi-amped true line source array column, which pro-
duces an extremely tight vertical sound eld. Photo courtesy of SLS.
169
extra mid-range push, they will also share some of the burden of covering
all the frequencies along with the tweeters and the woofers. While an
excess of mid-range sometimes sounds appealing to consumers, as engi-
neers we often nd that there is enough mid-range audio information
available without the additional mid-range driver.
Tweeters
Tweeters reproduce the highest frequencies of any given speaker array.
The frequencies output by tweeters can be as low as l kHz to 2 kHz and
as high as 16 kHz to 22 kHz. The tweeters are usually the most delicate
and sensitive speakers in a system, because they are required to make
ne, subtle movement in order to reproduce the highest frequencies. For-
tunately, it takes very little energy to reproduce higher frequencies, which
makes it more difcult to blow them up under normal operations.
Having said that, we must always take care not to overload them. The
occasional feedback, clients who want to hear mixes played back at 11,
and other unfortunate situations can affect tweeters adversely. When an
analog tape is played at faster speeds, the frequency of the recorded
information will be increased, as discussed in Chapter 3. This increases
the amount of energy handled by the tweeters. The engineer must be
careful when fast-winding a tape due to this increase; when a tape is
being fast-wound, the monitor level should be lowered to prevent the
tweeters from becoming overloaded and possibly blowing up. These are
all good reasons to install a fuse on the inputs to your speakers; it is far
easier and cheaper to replace a blown fuse than it is to replace a blown
tweeter.
Figure 6.7 A Seismic Audio Titanium Horn Tweeter supplies high-frequency informa-
tion. Photo courtesy of Seismic Audio.
170
Crossovers
The crossover, also known as the frequency dividing network, is the
circuit component that identies and splits an input signal into sep-
arate frequency bands. Each of the separate bands is then routed to a
particular speaker typethe tweeter, the woofer, the subwoofer, or
the mid-range driver. The crossover frequency, which is the dividing
point between frequency bands, is either preset or, in high-end appli-
cations, user-selected. Some audio professionals will adjust their
crossover frequencies to their advantage, especially in sound rein-
forcement situations, to improve the sound in the auditorium, club,
or stadium.
The crossover also allows control of the volume of each separate
frequency band; in particular, it will boost the output routed to
the tweeters. Because energy is not equal across the frequency spec-
trum, the crossover needs to amplify higher frequency signals to
keep them sounding equal to our ears. Sometimes the crossover
will supply a preset increase in amplication to higher frequencies;
other times the user will have access to a potentiometer (through a
dial mounted externally on the speaker cabinet), which will control
the amount of high-end boost supplied by the crossover. While
many audio professionals will agree to work at the speaker manu-
facturers recommendation by leaving that potentiometer at zero,
others will boost the amplication of the high end, either to com-
pensate for a room deciency or to compensate for their own high-
end hearing loss.
Studio Monitors
Audio professionals will generally have a preference regarding a par-
ticular speaker they like to use in a particular situation. Some engineers
will use many different speakers while mixing. All of these different
speakers, different models from different manufacturers, can be bro-
ken down into three types of monitor speakers commonly used during
production. Three types of monitor speakers are used so that the engi-
neer has a means of simulating various listening environments, such as
a theatre, a car radio, or a home stereo system, to name a few. Home
171
stereo systems vary tremendously in quality, as do theatre spaces. Part
of the engineers job is to make sure that the production will sound
good regardless of what type of speaker system the end user favors.
Using all three types of monitors allows the audio professional to
anticipate any listening environment in which the production will be
reproduced.
The rst of these three types are studio monitors, which are enormous
wall-mounted boxes containing numerous drivers. The studio moni-
tors provide the most accurate picture of the entire audible spectrum.
Their range is one to two octaves deeper than the average listener has
at home, and they simulate a large, well-equipped theatre efciently.
Clients love them. There is nothing ashier than high SPL shooting
out of those huge, soft-mounted boxes. The high-frequency response
is far better than that of a typical home stereo system. The studio
monitors sound so different from home stereo systems, which are usu-
ally designed to sound good rather than sound accurate, that it takes
training to understand how to listen to them correctly. Always keep in
mind, while listening to production on studio monitors, that you are
hearing a greater frequency spectrum than your end user will hear. For
this reason, studio monitors can be quite deceiving, yet they are
invaluable for detecting poor quality or defective gear, particularly
at the highest and lowest ends of the audible bandwidth. So these
monitors will leave no frequency unturned and will offer the audio
professional a true look at the audible bandwidth, warts and all,
but they will not tell us about the average listener and the experience
that listener can anticipate at home.
Figure 6.8 UREI 813 studio monitors. These 400-pound boxes
were state of the art through the 1970s and into the 1980s
and are still favored by many in high-end commercial studios.
Photo courtesy of UREI.
172
Bookshelf Speakers
Since it will be impossible to determine how the sound heard from the
mix or production will translate to the equipment found in other set-
tings if only studio monitors are used, we need to explore other
options. It is possible to make mixes that sound good only on studio
monitors. The extended high and low frequency response can be used
to make deeper and brighter sounding mixes that cannot be repro-
duced by the type of equipment typically normally available to the
end user. For this reason, more than one type of monitor must be
used in the studio or production room.
In addition to the studio monitors, home-quality bookshelf-style mon-
itors are used. This enables the audio professional to hear exactly how
a mix will sound on normal speakers when the consumer listens at
home, at work, or when production work is played back in a small
theatre. Bookshelf monitors give the engineer the best possible infor-
mation about how a mix will translate onto a wide variety of speakers,
and are therefore one of our most valuable tools while mixing. Their
accuracy also holds up at low listening levels, making them invaluable
when concerns arise about the relationships of the various frequency
bands at various listening levels, as mentioned previously during our
discussion of the Fletcher-Munson Equal Loudness Contours.
Figure 6.9 A Tannoy Precision 8D near-eld monitor offers the audio professional a
listening experience that mimics the end users experience. Photo courtesy of Tannoy.
173
Sound Cubes
Sound cubes are small cabinets housing a single speaker, ranging from
four and a half to six inches in diameter. This is not considered a high-
quality speaker; however, it fullls an important function. Sound
cubes are used so an audio professional can hear how a mix will
sound on inexpensive stereo systems, clock radios, boom boxes, tele-
visions, and car stereos. The sound cube is used so the engineer can be
sure the mixes will translate well in any listening situation. Generally
speaking, if it sounds great on all three sets of speakersstudio mon-
itors, bookshelf speakers and sound cubesit will sound great
anywhere.
Sound cubes are also frequently used for the mono compatibility test.
We will discuss mixing in great detail in Chapter 10, Mixing, and
we will discuss the need for an engineer to ensure that the mix will
work regardless of the playback format versus the original mix format.
To accomplish this, the audio professional needs to collapse the mixes
to ensure quality playback in every possible format. If an engineer has
mixed in stereo or surround, he or she still needs to check that the mix
will sound good if it is played back in mono (and that no unnoticed
phasing has occurred). This is commonly done on sound cubes through
a mono bus.
Figure 6.10 Avantone sound cubes are a higher-quality replacement of their prede-
cessor, the Auratones. Their purpose remains to inform the engineer of what the mix
will sound like on systems that do not live up to bookshelf or studio monitors, and they
are convenient for the mono compatibility test. Photo courtesy of Ampeg.
174
Enclosures
Some speaker designers will spend more time designing and construct-
ing the enclosures than they will on the speakers and the crossovers
combined. Enclosures are the boxes that hold the speakers. Designers
will consider the type of wood or other material, the nish of the wood,
and the manner in which different enclosure pieces are attached. All
places where wood meets will be glued and screwed to eliminate errant
vibrations. Much as a luthier will carefully consider every aspect of the
wood, glue, varnish, and so on as he or she creates a guitar or violin, any
decent speaker designer will focus on every detail of the construction of
the speaker enclosures. Much in the way that a great concert hall
becomes part of the orchestra performing within, enhancing the audi-
ences experience with the reections it adds, the enclosure enhances the
output of the speakers mounted within.
Choosing the right type of wood will help the designer to achieve a
particular sound, enhance the resonance, and accentuate certain fre-
quencies, which give a particular pair of speakers its own character-
istics. Often there will be a hole drilled in the front of the enclosure,
called a bass port, which will enhance the projection of the bass fre-
quencies. Enclosures are sometimes lled with specic materials, such
as insulation, to accent or attenuate certain frequencies. Other times
air is removed from the enclosure, creating a vacuum.
Line Arrays
So far we have discussed speakers and monitors with an eye toward the
production room. Lets take it out into the eld, or perhaps the audi-
torium, club, or stadium. A well-equipped production room may sport
one pair each of three different types of monitors, perhaps even a 5.1
surround sound system with six speakers of its own. When we go into
sound reinforcement or providing audio support for live sound, things
can get much bigger.
If an audio professional is providing sound reinforcement in a club or
a small auditorium, he or she may choose to go with a few well-
placed speakers. In the event that the audio professional is reinforcing
sound in a large auditorium or stadium, the line array is the current
choice (see Figure 6.12).
175
Figure 6.11 A line array installation at Royce Hall featuring SLS RLA/2s. Photo courtesy
of SLS.
Figure 6.12 Dual EAW KF760/KF761 line arrays utilized to provide main sound rein-
forcement on the 2006 world concert tour by Iron Maiden. Photo courtesy of EAW.
176
The move to line arrays has made large systems more efcient and
more articulate. Older systems featured more horns and compression
drivers that dispersed the sound widely. Newer line arrays offer a nar-
rower vertical pattern, allowing the audio professional to better tailor
his or her monitors to the needs of the house. These systems have
become so popular that even some smaller venues have invested in
compact line arrays (see Figure 6.13).
Efciency, Frequency Response, and Distortion
When comparing speaker systems, our rst responsibility is to our
ears. Does the system sound accurate? Is the response at (meaning
honest) or does it hype certain frequencies, adding its own coloration
or timbre? Do we like what we hear? Can we trust this system to
deliver the information we need? Beyond our aesthetic understanding,
judgment should be based on certain technical criteria as well. How
Figure 6.13 An EAW KF730 compact line array own at the new Fine Arts Auditorium
of East Kentwood High School (EKHS), East Kentwood, Michigan. Photo courtesy of
EAW.
177
much power is this speaker able to handle? How efciently does it
reproduce the signal sent? Are all the audible frequencies reproduced?
Will the signal sent to the speakers be reproduced with both adequate
bandwidth and dynamic range? Lets start with efciency.
Simply stated, an efcient speaker will utilize a greater percentage of
the signal sent from the amplier. We measure this signal in dB SPL, or
sound pressure level. The most efcient speakers will be rated with a
higher dB SPL. The rating relates to the amount of signal reproduced
when a 1-watt signal is fed to the speaker. A highly efcient quality
speaker rated at 100 dB SPL will reproduce 100 dB when measured
1 meter in front of the speaker. A less efcient speaker rated at
80 dB SPL will take the same 1-watt signal and reproduce 80 dB of
volume 1 meter in front of the speaker, while a speaker rated at 83 dB
SPL will reproduce 83 dB of volume. As previously discussed, an
increase of 3 dB is equivalent to a doubling of power, while a 6-dB
boost is required for a doubling of SPL.
Efciency is one way we judge speakers, but efciency does not always
accompany quality. Speakers sometimes lose efciency as a matter of
design. A speaker may be designed to absorb unwanted noise and distor-
tion through the enclosure construction, materials, methods of joining
components, or the crossover, making it less efcient but more pleasing
to the ear. Accuracy and clarity are more important to the audio profes-
sional than efciency. Furthermore, the size of the amplier may deter-
mine the need for efciency in speakers: A smaller amplier will benet
from a more efcient speaker while a larger amp with power to spare
may be better coupled with a less efcient but better-sounding speaker.
Perhaps the obvious conclusion is to get a bigger amp and speakers that
sound better.
Speakers can also be judged by their frequency response and distor-
tion. Speaker specications usually include a graph that represents
the response of the speaker at various frequencies (see Figures 6.14
and 6.15).
While a truly at response may be the ideal for which some audio pro-
fessionals will strive, it is also an unrealistic expectation. A more real-
istic expectation is that the various speakers in your enclosure will
178
compensate effectively for each others deciencies, resulting in a
pleasing sound throughout the audible bandwidth. If you play a mix
you know well through a pair of speakers and you hear both what you
want to hear and what you expect to hear, they are probably well
suited to your needs, and no chart will ever give you the level of infor-
mation you can obtain by listening.
Distortion is another consideration (see Figures 6.16 and 6.17).
Matching impedance and power between an amplier and speakers
is a good start in avoiding distortion. Checking diagrams supplied by
manufacturers describing the distortion in their speakers is also
65
20 50 100
[Hz]
[dB]
200 500 1k 2k 5k 10k 20k
70
75
80
85
90
Figure 6.15 The musikelectronic geithain (MEG) Basis 3, a subwoofer, shows a substan-
tially different frequency response than a studio monitor. Based on an original diagram
courtesy of MEG.
60
20 50 100
180
45
0
[Hz]
[dB]
200 500 1k 2k 5k 10k 20k
65
70
75
80
85
Figure 6.14 The musikelectronic geithain (MEG) RL 901k Studio Reference Monitor
shows an excellent frequency response. Based on an original diagram courtesy of MEG.
179
helpful. Ultimately with speakers, as in every other end of audio, use
your ears to make a nal, denitive determination as to the capabil-
ities and quality of the equipment, and regarding the speakers you
will ultimately choose for your production room, performance
space, or facility. We need to power our speakers, and many of the
same criteria apply for amps.
Ampliers
As the name would suggest, an amplier takes a signal and increases its
amplitude. We discussed mic and line preamps, voltage controlled
ampliers, and potentiometers in Chapter 3, as they are all featured
on the console in our production room. Just like these other devices,
0
30 40 50 60 70 80
K2
K3
[Hz]
[%]
1
2
3
4
5
Figure 6.17 THD in the MEG Basis 3. Based on an original diagram courtesy of MEG.
0
100 200 500 1k 2k 5k 10k
K2
K3
[Hz]
[%] 0.5
1
Figure 6.16 THD in the MEG RL 901k. Based on an original diagram courtesy of MEG.
180
ampliers are not transducers because the signal is not converted from
one form of energy to another; a signal embedded in electrical energy
comes in, and a louder signal contained in electrical energy is output.
The form of energy is not changed; it is only reproduced at greater
amplitude (see Figure 6.18).
In the consumer market, ampliers are typically coupled with preamps
and tuners and sold as integrated ampliers and receivers. In the high-
end audiophile market as well as in professional situations, the ampli-
er is not coupled with a tuner, and sometimes even preamp functions
occur in a separate device (see Figures 6.19 and 6.20).
Figure 6.18 A Bryston B-100 SST Integrated Amplier. Photo courtesy of Bryston.
Figure 6.19 The Ashly Powerex 6250 Integrated Amplier does not include a tuner or
preamp functions; it just amplies. Photo courtesy of Ashly.
181
Sometimes ampliers will be installed in speakers called self-powered
speakers, much in the way a guitar amp will contain both a speaker
and an amplier. Like speakers, ampliers can be rated by their fre-
quency response and distortion. Ampliers can also be rated based on
wattage. Wattage ratings on ampliers fall into two categoriesinput
wattage (a measure of power supplied to the device in order for it to
operate) and output wattage, which the device sends to the speakers.
For our purposes we will only discuss output wattage.
Wattage is a measurement of power. Greater power equals greater loud-
ness. Wattage can range from1=10 of one watt for personal stereos up to
hundreds or even thousands of watts for a public address, sound rein-
forcement, or theatre system. Consumer stereo systems will generally
fall between 28 watts and 60 watts, although audiophiles may choose
amps rated for 100 to 200 watts for their home systems. In professional
production rooms the wattage of the amplier will depend on many fac-
tors. If one is powering studio monitors, 100 to 200 watts is recom-
mended, while bookshelf monitors in a production room may only
require 30watts. Keepinmindthat, like decibels, wattage is a logarithmic
scale, not a linear one. In general, we double the loudness (our perception
of amplitude) through a tenfold increase in wattage. In other words, rais-
ing our power from 4 watts to 40 watts represents an approximate dou-
bling of loudness, as does an increase from 40 watts to 400 watts. Of
course this is different than doubling the power, which is a 3-dB increase,
as discussed earlier. Nonetheless, people often do not perceive a doubling
in volume until there has been a 10-dB increase, since perception often
varies from the technical reality.
Figure 6.20 The back of the Ashly Powerex 6250 Integrated Amplier. Photo courtesy
of Ashly.
182
As with speakers, when we rate ampliers on the basis of frequency
response, we look for a full, at response. Amplier specications will
enumerate the range of frequencies that the amplier will reproduce
reliably and will tell you when the frequency response begins to
decline. While inexpensive consumer systems or public address systems
may only reproduce 50 Hz to 12 kHz, audiophile-level equipment and
professional gear should be expected to exceed the human audible
spectrum, with something in the range of 10 Hz to 22 Khz or greater
(see Figures 6.21 and 6.22).
Figure 6.21 Frequency response in the Bryston B-100 pictured in Figure 6.17. This chart
demonstrates a healthy frequency response to above 20 kHz. Diagram courtesy of Bryston.
Figure 6.22 Distortion in the Bryston B-100. This diagram demonstrates that we experi-
ence an acceptable level of distortion until we exceed around 130 watts. Diagramcourtesy
of Bryston.
183
The other number to look for in amplier specications is the number
of dBs the signal drops off, and the frequency at which this begins to
occur. If an amplier drops 3 dB at 20 kHz, that will be ne because
few of us hear 20 kHz. However, a production room amplier with a
signal drop of 3 dB at 10 kHz would present a real problem for us
because our mixes would all come out too bright as a result of us over-
compensating for high-end loss in the amp.
All equipment produces distortion. When we rate ampliers we look for
a very low percentage of distortion. A distortion level of around 0.25%
is desirable because it is inaudible. A distortion level up to about 1% is
acceptable in most situations, but some equipment will have distortion
levels as high as 5%, which we will be able to hear and which is there-
fore unacceptable. Similar to distortion is total harmonic distortion, or
THD. This occurs when an amp colors a reproduced sound and adds
something at double or triple the original frequency, or at half the orig-
inal frequency, following the harmonic series discussed in Chapter 2.
This coloration is also quantied as a percentage, where up to 1% is
inaudible and acceptable.
Just as all equipment produces distortion, all equipment produces noise.
We discussed signal-to-noise ratio (S/N ratio) in Chapter 2 and con-
cluded that we like more signal and less noise. In ampliers this is rated
as a ratio, such as 80:1, where we produce 80 dB of signal and generate
1dBof noise as aresult. This ampcouldalsobe describedas havinganS/N
ratio of 80 dB, or 80 dB. A higher number represents more signal in
relation to the noise generated by the device, so a piece of gear with an
S/N ration of 96 dB will be far quieter than a device rated at 40 dB.
So far we have been going back and forth between digital and analog
technology. This would be a good time to add the last piece of the
puzzle regarding equipment, by taking a good look at the digital
domain in audio.
Exercises
1. Sit in a production room and play back a mix with which you
are familiar. Play it back on all speaker types availablestudio
monitors, bookshelves, and sound cubes. Observe the
184
differences in frequency response. Is one type of speaker more
complementary than the others? Is one lacking in quality?
Listen closely to the frequency response and timbre of each
monitor type. Identify the differences you hear.
2. Obtain three different pairs of bookshelf monitors and perform
the same listening exercise as above. Which monitors suit your
ears the best? Which ones suit the mix best? Try playing a
different mix. Are your observations and conclusions consistent
or have they changed?
3. Using the same three sets of bookshelf monitors as above, try
listening to each pair of monitors at an extremely low (barely a
whisper) level, a moderate level, and a loud level. Does the
character of the sound change in the monitor at different levels?
If you were to mix on monitors with a character that changed
at different levels, would it affect your mix? How?
Additional Reading
2004.
Springer, 2005.
Press, 1993.
CA: Alfred, 1989.
185
Press, 1989.
Press, 2000.
Sams, 1989.
186
7
Digital Audio
11010001010010100101001010101001010001010101001010101010100101010101001010101010
187
N
o overview of todays audio would be complete without an
examination of digital audio. Digital technology has been
mentioned repeatedly throughout this book in past chapters,
and it will be in upcoming chapters. Since it is inescapably intertwined
with analog throughout the eld of audio, it is benecial to examine
the basics, even if only in a perfunctory manner. An audio professional
may have a strong preference toward either digital or analog for any
number of reasons. Functioning successfully in the world of audio
requires a comfortable understanding of both.
One of the most interesting and compelling features of digital technol-
ogy is that it is far more complicated than analog, yet it makes our
interactions with equipment far simpler. As we will see in Chapter 9,
Signal Processing, digital signal processing devices are cheaper,
smaller, lower in temperature, more programmable, and easier to use,
yet the underlying theorieswith sampling rates, conversions, quanti-
zation, dither, and moreis far from simple. The simple part of digital
technology theory is this: Everything is reduced to ones and zeros.
While that gives us two clear optionsit is either on or it is offit
also allows us to circumvent some of the problems inherent to analog,
such as wow, utter, hum, generation loss, and so on. Because ones and
zeros can be copied, stored, and reproduced in an identical fashion
every time, we have far fewer errors in the digital realm (at least in a
theoretical sense). As an operator of digital gear, the audio professional
can really home in on a particular parameter and adjust that parameter
by far smaller increments than he or she can by using the analog coun-
terparts. The audio professional can also commit operator error on any
device.
Just as analog technology presents us with a certain set of problems,
such as generation loss and heat buildup, digital presents problems
of its own. We are limited by technology (and sometimes by our budget)
regarding the quality of the sample and the amount of storage required
based on the number of times we can slice through a particular sound
wave. We cannot afford to slice into it a million times every second with
a high bit rate, and if we slice into it twice in one second, our sample
will not be an accurate and acceptable representative of the original
wave. Lets examine this issue more closely.
188
Sampling Rates
Lets suppose we have a sound wave with a frequency of 1 kHz. As we
recall from Chapter 2, this means that the sound wave repeats itself
1,000 times each second. To bring this sound into the digital realm, it
needs to undergo an analog-to-digital conversion (AD) (see Figure 7.2).
In this process, the AD converter examines the sound wave by slicing
into it repeatedly. If the AD converter sliced into this wave 1,000 times
each second, which would be once per repetition or completion of the
wave, would the sampler have enough information to replicate this
wave? The answer is no, because the sampler needs to see a variety
of locations during the compression and rarefaction to create an accu-
rate and useable algorithm. Since the wave would be in the same place
in terms of its compression and rarefaction in each sample, a 1-kHz
sample of a 1-kHz wave could not create an adequate replica of this
wave because one sample per cycle does not give our sampler a clear
representation of the various components of the wave. This amount of
Figure 7.2 Although virtu-
ally all digital gear offers
internal digital-to-analog
conversion, many audio pro-
fessionals prefer using out-
board converters, such as
the Manley Reference Digi-
tal to Analog Converter.
Photo courtesy of Manley
Labs.
Figure 7.1 The Studer Vista 5 digital
console. Photo courtesy of Studer.
Chapter 7 Di gi tal Audi o
189
information would be inadequate to create an accurate digital sample
(see Figure 7.3).
If the converter were to slice into this wave 2,000 times every second,
which we will call a sampling rate of 2 kHz, we would have twice as
much information about each cycle of the wave. The sampler would
see the wave on the way up as well as on the way down, which would
be enough information to create the most basic valid digital equivalent
to the original wave (see Figure 7.4).
Suppose we doubled the frequency that we wish to sample to 2 kHz
now we must double our sampling rate again, to 4 kHz. Every time we
increase the highest desirable frequency in a sample, we must increase
our sampling rate to twice that frequency. Since our audible band-
width, or the range of frequencies that we can hear, spans 20 Hz to
20 kHz, a sampling rate of double the highest frequency we can hear,
or 40 kHz, is needed. As we increase the sampling rate, we increase
bandwidth (see Figure 7.5).
A CD samples at 44.1 kHz, which gives us a bandwidth of 22.05 kHz
enough for us to hear even the highest frequencies within our audible
A
m
p
l
i
t
u
d
e
1 Second
Figure 7.3 When an analog-to-digital sampler analyzes a wave, one slice through
one completion of the wave is inadequate to create a sample.
A
m
p
l
i
t
u
d
e
1 Second
Figure 7.4 Two slices through a wave gives the sampler a better idea and the mini-
mum of information needed to create an algorithm.
190
spectrum. As the resultant bandwidth rises, so will the need for memory
to store this ever more complicated series of ones and zeros (an algorithm)
(see Figure 7.6).
As we approach the realm of higher sampling rates we require more
sophisticated gear, which will cost more money. If one is recording
digitally to save money, he or she may be stuck with a lower sampling
rate or lower-quality converters to keep it affordable.
This begs the question, How high of a sampling rate do I need?
Once again, Bell Laboratories supplies us with the answer, this time
in the form of the Nyquist Theorem. The Nyquist Theorem states that
the sampling rate must be twice the highest frequency that we wish to
sample. In other words, in the aforementioned example, a sampling
rate of 2 kHz would be the minimum to achieve a usable sample for
a sound that occurs at 1 kHz. A sampler needs to slice into each com-
pleted wave at least twice in order to understand it. To sample a signal
at 8 kHz, one must use a sampling rate of at least 16 kHz; a signal of
12 kHz requires a minimum sampling rate of 24 kHz; and so on. As we
have discussed already, a higher sampling rate will provide a better
quality sample; the Nyquist Theorem only deals with the minimum
frequency required for an adequate sample.
A
m
p
l
i
t
u
d
e
1 Second
Figure 7.6 Doubling our sampling rate yet again, to eight times the sampled fre-
quency, gives us an even better quality sample.
A
m
p
l
i
t
u
d
e
1 Second
Figure 7.5 Doubling our sampling rate again, to four times the sampled frequency,
gives us a better quality sample.
191
Aliasing and Quantization Problems
As stated, we require a sampling rate that is twice the frequency of the
highest frequency we wish to sample. There is always a possibility that
frequencies higher than the highest one we are attempting to sample
will be present. An appropriate question would be, What happens to
all those other frequencies? In the case of a sampling rate of 48 kHz,
every signal up to 24 kHz in a sample will be ne. If there are signals
present in our complex wave or sound event above 24 kHz, this will
present a problem for us because we will be unable to take our two
slices per completed wave, as we have deemed necessary. If we try to
sample frequencies that are too high for our converter to understand
(because it cannot take two slices per completed cycle), we introduce a
nonmusical and highly annoying interference called aliasing. We solve
aliasing problems before they occur by placing an anti-aliasing lter at
the analog input of every digital device. This low-pass lter, a device
that allows low frequencies to pass unaffected but lters out all higher
frequencies, is set at half the sampling rate and allows all frequencies
through that fall below that number, while stopping all frequencies
that occur above that point (see Figure 7.7).
In the aforementioned example, with a sampling rate of 48 kHz, the
anti-aliasing lter is set at 24 kHz, preventing all signals above 24 kHz
from entering our device and thereby eliminating aliasing noise.
Another problem, called quantization distortion, presents itself in digital
recording when levels are too low. The small amplitude levels will
be rounded or approximated, which will produce strong harmonic
20kHz
15kHz
10kHz
5kHz
25kHz
30kHz
Figure 7.7 An anti-aliasing lter is a low-
pass lter set at half the sampling rate. In
this example, with a sampling rate of
48 kHz, only signals at 24 kHz or lower
are allowed to pass through and enter
the analog-to-digital converter.
192
distortion. Due to the chirping sound that results, these distortions are
sometimes called birdies or birdsinging. As counterintuitive as it may
sound, quantization distortion can be removed through the application
of a small amount of analog white noise, known as dither. Introduction
of dither to the digital noise oor makes the small amplitude of the dig-
ital signal more recognizable and stable and will reduce quantization dis-
tortion. It may seem odd to add noise to a nice clean digital signal to
make it cleaner, but dither is necessary and should be inaudible due to
the massive dynamic range available in digital. When it comes right
down to it, a few decibels of inaudible noise is far better than birds chirp-
ing in your mix. Audio professionals will frequently change their dither
point to improve the overall quality of sound in the digital domain.
Error Correction, Compression, and Formats
We discussed some of the errors that can occur in analog, such as drop-
outs due to loss of magnetic particles on tape. It is also possible to lose
small amounts of digital data. Digital equipment compensates for this
loss through error correction, by interpolating the data that has been
lost or corrupted. If a digital device suspects there is a problem, it will
resample the section and try to gure out what is missing. If the ampli-
tude in a sample went from 3 dB to bad to 5 dB, the error correc-
tion feature would interpolate that the bad section was 4 dB. Much
in the way our eyes and brain turn a series of static images or an array
of dots into the vision of movement in lm and on television, a quality
digital audio device will compensate for the pieces that arent there.
Like sampling, good-quality error correction requires a great deal of
power and the ability to store and manipulate large amounts of data,
all of which can become very expensive. In the audio professionals
drive to provide quality audio at a reasonable price, data compression
can be used to reduce costs on the storage end.
There are two kinds of data compressionnon-lossy, which is also
known as lossless, and lossy. Non-lossy compression can be used to
minimally reduce data. It is nondestructive and can cut the amount
of data to be stored in half, which we call a 2:1 compression. The
nice thing about non-lossy compression is that as the name implies,
nothing is lost; it is just packed better. Lossy compression is employed
193
when one wishes to store a great deal more data or have it move more
quickly, as in some Web applications. Our compression could go to
50:1, 100:1, even 200:1 with lossy compression; however, data is elim-
inated in this process, and the more compression one applies, the
worse the program will sound. Low sampling rates and high compres-
sion are both features of personal digital stereos. This goes a long way
to explain why they sound so poor.
There are several popular lossy compression methods that have become
part of every computer users day-to-day life. Real Audio is a common
way that audio is transported over the Internet. Utilizing a compression
ratio of 50:1, it reduces CD quality (which paces itself at 1,411 kilobits
per second in stereo) to only 28 kilobits per second. Quality is adequate
when the audio is transported over a high-speed line, but there is signif-
icant loss of quality on a 56K or slower modem. A better-sounding alter-
native is MP3, also known as MPEG-2 layer 3. Using a compression
ratio of 10:1, MP3 features low compression and a high data rate and
only throws away minimal data. Although losing any data may seem
unacceptable in some situations, when it is necessary the combination
of a lower compression rate and a higher data rate will result in less over-
all loss of quality. This combination is useful for high-quality audio- and
video-making. MPEG-2 layer 3 is popular for DVDs and digital satellite
video transmission, and its derivative, MP3, is useful for audio.
Sometimes we need compression methods to send data hither and yon;
other times we simply need to understand which le formats are used
to store audio. Mac-based systems tend to store audio as AIFF (Audio
Interchange File Format), while PCs use WAV format. BWF, or broad-
cast wave, is the muscular version of WAV that has taken over to
become the industry standard, thanks to its superior management of
data and other handy features, such as time stamps.
Figure 7.8 The AMS RMX-16 digital reverb. Photo courtesy of AMS.
194
Bit Quantization
We have all encountered the term bit in audio. Let us explore what it
means. We have already discussed that the algorithms that represent
sounds in the digital realm are comprised of ones and zeros. These ones
and zeros string together to form binary words. Our bit rate deter-
mines how long these words can be. If we had a bit rate of two, our
system would understand binary words with a maximum length of two
charactersin other words, zero (0), one (1), two (10), and three (11).
Because the number four (in our common Base 10 system) is 100 in
binary, we would need a three-bit system to interpret that number. Of
course if we had a bit rate of three, our system could also recognize ve
(101), six (110), and seven (111). As our bit quantization rate
increases, the number of values our system can recognize and interpret
grows. With a four-bit system we can interpret twice as many values as
a three-bit system16 instead of 8. Once we arrive at eight-bit, our
system recognizes 256 values, giving us minimally acceptable audio
quality. At 16-bit we achieve CD quality, with 65,536 values available,
and at 24-bit, the highest professional standard (currently), our system
can recognize 16,777,216 values. Bit rates can also be related to
dynamic range because each bit gives us 6 dB of volume. This means
that a CD recorded at 16-bit has a dynamic range of 96 dB, far larger
than the best analog recording. A 24-bit recording allows for a theo-
retical dynamic range of 144 dB, far more than we need, but we like it
all the more for that.
Much in the way that wider tape (-inch rather than -inch) and
greater speed (30 inches per second rather than 15 inches per second)
add together to dramatically increase analog sound quality, sampling
frequencies and bit rates combine to give us dramatically better digital
audio quality. As discussed earlier, recording at a high bit rate with a
high sampling rate requires a tremendous amount of power and stor-
age capability, but on the positive side, when these rates are high our
sound quality is excellent. Working in the world of audio requires that
we accept certain limitations and make compromises when necessary.
Just like times when an engineer might record at 15 IPS to save the
client money, an engineer may opt for 16-bit rather than 24-bit to
ensure that the production work will t on one hard drive or t into
195
the producers budget. Now that we have a basic understanding of the
nuts and bolts that hold digital information together, lets take a look
at the digital languages available to audio professionals.
MIDI
MIDI, or Musical Instrument Digital Interface, is the language used by
synthesizers and sequencers to communicate with each other. Much in
the way that composers use musical notation on sheet music to convey
their musical ideas to musicians, devices use MIDI as a common musi-
cal language. Composers and synthesizer players use MIDI to commu-
nicate musical information between instruments and computers
(including sequencers). In its simplest form, a part is played or entered
into the computer through a keyboard or other input device. The com-
poser determines the synthesizer patch, which will reproduce this part,
and tells the computer where to send it, then tells the synth (or other
playback device) to look for this particular part. In other words, a com-
poser will play the bass line of a song on a synth that is connected to a
computers sequencer software. Upon completion, the composer will
assign this part to a MIDI channellets say Channel 1. The composer
will then bring up the bass sound (or any other patch or sound) on the
synthesizer and assign that patch to Channel 1. Now, when the com-
poser presses play in the sequencer, the sequencer will send the bass part
to the bass patch, which will play it back. The composer will then move
on to input the chords, melody, and sweetening, assigning each a chan-
nel number in MIDI and then assigning that same number to the desired
patch or sound. The end result is that all parts will be played simulta-
neously, much in the way a band or orchestra would.
Figure 7.9 The HHB MDP500 Porta-
ble MiniDisc Recorder. Photo courtesy
of HHB.
196
MIDI is valuable for the amount of information contained in the lan-
guage. Along with obvious information, including which notes were
played and how long they were held, MIDI also stores and transmits
information about how hard (or loud) the notes were played, the
sharpness of the attack, the gentleness of the release, and more.
Much like notation on a musical chart indicating tempo and whether
the piece should be played legato or staccato, MIDI will instruct the
synthesizers on the detail of the desired performance, based on how the
composer entered the data. Along with notes and timing, MIDI stores
a great deal of information, giving the resulting performance a great
deal of nuance.
Along with instruments, MIDI can be used to communicate with dig-
ital signal processing gear. Multiprocessors, or devices that are capable
of replicating many different forms of signal processing, can be manip-
ulated through MIDI such that a multi-effects processor may be a
reverb in one part of a mix and then, upon a MIDI command, will
switch to a digital delay for another section of a song. MIDI could
also dictate a change within one patchin other words, the reverbs
RT-60 could change from 1.8 seconds in the verse to 2.4 seconds in the
chorus, then return to 1.8 seconds when the song returns to the verse.
One of the features of MIDI that makes it a great tool is its consis-
tency. Once a composer likes what has been entered, he can be assured
(computer crashes notwithstanding) of hearing the same thing repeated
every time the play button is clicked.
Figure 7.10 An ESI M8U 8-In/8-Out USB-to-MIDI interface. Photo courtesy of ESI.
197
SMPTE
Unlike MIDI, which communicates musical information between
instruments, devices, and their controllers, SMPTE is a time-based
code used by machines to stay synchronized. Originally created by
the Society of Motion Picture and Television Engineers (hence the
name), SMPTE is used by machines to stay in time with each other
and by composers who work with lm and video to ensure that
their work lines up with the action on the screen. Depending on the
devices used, SMPTE can be transparent and simple or extremely com-
plicated (see Figure 7.11).
All audio software programs run SMPTE. They do this without any
action on the part of the composer or engineer. When a composer
looks at the counter running as he inputs information into his sequencer,
that composer is looking at SMPTE time code as it is generated or read
from his program. Similarly, hard drive recorders and digital audio
workstations will generate SMPTE as they roll along. This is extremely
convenient. If the engineer or composer needs to have his sequence
Figure 7.11 The Alcorn McBride Digital Binloop uses both SMPTE and MIDI to control
different devices. Image courtesy of Alcorn McBride.
198
communicate with another hard drive recorder or a workstation, the
SMPTE is already there to ensure that everything will line up and stay
in time.
These machines can also be locked up to analog tape machines and video
decks through SMPTE, although that process is more complicated and
less reliable. In this instance SMPTE time code needs to be recorded on
one of the tracks of the analog recorder. The SMPTE time code from
another source, such as a sequencer, can be transferred onto the analog
track, or new SMPTE can be recorded from a SMPTE generator. Once
there is SMPTE on the analog device, this signal is fed into a SMPTE
reader card in the computer or a stand-alone synchronizer. One device
is assigned as the master, and the other device(s) are the slave(s), which
will chase the master and stay locked up. This process can also be per-
formed between two 24-track analog machines, giving us an inaccurately
named format: 48-track analog. This format is poorly named because, in
fact, one track on each 24-track tape needs to be striped, or dedicated to
SMPTE, and smart engineers leave a track empty next to SMPTE to pre-
vent crosstalk between the SMPTE track and the adjacent track, so in
essence a 48-track analog session only allows for recording on 44 tracks.
Other devices that can be locked up like
3/
4-inch and 1-inch video decks
Figure 7.12 Cubase SX4 Screenshots. Photo courtesy of Cubase.
199
are getting more difcult to obtain because they are such antiquated devi-
ces. They are useful when we are recording for video or lm and wish to
reference the music or sound effects against the video or lm to ensure
that everything is occurring in the correct place and at the correct time,
or in sync. Many machines can be connected simultaneously in this man-
ner. One can have a sequencer driving two 24-track analog machines for
source material while hooked up to an analog eight-track for mixing four
stereo pairs (the old way of doing lm), which is also hooked up to a
video deck for reference. This process is viable as long as there is only one
master (in this case the sequencer) and enough SMPTE card readers to go
around for the slaves.
Of course this complicated method is a bit outdated. Most lm sound-
tracks are currently performed and recorded on software such as Avid
and Pro Tools, which communicate well with each other through the
relatively transparent use of SMPTE. Now that we understand all our
bits of equipment and we have an idea of how they work and play
together, lets see what we can do with them.
Additional Reading
2004.
Springer, 2005.
Press, 1993.
200
CA: Alfred, 1989.
Katz, Bob. Mastering Audio: The Art and Science. Burlington, MA:
Focal Press, 2002.
Lehrman, Paul D. and Tim Tully. MIDI for the Professional. New
York: Amsco Publications, 1993.
Pohlmann, Ken C. Principles of Digital Audio, 5th ed. Blue Ridge
Summit, PA: Tab, 2005.
Press, 2000.
Sams, 1989.
201
PART
III
Audio Methods
and Operations
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8
Editing
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W
ith the advent of readily available digital technology, the art
of editing has undergone many changes in recent years, but
the basic theories remain the same. In its simplest form,
editing involves the removal of unwanted noise and the reordering
or replacement of recorded material. The art and science of editing
has evolved, and continues to evolve in a seemingly limitless fashion,
with the aid of computers and sampling, or storing of chunks of infor-
mation. Editing has become an artistic endeavor unto itself. In decades
gone by, editing was primarily thought of as repair worka method of
removing breaths and random room noise from voice tapes, creating
dub mixes or dance mixes by repeating sections of a song with small
differences and editing them in, or improving a single mix by combi-
ning sections from various mixes to achieve a superior composite.
Currently, a sound sampled into a computer can be regenerated at will,
repeated, modulated into different frequencies for effect, or truncated
as needed. The effects currently employed, while a radical departure
from ancient editing techniques that required physically cutting and
reassembling a piece of tape, are still based in these ancient techniques.
As such, it will be valuable to understand the basics of editing. An
audio professional at this point in time could spend his or her entire
Figure 8.1 A Nuendo 3 Media Production System by Steinberg Media Technologies.
Photo courtesy of Steinberg Media Technologies.
206
career working in a digital environment and never use a razorblade
for anything other than shaving, but by understanding the techniques
and approaches to editing of eras gone by, the engineer will have a far
easier time honing his or her craft.
Any audio professional starting out in the business who works at a com-
mercial studio faces the possibility of tail leadering a master tape, which
requires razorblade editing skills. Additionally, razorblade editing is
becoming a lost art, and being familiar with it could make a young
engineer a celebrated hero at the right place and the right time.
Razorblade edits can be done on virtually any professional tape for-
mat. They are most commonly performed on -inch and -inch mix
formats, but can also be performed on multitrack applications up to
2-inch tape. They can even be performed on digital tape, although
that process requires additional preparations and precautions. The
only forms of analog audio tape that cannot be edited are those non-
professional formats that allow the user to turn the tape over, such as
cassettes and -inch quarter track.
Objectives
To better understand the skills expected from a competent editor, it
would be useful to establish some skill objectives. Beginning with
the simplest of objectives, an editor should be able to locate a specic
word or sound event on a piece of tape or as a waveform on a screen.
As one becomes more skilled and adept, he or she should be able to
locate and identify specic musical segments within a song. Dubbing,
or recording music or other audio material from one machine to
another, is a skill that any entry-level audio professional should pos-
sess and master, and it is also necessary to the editor. Similarly, the
ability to route the signal as needed is germane to the editing process.
In other words, regardless of the environment, digital or analog, the
audio professional needs basic skills to edit successfully.
Once the student is able to route, record, and locate, the actual splicing
can begin. Analog, non-linear splicing is the physical action of cutting
the tape and reassembling itsometimes removing tape containing
undesirable material in the process, sometimes reordering it. An editor
Chapter 8 Edi ti ng
207
should be able to insert, delete, and/or assemble portions of narration
or music such that the nal result is pleasing to the ear and does not
appear to have been edited. One of the tests of successful professional
editing is the inability of others to detect the location and existence
of the edits. Much like the basic skills required, this is true regardless
of the platform or the technology used.
Another skill common to the best editors is the ability to handle the
paperwork. While paperwork may seem mundane by comparison, any
editor who has returned to an analog project to nd the nightmare of
hundreds of bits of unlabeled tape in no particular order understands
the value of accurate cue sheets and good notation. An entry-level
audio professional has the opportunity to prove himself worthy in this
situation by taking copious notes.
Terminology
Much of the terminology learned in Chapter 3 regarding tape machines
will carry over into our initial discussions of editing. Understanding
which head is the repro head and what differentiates it from the sync
head, also known as the record head (as covered in Chapter 3), are crit-
ical to certain procedures, but there is also some critical terminology to
master. Shuttling a tape is both a terminology and a methodology that
must be understood when editing in an analog environment. Shuttling
involves switching rapidly and repeatedly between the fast forward and
rewind switches on the tape machine. By engaging the cue or edit switch
on a tape machine before shuttling, we can hear the material recorded as
it passes the heads rapidly. This enables us to quickly locate a particular
section of a song, a long gap in dialogue, or an anomaly in the recording
that should be removed. Care needs to be taken in this process not to
damage the tweeters, because the increased tape speed while shuttling
outputs far more high-end energy. Always be aware of the control room
volume during this process to protect your speakers.
Rocking can be thought of as a slower, more accurate form of shut-
tling. While shuttling is performed in fast wind modes using the fast
forward and rewind controls, rocking is done slowly, by hand. With
the edit or cue switch engaged (defeating the head lifters and allowing
the tape to contact the heads), and with the right hand on the take-up
208
reel and the left hand on the supply reel, the editor simply rocks the tape
back and forth across the heads to accurately pinpoint a location on
the tape. In addition to allowing the editor to locate a particular section
of a song or speech, the low speed of rocking also allows the editor to
accurately mark the tape. The reason for this will become apparent
when we discuss methodology. In the digital realm, an equivalent to
rocking the tape is frequently performed by moving a wheel, mouse
ball, or another input device left and right over the desired location.
Methodology of Simple Edits
The method employed to edit tape, in its simplest form, involves cut-
ting a piece of tape out and rejoining the loose ends. Although this
sounds fairly simple, there are some guidelines that the audio profes-
sional must follow to ensure that the edit will be inaudible and that
maximum quality is retained on the tape at all times.
First of all, clean the heads of the machine (see Figure 8.2).
Use two sterile cotton-tipped swabs dipped in 91% pure (or purer) isopro-
pyl alcohol. Rub all three heads gently, and then rub all metal in the tapes
path, including guides and the capstan. The heads should be cleaned daily,
or at the beginning of each session if the room is used heavily. Never use
alcohol that is less than 91% pure, and do not use isopropyl alcohol to
Figure 8.2 A typical tape path. When cleaning tape heads with isopropyl alcohol, also
clean all metal guides and rollers in the tape path, along with the capstan. Based on an
original diagram courtesy of Otari.
Chapter 8 Edi ti ng
209
clean rubber in the tape path, such as the pinch rollerthis will dry it
out. Special cleaners should be used for the pinch roller.
To begin editing, nd the location of the rst cut. Suppose you have a tape
of a narrator counting at an even pace fromone to ten, but he repeated the
number ve. Your goal as an editor would be to remove the rst number
ve (assuming that he repeated it because he didnt like the way he said it
the rst time). As a general map of howwe will accomplish our goal, let us
say that we will make one mark on the tape immediately before the rst
time the narrator says ve, make a second mark just before the second
time he says ve, and remove all tape in between the two marks. By
making our marks just before each of the sounds, in a uniform manner,
the pacing or timing will remain the same as the original, and the sequence
of numbers will make sense.
Using the rocking method described in the Terminology section, nd
the location of the rst cut. First of all, be certain that the tape machine
is in repro mode, so when the tape is marked, you will know which
head is producing the signal. Keep in mind that sound is reproduced
from the repro head when the audio professional has set the machine
in repro, or playback. If the machine is set in sync, the signal is being
reproduced from the record head, and the engineers mark at the repro
head would be in the wrong location.
By rocking the tape, or rolling it across the heads with one hand on
each reel, feeding it with the left hand while taking up the slack with
the right hand, the sound of the narrator counting is heard. When we
hear him say four, we will be prepared for him to say ve next.
When we hear the rst sound of the word ve, before we hear the
full word, we will back up the tape slightly, feeding with the right hand
and taking up the slack with the left hand, until we are on that bit of
tape that is just before the rst sound of the word ve. Sometimes
we will have to rock back and forth three or more times to be condent
that we are where we think we are, just before the rst sound of the
word ve. Just like in carpentry, where it is better to measure twice
and cut once, the audio professional must be sure that the mark is in
the correct location before cutting. Once the editor is condent that he
or she is in the correct spot, a mark will be made on the tape. This type
210
of marking is similar to marking a region or creating an edit point in a
digital environment; the primary difference is that typically, the point
will be marked on the computer with a click of the mouse in software
applications or by pressing a button in hard drive editing.
When marking the tape, editors commonly use a white or yellow
grease pencil, also known as a china marker or a china white. Using
the white ones is easiest, since white marks are easier to nd (because
the mylar backing of tape is usually black). It is not necessaryand it is
in fact undesirableto make a thick, heavy mark on the tape, because
the back of the tape rests against the emulsion side of the tape a short
distance down as it winds on the reel, and any foreign substance on the
emulsion can compromise the sound. A simple vertical stroke on the
bit of tape that is at the center of the repro head will be enough to
accomplish the goal of marking the location. If this is a piece of tape
that will be removed and may at some future date be added back into
the master, there is one more step: Some editors nd it helpful to make
a small arrow on the piece of tape being removed; otherwise, there is
no way to determine at a later date which is the beginning, or head, of
the piece and which is the end, or tail.
It is smart methodology in analog editing to make both marks involved
in one edit before cutting, because it is extremely difcult to rock the
tape after the rst cut has been made. Now that we have marked the
rst, or out, point of the tape to be removed, let us nd the second, or in,
point. Since the amount of tape in question is short (we are only remov-
ing one word), we can rock the tape forward. As the tape is rocked for-
ward, the editor hears the narrator says ve (the part that will
be removed), then the narrator begins to say the word ve again
(the one that will be kept). As before, the editor will nd the point on
the tape just before the word ve and mark it with a single vertical
line. What we now have is a tape with two marks, one just before the
word we wish to eliminate and one just before that word is repeated
the word we wish to keep. All that remains to be done is to remove the
unwanted word.
At this point the editor will prepare to cut the tape using a new, single-
edged, demagnetized razorblade. Because recording utilizes a magnetic
Chapter 8 Edi ti ng
211
medium, the razorblades must be demagnetized, otherwise their mag-
netic charge could change the magnetic elds on the tape, causing an
audible glitch, or noise. We already know that edits must be inaudible,
so avoidance of glitches is critical to our success. We will discuss
other possible causes for glitches later in this chapter, in the Possible
Edit Problems section. The blade should also be new, or relatively
new, because an old, jagged blade will make a jagged cut, likely to
cause a dropout (also discussed in the Possible Edit Problems
section).
To cut the tape consistently, we rst seat the tape rmly in an editing
block with the emulsion side down. When seated properly, the tape
can slide to the left or right, but will have no vertical movement.
While editing blocks offer several options for cutting angles, including
90 degrees and 60 degrees, the 45-degree cut should be used exclu-
sively for all -inch and -inch tape (see Figure 8.3).
The 45-degree angle provides for a stronger splice and a smoother
transition from the old piece of tape to the new. In multitrack editing
the 22.5-degree angle may be used to avoid different transition points
between different tracks; however, this mostly applies to 2-inch
editing, which should not be attempted by the neophyte editor.
Returning to our previous example, lets position the mark for the rst
cut directly over the 45-degree angle cut on the edit block. Seat the
blade in the groove and cut the tape using a quick, solid swipe. Cutting
the tape slowly and tentatively, or sawing it, will sometimes result in
an inconsistent cut. Move the right side of the now cut tape (which
contains the audio information that the client wishes to keep) slightly
Figure 8.3 The industry-standard EDITall Splicing Block, showing the 45-degree angle
(center), which is recommended for all -inch and -inch splicing. Based on an original
diagram courtesy of EDITall.
212
to the right and out of harms way. Pull the left side of the tape out of
the block and look further down this side for the second mark. It
should not be far, since only a small piece of tape is being removed.
Position the second mark over the 45-degree angle cut on the edit
block as before, seat it rmly, and cut it with a solid stroke. The
piece that is currently on the right side of the block is the piece that
should be removed; pull this out of the block and place it off to the side
for now. Some editors will write 5 on it in grease pencil or in some
other way indicate what it is, just in case this piece needs to be edited
back in later. Unlike the non-destructive editing offered in the digital
environment, mistakes or changes of mind in analog editing require
more editing to repair the problem.
Meanwhile, having removed this piece, the two ends that need to be
combined are now in the editing block, the result of the two cuts. Slide
one of the pieces of tape toward the other such that they touch. It is
important that the edges touch without overlapping and without a gap
between them. When they are touching, apply a piece of splicing tape
or editing tape across the cut. Only use splicing tape that is specically
manufactured for this purpose; other tapes will be too thick to be
effective. Avoid excessive handling of both the audio tape and the
editing tape, since the natural oils on hands will weaken the edit and
the emulsion of the tape.
The editing tape should be exactly parallel with the audio tape; there
should be no excess hanging over either the top or bottom edge of
the audio tape. When the placement of the editing tape is deemed
accurate, secure it by pressing down with the back of the single-
edged razorblade or a clean, dry ngernail until the color of the edit-
ing tape matches the color of the audio tape. Never use a nger to
secure the editing tape, once again due to the oils on your hands. Be
sure to secure the editing tape securely to the audio tape. When
properly attached, an edit can last for decades; an improper edit
can fall apart the rst time it is played back.
Always check edits as soon as they are completed. When the edit is
secure, remove it from the block, load the tape back onto the machine,
and play it back. If done properly, the narrator will count through at
an even pace, and the edit will be inaudible and undetectable.
Chapter 8 Edi ti ng
213
Methodology of Complex Edits
We have discussed only short and simple edits thus far; lets take a look
at some more complex possibilities.
Suppose that, instead of only one word, we wished to remove several
paragraphs of speech. This is pretty simple in a digital environment,
where we just jump ahead and mark our points or regions. Its a bit
more involved in analog editing. The basics of marking, mounting in
the editing block, cutting, and rejoining remain the same as with a
simple edit. Finding the second mark, however, can be much more
complicated since it is a long distance down the supply reel, rather
than just a few inches away. To make nding that second mark easier
and to dispose of large quantities of tape more easily, tape machine
manufacturers created a mode called edit mode, or dump mode.
In edit or dump mode, the take-up reel disengages, so when the editor
presses play, the tape plays across the heads and then dumps out onto
the oor, rather than being gathered up onto the take-up reel. This
allows the editor to hear the audio program as it goes by, since the
capstan is still pulling it across the playback head at the normal
speed, while discarding that piece of tape either onto the oor or
Figure 8.4 Cubase 4 Advanced Music Production System. Photo courtesy of Cubase.
214
into a carefully positioned garbage can. This allows the editor to hear
the point approaching where the edit will be made, narrowing down
the area for the visual search for a mark. Lets review the methodology
in this situation, which is slightly different.
Instead of simply removing the word ve, suppose the narrator is
counting from one to one thousand, and we wish to remove all numbers
from 100 to 199 so that in the end result the narrator will count,
99200.This will be a large piece of tape toremove. Inour rst exam-
ple, we marked the out point rst; it didnt matter that the in point and
the out point were so close. In this example, lets mark the in point, or
second mark, rst. The reason will become apparent soon. Due to the
large size of the tape, we will shuttle ahead (as described earlier) rather
than rock the tape to nd the number 200. As earlier, the tape should
be marked just before the word 200. The tape can then be rewound
and shuttled to nd the point just before the number 100, where
we will mark the tape once again. After cutting the tape just before the
number 100, the tape is loadedthroughthe capstanandacross the heads.
Instead of mounting the tape onto the take-up reel, the free end, starting
with the number 100, will be left dangling over a garbage receptacle.
With the edit or dump button engaged (which initiates edit or dump
mode), pressing play will allow the editor to hear the numbers being
played while the tape is dumped into the garbage. When it gets close to
199200, the editor can stop the tape, search out the second mark
(this is why the secondmarkis made rst inthis method), make the second
cut, and join the ends. While this certainly seems more difcult than edit-
ing in a digital environment, if an engineer does analog edits every day,
he or she will quickly become comfortable and procient with it.
Although it is unlikely that a client will hire you to change the accepted
number system through editing, this same method is invaluable when
eliminating choruses, verses, or other musical sections that are too
large to be intuitively dealt with by the rst method. When a client
asks an engineer to remove one beat or one measure of music, the
rst method (for simple edits) should sufce; if a client asks an engi-
neer to remove an entire chorus, the second method is much more ef-
cient. The beginning and end of a chorus can be marked and
eliminated, or marked and spun off onto a separate take-up reel,
Chapter 8 Edi ti ng
215
then reinserted in a different spot in the song. Similarly, a chorus can
be dubbed or transferred onto a fresh piece of tape and reinserted after
an existing chorus, thereby doubling the chorus. All these techniques
and more are typical in dance mixes and dub mixes, although they are
far easier to accomplish in the digital environment.
As with editing speech, the editors ear for the correct spot to mark and
cut the tape will develop over time. No one is born with the ability to
understand audio information delivered in slow motion, but over time
anyone can learn to interpret this information, developing the ability
to distinguish between a bass drum hit and a snare drum hit as the
editor rocks the tape. The editor will also develop the nuance involved
in shading a cut a little to one side or the other, just as a bass player
might give a song a different feel by playing an imperceptible amount
ahead of or behind the beat.
The subtle decisions as to the exact location of the cut are also devel-
oped over time through experimentation. We discussed earlier how
both cuts in speech should immediately precede the sound. The reason
for this is to keep the pace similar to the original. Similarly, in music one
Figure 8.5 WaveLab 6 audio editing and mastering software by Steinberg Media
Technologies. Photo courtesy of Steinberg.
216
normally cuts fromone downbeat to another downbeat, fromthe begin-
ning of one section to the beginning of another. This helps to keep the
musical integrity and timing intact. Occasionally an editor will be asked
to work either off the beat, creating bars with unconventional time sig-
natures, or on experimental works, in which conventional time is less of
an issue. These projects are usually difcult and should only be
attempted by experienced editors. Another common practice is to cut
from one upbeat to another upbeat in a situation in which a cymbal or
some other sound is continuing over a particular downbeat, making that
particular downbeat a bad place to cut. Because it would sound unnat-
ural for the cymbal to suddenly disappear, creating an unacceptable,
noticeable edit, the engineer will sometimes choose to cut from one
upbeat to another upbeat, keeping that troublesome downbeat intact
and cutting a subsequent beat.
As discussed in Chapter 3, all tapes should be stored tails out. The edit
pieces that have been removed should be labeled as noted earlier and
spun onto an outtake reel. If an editor never needs to reassemble a
piece that has been edited, the outtake reel will never be used. Here in
the real world, once in a while we need to either reassemble a piece or
slug one edited piece back in. The rst time an editor is called upon to do
that, he or she will be happy for the fewminutes spent at the end of each
editing session putting those little bits of tape onto an outtake reel.
Before leaving a discussion of editing methodology, some mention
should be made of leader tape. White or yellow leader tape is used
primarily at the head and/or tail of the chosen takes. It clearly marks
the choice take for any audio professional who pulls that reel out of
the box. It comes in paper, which offers better traction, and plastic,
which does not offer the opportunity for paper cuts, which can be
quite severe when delivered by a tape ying by in fast wind mode.
Leader tape can also be used for timing as it is marked every 7-inches,
(one second at 7- IPS, second at 15 IPS, and so on); however, it
should not be used to create silence. The difference between the ambient
sound of the recorded material and the lack of ambient sound on the
leader will be too apparent. It is better in this situation to have recorded
some ambient room noise in the same place as the desired material was
recorded and to insert this when silence is needed. Often the true
Chapter 8 Edi ti ng
217
silence of leader tape is inferior to the supposition of silence supplied
by ambient room noise.
Possible Edit Problems
Edits should be an audio professionals little secret. They should pass
unnoticed if performed correctly. There are a variety of problems that
can interfere with that silence, and there are certain conditions that
will lead to specic problems while editing. We already mentioned
glitches that can occur as a result of a blade containing a magnetic
charge or an old blade, which results in a jagged cut. A glitch will
sound like either a pop or an electronic jolt.
Glitches can also result from improperly matched edit pieces if there is
space between the pieces when they are attached or if they overlap.
Always make sure the two pieces touch, but do not overlap. Another
common cause for glitches is a level difference between the two pieces
edited together. If the band is playing loudly in one section and the
editor attempts to edit into a spot where the band is playing softly,
the edit will be apparent. Choosing a different edit point where the
sounds are better matched in level can sometimes cure this. Otherwise,
the material needs to be remixed to better match the levels between the
two pieces.
Another set of possible problems surrounding edits involves dropouts,
where the level literally drops out momentarily at the edit point. The
most common reason for dropouts around edits is jagged cuts that do
not quite match up. The other common reason has to do with bad
tape emulsion, which is usually caused by dirt, dust, grease, and oils
deposited from ngers onto the emulsion, the result of excessive tape
handling. As mentioned earlier, tapeparticularly tape emulsion
should always be handled with the greatest of care and as little as
possible.
Digital Editing
Now that we have a grasp of the basics of razorblade editing, lets
briey examine the differences and similarities between that and digital
editing. Although some may claim that digital editing is so radically
218
different than razorblade editing that they defy comparison, others
believe that digital editing is a natural extension of razorblade editing,
much in the same way that digital signal processing is a natural out-
growth of analog signal processing, despite the inherent differences in
the underlying technology. Analog editing and digital editing are simply
different tools at the audio professionals disposal, both of which
accomplish the same goals. As one wise student of mine pointed out,
both he and his grandfather have hammers. They may look different,
but they both bang in a nail.
Many digital tape recorders offer a razorblade option, but there are
two main difculties in this process. First, as sensitive as analog
magnetic tape is, digital tape is far more sensitive to oils and dirt
that occur on human skin. As such, some manufacturers recom-
mend wearing latex gloves and sometimes face masks while editing
digital tape with razorblades. When these machines were rst intro-
duced in the early to mid 1980s, some manufacturers even sug-
gested editing in a white room, a room free of dust or dirt of
any type. This is an uncomfortable and inconvenient way to edit.
As digital tape technology has improved, many of these restrictions
have eased.
Figure 8.6 A screenshot of Pro Tools LE 7. Photo courtesy of Digidesign.
Chapter 8 Edi ti ng
219
Furthermore, methods such as rocking and shuttling are ineffective
with digitally recorded material. Analog recordings can be played at
various speeds, and the heads will read the magnetic eld and repro-
duce the data recorded. Digital recordings, on the other hand, can only
reproduce when played at the speed at which they were recorded, since
an analog of audio information is not stored, but a series of ones and
zeros that represent audio information. To get around this problem
and allow the engineer to shuttle and rock digital tape, most manufac-
turers of digital multitrack machines add a few analog tracks, which of
course reproduce stored audio information at any speed. An editor will
send a rough mix to these tracks, or perhaps the two or three most
important instruments to hear that are critical to identify the in and
out points for the edit. When they rock the tape, they will reproduce
the information stored on these analog tracks.
While razorblade editing of digital masters is difcult by comparison,
digital editing on a computer may be the simplest form of editing imag-
inable. Unlike razorblade edits, computer editing is non-destructive;
the engineer can always return to a previous edit or the beginning of
the project with a simple keystroke, rather than trying to gure out
Figure 8.7 The Sony 3324 Digital Multitrack Recorder recorded 24 tracks of digital
information on a 1-inch tape. Photo courtesy of Sony.
220
which bits of tape go where. Much like the way a synthesizer patch can
be truncated in various ways, then returned instantly to the original
sound, computer edits allow the editor a great deal of freedom. This
form of digital editing has become the most popular method of radio
production due to its simplicity and speed, and it has a substantial and
still-growing following in recording studios and among sound
designers.
Although the variety of software currently available makes a rundown
of every command and method impractical in a book of this scope,
there are some universal methods, which are based on the same
theories as razorblade editing. Of course the possibilities of digital
editing far surpass that of razorblade editing, both in ease and quality.
If an analog editor wanted to repeat one word on the beat 10 times, he
or she could start by dubbing that word. Then the editor would cut
that word to t the beat, and then make small adjustments so the tim-
ing was perfect. Then he or she could repeat the process nine more
times. Or the editor could do it digitally by copying the word and
regenerating it with 10 keystrokes. While the former process could
chew up a few hours, the latter process would be accomplished in a
matter of minutes.
Figure 8.8 A screenshot of Pro Tools HD 7. Photo courtesy of Digidesign.
Chapter 8 Edi ti ng
221
All digital editing software will enable the editor to copy, delete, or
move any section, and then undo anything he or she wishes. Once cop-
ied, any section can be regenerated at will in any location. Cuts can be
changed by fractions of time much smaller than the best editor can cut
with a razorblade. Then they can be moved back, and then moved again
the other way. If the client says, Lets see what it sounds like coming in
a hundredth of a second earlier, a digital editor can do it. The best
razorblade editor can neither add nor subtract a hundredth of a second.
While the methodology differs in that you are using a keyboard and
mouse rather than a razorblade and block, the basic functions remain
the sameremoving unwanted noise and reordering recorded material
in a way that is not noticeable to the end user. Regardless of whether
you work in a digital or an analog environment, the next aspect of audio
we need to add is how we manipulate those signals through various
types of signal processing.
Exercises
1. Record yourself as you count backwards from ten to one at an
even pace. Edit what you have recorded to reorder it as
counting from one to ten. Add ambient noise between ve and
six so there is double the space between them as between the
other numbers. (Hint: If you count fast, this exercise will
become more difcult.)
2. Pick a song with which you are familiar, preferably a song with
a pretty standard verse chorus verse chorus structure.
Reverse the choruses and verses. If there is a bridge, remove it.
Try dubbing one chorus and doubling up the nal chorus.
Additional Reading
2004.
222
Springer, 2005.
Press, 1993.
Focal Press, 2002.
Lehrman, Paul D. and Tim Tully. MIDI for the Professional. Amsco
Publications, 1993.
Press, 1989.
Utz, Peter. Making Great Audio. Mendocino, CA: Quantum,
1989.
Chapter 8 Edi ti ng
223
Press, 2000.
Sams, 1989.
224
9
Signal Processing
P
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t
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e
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g
i
c
.
225
W
e learned how an audio professional manipulates the whole
track through editing, by eliminating unwanted moments in
speech or music, or by reordering an entire composition.
Now lets examine more subtle ways in which the engineer can manip-
ulate the individual signals or the mix as a whole. This requires a grasp
of the basics of signal processing.
The proper use of signal processing gear is critical to the success of
todays audio professional. As such, it is necessary to possess a theo-
retical and practical understanding of the use of audio signal processing
devices, analog, digital freestanding, and plug-ins. This chapter will
enable the engineer to competently and condently employ a variety
of effects and processors in their proper context, as dictated by the
demands of the production. Ideally, the audio professional will do
more than merely learn to operate signal processing equipment; a
young engineer should begin to learn to make critical decisions about
when the use of signal processing is needed and what specic type of
signal processor is appropriate. As with much of audio engineering,
this decision-making process can be very creative and highly individu-
alized. Signal processor choices can help create an engineers signature
sound. Conversely, a large portion of many engineers signal processing
repertoire is very common, straightforward, often repeated, and easily
quantied for the edication of the beginner.
Operating compressors, expanders, reverbs, delays, and digital multi-
effect processors are standard duties for an audio professional. The
multitrack recording environment, production room for sound design,
control room for broadcast, or stage for sound reinforcement is almost
unimaginable in todays audio culture without these devices. Anyone
serious about entering the world of audio engineering needs to have a
strong theoretical and operational understanding of a wide variety of
signal processing devices and applications. In many ways, sound
design and the sound of modern music is the sound of signal process-
ing. This is clearly understood by electronic musical instrument manu-
facturers and software designers, since virtually all audio software,
synthesizers, and drum machines have built-in signal processors. Its
the reason why guitar players buy pedals and the reason why spring
226
reverb, an early form of signal processing, is installed as standard
equipment on most guitar amps.
For the purposes of this chapter, signal processing applications have
been divide into four categories:
1. Signal processing used to overcome the limitations of
equipment.
2. Signal processing used to recreate ambience lost to close-miking
techniques.
3. Signal processing used to change timbre.
4. Signal processing used for fun or to create new sounds and
ambiences.
It is hoped that the creative and the mundane will both be given their
due. There must be a good balance between theoretical background
understanding, step-by-step operational savvy, and creativity when
determining that a signal processor is necessary and when selecting
which signal processor to use.
Historical Overview
Muchof the reasoning behindthe development of early signal processing
lies in the changes that the recording industry has undergone. Early
recordings, before multitracking was available, relied heavily on the
room in which the performances were played for ambience. The
size, shape, and materials from which a concert hall, ballroom, or stu-
dio were made, as well as the microphones used to make the record-
ings, were key factors in determining the character of the sound of a
recorded ensemble. Early monophonic recordings used few micro-
phones. In order to pick up all the musicians, these microphones
were somewhat distant from the ensemble, simulating the position of
a listener in the room. The microphones, because of their distance from
the ensemble, captured sounds reected by the rooms surrounding
surfaces. Often the band would simply use one microphone set up in
the center of the room, and the musicians would step up to the
Chapter 9 Si gnal Processi ng
227
microphone for their solo. Various concerts halls, ballrooms, and stu-
dios around the world became famous for the character of their sound,
such as Carnegie Hall in New York, and were sought out by artists
and producers for radio broadcasts, live performances, and recording.
Virtually no signal processing was used on these recordings due to
the limitations of the existing technology and the lack of need for
processing, since the recording process already incorporated a natural
room sound.
As technology improved and the sophistication of audio consoles and
tape machines increased, more microphones were used. By using mul-
tiple microphones, audio professionals were able to better isolate one
instrument, which could then be recorded on one track of a multitrack
recorder. The advent of stereo and multitrack recording resulted in
experimentation and brought about new philosophies in using and
eventually losing room sound. Close-miking techniques became the
prevailing style. When a sound source is miked closely, the acoustic
phenomena caused by the surrounding environment have little or no
effect unless separate microphones are used to capture them. Unfortu-
nately, these close-miked signals lack the natural ambience of an envi-
ronment, and as a result they sound at and one-dimensional. To a
great extent the development of signal processing equipment is a result
of close-miking technique and multitrack recording, something of an
effort by audio professionals to recreate the environment they elimi-
nated while trying to maximize the benets of new technologies.
Spring reverbs, now commonly contained in guitar amps, were the rst
of the devices created to simulate room sound. While it is not recom-
mended to either kick or pick up one corner of a guitar amp and drop
it less than an inch onto the oor, either of these acts will produce a
springy sound; this is the spring bouncing around. Spring reverbs are
not very effective at simulating rooms because they are very bright and
have an unnatural sound; however they do give a dry, or unaffected,
sound some sense of depth.
The next development in simulating room sound had far more impact
on the industry, and its technology is still commonly used today. In a
plate reverb, a thin metal sheet is mounted under tension in a box.
When signal is fed to the plate, the waves travel through the plate,
228
bouncing back from the sides, simulating the way sound waves bounce
through a room. By mounting a pickup on the far end of the plate, the
resulting signal will be analogous to the same wave traveling through a
room. Two pickups can be installed on the far corners of the plate from
the side where the signal was originally fed for a stereo plate reverb. The
decay time, or the simulated size of the room, can be adjusted by adjust-
ing a panel called a damper. When the wheel on top of the enclosure is
turned, a panel swings toward or away from the plate. If the damper is
swung away, the plate can vibrate more freely, creating a longer reverb
time that simulates a larger, more reective room. If the damper is swung
close, the airspace between the plate and damper is reduced, preventing
the plate from vibrating freely and reducing the decay time, which sim-
ulates a smaller, more absorptive room. Plate reverbs are analog signal
processors that provide a warm, natural sound, which explains why they
are still sought after. Good-quality plate reverbs have entered the cate-
gory of revered vintage gear (see Figure 9.1).
Another early technique for simulating room size that was popular in
the 1950s and is still in use today is tape delay. This effect dened the
rockabilly sound and, along with its digital counterpart, it is equally
popular in punk and hip-hop. Tape delay utilizes the physical distance
Figure 9.1 An EMT 140 analog plate reverb,
with the front of the case removed. Photo
courtesy of EMT.
229
between the record head and the playback head on a tape machine, as
described in Chapter 3. An audio signal is sent to the tape machine,
recorded, then played back on the repro head and returned through the
console, where it is mixed in with the original signal.
This difference in physical location between these heads caused a prob-
lem with keeping overdubs in sync, which we solved thanks to sync
mode, discussed in Chapter 3. Now we are using this physical distance
to create a time difference between the signal arriving at the record
head and the signal being played back from the repro head, which
simulates the early reection of a sound. This establishes the distance
from the sound source to the closest wall; in other words, delay helps
to describe the shape of the room. Tape delay can be adjusted to sim-
ulate different-sized rooms or just for fun and thickness by adjusting
the tape speed; faster tape speeds return the signal faster, describing
smaller rooms with closer walls. While digital delay has effectively
replaced tape delay, sometimes an audio professional will be deep
into a mixdown, looking for just one more delay. In these instances,
as well as in home studios where the digital delay is being used else-
where or is nonexistent, tape delay is still a ne option.
As multitrack recording and close-miking techniques became the stan-
dard, the acoustics of the roomin which the music was recorded became
less of a factor in the nal product. The development of synthesizers
and drum machines also contributed to a loss of natural acoustics.
Increasingly, the sound of contemporary music (particularly pop,
dance, and hip-hop) became the sound of processing devices used to
simulate environments and create special effects.
As a quick rundown of the changes in subsequent decades, the psyche-
delic late 1960s spurred the use of special effects in music production.
Artists such as the Beatles, whose efforts were groundbreaking in early
multitracking (thanks to the pioneering efforts of Les Paul), experi-
mented with effects. Jimi Hendrix was also extremely creative with
effects. Artists like these helped to create the foundation for decades
of experimentation with delay, feedback, and reverb. These effects,
along with early preamps, equalizers, compressors, and limiters,
were executed with tube processors, which are still sought after due
to their warm sound (see Figures 9.2 through 9.7).
230
Solid state electronics arrived in the early 1970s. Although the solid
state equipment was quieter and more compact, sonically it provided
a much colder sound. Some audio professionals combine the two,
achieving a cold, tense edge in their music through the solid state
equipment blended with the warmth of tube processors.
Figure 9.2 A classic tube signal processing device that is still used today, the Universal
Audio 2-610 Tube Preamplier. Photo by Gary Gottlieb.
Figure 9.3 Among todays most sought after tube signal processing gear are the Pultec
EQH-2 and the EQP-1 (pictured) equalizers. Photo by Matt Allen, courtesy of Blackbird
Audio Rentals.
Figure 9.4 Famous for its smooth, warm sound, a classic tube signal processing devices
is the Teletronix LA-2 tube limiter. Photo by Matt Allen, courtesy of Blackbird Audio
Rentals.
231
The rst digital processors also arrived in the early 1970s. In 1970,
Eventide Clockworks produced the rst digital delay, the 1745, which
was one of the rst digital products on the market. Unlike its analog
predecessor, it allowed the selection of specic delay times. As the
digital onslaught continued, units such as the Lexicon 224 digital
Figure 9.5 The Fairchild 670 compressor is still one of the best-loved tube signal proc-
essing devices. Photo by Matt Allen, courtesy of Blackbird Audio Rentals.
Figure 9.6 A classic signal processor still found in many studios, the dbx 160 compressor.
Photo by Gary Gottlieb, courtesy of Four Seasons Studio.
Figure 9.7 The Urei/Teletronix LA-3A limiter, another classic tube signal processing
devices. Photo by Matt Allen, courtesy of Blackbird Audio Rentals.
232
reverb arrived (see Figure 9.8). These devices offered users control over
many parameters, and, at rack-mountable size (about the size of two
large telephone books), they were somewhat more compact than the
seven-footby-four-foot plate reverbs (see Figure 9.9).
In the early to mid 1980s, Solid State Logic consoles offered inboard
compressors, limiters, and gates, bringing much of the signal process-
ing to the engineers ngertips (see Figure 9.10).
Digital Signal Processing Technology
Here in the early twenty-rst century, digital technology has arrived
and been accepted and incorporated in full force. Advances in digital
audio have resulted in dramatic changes in recording hardware and
software. These advances, however, are far outpaced by the develop-
ment of high-quality digital processors and plug-ins in every price
range. Some audio professionals believe that regarding all types of sig-
nal processors, analog is little more than recording history. Virtually
all signal processing can be done in the digital domain, although engi-
neers will frequently attempt to introduce an analog mic preamp, a
tube mic, a tube compressor, or a plate reverb to increase the warmth
of the track and counteract what some engineers perceive as an inher-
ent coldness in digital recording.
Regarding combining digital and analog, it is worthy to note at this
point that signals deteriorate as they transfer from digital to analog
Figure 9.8 One of the most popular digital delays is the Lexicon PCM 42. Photo by Gary
Gottlieb.
Figure 9.9 The Ensoniq DP/4 Parallel Effects Processor, a digital reverb. Photo by Gary
Gottlieb.
233
and vice versa. To maintain signal quality, remain in the digital domain
after an analog-to-digital conversion has been performed, and remain in
analog after a digital signal has been converted to analog. This is part of
the reason why tube microphones and tube preamps are so popular
they are in the part of the signal ow that occurs naturally, as a singer
sings into a microphone or a violin plays into that microphone, before
Figure 9.10 Solid State Logic brought the outboard rack into the console in the 1980s
by putting signal processing in every module. Photo courtesy of Solid State Logic.
234
the signal is converted to digital at the console, thereby introducing
warmth with a minimum of conversions.
Lets take a closer look at analog and digital signal for a moment and
explore their differences and the way we manipulate those differences.
Analog signals are continuous and proportional to the original acous-
tic waveform, while in their most simplistic form digital signals are
modulated pulse waves representing a series of ones and zeros,
which in turn represent an amplitude value and a time address for a
series of samples of the original analog waveform (see Figure 9.11).
Processing of digital signals involves manipulation of the numbers that
represent the sampled waveform. In other words, we may manipulate
the algorithm by imposing another algorithm upon it that states that
every time there are three consecutive zeros followed by a one, the
number following that one should always be a zero. By doing this
we are creating a new algorithm, which is the effected or processed
signal. Digital processors outperform analog processors with respect
to signal-to-noise characteristics, size, cost, and programmabilityin
other words, they are smaller, cheaper, and quieter and they offer
Figure 9.11 Digital signal processing in Cubase. Photo courtesy of Cubase.
235
more options in manipulating the sound wave. The only disadvantage
is that some engineers think of these processors as sounding cold.
While it is ideal, as mentioned earlier, to select and remain in either the
digital or the analog domain as much as possible, it is often impractical.
A project studio may contain primarily analog equipment with one or
two digital processors and a digital recorder. A home studio may consist
of a computer with software operating as a console and recorder and
one or two pieces of analog processing gear. With all the equipment
available today at reasonable prices in both the digital and analog
domains, any combination is possible. As an audio professional, if
you nd yourself in this situation, it will be valuable to keep in mind
that successive digitalto-analog and analogto-digital conversions will
deteriorate the signal. Try to arrange the signal path such that all digital
components are connected to each other through their digital I/O ports,
and that the path stays in that domain as long as possible before being
converted to analog. Similarly, if there are several analog devices along
the way, try to set the signals path to encounter them one after another,
such as the tube microphone to analog preamp mentioned earlier,
before entering the digital domain of the console.
Classication of Processors by Types
of Perception
There are many options when trying to group, or classify, signal pro-
cessors. Classication by types of perception is given only for the sake
of cataloging processor types. It may be helpful in deciding when and
what type of processor is needed to solve a given problem by relating
the type of problem to one of the ve basic perception types discussed
in Chapter 2, such as problems related to loudness requiring dynamic
processors or problems relating to timbre, pointing the audio profes-
sional in the direction of equalization.
We will also break down processors that affect pitch, envelope and
proximity, or location. While some processors could appear in more
than one category since the listings are related to applications rather
than operating principal, we will try to keep things clear and simple.
We will also discuss processors that we use for fun.
236
Processors That Affect Loudness
Processors that affect loudness, or dynamic processors, will alter the
natural changes in volume of an audio program by either increasing or
decreasing its dynamic range. These include compressors, limiters,
expanders, and noise gates. Lets begin with compressors.
Compressors are devices that decrease the dynamic range of program
material (see Figure 9.12).
A compressor will output a scale model of an input signal and is used
in applications where the natural dynamics of a program can be prob-
lematic. Its applications include:
n
Optimizing level
n
Preventing masking
n
Reducing or eliminating peaks
n
Protecting speakers
Compressors can be used to optimize tape saturation and compensate
for the limitations of the storage medium. By controlling the level of
peaks, the overall program can be brought closer to optimum tape sat-
uration level, thereby more effectively masking the noise oor and
improving signal-to-noise ratio. In radio, this translates into packing
maximum signal into a transmission without exceeding federal permit
restrictions.
Compressors are also used to prevent backing tracks from masking a
dynamic vocal performance. When a vocal performance is dynamic, or
Compression with 2:1 ratio
+4 db
+2 db
Threshold
Figure 9.12 A compressor reduces the dynamic range by reducing signal above the
threshold according to the compression ratio. In this case, a signal that was 4 dB above
the threshold was reduced to 2 dB above the threshold, due to a 2:1 compression ratio.
237
varies in volume, and the soft moments are being masked by the back-
ing tracks, a compressor can smooth out the vocal performance, keep-
ing its level consistent in relation to the band. Conversely, if a vocals
loud moments are masking the band, a compressor can smooth these
peaks and valleys of level (see Figure 9.13).
Transients, as discussed in Chapter 2, are insidious. Transients and
other peaks in the dynamics of musical instruments can be evened
out with a compressor. Electric bass guitars output uneven volumes
as a player moves from string to string or up and down the neck.
Bass players employ different techniques, such as picking, plucking,
slapping, and popping strings, all of which may result in an unaccept-
ably large dynamic range or transients. Electric guitarists also employ
a variety of techniques that may require compression to smooth out a
performance, including switching from power chords to a clean
rhythm sound, which may cause extreme volume changes.
Compressors are excellent for speaker protection. By disallowing a
signal to exceed a preset level, compressors will prevent speakers from
being damaged by peaks in the overall level of a program. This is com-
monly used in theatre and sound reinforcement situations, where an
excess of level could damage the speakers and a fuse is not an adequate
Figure 9.13 Cubase compression. Photo courtesy of Cubase.
238
solution, since the audio professional cannot stop a performance to
change a fuse (see Figure 9.14).
Whether you are recording digitally or in analog, recording has a limit
to its dynamic range. Program material is often compressed to opti-
mize signal-to-noise and distortion characteristics, because exceeding
the high-end limits of reproduction equipment or attempting to pack
too much level onto a storage medium will result in distortion in either
digital or analog. Reproducing a compressed recording sometimes
requires expansion to restore dynamics lost to compression.
In essence, compressors, also known as leveling ampliers, are used to
decrease dynamic range. A compressor is an amplier whose output
level decreases when an input signal rises above a user-dened threshold.
One of many adjustable parameters on a compressor, the threshold, is
the level measured in volts or decibels, at or above which gain reduction
begins. The other parameters we need to understand are compression
ratio, which is a comparison between changes in volume in the input
and output signals once the input signal level rises above the threshold;
attack, which is the time in milliseconds it takes for gain reduction to
begin once the input signal rises above the threshold; and release, the
speed in milliseconds in which gain reduction of the output signal ceases
once the input signal falls below the threshold. In other words:
1. A signal enters our compressor.
2. We have set the threshold 8 dB below its highest peak.
3. We have set our ratio at 2:1.
4. We have set a fast attack and a slow release.
Figure 9.14 An Eventide Clockworks 2826 Omnipressor compressor, circa 1971. Photo
courtesy of Eventide Clockworks.
239
5. When the signal enters the compressor, it will react quickly
(due to the fast attack).
6. The signal will be reduced by 4 dB, because we have a 2:1 ratio
and the signal entered 8 dB above the threshold.
7. After the signal leaves the compressor, it will return to normal
slowly, due to the slow release.
If we changed our ratio for the same input signal to 4:1, the output
signal would be only 2 dB above our threshold.
Limiting is an extreme amount of compression. Any compression at a
ratio of 10:1 or greater is called limiting (see Figure 9.15).
When limiting, the threshold is usually set comparatively high so as to
only stop extreme peaks in program material. When compressing, the
threshold is frequently set lower so that a lesser degree of compression
is present more often. Since many compressors cause an audible change
in the tone of a signal, limiting (with the higher threshold) is sometimes
the more desirable approach. When dealing with a musical program,
the choice between limiting and compressing is a matter of taste. There
are some applications in which limiting is the standard, such as limiting
for speaker protection in sound reinforcement and sound design, and
limiting of broadcast signals to serve a larger audience. To start an
argument in a room full of engineers, ask one of them whether limiting
or compression is better on vocals. This is a hot topic among engineers;
although I believe vocals should be compressed but never limited, I
know many engineers who stand opposed to me on this issue.
The opposite of a compressor is an expander, which is used to increase
dynamic range. An expander is an amplier whose output level
Figure 9.15 The Manley ELOP Stereo Electro-Optical Limiter. Photo courtesy of Manley
Labs.
240
decreases by a preprogrammed amount when an input signal falls
below a user-dened threshold. In other words, expanders increase
dynamic range by making the softest moments softer or pulling
those quietest sounds down toward the noise oor. This can be par-
ticularly useful in situations where there is leakage from one micro-
phone to another, such as in a drum kit where the snare drum is
being picked up by the hi-hat microphone. The snare drum can be
made softer on the hi-hat track with an expander, thereby cleaning
up the drum track and allowing the snare drum to appear in the mix
only from its own microphone or track (see Figure 9.16).
As with the compressor, the parameters for an expander include the
threshold, a level (set by the user and measured in volts or decibels,
below which expansion begins), the expansion ratio (a comparison
between the input and output signals once the input signal falls below
the threshold), attack (the speed in milliseconds at which an input signal
below the threshold will cause the expander to react), and release (a
user-dened time period, in milliseconds, before expansion ceases
once the input signal rises above the threshold). Since the expander is
the opposite of the compressor, the unit engages when the signal falls
below the threshold rather than when it exceeds the threshold.
Expanders also include two other parameters that dont exist on com-
pressors. Range is the amount of level reduction, in decibels, that will
be applied to the output signal once the input signal falls below the
threshold, and slope is a choice between a linear or exponential
Figure 9.16 The Nuendo
Finalizer offers signal pro-
cessing, including expand-
ing, compressing, limiting,
and more. Photo courtesy
of Nuendo.
241
(logarithmic) rate applied to expansion, which is a technical way of
saying a choice between expansion proceeding at a constant or accel-
erating pace. The best way to choose between these often lies, as usual,
in your ears. If you are unsure as to which setting will sound best, listen
to both and trust your ears.
Just as limiting is the extreme form of compression, gating, or using a
noise gate, is an extreme amount of expansion. Any expansion at a
ratio of 1:10 or greater is considered gating. Gating is used for auto-
matic muting of percussive or seldom-used tracks; it can eliminate tape
noise introduced into the mix while there is nothing happening on that
track. It can also eliminate the snare drum from the hi-hat track, as
mentioned previously, or eliminate the hi-hat from the snare track if
you prefer.
Gating has another interesting option, called external triggering or
keying (see Figure 9.17).
Keying allows the engineer to take an unrelated sound and open the
gate only when that sound occurs, or more accurately, prevent the
sound being modied from entering the mix when the keying signal
is not occurring. For example, if you are imitating Trent Reznor of
Nine Inch Nails, you will want to replace the snare drum with white
noise. By running a track of constant white noise through a gate, key-
ing it off the existing snare drum track, and removing the snare drum
from the mix, you will effectively replace every snare drum hit with a
burst of white noise. Here is how it works: The gate is shut, not
(Gate Will Only Open
When This Signal Occurs)
Noise Gate
(Expander)
External Source
Signal In Signal Out
Key Input
Figure 9.17 External keying of noise gate.
242
allowing the white noise into the mix, since the signal of the key (snare
drum) is below the threshold. When the snare drum is hit, the signal in
the key exceeds the threshold, and the gate stops reducing the level
of the white noise, or opens, allowing the white noise into the mix.
When the snare drum ends, its level drops below the threshold, and
the gate reengages, reducing the level of the white noise and removing
it from the mix. The attack and release can be manipulated to make
the white noise hit sharper, or last longer. In addition to replacing a
snare drum with white noise as discussed, an engineer can supplement
the sound of the snare drum in this example by leaving the snare drum
in the mix and blending the white noise in with the original snare. The
audio professional can choose the relative volume of the original signal
that is keying the device and the volume of the sound being added by
the keyed gate. The triggering sound that is plugged into the key does
not need to enter the mix; the path for the key and the path for the
signal being modied are discrete within the unit. The relative volume
of the two is entirely at the discretion of the engineer.
Processors That Affect Pitch
Processors that affect pitch are typically used to double the sound of
vocals or for effects on guitars, basses, and keyboards, where a slight
de-tuning will create an interesting character. Harmonizers are extreme
pitch shifters that are also used where pitch shifting will create a very
unusual and mechanical trick vocal sound (see Figure 9.18).
Figure 9.18 The Yamaha S Rev1 multi-effects processor is one of many signal process-
ors that offers pitch shifting capabilities. Photo courtesy of Yamaha.
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Prince often uses these vocal characters. Another uncommon usage
employed by Prince (or the artist formerly known as Prince, depending
on what year in the 80s you happen to beam into) is to shift the pitch
of kick and snare drums to create strange and unique percussion
sounds when mixed in with the originals. Harmonizers are also a
mainstay of radio production where a voice needs to be made
extremely high or low in pitch. Harmonizers electronically pitch
shift an input signal. The further away from the original pitch we
get, the more mechanical and electronic it will sound.
Chorusing is another common method to shift pitch more subtlya
combination of pitch shifting and short delays. A chorusing device is
used to make an individual voice or other input signal sound doubled
as if more than one instrument is present. There are other applications
for chorusing where doubling is not necessarily the goal, such as tightly
chorusing bass, which results in a fat, anged sound, or a chorused
guitar, which thickens the sound and makes it sparkle. As with so
many effects these days, chorusing, harmonizing, and other forms of
signal processing can be found in simple, rack-mountable boxes called
multi-effects processors (see Figure 9.19).
Processors That Affect Timbre
The primary processor that affects timbre is the equalizer. Equalizers
are frequency-selective ampliers. Like the tone control on a stereo, an
equalizer will allow select parts of the frequency spectrum to be
increased or decreased in level, without affecting the pitch. This is
because equalizers change the harmonics, while affecting the fundamen-
tal frequency minimally in level, if at all, as described in Chapter 2. Most
equalizers we use in audio production have controls that allow for the
selection of the desired frequencies and a volume control to increase or
decrease the volume of these frequencies. Equalizers can make a sound
Figure 9.19 The Yamaha SPX90II Digital Multi-Effects Processor. Photo by Gary Gottlieb.
244
darker or brighter, helpsounds t together ina mix, andreduce masking,
all of which will be discussed in more detail later in this section.
A de-esser, a combination effect made up of an equalizer and a com-
pressor, also affects timbre. A bandpass lter set for the sibilant range
is installed at the input of a compressor acting as a trigger, so that the
energy from this part of the frequency spectrum modulates the com-
pressor, which in turn compresses the audio passing through more
heavily when there is a lot of this energy in the sibilant range of vocals
present. The result is that compression increases when particularly sib-
ilant frequencies hit, and therefore the audio containing the sibilance is
reduced. De-essers are particularly useful with singers who are partic-
ularly sibilant, those who hiss their Ss.
Like so much audio equipment, equalizers originated from the Bell
Laboratories for telephone use. Long before digital equipment, micro-
wave technology, or sophisticated repeaters were utilized, equalizers
were created to compensate for signal loss when a voice had to travel
a long distance over wires. It was found that attenuation of a wire-
bound signal was most extreme at select frequencies. Special devices
were made to make the energy level of all the transmitted frequencies
more equal and stress the vocal frequencies. This is where the name
equalizer comes from.
There are several types of equalizers, all with their own features. The
most common equalizers in studio use are parametric equalizers,
which offer the most comprehensive controls and the most versatility.
They allow the user to manipulate more parameters, including the
selection of virtually any frequency within the audible range and up
to four bands at a time. They also allow us continuously variable band-
width, a range of frequencies that will be affected by the amplication
circuitry measured as a slope in decibels per octave from a user-
selected center frequency. While most onboard peaking equalizers fea-
ture xed bandwidth, the parametric equalizer allows the bandwidth
to be varied continuously from a fraction of an octave to three or more
octaves, by adjusting the Q (see Figures 9.20 and, later, 9.23).
All equalizers (with the exception of high-pass and low-pass lters)
contain boost and cut controls, which are the level controls for the
245
Figure 9.20 The two middle bands of the four-band EQs found on the Solid State
Logic 6000 consoles are parametric. The bottom knob of each allows the engineer to
control the Q, or bandwidth of the frequencies effectedin other words, we can select
the width of the bell shape of our EQ. Photo courtesy of Solid State Logic.
selected frequencies. These controls usually span a range of plus or
minus 12 to 18 decibels, allowing the engineer to either increase or
decrease the volume of the harmonics of a sound in that range. These
knobs essentially perform the same task as the bass and treble controls
on your home or car stereo; the main difference is that in your home or
car the frequencies are preset, while audio professionals can usually
select the center frequency of their harmonic manipulations.
The mid-range equalizer(s) on most audio consoles are peaking equal-
izers. The actionof a peaking equalizer is greatest at the selectedor center
frequency, where the center frequency is the peak of a bell-shaped curve.
The effect of the equalizer decreases as you move farther away from the
center frequency. The degree to which this effect decreases is determined
by the bandwidth of the equalizer. The bandwidth of a peaking equalizer
is usually xed, although some equalizers offer a Q control, which
allows the user to either choose or sweep between two xed bandwidths,
one wide and one narrow. A wide band is known as a low Q, while a
narrow band is known as a high Q.
246
Another common type of equalizer is the shelving equalizer. On a
shelving equalizer, all frequencies either above or below the selected
frequency, depending upon whether it is a high- or low-frequency
equalizer, are affected (increased or decreased) by an equal amount,
forming what looks like a shelf. If a high-frequency shelving lter is
set for 14 kHz and the signal is attenuated, everything above 14 kHz
will be reduced, not just 14 kHz (see Figure 9.21).
When attenuating a signal with a high-frequency shelving equalizer,
the audio professional is performing a task that is similar to a high-
pass lter. A high-pass lter will attenuate the level of low frequencies
and let high frequencies pass unaffected, while low-pass lters attenuate
the level of highfrequencies andlet lowfrequencies pass unprocessed. These
can be convenient for eliminating tape hiss (low pass) or a 60-Hz hum
(high-pass) (see Figure 9.22).
Figure 9.21 In this high-frequency shelving EQ control from an SSL, all frequencies
above the selected frequency will be affected, unless the bell button is pressed,
which converts this to peaking EQ. Photo courtesy of Solid State Logic.
Figure 9.22 The high-pass lter (left) will eliminate all frequencies below the selected
frequency, allowing high frequencies to pass through. The low-pass lter (right) will
eliminate all frequencies below the selected frequency. In other words, high-pass lters
affect low frequencies, while low pass lters effect high frequencies. Photo courtesy of
Solid State Logic.
247
Bandpass lters are made up of a combination of low-pass and high-
pass lters. As the cutoff frequencies of the low-pass and high-pass
lters are manipulated, a region between the two is dened. Frequen-
cies within this region or band are allowed to pass unaltered. This is
how the name is derived, as one set of frequencies, or bandwidth, is
allowed to pass through unaffected.
Graphic equalizers are less common in professional audio settings,
except to x room deciencies, where they are set to compensate for
excesses or lacks of certain frequencies as a result of room design, then
locked up out of everyones reach. They consist of a series of sliding
faders, each representing a xed frequency and xed bandwidth.
These faders can boost and cut only at the frequency at which they are
xed. Some graphic equalizers have a bandwidth as small as
1/
16 of
an octave. These are often used in theatre and sound reinforcement
applications to equalize speaker systems and compensate for room
characteristics. Graphic equalizers get their name from the fact that
they form a graphic representation of the way the audio professional
is altering the frequency response of an input signal.
As audio professionals, we nd there is a need for subjective terms to
describe the parts of the audible frequency spectrum. When working
with musicians and producers, an engineer often has to interpret sub-
jective terms or descriptions of feelings into a technical action. It is rare
for a producer to say, Add 3 dB at 800 Hz to the rhythm guitar.
More commonly, a producer might say that the drums need to be slam-
min or that they need more thump. An engineer might be requested to
give the guitar more bottom, or to put more edge on the vocals. There
are no controls on the console labeled slammin, thump, bottom, or
Figure 9.23 Vintage outboard EQ, an Orban/Parasound Parametric EQ. Photo by Gary
Gottlieb.
248
edge. The engineer must, therefore, interpret these terms. This is why it
is necessary to be able to relate these phrases to a part of the frequency
spectrum. Regardless of whether you, as the audio professional,
believe in the validity of these subjective terms, they are the terms
that many of our artists and clients will use to communicate with us.
The subjective terms that we will assign to certain regions of the fre-
quency spectrum do not begin and end at precise frequencies. As the
frequency is increased, a rumble gradually becomes boomy. There is
no set frequency where this suddenly and completely happens. The
subjective terms are used to describe the feeling or experience of
selected harmonics, which is entirely appropriate for the subjective
area of timbre.
Here are some brief andarguable (due totheir subjective nature) descrip-
tions of the subjective terms used in describing frequency ranges:
n
Rumble. 2065 Hz. Sounds that are rich in energy from this part of
the frequency spectrum can be described as rumble. Rumble sounds
like distant thunder or an earthquake. The sound of rumble is
commonly experienced late at night in urban environments, when
the city is quiet, waiting in an underground train station. The sound
of a train several stops away is the sound of rumble. In the country,
the sound of distant thunder can also be described as rumble.
Frequencies in this range add warmth and size to a sound.
Removing these frequencies will make a sound smaller and harder.
The frequencies that produce rumble are so low that it is often
difcult to tell the location of the source due to their long wave-
lengths. Rumble can also be described as a vibration felt under the
feet. Part of the perception of low frequencies has to do with the
feeling of them vibrating the body, which can be experienced in a
dance club where one can feel the bass or kick drum vibrating in his
or her chest. While rumble is felt under the feet, the bass frequen-
cies that are felt in the chest are produced by somewhat higher
frequencies. These frequencies can be described as boomy.
n
Boomy. 60180 Hz. As the frequency increases above the rumble
range, the sound can be said to become boomy. The sounds of
nearby cannons or of thunder directly overhead can be described as
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boomy. These sounds are easier to locate than rumbly sounds, since
their higher frequencies have shorter wavelengths and are therefore
more directional. As an airliner approaches you from a long way
off, the sound is rst heard as rumble, and, as it get closer and then
directly overhead, the sound becomes increasingly boomier. The
feeling of the bass pounding against your chest in a dance club is a
result of the added directionality of these boomy frequencies. At the
low end of the boomy range, these frequencies are more dispersive
and therefore sound softer. As the frequency increases, and the
sound becomes more directional, the boomy range of frequencies
takes on a harder sound and becomes increasingly punchier. Boomy
components in a sound can enrich a kick or snare drum; at the same
time, the higher end of the boomy spectrum can add a certain
oppiness to a kick drum or snare drum.
n
Punchy. 100250 Hz. This range of frequencies overlaps the upper
end of the boomy range. Again, as the frequency increases and the
sound waves become more directional, this range of frequencies
takes on a harder character. The upper end of the boomy range and
the punchy range of frequencies are responsible for the hardness or
denseness, not the edge, of a low-frequency sound. It is this range of
frequencies that will make things thump or kick. This punchy or
thumping, kicking sound is created by these frequencies in con-
junction with the boomy range frequencies. Scaling these two
ranges will produce a variety of hard kick drum and bass guitar
sounds, while removing this range of frequencies makes the sound
of a snare drum t in better with a kick and bass guitar. Removing
too much, however, will result in a papery-sounding snare. As the
frequency increases toward the top of this punchy range, the sound
becomes increasingly harder and takes on a boxy characteristic.
n
Boxy. 200750 Hz. The boxy range of frequencies changes its
character, sounding cardboard-like at the lower end and becoming
increasingly wooden as the frequency rises. Most drum sounds
require removing the lower end of this range of frequencies.
Removing the frequencies that make the drums sound boxy and
cardboard-like gives the drums a warmer, punchier character and
makes their sound more clear. Generally speaking, judiciously
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scaling this range is the key to drum sounds and an overall added
clarity to most mixes. Most often this range of frequencies, which is
generally unpleasant-sounding, is reduced in volume. For some
drums and bass sounds, which have a wooden character, the upper
end of this range is increased. If too much energy is removed in this
range, the sounds will become softer and ufer, especially if there
is a reasonable amount of energy present in the boomy and rumble
range. Reducing the amount of energy in the middle of the boxy
range by a moderate amount will make most of the rhythm section
instruments sound more round or hollow. Again, as the frequency
rises, the directionality increases, and the sounds take on a harder,
more present character. This is also due, in part, to the fact that the
upper part of this range is approaching the area of the ears greatest
sensitivity, where speech lies.
n
Metallic, glassy, papery. 0.755 kHz. In this range the trend
toward hardness as frequencies rise reverses itself. This is because
as we increase the frequency, we move above the range of the ears
greatest sensitivity, the vocal range. This is also due to the fact that
we are entering the range where the fundamental frequencies of
most instruments end. Increasing the amount of energy on sounds
at the lower end of this range will give them a gritty, edgy, and
metallic sound. As the frequency is increased, the edgy character
initially remains, and the sounds become smoother. Further
increasing the frequency reduces the edginess, and the sounds
become softer and papery. Decreasing the amount of energy almost
anywhere in this range will makes things sound darker and
mufed. This effect becomes subtler as the frequency increases.
Because the ear is so sensitive to this range of frequencies, a slight
boost almost anywhere in this range will make an instrument seem
to jump off of a track. This effect will be cancelled, or washed out,
if it is used on too many instruments. It is a real temptation for a
beginning audio professional to increase this range on many
instruments, since each instrument, when listened to alone, sounds
better with an increase in this range. The end result, however, is a
huge frequency bulge that masks many instruments and vocals. The
energy in this range of frequencies is also somewhat fatiguing.
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Boosting this range on many instruments will make the track hard
to endure for repeated listening. Used sparingly, this range can add
a compelling, attention-grabbing quality to a track. The sound of
the attack of many instruments falls within this range. The edge on
rock guitars, bass guitars, and vocals can be highlighted with this
range of frequencies, if done judiciously.
n
Shimmery. 520 kHz. As the frequency is increased to the top of
the metallic, glassy, papery range and beyond, only the upper
harmonics of a sound source will be affected. This gives the feeling
that the aura around the sound, rather than the sound itself, is
changed. This adds a shimmery, halo-like effect to most tracks.
This range may seem to cover a great many frequencies, but it is
actually only two octaves. Cymbals, utes, and instruments playing
in their highest registers are most affected by this range. The
character and ringing quality of the cymbals can be altered here, as
can the edge of a piccolo.
Note: Two or Three More Things About EQ There are two more
things that need to be mentioned about equalization. First of all,
respect subtlety. It is tempting for new audio professionals to crank
the EQ when they find a frequency that they like in a sound. It is
better in most situations either to add no EQ at all or to find that
certain frequency and then add just a hint of it to enhance the overall
feel, rather than bombard your listener with a frequency you just fell
in love with. If it is apparent to the listener that you cranked it up, it is
too much, and you have not done your job to respect all elements of
the mix (not just the one you love). Also, turn it down before you
turn it up. If a sound is too dark, many first-time audio professionals
will automatically assume that they need to add high end. This may
not be the best approach. Adding a lot of EQ to a lot of tracks can
cause them to fight each other. If a track is too dark, start by trying to
pull something out in the low end rather than adding highs. This is
often easier to do anyway, because finding a bad frequency is a
pretty straightforward operation.
First, boost frequency on your EQ in the vicinity of the offending
frequency, then sweep around until you make it sound as bad as
252
you possibly can. Once you have made the sound as odious and
offensive as you can, dial out the frequency you have been adding.
This method is based on the well-accepted idea that it is far easier to
make something sound bad than it is to make something sound
good. If you isolate the nastiest of frequencies by cranking them
way up, and then trim them out, you will have an easier time of
finding those frequencies and you will be following the turn it
down before you turn it up theory.
In context of these subjective discussions, this would seem a perfect time
to discuss masking in more detail. Masking is the most common prob-
lem with early students and beginning audio professionals mixes. As
discussed in Chapter 2, masking is where one sound covers up another,
burying it slightly, blocking it completely, or changing it into a less clear
sound than the engineer desires. Since masking is at its most extreme
where frequency, timbre, and location are similar, to cure it we need
to change at least one of these parameters. Kick drum, snare drum,
and bass guitar overlap in the same parts of the frequency spectrum,
are all usually centered in the mix, and are commonly at about the
same level. This causes a typical problem. The sound of the kick
drum, snare drum, and bass guitar may be acceptable individually,
but when they are added together, one will obscure the other.
Many beginning audio professionals will become frustrated as they push
up the bass to hear it more clearly; then push up the kick, which has now
become buried; then push up the snare, which is masked by the bass and
kick. This becomes a vicious cycle. Unlike the guitars and keyboards,
which can be panned to opposite sides to open them up, or reduce
their masking (as we will explore in Chapter 10, Mixing), all three
of these items want to be centered. Dovetailing them together, or reduc-
ing the amount of energy at a given frequency on one instrument and
then slightly boosting that same frequency on another instrument,
lling the vacated space, will often make both instruments stand out
better and speak more clearly in the mix. For instance, suppose we add
somewhere between 40 and 80 Hz to the kick drum while removing
200 Hz from it, then reduce slightly 70 to 80 Hz from the bass guitar
while adding 200 Hz to it, and remove 200 Hz from the snare, while
253
adding a bit of 5 kHz to it. We have essentially cut a little notch at
70 to 80 Hz in the bass, where the kick and the bottom of the snare sit
comfortably, while cutting a notch at 200 Hz in the kick and snare in
which the bass can shine. Meanwhile, we have helped to make the
kick and snare more distinct to each other by pushing up the snares
high end (5 kHz).
Processors That Affect Envelope
With the increased sophistication and availability of synthesizers and
other computer-generated sounds, an entire generation of musicians
and engineers has entirely new options for affecting envelope (see
Figure 9.24).
In the past there were relatively few ways to affect the envelope of a
sound. It could be done with a razorblade during editing, by simply
truncating an attack or release, although this is a fairly clumsy method
of changing the envelope. It could also be done by changing the attack
and release settings on an expander or compressor. At rst thought to
be undesirable, since it made the use of a compressor more noticeable
if the attack was not set subtly, it was used later for unusual effects.
One such effect involved delaying the attack of a drum for simulated
backwards effects; another involved changing the envelope of a reverb
plate or program, which could create a disorienting effect.
Time
A A S S D D R R
A
m
p
l
i
t
u
d
e
Envelope of a plucked
string instrument.
Time
A
m
p
l
i
t
u
d
e
Envelope of a bowed
string instrument.
Figure 9.24 Envelopes are far easier to manipulate now. Thanks to digital technology,
it is as easy for the engineer to change the envelope as it is for the musician.
254
Now anyone with a synthesizer can truncate or extend any sound, cut
off an attack, switch the decay and sustain, or otherwise manipulate
the envelope to create a sound that is new and different. Engineers,
too, in digital workstation situations, can dissect and reassemble
sounds to their own personal specications, snipping a bit here and
placing it there, changing the envelope in the process.
Processors That Affect Location
As discussed earlier in this chapter, close-miking removes the natural
ambience that an environment adds to the sound of a recording by
focusing closely on the sound source. The room in which an audio
event occurs played a larger role in shaping the character of the
sound before the advent of close-miking, as did the position of the lis-
tener in the environment. When close-miking, it becomes necessary to
articially create an ambience, or perception of relative proximity, in
order to produce recordings that have depth and a sense of location,
and that sound more interesting. There are many devices available that
may be used to simulate lost ambience. First, let us review exactly what
has been lost.
Early reections are part of the sound we associate with reverb. The
rst few reections to arrive at the listeners ears just after the direct
sound can sometimes be discerned as discrete from the reverberation,
although even if they are not, we pick up on these subconscious cues to
learn about the surfaces nearby that created those reections. These
are the early reections. The volume difference between early reec-
tions, the direct sound, and the reverberation increases, as does the
time of incidence, with the size of the enclosure creating the reverber-
ation. Early reections can be simulated by the built-in parameters of a
digital reverb, or they can be created by using a pre-delay inserted in
the path as you send a signal to a reverb unit.
Made up of multiple reections of sound waves from surrounding sur-
faces, reverberation is a sound that persists after the driving force, the
direct sound, has been removed (see Figure 9.25).
The reections occur so close together that they are indistinguishable
from each other and appear to be constant. Reverb devices simulate
255
this sound. An input or monitor signal can be made to sound as if it is
in a variety of known listening situations, such as concert halls, stadi-
ums, churches, gymnasiums, small rooms, and so on. The audio pro-
fessional can also place the listener in a specic location in the room or
halltoward the front, toward the backby adjusting the reverb. Dis-
tance from the observer to the sound source within the environment
can be simulated by altering the balance of the wet and dry signals or
by manipulating the early reections. In addition to all these possibil-
ities, available in either analog or digital, modern digital reverbs allow
simple creation of rooms that could not possibly exist, such as inverse
rooms or rooms with huge pre-delay and short decay times.
As a method of quantifying reverb, we have RT-60, which stands for
Reverb Time Minus 60 Decibels. This is a way of measuring reverb,
and it represents the amount of time it takes for the amplitude of the
reections to drop 60 dB once the driving force has been removed. The
average listener is familiar with this phenomenon from high school or
college gymnasiums, indoor swimming pools, or concrete staircases.
Generally speaking, the larger the room, the longer the reverb lasts
after the driving force has been removed, and the higher the RT-60.
The timbre of natural reverb is determined by the substances from
which the surrounding surfaces are made. Harder substances will pro-
duce a greater number of reections, while more rigid substances, such
as concrete as opposed to wood, create brighter reections.
Figure 9.25 Reverberation, or reverb, is a series of indistinguishable random reec-
tions, growing denser and diminishing over time. Based on original diagram courtesy
of Tannoy.
256
Echo, like reverb, is made up of reections off of surrounding surfaces.
The difference between reverb and echo is that each echo can be dis-
cerned as a discrete individual reection, indicating a larger space. Dis-
tance or room size may be simulated with echo by adjusting the
amount of time it takes for a reection to follow the original sound.
Because sound travels at a constant rate in a uniform medium, longer
delay times suggest greater distances, or larger rooms. If you were to
ask a child to emulate the echo from the Grand Canyon, the child
would intuitively wait a long while before making the sound of the
reection. That is because the sound must travel a long distance to
reect off of a surface and then return to the listener.
While you will rarely be called upon to simulate a roomof a precise size,
you can approximate an actual roomusing the speed of sound learned in
Chapter 2. Depending upon elevation and other atmospheric condi-
tions, sound will travel approximately 1,130 feet per second. If we
break this down to more understandable distances, we could say that
sound takes approximately 9 milliseconds to travel 10 feet or that sound
takes approximately 90 milliseconds to travel 100 feet. In other words,
if a client species that the room this band is playing in should sound as
if its 200 feet long and the observer is positioned at the far end, try a
pre-delay of 180 milliseconds.
The location of a sound source relative to the walls of an enclosure
may also be simulated by the positioning, or panning, of that sound.
Panning the original sound against the positioning, or panning, of the
reections offers the audio professional all kinds of fun and the ability
to further manipulate the listeners perception of the space. Properly
positioned, the original and delayed signals can create the impression
of three-dimensional ambience, or depth, the greatest achievement in a
beginners mix. While were discussing panning, some mention should
be made of dynamic panning, which may be used to simulate the
movement of a sound source within an environment. An auto panner
is a device that can be programmed to alter the position of a signal
between the left and right channels at predetermined rates and
times. For a truly bizarre effect, try using an auto-panning program
on the output of a reverb program. Your client will probably hate it,
but youll never know unless you try it.
257
Processors That Are Fun
Phase shifters and angers fall into a somewhat different category
from other processors in that they are almost always used for fun. Pre-
viously mentioned effects, such as chorusing and harmonizing, can
also be used to make a sound more fun. Through different methods,
phase shifters and angers split an input signal and then recombine it
with a dynamically changing phase relationship, causing a sweeping or
swishing sound. At one time phase shifters and angers were a staple
for electric guitar and electric bass sounds, and they are still common
pedals used by guitar players to process their signals on the way to
their amplier.
So who really needs a anger or a phase shifter? This point may be
arguable, but there are many instrumentalists whose sound lacks
air and character. Some sounds just lie there in the mix and need a
little help to come to life. Then there are these musicians and singers
who rely on signal processing to add character and depth to the sound
of their instrument or voice. Sometimes its the style of the music; other
times it is the part they are playing. Perhaps this can be thought of as
cosmetics. Signal processors that have practical applications can be
used in creative or seemingly frivolous ways. While reverberation
and delay devices are primarily used to simulate space and create the
illusion of depth, they can also be used to create some truly bizarre and
interesting effects. Keep in mind that there are many instances where a
strange or new sound on a record helps make that record sell. For as
much as strange new sounds can help sell records, signal processing
can do more than create novelties. Often new sonic idioms are the
result of exploration and creative application on the part of audio
Figure 9.26 The Cubase HALion One is full of fun effects. Photo courtesy of Cubase.
258
engineers. Examples include huge exploding reverbs on snare drums,
gated reverb on drums, synchronized echoes in dance music, leslie
(rotating speaker)-like effects on guitar sounds, chorused bass, and
sampled or electronically doubled vocals. All of these sounds, once
thought of as novelties, have become conventions.
Signal Flow or Where to Use a Processor
We have been discussing when to use signal processing; however,
where the processor goes in the signal path is of equal importance to
when a processor is used. The placement of the processor in the signal
ow is often an important factor in how it will be used, and the loca-
tion can change the overall effect, sound, or perception of the pro-
cessor. There are few instances in which there is only one acceptable
point in the signal ow for a processor. More often than not, there are
generalized guidelines that lead to experimentation, and tradeoffs to be
understood.
As an example, the placement of a compressor in the signal ow, gen-
erally speaking, should be as close to the sound source as possible.
According to some engineers, it should be done while recording to
the multi-track tape. Other engineers routinely compress signals that
are already on tape. To further complicate this simple rule, some audio
professionals prefer using an insert send/insert return, pushing the
compression farther back in the chain than those who compress
between the microphone and the preamp, or between the preamp
and the console line in. There is no absolute rule. When we compress
a signal on tape, we add more noise than we do when compressing a
signal from a microphone. Therefore, we have a guideline. As guide-
lines go, however, it is a weak one, so lets examine one of the under-
lying questions.
The rst real question to ask is whether to use a processor in mixing or
tracking (see Figure 9.27). This is the old argument of x it in the
mix versus set it and forget it. A simple way to view it for now
is that the rules are different for beginners. It is innitely more difcult
for a beginning audio professional to set it and forget it, because this
259
approach takes experience. The reason this is a more difcult approach
is that one cannot take an effect off once it has been recorded on the
multi-track. If an effect is used in the mix and the results are deemed
inappropriate, all you need to do is get some more 1/4-inch tape or pop
in a blank CD. The drawback to the x it in the mix approach is
that it sometimes will add more noise to a recording, such as anging
tape hiss or a hiss from a compressor raising the noise oor. Until an
engineer gathers some experience, the x it in the mix approach is
suggested. That way no mistake will be permanent. The exception is
the use of dynamic processors to overcome the limitations of storage
Microphone Compressor Console
Mic
Pre
In Out
Recorder
Microphone Console
Mic
Pre
Line
Pre
Line
Pre
In Out
Recorder
Compressor
Figure 9.27 A compressor can be inserted while recording, between the microphone
and the consoles microphone preamp (top), or while mixing, between the recorder
and the consoles line preamp. Each method has advantages and disadvantages.
260
media, where the limiters threshold should be set so high that it will
probably not engage anyway.
There are two basic ways to add a signal processor into the signal ow,
in an effects loop or by direct insertion (see Figure 9.28).
Effect loops are used when the signal needs to be split into a processed
andunprocessedsignal. The processor is insertedat the output of anaux-
iliary send, allowing signals from any I/O module to be processed. The
output of the processor is recombined with the dry signal at the master
fader by using either an effects return or an available line input. Direct
insertion is usually used for one signal at a time, where the output of the
processor can be used without recombining it with the dry or original
signal, such as in the case of compression (see Figure 9.29).
The exceptions to this generalization are insertions at the master fader
or at a bus output, when the bus output is used as an additional aux send.
One last thought on signal processing and mixing: It takes years to
become a world-class mixer, and it also takes years to gel your own
Send
Master
Trim
FX
Return
or
Line Input
Send
Channel
Trim
Post
Fader
Signal
Fader
Master
Fader
Speaker
Volume
Amp
2 Track
Monitor
Speakers
Equalizer
1 2
3
Figure 9.28 Processors can be inserted in an effects loop when more than one signal
needs to be processed by the same device, and that composite signal will be blended
back into the mix.
261
personal style in terms of how and when you use signal processing.
Rather than seeking instant results and immediate improvements in
your ability, your time will be better spent by observing other, more
experienced engineers. Learn their techniques, and then experiment
with them to discover your own techniques, to develop the engineer
and mixer within you. Your techniques will ultimately dene your
style, and if they are applied patiently, humbly, and conscientiously,
with the understanding that you have a long way to go, you will arrive
as a mixer, original and creative in your own right.
Exercises
1. Record a kick drum playing quarter notes along with a snare
drum playing half notes (a backbeat). Engage EQ on the
module with the kick drum and try to make the drums more
clear to each other; try to eliminate all masking so the drums
each speak clearly.
2. Add reverb to the snare drum. Observe whether this changes
the masking relationship between the snare and the kick.
3. Try processing the reverb chamber. Try EQing the input to the
chamber, then try EQing the output from the chamber. Do
Line
Input
Mic
Input
Fader
to
Multi
Mon
Fader
Track
Select
Multitrack
Master
Fader
1 2
4
Trim/
Preamp
EQ
Compressor
Compressor
5
Figure 9.29 Processors can be inserted by direct insertion when only one signal will be
processed, and it will replace the original signal.
262
these actions result in different sounds? Is one warmer than
the other? Is the masking different with these different
approaches?
4. Now try adding pre-delay to the chamber by putting the send
through a digital delay or tape delay. How did the chamber
change? Did it seem to get bigger with the same RT-60? Did it
affect our masking?
Additional Reading
2004.
Anderton, Craig. The Digital Delay Handbook, rev. ed. London:
Music Sales Corp, 1985.
Benson, Blair, ed. Audio Engineering Handbook. New York:
McGraw-Hill, 1988.
Springer, 2001.
Davis, Don and Eugene Patronis, Jr. Sound System Engineering,
Springer, 2005.
263
Press, 1993.
Fraser, Douglas. Digital Delays (and How to Use Them). Sherman
Oaks, CA: Alfred, 1989.
CA: Alfred, 1989.
Jorgensen, Finn. The Complete Handbook of Magnetic Recording, 4th
ed. Blue Ridge Summit, PA: Tab, 1995.
Focal Press, 2002.
Press, 1989.
Pohlmann, Ken. Advanced Digital Audio. Carmel, IN: Sams Publish-
ing, 1991.
264
Press, 2000.
Sams, 1989.
265
10
Mixing
P
h
o
t
o
c
o
u
r
t
e
s
y
o
f
S
o
l
i
d
S
t
a
t
e
L
o
g
i
c
.
267
N
ow that we have explored signal processing, lets gure out how
to process many signals simultaneously and combine them.
Much like signal processing and editing, mixing is a hands-on
endeavor, and one that can only be truly learned by an aspiring audio
professional turning the knobs, pushing the faders, and hearing the
results. Nonetheless, mixing offers us some universal guidelines, which
we will explore in this chapter.
One of the inherent problems with learning to mix, from both a hands-on
and a textbook perspective, is that there are fewabsolute truths. Ten engi-
neers will produce ten different mixes from the same elements, which are
all acceptable; they are all good mixes. One mixs superiority over
another is often a matter of personal taste, assuming certain basic techni-
cal aspects have been fullled. In a heavy metal mix the guitars will usually
appear to be placed way up front, or loud, and will sound very full, some-
times partially masking the vocals. In other forms of rock, this would be
unacceptable. While mixing a soundtrack for a lm in which guns and
explosions are featured, extra volume on these elements would be appro-
priate, because we perceive gunre and explosions as louder elements
within a soundscape. In all forms of audio production, the mixer attempts
to balance the elements, effects, or instruments such that they can all be
heard and are all at appropriate levels in relation to each other. In some
lmmixes and radio productions, certain elements will be heavily favored
over others, just as in some forms of music, such as heavy metal, pop, and
country, where the convention of our expectations of this form of music
can supersede issues such as masking. While understanding conventions is
useful, it will be helpful for us to begin by understanding what constitutes
a good mix.
A Good Mix
What constitutes a good mix? Some engineers will answer that question
with a list of subjective or quantitative qualities, including descriptions
of relative timbre, loudness, and placement. Audio professionals will all
agree that a mix needs to be free of noise and distortion. There will be
general agreement that the end users or observers should be able to hear
all elements. When we get to the part about just how clearly all these
elements need to be heard, their opinions will depend on both their
268
personal taste and their work history. Someone who loves country music
may be predisposed to like a tune with ddles way up front, just as an
engineer who has been specializing in dance remixes for the last 10 years
will probably appreciate a strong backbeat, a kick drum that makes
your chest thump, and a powerful bass sound. Are any of these audio
professionals wrong? Not if their mix is appropriate for the style they
are attempting to create or imitate. We have a lot of leeway in creating
the relationships within a mix from a purely aesthetic standpoint.
Other engineers may answer that the best mixis the one that sells the most,
whether its records or tickets at the box ofce. Both answerssounds
good and sells a lotare correct. The rst type of answer is correct
because taste is personal. The entire experience of listening to music or
watching a movie is subjective; therefore, whatever anyone likes cannot
be considered wrong. Never be a snob about a style of music you dont
like; if someone else appreciates it, if it ts their personal aesthetic, it is
valid. The second answer is also correct because, as a famous record exec-
utive once said, We are in the business of selling records, not making
records. Successful projects result in greater opportunities for those
whowere a part of the teamresponsible for its success. Whenyouengineer
a hit record, your name gets out there, and more producers will be inter-
ested in working with you. Also, producers will often want to match the
sounds of a successful record or production, so they may come looking for
Figure 10.1 The Cubase 4 Mixer. Photo courtesy of Cubase.
Chapter 10 Mi xi ng
269
youtoget that [ll inyour name here] Sound.Furthermore, historically,
there have always been some musicians and engineers who have been
thought of as lucky, where a producer believes that by using his lucky
engineer or his lucky drummer (the one with whomhe had his last hit), he
will have another hit. It may seema bit silly, but there are some very super-
stitious people in the music business, and if they believe you are a hot engi-
neer, then you will be a hot engineer.
If you are now wondering which of the two answers weighs more heav-
ily, consider this: It is necessary for an engineer to be able to deliver
whatever type of mix the client wants. Being able to please a client
will ultimately determine the success of an audio professional. To do
this, it will be necessary to know the mixing techniques involved in
many types of music and production. Be open to other types of music
and lm than what you usually listen to or watch. Listen to all types of
music, and listen to them closely. Keep in mind that there is no bad
music and no bad taste. Never consider yourself to be above certain
types of music; if youve heard of a particular form of music, its
because people have bought it, and if people have bought it, then people
have made money from it. Similarly, it is easy to poke fun at commer-
cial feature lms or commercials on the radio, but there is a good living
to be made by audio professionals in these areas.
To be able to capitalize on the opportunities that become available to
you, you should be ready for anything. Being at the beginning of your
career and having little control over the type of client, studio, station,
or lm house that will offer you an opportunity, you should develop
the skills needed to make the most of whatever type of entertainment is
being created in the facility where you nd yourself and beyond. To
accomplish this, it will be necessary to study different styles of music
and different genres of lm, listening as an engineer does. This means
listening for the techniques used to shape the character of the sound.
While you may object on some level to music that you think you dis-
like or a lm that you believe to be in some way inferior to that which
you normally watch, you now have another reason to pay attention.
Aside from personal enjoyment, you are now listening to become
aware of the stylistic and idiomatic trends employed by engineers
270
and producers. In other words, you are listening professionally and
critically, not merely for your own enjoyment. When you go home
you can pop any CD you want into the player, and you can listen to
or watch whatever you want. At work you will need to develop the
ability to operate in a successful and nonjudgmental manner in any
form of music, no matter how superuous you may think this genre
is when you are not at work.
Using myself as an example, I have never favored heavy metal. The
rst time I was asked to mix a heavy metal album, I listened to a
great deal of heavy metal the night before. I asked questions such as,
What does the vocal sound like? The guitars? What is the balance
between the bass and the drums? When I mixed the album, the
band loved it. They had no idea that it was the rst heavy metal
album I had mixed; based on the product they believed I was experi-
enced, and they had no idea that I held heavy metal in anything other
than the highest regard. Of course, I did not tell them my personal
opinions about heavy metal. There was no point in giving them a rea-
son to dislike my mix. If they thought I didnt respect them as artists or
the genre in which they work, it might have colored their opinion of
the mix. I did not see any reason to give them that opportunity, and the
fact is those mixes came out well. They were appropriate for the style,
Figure 10.2 The Nuendo Mixer. Photo courtesy of Nuendo.
Chapter 10 Mi xi ng
271
and the client loved them. Keeping a poker face is important in these
situations. As an engineer, you are constantly dealing with huge, frag-
ile egos. There is no excuse for insulting their music. After all, they
have fans who buy their albumsthat makes it a valid form of
music, and you have no basis to judge them negatively, especially if
their check clears the bank.
I ran into the same situation a short time later, when I was booked to
overdub vocals and mix an opera album. I had a neighbor who fre-
quently had the sound of opera discs leaking out of her apartment. I
knocked on her door and asked her to play me some opera the night
before the gig. She thought it was all very amusing. I knew what opera
sounded like, but not as an engineer. Once again, I listened in detail to
the vocal quality, the relationship between the vocal and the orchestra,
and the internal balances of the orchestra. The producer enjoyed my
mixes so much that he booked me for three more opera projects over
the next few years. As a freelancer, this was a wonderful opportunity to
fatten my bank account while learning about a style of music I had pre-
viously gone out of my way to ignore. And never once in all that time
did the producer or artists think that I held opera in anything but the
highest esteem. Stay open to different styles of musicthey are all
potential income streams for the audio professionaland keep that
poker face on at work!
As a sound designer I have also found myself in similar situations. No
surprise here; I do not like operettas, yet I have been sound designer for
two plays by Gilbert and Sullivan. I may personally think they are the
pits, but I did not tell the directors, actors, or anyone else of my per-
sonal dislike of that genre. Instead, I did my job and did it well. I must
have done it well because they hired me to do a second one. There is a
theory that audio professionals may do better work in genres we like
less, since we can be more objective about a genre that we marginally
understand than we can be with our favorite artists next album. The
sad truth is that, as an engineer or sound designer, you do not have the
opportunity to work with your favorite band or on your favorite play
every day, and the rent still has to be paid. Besides, it builds character
and expands your repertoire to work in forms of music that are beyond
your comfort zone.
272
I/O Modules
In Chapter 3, we took a brief look at consoles, and in Chapter 4, we
examined the different components in the console and the way they
interacted in detail. To really understand mixing and everything we
can do in a mix, it will be helpful to briey review the way the signal
ows through a console and what the main components will do for
us. While every console is different, we will review the most common
elements found on most consoles. As you will recall, the input/output
(I/O) module is the location where we will perform most of our oper-
ations; it contains the components in Figures 10.3 and 10.4.
The microphone/line input-selector switch chooses between mic and
line input. While we would be set to the mic position while recording
or overdubbing through a microphone source, this switch is always set
to line to receive the output of the multitrack while mixing. The mic/line
trims are used to alter the level of an input signal (see Figure 10.5).
One way an audio professional can ensure good gain-staging between
the trim and the fader is to set the fader to zero and use it only for
dynamic changes during the mix, such as bringing up an actors
voice or a guitar during its solo, raising a bass when it does a partic-
ularly interesting ll, or helping a string to swell a little stronger, while
the trim may be used to set the initial level in each module.
Figure 10.3 The components of a typical input/output (I/O) module from a Solid State
Logic console. Photo courtesy of Solid State Logic.
Chapter 10 Mi xi ng
273
Figure 10.4 A typical input/output module from a Soundcraft console. Photo courtesy
of Soundcraft.
Figure 10.5 Line and mic trims, with options (from top) to ip or switch the signal
from its normal ow and make it ow to the monitor pot as opposed to the channel
fader, add to sub-group, reverse phase, and pad (reduce) input by 20 dB. Photo cour-
tesy of Solid State Logic.
274
The next commonstopinthe signals pathis the equalizer (see Figure 10.6).
Equalizers are frequency selective ampliers. As discussed in Chapter 9,
they will increase or decrease the volume of a user-selected range of the
Figure 10.6 The equalizer section of an SSL I/O module. The high- and low-frequency
EQs have an option to switch from shelving to bell, while the two mid-range frequency
EQs have an additional pot to adjust the Q, or width of the bell. The switches on the
bottom give the engineer the option of inserting the EQed signal into the path of the
channel signal, sending it to the dynamics side chain to act as a key, or inserting it into
the monitor signals path. Photo courtesy of Solid State Logic.
Chapter 10 Mi xi ng
275
audible band. The chief function of an equalizer is to alter the timbre of
a signal, which means they are commonly used to reduce masking due
to conicting frequencies, or to dovetail one sound into another. While
equalization doesnt actually alter the pitch of a sound, altering its har-
monics can be an extremely effective way to make things t together
better in a mix.
The channel assignment switching matrix, or multitrack bussing, has
several functions, as discussed in Chapter 4 (see Figure 10.7).
Normal signal routing in mixing with a simple console involves routing
the signal from an I/O module through the stereo bus to the master
fader. Often the signal will pass through passive sub-faders, which are
actually bus faders. They are not truly sub-masters because they will not
alter the level of a signal routed to a post-fader aux send, even though
they will alter the level of the signal itself as it feeds from the sub-fader
into the master fader.
The channel assignment switching matrix in a sophisticated console
can be used for many things. When the audio professional has used
up all the available aux sends, the channel assignment switching matrix
can be used to provide assorted stereo mixes during a lm mix, such as
Figure 10.7 This channel assignment matrix features 32-track bussing, plus three ways
(A, B, and C) into the stereo bus. There is also a pan pot used to pan between odd and
even sends in the multitrack bussing. Photo courtesy of Solid State Logic.
276
a pair for speech, another for effects, another for underscoring, or for
additional sends to outboard equipment during mixing. A signal can be
sent from a fader, through the bussing, then picked up at the patch bay
and sent anywhere for signal processing, then returned to the console
through any available fader.
The pan pots place the signal, left to right, within the stereo eld (see
Figure 10.8).
This can also help lessen masking, since masking is at its worst when
sounds of the same or similar loudness and frequencies come from the
same location. By panning similar sounds away fromeach other, masking
is reduced. Keep in mind as you use your panning that there are far more
options than center, left, and right. Frequently, panning a sound just
slightly out of center will sound best to the audio professional, or perhaps
a little further out of center but not all the way to left or right (hard-
panned) will be best for a particular sound. Always be ready to use your
ears to determine the best placement for a particular element in your mix.
The sub-masters and bus faders are masters on different summing net-
works, just as the stereo master is the master of the stereo bus. There are
other summing networks called auxiliary summing networks, which are
used in mixing for reverb, delay, and other effects, which the engineer
wishes to add to the signals to change the spatial relationship of the mix
(see Figure 10.9).
These summing networks, also called aux sends, are used in live situations
for everything from cue mixes for the musicians to backstage mixes for
actors. In a mix, reducing a signal while adding reverb will make a sound
appear more distant, while reducing the reverb and making a sound
louder will make it appear closer, or in front of the mix. Before delving
any deeper into the character of the mix, lets briey discuss the master
section of the console.
Figure 10.8 A typical pan pot. C, for center, feeds both sides of
the stereo bus equally, while L and R feed only the left and right,
respectively. Photo courtesy of Solid State Logic.
Chapter 10 Mi xi ng
277
Master Section of the Console
Along with the I/O modules, which are redundant and repeated
through the console, the sound will pass through its last stage in the
console, the master section (see Figure 10.10).
Figure 10.9 The aux sends provide additional mixes from the console, which can be
sent to signal processing, used for a stage monitor mix, to cue actors in the green room,
for a satellite uplink, or anywhere an alternate mix is needed. Photo courtesy of Solid
State Logic.
278
The master section affects all signals passing out of the console and
includes the master fader, aux send masters, source selection, speaker
selection, and volume.
The master fader, or any master trim, will control the output level of
the summing network that it modulates. The output of the master
fader leads to the two-track machines, CD burners, and the speaker
volume controls. In sound reinforcement or sound design situations,
the output of the master fader will usually lead to the house monitors,
the speakers that face or surround the audience.
The monitor source selector switch determines which of the available
output busses or machines will feed the monitors (see Figure 10.11).
While normal options include the stereo bus (sometimes called the two
mix or mix bus) and various other busses in the console, including the
aux sends, many consoles will offer options for machines, such as a two-
track or CD player to be brought up and fed to the monitors. The level
at which the selected bus or machine will enter the monitor amplier is
determined by the speaker volume control (see Figure 10.12).
While it may seem elementary, I have seen good engineers in the mid-
dle of complex setups sh around for what seemed like an eternity
Figure 10.10 The master section of a Solid State Logic console, featuring subgroups,
monitor source selection, the master fader, aux send masters, and a mix computer.
Photo courtesy of Solid State Logic.
Chapter 10 Mi xi ng
279
Figure 10.11 The master section of a DDA DMR-12 featur-
ing master faders (lower left), monitor source select
switches (above master), and aux send masters (on right).
Photo by Gary Gottlieb.
Figure 10.12 This master monitor pot manipulates the overall
volume in the control room. Photo courtesy of Solid State
Logic.
280
trying to gure out why they couldnt hear anything, until someone
pointed out that the volume was down.
Normaling
While it may seem a little early in our discussion of mixing to be dealing
with the proper way to leave the console when you are nished, normaling
is every bit as important as any other operation you will perform behind
the console. At the early stages of ones career especially, one is judged
based on the little things one does. There are many engineers, myself
included, who will quickly judge a studio and its employees as substandard
upon walking into work and nding the console improperly normaled. As
an audio professional, I want to start every session fresh; I certainly do not
want to start off with someone elses EQ settings left over from last night.
Every component just mentioned will have a normal position, which
will usually either be off or null (see Figure 10.13).
At the end of each session, every workstation, console, and piece of
gear must be normaled. While there will be some variance from station
to station and studio to studio, there are some common rules. There
are exceptions, particularly when the same engineer always works in
the same room. (This engineer may always plug the hi-hat into input 7
and may like to keep his input equalization set up.) Despite these
exceptions, it is valuable to know the proper way to normal a console.
Some devices will be powered off overnight, such as effects units and CD
burners. Small consoles and ampliers will also be powered down usu-
ally, but large consoles and amps will generally be left on constantly,
since the shrinking and expanding caused by heating up and cooling
down every day will shorten the life of the equipment. This will vary
from one production room to another, so unless you are the owner,
be sure to consult a fellow employee before hitting the power switch.
Before powering down any piece of equipment, make sure its volume
is down at the console. Powering an effects unit up or down while its
returns are up can send a spike of signal to the amp, which will destroy
or shorten the life of the speakers.
All faders must be pulled down. This includes channel faders, monitor
faders, sub-masters, and master faders. All volume controls and pots
Chapter 10 Mi xi ng
281
(except pan pots) should be turned completely to the left (off). This
includes the monitor volume pot, individual monitor pots, and all
aux sends. All pan pots, boost/cut controls on equalization, and fre-
quency select switches on equalization should be set centered, at
12:00. All push buttons, such as the pre/post switches, bus switches,
Figure 10.13 A normaled module. Photo courtesy of Soundcraft.
282
channel on switches, and solo switches, should be turned off. Usually
this means they are left in the up position. When you have your rst
mixing experienceoften the rst opportunity you will have to sit
behind the consoleyou should practice normaling each section
immediately after experimenting with it. Its a good habit to get into.
Using Monitors
The difference between the sound of music in a dance club and the sound
of music coming out of a clock radio or a car stereo is obvious. As dis-
cussed in Chapter 6, the difference in the quality of the sound as the
monitor system changes is something of which engineers must have an
acute awareness. Lets examine in more detail how this sound difference
impacts the audio engineer.
Most engineers, early in their training, have experienced being pleased
with the quality of the sound of a project while in the studio, and then,
when listening to their mix elsewhere, being very disappointed in the
quality of the sound. The reason for this is often a lack of insight into
the character of the control rooms acoustics and of the monitor speakers
being used for the mixdown. All aspiring engineers, therefore, must learn
how to understand the character of any of the many monitor speakers
they may encounter while creating their mixes.
When mixing, the engineer must have an understanding of the character
of the speaker system that is being used. If the engineer does not under-
stand the character of the speaker system being used, he or she may
disproportionately compensate, especially with equalization, for some
quality inherent to the speaker system. If the speaker system is inherently
boomy, an engineer, unaware of this characteristic, may feel that there is
enough low end on a recording when this is not the case. This could
result in a recording that sounds thin when played back on other speaker
systems. Conversely, if the inherent characteristic of a speaker system is
to be overly bright or have a shallow low end, a well-balanced recording
may sound in the studio, to the unaware engineer, as if it is lacking in
low frequencies. This can cause an engineer to compensate by increasing
the volume of the lower frequencies on a number of instruments. The
result of this could be a recording that sounds muddy when played back
Chapter 10 Mi xi ng
283
on another speaker systemor a production that sounds boomy in a theatre.
Eachmonitor systemhas its owninherent character; therefore, the soundof
amixplayedondifferent monitor speakers will change. Themonitor system
is the onlycomponent inthe studiosystemthat gives the engineer anaudible
insight into the character of the sound being recorded. If the inherent char-
acter of themonitor systemis not known, theengineer is onlyguessingabout
the nature of the sound that is being recorded onto tape and may be setting
himself and his clients up for some unanticipated additional remix time.
As stated earlier, some engineers feel that the best mix is the one that sells
the most records. If a mix does not sound good in every environment in
which a consumer may listen, potential sales are lost. There is a wide vari-
ety of systems and speaker types that consumers use when listening to
music. The rst time that someone hears a record, he or she may be listen-
ing to the sound system of a nightclub, a car, or a home stereo. Many
people listen to personal stereo systems or a boom box. Each of these
systems has dramatically different sound qualities. It is the engineers
job to ensure that the sound of the recordings works well in all of these
types of systems. Each of these systemtypes represents millions of listeners
or viewers and millions of potential sales. If the production teamhas done
its job properly, a consumer should want to buy a record the rst time that
he or she hears it, regardless of the type of system on which it is heard.
When mixing, engineers use several speaker types. Using studio monitors,
home-quality bookshelf speakers, and sound cubes, the engineer can have
a good idea of how the sound of a mix will translate on almost every type
Figure 10.14 A classic
Neve 8068 console. Photo
courtesy of Neve.
284
of commonly used system. Some studios, including the former Sigma
Sound Studios in New York, went so far as to wire a clock radio to
the console. In other studios, an engineer and producer will run a
rough mix off onto a CD, listen in the managers ofce on his or her
computer, then run down to the car and listen to see what it sounds
like on a different system (one that is commonly used). An audio profes-
sional should listen to a mix on as many sets of speakers as he or she can
before calling it a good mix; otherwise, he or she cannot be assured that
the end users experience will be a positive one.
Through the process of switching between speaker types, then compen-
sating for incongruities, then switching and compensating again, the
engineer will come upon a blend that works well in all of the speaker
types, and therefore all possible environments. In each case, however,
the engineer must be intimately aware of the special character of the
speaker being used in order to avoid the pitfalls described earlier.
To learn the characteristics of a set of speakers that an audio profes-
sional will be using for the rst time, it is helpful to always carry
around three pieces of music. The source of the music should be of
the highest possible quality. Each of the three selections should be
very familiar to the engineer; the engineer should have a clear idea
of what he or she expects to hear. The greater the number of systems
on which the music has been heard, the better it will serve. Choosing a
recording that the audio professional has heard at home, in a car, in a
club, in other studios, and in the homes of one or more friends will
ensure that we are as aware as possible of how this recording will
sound under different listening situations. To save time, it is not neces-
sary to use the entire composition; the object is to listen to sound quality,
not musical composition. The engineer should bring along a pair of
headphones he or she trusts for reference, and then listen to the three
selections on all speakers available at the new facility. On each of the
monitors, the engineer should compare the sound in the speakers to
the sound in the headphones, comparing back and forth several times.
Note the differences between your expectations of the sound, based on
your knowledge of the mixes, and the reality of what is output from
the speakers. Are the low frequencies, mid frequencies, and high
Chapter 10 Mi xi ng
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frequencies consistent with your expectations based on your familiarity
with these mixes? Is the character of the music different from what you
expected? Is it different from the sound you are referencing in the head-
phones? If you were to mix on these speakers, can you identify fre-
quency bulges or deciencies for which you may need to compensate?
Comparing different systems, from headphones to car audio to cheap
bookshelf speakers to audiophile speakers, is the beginning of develop-
ing your ears to listen critically, the key to mixing.
Methodology
Now that we have a basic idea of what to look for, lets discuss how
we actually put a mix together. If this is a project that the audio pro-
fessional has been working on, there will already be ideas that have
been tried during tracking and overdubbing. The engineer would
have already tried several reverbs and other effects, and will already
have an idea as to the relationships of the elements and the producers
and artists opinions as to different approaches. Starting a mix like this
is easy; the audio professional simply begins blending the tracks, as he
or she has heard them blended before.
An audio professional who comes in as a remixer faces a different chal-
lenge. Sometimes the engineer will receive a copy of monitor mixes or
other mixes; other times the engineer will hear the song for the rst time
when he or she pushes up the faders. In this second situation, the best
approach for the audio professional is to push up the faders and listen
through to a couple of passes of the song. We will think up ideas at this
point, consider some approaches, identify the genre and consider how
to make this song t the genre, consider subgroups and effects, and con-
sider different approaches to the mix. It is best to resist the urge to dive
right in and start EQing something at this point; the mix will benet
greatly from a few minutes of contemplation and evaluation of the tracks.
Once we have settled on an approach, we can begin the mix in earnest.
Beginning with some fairly standard instrumentation for rock, pop,
dance, and country, and starting with the mechanics, most engineers
(though not all) build a mix from the bottom upin other words, they
start with the drums. Experimental mixes and unconventional music
286
styles aside, the kick drum (or bass drum) and snare drum should
always be centered. A mix with the kick and snare anywhere but center
can be somewhat disorienting; most listeners will seek the kick and
snare in the center as an anchor for the mix, something to hold every-
thing else together.
Assuming the rest of the drums have been miked in stereo, follow the
audiences view of the drums with the pan pots. In other words, place
the cymbal that appears on the left as you stand in front of the kit on
the left side of the mix. The cymbal on the right goes on the right side
of the mix; the hi-hat also goes on the right (while it is on the drummers
left, it is on the audiences right). The tomtoms cango right to left for high
to low, which is how you see them. Or, for something different, try going
hard right for the high one, hard left mid tom, and hard right low tom. It
gives a different sense of movement, and this hard-panning method works
even better if the drum kit has four toms. Drums are a good place to
experiment with panning as well as with reverb, since reverb establishes
the character of the roomin which the mix is taking place, and placing the
drums in the room gives the engineer the foundation around which the
rest of the room is builtaround which the other instruments are placed,
as we discussed in great detail in Chapter 9. Lets place some more instru-
ments in this room.
Figure 10.15 The Wavelabs Mixer with Mix Bus and Nuendo Audition Bus. Photo cour-
tesy of Wavelabs.
Chapter 10 Mi xi ng
287
Continuing to build from the bottom up, lets add the bass next. Like the
kick and the snare, the bass should be dead center unless heavily effected
or used experimentally. Use some of the tips on EQing from Chapter 9
to blend the timbres of the bass, kick, and snare. Pay attention to the
style: Are you mixing pop or dance, where the bass high end tends to be
exaggerated, or are you mixing jazz or bluegrass, where the bass tends
to be darker, fuller, and rounder? Remember to stay focused on the style
you are mixing and keep the information entering your ears consistent
with your understanding of this particular sound.
Lets put in some guitars and keyboards next. If there is more than one
guitar or if a guitar part is doubled, it can be very effective to split them
using the pan pots. Should they be split slightly, extremely, or some-
where in between? That depends on the song, the style, and the mix.
Listen to the guitar in many locations as you sweep it from one side
to the other and ask yourself, Where does it sound best? Wherever
it sounds best is exactly where it should be. Splitting a doubled guitar
part slightly out from the center can be an effective way to thicken the
sound while separating the guitar sounds and making them more dis-
tinct, but you do not need to have two of something to pan it. Anything
can be placed anywhere if that is where it sounds best, and there doesnt
have to be something panned against it on the other side.
Keep in mind that hard panning is not necessary, and in this situation it is
often not desirable; often a slight split is most effective. Experiment with
a doubled guitar split to varying degrees off center; see what sounds good
to you. A lead guitar part can also be split against a keyboard line that
is full of lls or answers; any two parts can be juxtaposed or a part can
be panned off by itself. Once again, it is a process of experimentation,
especially at the outset of a career. Stay open to what your ears are telling
you. Does this keyboard part sound better against this guitar part? Is
there another part that would sound better against it? Does this part
sound good panned to one side without anything balancing it on the
other side? Anything is possible in a mix, and every piece of music is
unique; use your ears and start to really listen to them.
The other sweetening is added next. This could include more keyboard
parts, horns, strings, woodwinds, or some light percussion. Keep in
288
mind that sweetening is frequently relegated toward the back of the
mix or is low in volume, and often more reverb or delay is added to
increase the idea in the listeners mind that the sweetening is at the
back of the stage, rather than up front the way a lead guitar or a
lead singer would be.
Next come the vocals. Vocals can be extremely tricky, because the fre-
quency range of a vocal falls in a similar range to guitars, many key-
board parts, and even the high end of the snare drum. To make the
vocal shine out in front of the mix without making it so loud that it
sounds like the singer is in a different room, try equalizing the vocal
to dovetail into the other sounds. Try boosting the high end of the
vocal while pulling some of those same frequencies out of other instru-
ments, such as the guitars, which are also occurring in this range. Also,
since vocals tend to be centered at 1 KHz, try pulling some 1K out of
any instruments that will potentially mask the vocals. Vocals often
sound better with a certain amount of signal processing. Try a different
reverb on the vocal than you have on the drums, guitar, or keyboards.
Vocals often sound better with a short pre-delay (indicating that the
singer is in front of the band) and a medium decay time (indicating
again that the singer is closer to the audience than the rest of the
bandin other words, in front of the band). Like everything else in
mixing, nding the right balance between the instruments, then the bal-
ance between the instruments and the vocals, and the balance between
each instrument and its reverb or effect, requires experimentation.
Having said all this, there are many extremely successful mixers who
take a very different approach. They may start with the vocals, con-
sidering that to be the most important element in the mix, and build
everything around the voice. They may consider the guitars to be crit-
ical to this style of music and start with a smoking-hot guitar mix, and
then ease everything else into that blend. There is no wrong place to
start if the result is a good mix. Personally, I tend to start with drums,
but if I reach a point where I am unhappy with my mix I will pull all
the faders down and start over, usually starting with a different
elementthe guitars, the vocals, even the horns if they are prominent.
Audio professionals should never be afraid to pull down the faders and
start over, keeping the EQ settings and effects. The balancing of levels
Chapter 10 Mi xi ng
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in the mix is probably the easiest part, so never be afraid to pull the
faders down.
While we are on the subject of EQ, it is worth mentioning that it is usu-
ally better to subtract than to add. This is a gross generalization, but
there is a tendency among beginning engineers to add EQ to everything,
when taking away certain regions of timbre can be far more effective at
helping everything in the mix sound clear and cut through. If a particular
instrument is too thin, try EQing out some high end before you add
bottom. If you try to cure a boomy sound by adding high end, the boom-
iness may be less apparent, but it is still there. If you can cure timbral
problems by subtracting EQ rather than by adding EQ, you have less of a
chance of creating a nasty frequency bulge.
Something related to the idea of subtracting EQ rather than adding it is
that some sounds get bigger in the mix when we make them smaller
with EQ. Heavy metal guitars are a great example. If we followed our
inclination to make heavy metal guitars big and fat with effects and
EQ then make them loud in the mix (appropriate for the style) we
would never be able to hear the vocals because the big guitars would
completely mask the vocals. Instead, lets do the opposite and use our
EQ to make the guitars tiny from an EQ standpoint. Guess what?
When we make them loud in the mix, they still sound really loud,
but we can hear the vocals just ne because they are no longer masking
in the 1-kHz to 3-kHz range. This leads me to something I learned a
long time ago about mixing: If what you thought should work didnt,
then try what shouldnt work. Oddly enough, things that shouldnt
work sometimes do, and usually its when the things that should
work dont. Go gure.
Now suppose for a moment that there is no drum kit. If you have the
opportunity to mix a string quartet, a full 60-piece orchestra, or a folk
singer with just a guitar, the aforementioned principles still hold; either
start from the bottom and build your way up or nd your focal point
in the instrumentation and build the mix around that sound. If you
have nothing but strings, start with the double bass. Add the cello.
Seat the viola into the mix, and then add the violin. If you like building
from the bottom up, then establish it as your convention and try to
290
always start by building from the bottom up. In the case of an orchestra,
begin with the percussion section, building it within itselfin other
words, start with the tympanis, add congas or other low- to middle-
frequency instruments, then work your way up to the high-frequency
sounds, such as cymbals and bells. Now work on the horn section
unto itself from the bottom upthe tubas, baritone horns, bass trom-
bones, trombones, and nally trumpets. Seat the entire horn sub-mix
into the percussion sub-mix. Next, balance the woodwinds, bass clari-
net, baritone sax, tenor sax, clarinet, oboe, soprano sax, ute, and pic-
colo, bottom to top. Once again seat this entire sub-mix into the existing
percussion and horn mix. Finally, sub-mix the strings, bottom to top as
earlier, and seat this sub-mix into the existing mix. If you listened to
classical music among other forms, as suggested earlier, you would
have an idea of what the end result should sound like, and mixing is
often the process of lling in those gaps and mimicking those sounds.
Of course, if you feel that the bassoon drives this whole 60-piece orches-
tra, try getting a really nice sound on the bassoon and building every-
thing else around it.
Mixing in a Sphere
Having dealt with the nuts and bolts of the technical end of the mix, lets
discuss an aesthetic approach to mixing. Ask a dozen engineers about their
aesthetic approach, and you will get a dozen different answers. There is no
simple right or wrong, since twoengineers canmixthe same piece of music,
take radically different approaches, end up with radically different mixes,
and both could be right. I have heard engineers describe their aesthetic
approach in many different terms. Some audio professionals will think
of their mixes in terms of construction (building a foundation and building
on it brick by brick), while some view it like a pyramid, still construction
but with a stronger base and a narrower top. Ive heard it described as
opening a window into a piece of music (with each element lling in
part of the aural view), and Ive heard it described as lling in a circle. If
any of these visualizations work for you, then they are right for you.
My personal approach involves visualizing a sphere and lling it in.
The advantage in my mind to a sphere is that it has three dimensions,
Chapter 10 Mi xi ng
291
as do all the best mixes. It allows for movement on three axes, which
ultimately produces a mix with more depth in more directions. To
break down the three axes, let us consider the side-to-side axis as con-
trolled by panning; the top-to-bottom axis as controlled by pitch and
timbre, effected by equalization; and the front-to-back axis as con-
trolled by the relationship between loudness, delay, and reverb.
Panning has already been discussed at length in this chapter, and equal-
ization and reverb have been dealt with in detail, both in this chapter and
in Chapter 9. Aside fromthe mechanics of placing sounds where you want
them, there is an aesthetic associated with where they belong or seem to
belong. Part of the idea is that if everything is bunched up in the center, the
result will be a at, lifeless mix. Think of it as two-dimensionalthe kind
of mix that, if you tried to touch it, would reject your hand like a pane of
glass. This bunching may occur on only one axis; for instance, suppose
you have spread all the elements out well through panning and equaliza-
tion, but everything is at the same loudness level with the same amount of
reverb. The resulting mix will be at and two-dimensional. The listener
will perceive the musicians to be standing crowded together or right on
top of each other, and the individual elements will be indistinct, lacking
depth, and masking each other.
Similarly, if the panning is effective and different reverbs are used on
elements presented at different levels, but the frequencies are bunching
up due to lack of effective equalization, masking will occur. These
frequency bulges are very common, especially in mixes executed by
Figure 10.16 A Neve 88d digital console. Photo courtesy of Neve.
292
neophyte engineers, since the ear must be developed through experi-
ence and critical listening to pinpoint and correct the bulges. Most
frequently, these bulges will occur either around the drums and bass
in the 80- to 200-Hz area or in the vocal range, around 1 kHz to
3 kHz. As mentioned earlier, masking and the ways to cure it were
discussed at greater length in Chapter 9.
Now suppose we were to spread out our elements, or create the percep-
tion that they were spread out, on all three axes. We have placed different
elements left to rightsome centered, some a little out fromcenter, some
further out, all complementing each other. We have EQed where it was
necessary to avoid masking, and the result is, top to bottom, theres a lot
happening but not too much in any one place. And front to back? There is
good depth; the relationship between the elements, their delays, and
reverbs places them in the same room but with a little space between
them, giving the illusion that there is good movement front to back.
The result is more than just an aesthetically pleasing mix; it is a mix
that you can reach right into, perhaps even stand in the middle of.
The ultimate goal in visualizing a sphere and adding to it as the mix pro-
gresses is that no part of the sphere should remain empty when the mix is
complete. Perhaps one spot up and off to the right doesnt have much
happening in it, but suddenly in the bridge, a glistening sound appears
there. That can be a wonderful spatial surprise. The entire sphere does
not have to be full throughout the piece, but ultimately every part of
Figure 10.17 A Pro Tools M-Powered 7. Photo courtesy of Digidesign.
Chapter 10 Mi xi ng
293
the sphere should have some activity, preferably at just the right time. As
with any aesthetic decision, there is no clear-cut right or wrong; there is
only experimentation leading to the development of skills upon which
you will eventually, through experience, develop your own aesthetic
sense. Understanding how to develop your aesthetic sense is valuable,
but you will not have an opportunity to do so if you dont survive your
rst studio gig. Lets familiarize ourselves with the way things are done in
a studio and take a look at some keys to success in that arena.
Exercises
In the Using Monitors section, we discussed a method to familiarize
ourselves with different monitor speakers. Start training yourself to do
that now. Pick three pieces of music with which you are very familiar.
They can be songs you have mixed yourself or they can be commercial
recordings. Go to different production rooms that are available to you,
or go to friends houses. Bring along a pair of headphones and listen to
the three selections on at least three different systems. It may be nec-
essary during this process to wear out your welcome with some of your
friends. On each of the systems, compare the sound in the speakers to
the sound in the headphones, comparing back and forth several times.
At each comparison for each system, write down your impressions,
noting the differences in the sound in the following manner:
1. Listen for, and write down, any changes in the sound of the low
frequencies.
2. Listen for, and write down, any changes in the sound of the
mid-range frequencies.
3. Listen for, and write down, any changes in the sound of the
high frequencies.
4. For each system, write down how the character of the music
changed and how the sound in the headphones differed overall
from the sound of the speakers.
5. Add a written conclusion about the character of the headphones.
What is their inherent character? How do they color the sound?
294
Next, while the sounds of the different systems are fresh in your
mind, write down a comparison of the different systems based on
the same criteria. Which system had the punchiest bass? Which
systemhad the clearest high frequencies? What would you have to
compensate for if you were to mix on these speakers?
Additional Reading
Alten, Stanley R. Audio in Media, 7th ed. Belmont, CA: Wadsworth, 2004.
Anderton, Craig. Digital Delay Handbook, rev. ed. Music Sales Corp.,
1985.
Springer, 2005.
Press, 1993.
CA: Alfred, 1988.
Chapter 10 Mi xi ng
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Katz, Bob. Mastering Audio: The Art and Science. Burlington, MA:
Focal Press, 2002.
Moylan, William. Understanding and Crafting the Mix: The Art of
Recording, 2nd ed. Focal Press, 2006.
Press, 1989.
Pohlmann, Ken. Advanced Digital Audio. Carmel, IN: Sams Publishing,
1991.
Press, 2000.
Woram, John. Sound Recording Handbook. Indianapolis: H. W. Sams,
1989.
296
11
Recording Studio
Operations and
Procedures
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A
s audio professionals, we may nd ourselves working in a wide
variety of situations and fullling an ever-increasing number of
roles. As a freelance engineer, my career has developed such
that I may sound design a play one day, record a jingle the next day,
and spend the following day sending bills, making phone calls to cli-
ents, and otherwise fullling the administrative end of the business.
Even within a particular type of facility, the lines drawn by job descrip-
tions often blur. In one studio a maintenance engineer will align the
tape machines before each session, while in another studio the assistant
engineers will perform the alignments, while in yet another studio the
chief engineer will do the alignments. Similarly, billing, typing labels
for cassette dubs, answering phones, making coffee, booking sessions,
making peace with a client, doing paperwork, getting lunch, emptying
ashtrays, and setting up microphones could fall upon any studio
employee. As such, lets set up some rough guidelines and job descrip-
tions, all with the understanding that no studio employee is above or
below any task, and with the further understanding that these descrip-
tions will vary greatly depending on the size, structure, and type of
facility. A post-production facility or broadcast facility will be laid
out very differently than a recording studio or a project studio, and
a house studio will draw very different lines between job descriptions
than a commercial facility will.
Production Facility Job Descriptions
No productionfacility couldexist without owners, because someone has
got to take nancial risk and responsibility. An owners goal is clear and
simpleto turn a prot by owning a nancially successful facility. There
are many obvious differences between the owner of a major, metropol-
itan, multimillion-dollar facility and the owner of a home studio based
around a G5 loaded with Pro Tools, yet they share the rights and respon-
sibilities of ownership, such as liability and risk, and they each hope to
reap the benets of prots.
The owner is responsible for the nancial obligations on one end, such
as the mortgage or lease payments, utilities, taxes, and weekly payroll.
On the other end, the owner is also the main recipient of any prots
drawn from the facility, and therefore he has, more than anyone else
298
involved in the studio, a vested interest in keeping costs low and produc-
tivity high. If you ever work in a studio where the owner seems petty or
cheap, it may have something to do with the day-to-day concerns of this
formula as he or she attempts to keep expenditures in balance with
income. Consider the fact that the owner is privy to both information
and pressures that the rest of the staff is not.
Although there are many different types of owners, we can easily break
studios and their ownership down into two typesthose that are owned
by individuals or partners and those that are owned by corporations. In
the rst group, one personsometimes an engineer, sometimes a com-
poser, or sometimes just a smart businessmanwill open a studio. As an
employee in this type of studio, most people feel as if they are scrutinized
more closely. Employees are sometimes red for seemingly frivolous rea-
sons. Sometimes demands that appear to be unreasonable are made. The
owners actions can appear arbitrary and capricious to the staff at times.
While some of these impressions may be true in some studios, there are
still many advantages to being an employee under individual ownership.
If an employee has a suggestion regarding improving the layout or work-
ow in the control room, an individual owner generally will listen to the
suggestion and say yes or no in a timely fashion.
Figure 11.1 Crescent Moon Studios; Miami, Florida. Photo courtesy of Solid State Logic.
Chapter 11 Recordi ng Studi o Operati ons and Procedures
299
In a corporate environment there is sometimes the appearance of
anonymity, but the tradeoff is that it can often take three or four
weeks before a no is issued to a suggestion for an improvement.
This slow turnaround is because every decision is made by committee.
Also, corporations have a nasty habit of closing facilities that lose
money for one quarter, a difcult reality in a business that has its
ups and downs. Since they are driven by quarterly prots for their
shareholders, they will sometimes re several employees to make
their bottom line more attractive to shareholders and investors. Even
if you are not the one who has been red, this will impact you, because
someone needs to make up that persons work. Individually owned
studios can go out of business too, but they dont go out of business
simply to make other ends of their business look more attractive to
investors, and usually a momentary blip in the business will not
cause an individual owner to close his doors.
Studio employees frequently poke fun at their studios owner. There are
many situations in which a group of people with something in common
will nd one person on whom they can blame their problems. A studio
owner is an easy target. Engineers and administrative personnel make
the studio work and keep the clients coming back for more, yet the
owner is the one who prots. On top of that, he may keep imposing
seemingly arbitrary rules on the staff. If you work at this type of studio,
be aware that without the owner, the studio wouldnt be here. The
owner may be reaping the prots, but he or she is also taking all the
risks, and if something were to happen requiring that the session bill
needed to be forgiven, the owner would take the loss.
As far as the daily operations of the facilities, an owners involvement
and responsibilities run the gamut. In some facilities the owner will be
ensconced in an ofce, dealing with situations only as they arise or when
boredom sets in, with other staff members dealing with all the details
involved in running the operation. In other facilities the owner will
also be the chief engineer, general manager, or maintenance engineer
(or possibly all of these), fullling all the duties associated with those
positions. As with everyone associated with a studio, an owner should
be as willing as anyone to empty an ashtray, make coffee, or help out
with a quick turnaround from one session to another.
300
The same holds true for managersthey should be willing to jump in
and help out in any situation for the good of the studio. The general
manager primarily acts as a liaison between the owner(s), the clients,
and the employees. Normally, the manager oversees the business end
of the studio and has input about setting rates; discounting the rate for
certain sessions and certain clients; handling banking, advertising, and
marketing; and scheduling, hiring, and ring of personnel. While the
owner often likes to keep a hand in these items, the manager is just as
often the decision maker. Because the managers responsibilities regard-
ing hiring, ring, and scheduling directly impact entry-level employees,
it should be obvious that in your rst few years of studio work, you
want to be very good friends with your manager. Later on, as a free-
lance engineer, producer, or other audio professional, you will want to
maintain that friendship in order to get more work or qualify for dis-
counted rates. You may as well start learning how to establish these
relationships now.
Another critically important skill for the manager to master is client
handling. Since managers usually take the bookings from clients, they
must become adept at juggling these bookings. Invariably, all clients
Figure 11.2 Studios Guillaume Tell; Paris, France. Photo courtesy of Solid State Logic.
301
want to work at the same time. A good manager can get one client to
start an hour earlier, another client to start two hours later, and still
another client to work in the B room. By doing this, the manager not
only accommodates more clients, he or she also increases the studios
billable hours, thereby maximizing prots, which helps everyone asso-
ciated with the studio, from the owner down to the interns.
The manager also functions as the liaison between the different depart-
ments of the studio, making certain that work orders go from the assis-
tant engineer to clerical after a session so the bill goes out quickly, and
making sure that trouble reports go from engineers to maintenance. A
good studio manager will get a trouble report, communicate with the
maintenance engineer to ascertain the amount of time needed to either
assess the problem or make the repair, and then, since the manager has
access to the bookings, he or she will schedule that necessary time
for maintenance. The manager will then follow up to ensure that the
necessary repairs were made. This is a position that requires great
organizational and people skills, and a little audio knowledge doesnt
hurt either.
While assistant managers (and night managers) are not responsible for
the same level of decision making as the managers, they are responsible
for implementing the managers directives and general day-to-day
operations. Other duties may include ordering supplies, such as tape
and rental equipment; ordering food and drink for clients; dealing with
instrument cartage; overseeing session paperwork; typing labels for
dubs and rough mixes; and, of course, making coffee. Assistant man-
agers may double as receptionists or help break down a room after a
session, and they work closely with engineers, assistant engineers,
gophers, and maintenance engineers to ensure smooth minute-to-
minute operation of the studio.
The number of engineers associated with a facility and their status will
vary greatly, depending on the size and structure of that facility. As
mentioned earlier, the owner is sometimes an engineer. This is almost
always the case with home studios, project studios, and small studios.
In larger facilities, there will often be a chief engineer, the member of
the staff who engineers the larger projects and usually has his or her
302
own client base. In other words, this is usually an experienced, suc-
cessful engineer who is an asset to the studio, so the owner has struck
a deal with this engineer to bring his or her clients to the studio in
exchange for a higher hourly rate, a percentage of the prots, or both.
The chief engineer will often be involved in decisions regarding the selec-
tion of new equipment, improvements to control rooms and studios,
which brand of coffee is purchased, and the overall direction of the studio.
While the chief engineer may or may not choose to be involved in hiring
and ring of maintenance and ofce personnel, he or she is usually
involved in hiring of staff engineers. The chief engineer almost always
selects his or her assistant engineer and has input on all assistant engineers
hired by the facility.
Many larger studios will provide staff engineers for clients at no addi-
tional charge. These are usually competent, experienced engineers
who simply dont command the client base of a chief engineer, or at
least not yet. Of course, working as a staff engineer is a great way to
build a client base. Staff engineers are well compensated, but less so
than the chief engineer. They are also lower in the pecking order than
Figure 11.3 Imaginary Road Studios in Dummerston, Vermonta state-of-the-art
digital recording facility. Photo courtesy of John Cooper.
303
a chief engineer. In many studios, particularly smaller studios, staff
engineers will double as assistant engineers when the client brings in
a freelance engineer. While staff engineer may not be a prestigious
title, and the idea of moving from the engineers chair to the assistants
chair on occasion may seem demeaning, the staff engineer usually
aspires to bigger and better things. As a staff engineer, one learns to
make better coffee and is exposed to a variety of different work styles
and clients. This work, on ones audio skills and people skills, helps to
create a well-rounded audio professional. A good client base, a good
work ethic, and a good work attitude, all necessary to a career as
either a chief engineer or a freelance engineer, are built through this
process.
Freelance engineers are usually booked by clients such as record pro-
ducers, music production companies, lm houses, or ad agencies, who
are working at a studio that does not provide engineers or who are
unhappy with the staff engineers. Freelance engineers are also booked
when the client wants a particular sound that they believe a particular
engineer can deliver. Studios will assist in booking freelancers if they
do not provide staff engineers or if they are overworked or under-
staffed due to vacations or illness. Freelancers are also booked for
other reasons; I developed a reputation for being able to handle problem
clients, so if no one wanted to work with a particular client because
the client was being difcult, the studio would call me. It may not
sound pleasant, but I worked steadily. There is no shortage of dif-
cult clients, and I enjoyed the variety of work offered by this diverse
group.
While freelancers have little or no direct responsibility to a particular
studio, as a small business it is always smart to stay on everyones good
side. No studio owner likes to hear a freelancer extolling the virtues of
another studio to the owners clients, and if that owner has an oppor-
tunity to recommend a freelancer, you can bet they will recommend
someone who thinks and speaks highly of their own facility (and is
willing to make coffee). In general, it is bad form for anyone at any
studio to speak negatively of any audio professional or musician.
There are two reasons for this. As previously mentioned, it is a small
business, and there is a good chance that the person you badmouthed
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will hear about it. Equally important, if a client hears you speaking
poorly of another client, they may wonder what you say about them
when they are not in the room. The best policy is to stay positive and
avoid negative comments about anyone in the business.
Similar to freelancers, sound designers slip into the cracks of our other
denitions. Sound designers do not work for studios, yet they are often
engineers. They are like clients in that they book a production facility to
create the soundscape for a show, yet they know their way around a
production room as well as anyone. They are sometimes client, producer,
and engineer all rolled into one. They are less beholden to studios than
freelancers are, since their clients are directors and producers and they
rarely derive work from production facilities. In projects for which I have
been sound designer, I have found myself working just as often in a radio
station production room (because it was convenient) as a recording stu-
dio, since a basic production room is really all thats required. As such,
the sound designer has no obligation to the studio and may not even
have to make coffee. Imagine that.
Having discussedfreelancers andsounddesigners, both of whom escape
the typical pecking order, lets discuss maintenance engineers. Just as
the pecking order will change from studio to studio regarding studio
manager versus chief engineer, maintenance engineers t sideways into
the pecking order. They may or may not have more power than a staff
engineer. When something is broken and has stopped the session, their
presence has a tremendous amount of weight, and at that moment
their power exceeds anyones. Large studios will keep maintenance
engineers on staff whenever there is a session booked, while some
smaller studios will do whatever they can themselves and call in a free-
lance maintenance engineer when they get in trouble.
Maintenance engineers may have the most stable position in a produc-
tion facility. Musical and industry trends may change, making one
engineer hot for a minute, and then pushing that engineer out of
demand. Studios spring up and then go out of business. Regardless,
everything eventually breaks, and the ability to x equipment is always
sought after. Maintenance engineers are accountable to the studio
manager, who follows up to ensure that scheduled maintenance and
305
other repairs have been completed, and they are accountable to all
engineers, insofar as the room that the engineer walks into should be
in functioning order with the machines properly aligned. Otherwise,
most staff maintenance engineers Ive known spend much of their
time in their little shop, guring out new and creative ways to use sol-
dering irons and sledgehammers (some of which have been namedI
once met a sledgehammer named Mother in a maintenance shop).
And yes, maintenance engineers must make coffee, too.
Figure 11.4 Skywalker Scoring in Marin County, California. Photo courtesy of Neve.
Entry-Level Positions
Please dont get frightened as we descend through the pecking order.
Assistant engineers are the heart and soul of a studio, and they are also
among the most abused people in any line of work. As a freelancer, a
substantial part of the basis on which I judge a studio has to do with
the quality of the assistant engineers. Their efciency, awareness, and
helpfulness can make the difference between a good session and a bad
session. At the same time, they are sitting next to me throughout the
session, often performing fundamentally the same tasks, and earning
between 5% and 10% of my earnings. It may not seem fair, but its
part of the training process.
Every business has its training and weeding-out process. Lawyers have
to go through law school, and a high percentage of them do not make
it through the rst year. Doctors have to complete years of college,
then internships and residencies, and many dont make it. Every
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profession nds a way to eliminate those who arent dedicated (or
crazy) enough to succeed, and for many of us the audio professionals
baptism of re is the time spent as an assistant engineer.
Long hours, low pay, no respect, no social life, and not enough sleep is
how the job description reads. Take the blame for everything that goes
wrong, apologize when it isnt your fault, never turn down a session even
if you havent slept for two days, and smile the whole time is how the
description continues. Sounds too good to be true? Wait, you havent
heard the best part. After two to ve years of demeaning yourself for
minimum wage (or barely above it), you may become a staff or freelance
engineer. About one out of a few hundred does. All thats required is a
tremendous amount of dedication, luck, and the talent to back it up.
Sounds enticing, doesnt it?
Like most jobs in the arts, audio is extremely competitive. For every
engineer who actually makes a living at it, there are 10,000 people
who call themselves engineers and actually believe they can do it.
Now would be a good time to ask yourself how badly you really
want to be in this business. If youre smart, youll say to yourself,
Maybe I should go into the family business or open a restaurant. If,
like me, youre not that smart, but you have a burning feeling in the pit
of your stomach that you wont be happy doing anything else, then pre-
pare yourself for years of hardship and torture, and also prepare yourself
for the possibility that you will end up somewhere in this business other
than where you see yourselfdubbing tape copies rather than recording
your favorite band, or possibly even waiting tables in a restaurant. Also
recognize the possibility that you will work hard and end up exactly
where you want to be. Stranger things have happened, though few
come to mind.
I am the rst person to admit that Ive been lucky in my career. I assisted
in world-class studios where I was able to build a good client base. I
moved easily into freelance, where I was supported by that base, and
when I decided to leave New York, I had the opportunity to work as
a sound designer on plays outside the city, teach at various colleges, and
still return to the city for a day of highly paid work as needed.
I acknowledge luck as a factor in my career and my success. Feeling
lucky? If not, this may not be the business for you. Please dont get
307
me wrong; it also takes years of hard work and dedication, and both the
audio talent and the people skills to back it up, but luck is undeniably a
factor in a successful audio career.
Since youre still reading this, I guess the last few paragraphs didnt
scare you off. Either that or youve been assigned this reading and
you have no choice. Either way, your reward for sticking it out is
that you now nd out exactly what an assistant engineer does
besides, of course, making the coffee.
The assistant engineer is the rst one to arrive at a session and the last
one to leave. Responsibilities include setting up microphones, amps,
chairs, music stands, and headphones for the musicians; setting up
tape machines, the console, and the patch bay for the engineer; and
cleaning up the control room and/or producers room for the client.
After the session the assistant engineer is responsible for breaking
down, putting away, and cleaning up everything just listed. Often
the assistant will have help before and after the session; other assis-
tants, gophers, and assistant managers will often pitch in, especially
for a large setup. As mentioned earlier, anyone, right up to the
owner, may pitch in and help in a big setup or when switching the
room from one big setup to another, but ultimately the responsibility
Figure 11.5 The SSL 9000 in Studio Davout, France. Photo courtesy of Solid State Logic.
308
for the setup and operation lies solely on the assistant. And if a micro-
phone is plugged into the wrong input or a track is bussed improperly,
the assistant is to blame, regardless of who made the actual mistake.
During the session the assistants duties will vary, depending on the
type of session and the engineer. Some engineers like to do more them-
selves; others like the assistant to do more. During tracking, the assis-
tant will usually be responsible for adjusting microphones once the
musicians have arrived and positioned themselves, double-checking
audio levels, throwing patches in the patch bay, and running the multi-
track. Other duties often include answering the control room phone,
ordering food for clients, giving directions to the bathroom, taking the
blame if something goes wrong, and making coffee. The assistant is
also responsible for all paperwork, including the work order, track
sheets, and take sheets. All the duties listed in this paragraph continue
through the session, whether tracking, overdubbing, or mixing.
During overdubs, additional duties may include recording the overdubs
while the engineer takes a nap or runs to the track to place a bet. While
mixing, the assistant may be expected to run a mix computer, teach an
engineer how to run a mix computer, or carry rented signal-processing
gear up ights of stairs to the control room. Welcome to the romantic
life of an audio professional.
Another primary function of an assistant engineer is to act as a represen-
tative of the studio. An assistant is familiar with the quirks of a room and
should pass that information on to a freelancer who is unfamiliar with the
room, including which faders are not working well, which effects are nor-
malled to the console, and where things that arent labeled come up in the
patch bay. Many believe the most important qualication in an assistant is
attentiveness. If the assistant already knows an engineers working style,
he should be anticipating the next move; if he is not, he should be looking
at the engineer, awaiting instructions (as he continues to visually scan
everyone else in the room in case anyone needs anything). Either way, it
is always bad formfor an assistant to challenge an engineer. I love it when
I tell an assistant to set up a Neumann U 87 for vocal overdubs, and he
says, I did it already. I hate it when he says, Are you sure you want to
use the Neumann? I like AKG 414s.
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Before we dive any deeper into studio etiquette, lets briey mention the
only person who the assistant engineer gets to boss aroundthe gopher
or intern. I group these two together because their functions in a pro-
duction facility are essentially the same. Gophers will perform many of
the assistants tasks; they may be called upon to make coffee, clean a
bathroom, take out the garbage, act as a messenger, help an assistant
with a setup or breakdown, answer phones, type labels for dubs, or do
just about anything else. In other words, the gopher does whatever needs
to be done around the studio. Often, gophering in a studio is a good way
to get your foot in the door; my rst studio job was as a gopher.
As with any job in this business, you are being measured and judged for
your next position at every moment. Dont be paranoid, but you are
being watched. In other words, only the best gophers will be offered
the job of assistant engineer, just as only the best assistant engineers
will be offered the job of staff engineer. Being a better gopher may
sound silly to youafter all, how hard is it to make coffee and take
out the garbagebut there are different ways to do things. Just like
assistant engineers, anticipating situations plays well for interns. Instead
of waiting until youre told to dump the garbage, check it and dump it if
needed. Clean the bathroom before youre told to and make more coffee
if the pot looks close to empty. And do it all with a smile. Nothing will
get you to the next level faster than anticipating situations and dis-
patching them with a smile and a good attitude.
Figure 11.6 The classic Neve 8068 at Sorcerer Sound in NewYork. Photo courtesy of Neve.
310
Studio Etiquette
Regarding etiquette, some commonsense rules apply. Never argue with
a client. Even if they are rude and abusive, as a representative of the
studio it is your place to remain calm and rational. The customer is
always right applies as much to production facilities as anywhere
else. Few things are truly impossible to do in this business. If a client
requests something be done that is impossible due to equipment limita-
tions, explain it to the client and offer to rent the equipment, being very
clear about the costs involved. If a client requests something which is
absolutely, physically impossible, calmly explain to them why it cannot
be done, without being insulting or condescending. Keep in mind that
you are the representative of both the studio and yourself, and that a
client who occasionally asks for the impossible is still a client, a source
of income for the studio, and a potential future source of income
for you.
Always think before you speak in the control room. Few things will
scare a client more than an engineer who says, Oops or Uh oh
during a session. Equally important is not insulting the client. Perhaps
the song you are working on is in a style you do not like, or maybe you
think the singer is just plain awful. Keep it to yourself; the client and
producer may have a lot of time and money invested in this project,
and your opinion is just thatan opinion. If you were consistently
right about popular opinion, you would be a record company instead
of an aspiring audio professional.
Here are some other things you should never do in the studio. As men-
tioned earlier, never insult any studio, musician, producer, engineer,
and so on. You should never insult the studio you are in. Insulting
other studios or engineers has a twofold disadvantagerst, it sounds
petty to insult your competition; second, they will hear about it, since
it is a small business. Never interrupt, argue, or offer a musical opinion
to the producer unless you have a longstanding relationship or this
producer has specied that he or she wants to hear your opinions.
Never have plans you cannot break or a social life of any type when
the producer or artist you are working with wants to work late. (In
other words, nd a very understanding partner.) Never say, Oh,
this sounds just like that Beatles tune. It can trivialize the artist to
311
regard him or her as unoriginal. Never argue with an outside or free-
lance engineer about his or her style of working; this style may be the
reason why the client is paying extra to use that engineer instead of you.
Finally, never take drugs or drink alcohol during a session. If the pro-
ducer or musicians choose to do so, thats their choice. Your only
choice is, No, thank you. As audio professionals, we always need
to appear to the client to be in complete control. Even drinking one
beer in the control room can negatively affect your clients perception
of you and your abilities. Drugs and alcohol change your perception of
pitch and timbre, thereby reducing your effectiveness as an engineer. If
you dont believe that drinking alcohol affects your pitch perception,
go to karaoke night sometime and listen to drunken people trying to
sing in key. Drugs and alcohol also hurt your efciency. You may think
you are doing your best work under these circumstances, but you are
not. Furthermore, they give the client an excuse to get something taken
off the bill. Your best friend during the session, the one who offered
you a beer or some cocaine, will turn around the next day and tell the
studio owner that some of the bill should be deducted because the
engineer was on drugs and therefore working inefciently. Your best
choice if a client, producer, or musician offers you drugs or alcohol
during a session is and will always be, No, thank you. Remember
to always be polite to a client.
The best advice I can give anyone who intends to be in control rooms is
to remain calm. Often things go wrong in sessions. The client and pro-
ducer can be tearing their hair out because something in the music isnt
matching up to the video, because their artistic visions conict, or for
any reason. It is easy to get sucked into that tension. As the engineer,
you are expected to be the calm voice of reason. Try to relax everyone in
the room by infusing them with your condent, calm, relaxed demeanor
and attitude. It works amazingly well and it helps everyone to get back
to work and actually accomplish something during the session. Work
efciently, be condent, smile, anticipate, be attentive, and always
look busythe client is paying good money to see you working hard.
And one more thing: Always be willing to make coffee, dump an ash-
tray, pick up a piece of scrap paper off the oor, or help out if the
312
studio is in a jam. No one in this business is too big to help, and the
ones that become the biggest seem to be the ones who never forget to
pitch in and never let their ego get the better of them. We all share a
common goal, from the gopher to the owner, and that is to serve the
client, to give them what they want, and to do it with a smile. We are
all here to have an enjoyable and rewarding work experience and to be
proud of our work, and that is only enhanced by our clients approval.
Recording Studio Operations
As the old saying goes, the jobs not over til the paperworks done. In
recording studios, the job doesnt even start until the paperwork arrives.
The rst thing an assistant must do before setting up for a session is
obtain the work order. This work order contains pertinent informa-
tion, including the names of the client, producer, and artist; the time
the session begins and ends; and spaces for purchase order numbers
and job numbers (to make life easier for administrators). Another
extremely useful piece of information contained on the work order is
the time the next session begins, which is very helpful if the client
wants to run longer than originally planneda regular occurrence in
studios (see Figure 11.7).
The work order also contains setup information for the assistant
including the instrumentation, types of microphones requested by the
engineer, desired locations for instruments or sections in the room, and
bus assignmentsso the assistant can have the room fully set up before
anyone else arrives.
Remember, it is the assistant engineers job to set up the entire room
before anyone else appears. The assistant places chairs, music stands,
headphones, and microphones for each musician. The assistant then
sets up the console, bussing all inputs to their assigned tracks and patch-
ing in anything the engineer might have requested, such as specic effects
off of specic aux sends or compressors inserted on specic instruments.
A good assistant will then buzz out every input and headphone in the
room to ensure that every microphone, its cable, and its input are work-
ing, they are bussed to the correct tracks, and the returns of those
313
Figure 11.7 A typical work order; this one is from National Edison Studios in New
York. Work order courtesy of National Edison Studios.
314
microphones come up through the monitor section of the console and
through the headphones. Ideally, when the rst person other than the
assistant and anyone helping him with the setup arrives, the assistant
should be sitting there, relaxed and condent (or at least looking con-
dent), knowing everything is ready to go.
Most importantly, the work order contains slots for the assistant to keep
track, as the session moves on, of hours spent in various operations,
including recording, overdubbing, mixing, editing, and dubbing. These
categories are broken down on some studios work orders because many
studios charge different rates for different operations. This part of the
work order is the most important because the client is billed on the basis
of entries in this section. If this section is improperly lled out and the
client is improperly billed, there will be hard feelings all around, and the
assistant responsible may be red. Accuracy of work orders that results
in accuracy of billing is critical to the continued success of a studio.
Everyone in the studio gets paid as a result of clients paying their
bills, and their bills are generated from these work orders. That
makes work orders incredibly important.
Other items noted on the work order include rental equipment, tape,
video playback, lock-up, automation, transfers, discs for storage, CDs
for client dubs, reels and boxes, messengers, and food, all of which
may be marked up and are therefore potential additional sources of
income for the studio. It is the assistants job to keep careful track
on the work order of everything that happens and everything that is
used in the session so a proper bill can be sent out. As stated a moment
ago, the continued success of a studio depends on this accuracy.
Studios will frequently include tape release forms on their work orders.
This form is part of the work order because some clients will request
that their tape leave the studio at the end of the session. Tape release
forms must be signed when a client wishes to remove a master from the
studio or have it sent to another studio. The reasoning for this is that an
active studio generates a tremendous number of masters, and tape
release formswhether separate forms kept in a tape library book or
whether part of the work ordertell the staff where a master is. Many
studios have had the experience of a client calling up for a master six
315
months after a session, and being unable to nd the master. A quick
check of the tape release forms will tell the staff where that master
was sent and who authorized it. Since a studio is responsible for the
masters it stores, this information is critical.
Many work orders will require a clients signature even if the master
remains on premises. In this case the client is signing to conrm that
the hours and materials listed are accurate, to prevent them from com-
plaining later that they used fewer hours or less material than was indi-
cated in their bill. As you can see, this work order accomplishes a great
many things. It gives the assistant the information he or she needs to set
up the session. It gives the ofce the information they need to prepare the
bill correctly. It also conrms the accuracy of the bill and claries the
location of the master that resulted from the session. Overall, it is easy
to see why this is the most critical piece of paperwork in the studio.
Along with the work order, the assistant is responsible for two more
pieces of important paperwork on every sessionthe cue sheet (or take
sheet) and the track sheet. In addition to a head section listing the client,
producer, artist, engineer, date, reel number, and format, the body of the
cue sheet is a running list of the beginning and end of each take, or
attempt to record a song or production piece (see Figure 11.8).
There are spaces for the start and end times, title, take number, code
(such as CT for complete take, FS for false start, or PB for playback),
and comments.
The comments section of a take sheet is where the assistant engineer
has the opportunity to distinguish himself or herself. The more com-
ments the assistant makes on the cue sheet, the better. Suppose a band
does 20 takes of a song, and the producer asks for playbacks of six of
those takes as you were rolling along, then at the end says, I liked the
second one we played back best. Which take number was that? A
good assistant engineer will have the answer if he or she wrote PB
for playback in the comments section of the take sheet. This situation
and ones like it happen frequently. Take good notes on even random
comments. For example, if the producer says, I liked the bridge on
that one, write down producer liked bridge under comments, or if
the bass player didnt like his or her performance but everyone else
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Figure 11.8 A typical take sheet; this one is from Webster Universitys studios. Take
sheet courtesy of Warehouse Recording.
317
seemed happy, write down bass player unhappy. Later, after 63 more
takes, when the producer says, That one where the bass player messed
up was probably the best take we got, and he could have xed his part; I
wish we knew which take that was, a good assistant engineer will
become the hero of the day by saying, That was take nine. The cue
sheet is only used when cutting tracks; once a choice take is selected, it is
circled and marked as choice, and that is the only take you will work on
during overdubs and mixing.
Track sheets are a method of cataloguing track locations of instruments
in multitrack formats (see Figures 11.9 and 11.10).
Along with a heading, which contains the date, client, artist, engineer,
reel number, and title of the song, there are boxes representing the dif-
ferent tracks on the multitrack, in which the assistant enters the name
of the instrument or production element recorded there and often some
comments. It is common to note the type of microphone used, perhaps
Figure 11.9 A typical track sheet, from the National Edison Studios. Track sheet cour-
tesy of National Edison Studios.
318
the preamp, and the panning. If there were four guitar tracks, it would
be helpful to know what each one is. The assistant may note rhythm
on one, rhythm double on another, lead on a third, and solo on the
fourth. Similarly, it is common to record several vocal tracks in an
attempt to get one good one. Notes such as strong rst chorus or
good at end may help later when trying to combine six or seven
vocal tracks into one good composite track, which is noted on the
track sheet as vocal composite. Just like the cue sheets, too much
information is better than too little.
Both cue sheets and track sheets remain in the box with the masters, so
that if the master gets pulled out one or two years later, the assistant and
engineer immediately have all the information they need, including
which is the choice take, which tracks are already used, and which of
those tracks were used in the original mix. Properly lled-out paperwork
helps to make remixing old tracks, whether two years old or twenty
years old, easy (see Figures 11.11 and 11.12).
Figure 11.10 Another typical track sheet; this one is from the Center for Media Arts.
Track sheet courtesy of Center for Media Arts.
319
The other paperwork that the assistant engineer will sometimes ll out
is the trouble report (see Figure 11.13). When something doesnt work
or doesnt sound right in the studio, a trouble report is lled out and
sent to the studio manager to notify everyone of the problem. As pre-
viously stated, the trouble report goes to the manager (rather than
directly to the maintenance engineer) because the manager has the
bookings right in front of him on his desk. In the event that the
room is downin other words, the project cannot continue because
there is a major problemthe manager will call the maintenance engi-
neer in immediately to try to solve the problem. If the assistant and
engineer can work around the problem, the manager is notied because
he can best determine when a repair can be made, since he can see
openings in the studio bookings.
Figure 11.11 Well lled-out track sheets, like these from the original Derek &
the Dominos sessions, make remixing easier. Track sheets courtesy of Polydor
Records.
320
As an example, if there is no session booked between 3:00 and 4:00,
the manager will suggest that maintenance take a look at the problem
during that hour. Maintenance will come in while the studio is open
and either x or assess the problem to be xed later. If the problem is
simple, maintenance will typically x it on the spot; if the problem is
too complex to x in one hour or if maintenance needs to order parts
to make the repair, they will notify the manager. In the rst situation,
the manager will look for a day when there are enough hours open for
maintenance to make the repair; in the latter, they will schedule the
repair once the part comes in. Either way, maintenance will keep the
manager updated on the progress of the repair. It is maintenances
responsibility to x the problem; it is the managers responsibility to
track the progress and status of the repair and conrm that the repair
has been performed.
Figure 11.12 Another properly lled-out track sheet from the original Derek & the
Dominos sessions, this one of Layla. Track sheets courtesy of Polydor Records.
321
Being the Best Assistant
What does it mean to be the best assistant? What does it take to be the
best assistant? We have already discussed many of the traits common
to successful engineers, so lets put all this information into one place.
Have you ever heard the expression that the three most important
things in real estate are location, location, and location? Here, in no
Figure 11.13 A typical trouble report.
322
particular order, are the most important traits for success as an audio
professional:
n
Luck
n
Luck
n
Luck
n
Talent
n
Awareness
n
Willingness
n
Anticipation
n
Condence
n
Personality
n
Client-handling ability
n
Ability to smile through menial tasks
n
Ability to leave all baggage at the door
n
Ability to take the blame
n
Ability to never complain
I cannot overstate the importance of being in the right place at the
right time. Unfortunately, this is something over which you have no
control. You can put yourself close to it by working in an entry-level
position in the type of studio in which you would like to work or in a
studio that does the type of work that you would like to do, but there
is no guarantee that you will be the assistant on the perfect session the
day that the staff engineer or freelancer has a freak accident and the
assistant gets to step up to the engineers chair. You can improve your
luck by being close to where you want to be, but luck is not a given.
To succeed as an audio professional, you need talent. When you get that
break and you sit down in the engineers chair, you will need to have the
stuff to back it up. Start building that now. Engineer every project you
can, even if it doesnt pay you one penny. Play with equipment to the
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point where you look comfortable doing so. Develop your ears. Develop
a personal aesthetic.
Engineers who succeed are aware of their surroundings, which enables
themto anticipate the clients needs. Think about the last time you were
in a restaurant. Did you have a good waiter? What does that mean? A
good waiter scans the room after serving a table, noticing whether
another table needs something; a bad waiter stares at the oor as he
returns to the kitchen after serving. The audio equivalent to the good
waiter is the engineer who scans the room to see whether anyone needs
anything. As an assistant or an intern, never bury your nose so deep in
the paperwork or the keyboard that you are unaware of the room.
Every 10 to 15 seconds, an assistant should look up and sweep from
one side to the other, looking at the client, producer, engineer, and
musicians to see whether anyone is trying to catch his or her attention.
It could be something as important as a musician who cannot hear him-
self in the headphones, or something equally important, such as a pro-
ducer looking for a menu or a client looking for a sharp pencil to play
with. One mark of a good engineer is this awareness, which inevitably
leads to the ability to anticipate situations.
Anticipating situations is one of the hallmarks of a ne assistant, one
who is likely to have a great future as an engineer. If we are switching
from tracking a band to overdubbing vocals and I say to my assistant,
Go put up an 87, my favorite response is when the assistant says, I
already did. How did the assistant know to do that? First, he has the
work order, which states that we will overdub vocals when we nish
tracking. Second, top studios keep les on freelance engineers that
contain information on everything from microphone preferences to
how they like their coffee, which is how the assistant knows to put
up an 87. Its also how the assistant knows to meet me with a cup
of coffee, light, with three sugars when I walk in; it is all in the le.
If your studio does not keep this information on le, start keeping it
yourself. This kind of behavior turns assistants into engineers.
Clients and producers look to engineers to be in control of the situa-
tion at all times. As such, we must always look condent. Some of the
busiest engineers I have met got that way by looking condent when
324
confronted with gear or situations they had never dealt with before. If you
look nervous to the client or the producer, they will get nervous. It is
important to be condent, although it is far more important to appear
condent to allay any fears your clients may have. If the engineer is
relaxed, the session goes more smoothly. Start practicing that as an intern.
If you were a producer, would you want to spend 12, 14, or perhaps
16 hours a day in a small room with an engineer who appeared nervous
and withdrawn? Probably not. Start developing your people skills now.
Work hard at the console, but let your personality shine through. Be con-
dent and secure, but never let your ego get the better of you. We deal
with inated egos all day long in the music business, between musicians,
producers, and clients. There is just not enough space in that small control
room for us to bring our egos along. Leave them at the door when you go
to work. Also, leave all your baggage there. A client or producer may say
something to you like, How are you? He or she is being nice and usu-
ally does not really want to know. Unless you have a longstanding rela-
tionship with this person, the proper answer is, Fine, howare you? The
wrong answer is any one that includes cars breaking down, pets dying,
relationships ending, or IRS audits. In other words, they do not want to
know about your problems. We all have them. Professionals do not bring
them to the session. Feel free to tell a joke during a few seconds of turn-
around time, but if the client is paying a few hundred bucks an hour,
make sure the joke is a one-liner. If you need some good one-liners,
read Paul Lehrmans The Insider Audio Bathroom Reader.
At entry-level positions, we have more considerations. Part of an intern
or assistant engineers job is to take out the garbage, make the coffee,
clean the bathrooms, and do anything else that you may think is beneath
you. Do these jobs with the same positive attitude you have in the control
room. Do these tasks with a smile, and do them before you are asked to.
Remember: We are all watching you. If you complain about dumping the
garbage, you will probably complain about musicians, producers, and
maybe even the studio in which you work. Why would we promote a
complainer? If you have to be told to make coffee or empty a garbage
pail, then you probably wont anticipate your clients needs either. That
means you are not audio-professional material. If you pass a garbage pail
that is more than half full, empty it. If you pass a coffeepot that only has
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one cup left, dump it and make a fresh pot. Dont tell us you did it (despite
your egos suggestion to do so); well knowyou did it and well appreciate
that you did the right thing without being asked, without making a big
deal about it, and with a smile. Thats audio-professional material!
Another thing interns and assistants are called on to do occasionally is to
take the blame for someone elses mistake. We are all responsible for
making the people above us in the hierarchy look better to the people
above them in the hierarchy. As an engineer, I always try to help my
producer look good to his client. As an assistant, you may be called
on to keep your engineer looking good to his or her producer or client
by taking the blame for the engineers mistake. This is not fair, and as an
engineer I take the blame for my own mistakes, but many do not. If you
work with an engineer who blames you for his or her error, say you are
sorry and you wont do it again, look contrite, and move on. Sadly, that
is part of the job. As you do, remember the bright side to all thisthe
producer and the client are right there on the engineers other side. Odds
are they saw who made the mistake and they gained respect for you (for
doing the right thing) and lost respect for the engineer (for not owning
up to his or her mistake). This producer could be among your rst clients
as an engineer as a result of you playing the game properly.
Now that we know what everyone does in the studio and we understand
what is expected of us and everyone else, lets t the last piece into the
puzzle: sound design.
Exercises
The only way to be a great intern and become a great assistant engineer
is to work in a studio. Nonetheless, prepare yourself by learning how to
make coffee if you do not already know how. Practice awareness of your
surroundings at home or in your school. You may not be inclined to
actually empty a garbage pail at school, but practice scanning the
room to see what could be done. Is the garbage pail full enough,
where I would empty it if this were a studio and I was an assistant? Is
there food garbage that I would throw away if I were responsible for the
appearance of the room? Are there enough pens and sharpened pencils
available? Start training yourself now to be more attentive.
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Additional Reading
Alten, Stanley R. Audio in Media, 7th ed. Belmont, CA: Wadsworth, 2004.
McGraw-Hill, 1988.
Camras, Marvin. Magnetic Recording Handbook. New York: Springer,
2001.
Clifford, Martin. Modern Audio Technology. Engelwood Cliffs, NJ:
Springer, 2005.
Press, 1993.
CA: Alfred, 1989.
327
Jones, Steve. Rock Formation: Music, Technology, and Mass Commu-
nication. Newbury Park, CA: Sage, 1992.
Focal Press, 2002.
Lehrman, Paul. The Insider Audio Bathroom Reader. Thomson
Course Technology, 2006.
Press, 1989.
Press, 2000.
Woram, John. SoundRecordingHandbook. Indianapolis: H. W. Sams, 1989.
328
12
Sound Design
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S
imilar to many endeavors in audio that we have already discussed,
perhaps the culmination of them all, sound design, involves creat-
ing effective cues, producing a believable sound environment, or
creating a mood within a live audience. More signicantly, sound design
is an amalgam of all the technical and artistic skills we have previously
put forth, with an ultimate goal of becoming an integral part of a lm,
play, live installation, or other performance. One of the most difcult
aspects of sound design is that the audio professional needs to accomplish
all this without dominating the production. It is often tempting for a
sound designer who nds an exceptionally ne sound to pump up that
sound in order to show it off, just as an inexperienced mixer will put too
much of a good effect in as a matter of pride, a disc jockey in love with
the sound of his voice will be too verbose, or a writer in love with his
words will be less than concise in his writing. These actions are usually
detrimental to the artistic concept as a whole. Sound should always be
regarded as subtly supporting the project along with all other design ele-
ments, and never thought of as the star while creating a sound design.
In a business where egos run strong, a sound designer must release his or
her ego and view the overall aesthetics of the project rst. Viewing the
project as a whole and sound as only one component is germane to the
success of the project. Sound can be an important tool to establish a
location, create or increase a mood, or reinforce the action; however,
it typically accomplishes these goals in conjunction with other effects,
such as lights, sets, costumes, blocking and other directorial decisions,
and of course the script. Sound should always be appropriate and in
balance with effects that affect our other senses, resulting in a totally
believable environment, rather than a disparate set of creative designs.
As sound designers, we can sometimes feel as if we are the unappreciated
designers. Although some in the visual arts believe that sound is second-
ary, the most distinguished and successful lm directors of all time were
and are extremely focused on sound. Orson Welles, Alfred Hitchcock,
Ingmar Bergman, Steven Spielberg, Francis Ford Coppola, and Woody
Allen, to name a few, were all fully aware of the impact of their lms
soundtracks. Every good director sees the whole lm in his head before
he has shot a single frame. The best directors hear the sound design
too. It is easy to tell a story with pictures. It is more deeply emotional
to tell (or reinforce) that story with sound.
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Remember, sound without picture is called radio; picture without
sound is called a technical difculty.
Believability of Sound Cues
One of the sound designers primary concerns is the believability of the
sound cue. The sound of crickets occurring while a single light shines
onstage will establish the location of the scene as somewhere in the
woods on a moonlit night, and the sound of buses, trucks, and cars
passing by will establish a city scene, but how does one differentiate
between the sounds of New York and the sounds of London? Do
buses, trucks, and cars sound different in these two cities? Are other
background noises different? Is the theatergoer aware of these differ-
ences? To a sound designer interested in creating an effective and
believable cue, the answer is always yes.
Although everyone seated in the audience may not recognize the differ-
ence between the sound of a NewYork bus and a London bus, the sound
Figure 12.1 Sorcerer Sound is a ne production room. Photo courtesy of Neve.
designer must assume that at least some segment of the audience knows
the difference and must therefore provide the correct sound to establish
the location. If, as a theatergoer, a scene taking place in London had the
more guttural and less oscillating sound of a NewYork bus passing by, it
would be jarring. Similarly, if the actors in a lm are discussing a
Chapter 12 Sound Desi gn
331
Volkswagen passing by and the sound cue is of a truck, it would not be
believable. Often these cues are received and processed unconsciously,
and the audience is not even aware of why they do not nd the scene
believable, but if the result is not convincing, then the sound designer
has failed, regardless of where the audience places the blame.
To return to London versus New York for a moment, consider a scene
occurring indoors with a telephone ringing. English telephones ring in a
distinctively different manner than American telephones, in both fre-
quency and meter. What would happen if a sound designer used the
sound of an American phone during a scene occurring in London? The
American phone ringing during the scene in London would be as jarring
as the double-ring of an English phone ringing during a scene occurring
in NewYork. To further illustrate this point, consider the difference bet-
ween an outdoor, nighttime scene occurring on the bayou in Louisiana
or one on a lake in New England. Both would have crickets, but the
sound would be substantially different. On the bayou, the increased
humidity dampens sound, which shortens the distance that the high-
frequency components of the sound can travel, resulting in stronger
components in the lower frequencies. In the cooler, less humid night-
times of the north, higher frequencies travel better, causing crickets to
sound brighter. Furthermore, there are other sounds. In New England it
is common to hear tree frogs, whose sounds repeat rapidly and at high
frequencies; in the bayou one would be more likely to hear lower-
frequency bullfrogs with their mellow croaks. The sound designer must
be ever conscious of these differences when establishing locations.
To recognize and utilize these differences to our advantage, we should
incorporate the ve perceptions of sounds discussed in Chapter 2. Loud-
ness, pitch, timbre, location, and envelope are all incorporated in the
believability of any cue, just as they are part of our ability to perceive
sounds in any environment, real or fabricated. The difference between
northern frogs and southern frogs is mostly related to pitch and timbre;
the difference betweena truckanda Volkswagenalso involves both pitch
and timbre. Location is critical in all cues. If the script calls for a car to
pull up offstage, stage left, stop, and shut its engine off, would the sound
of a car on an overhead speaker cluster be believable? Certainly not.
Of course, the exception regarding establishing location is when the
location is one with which few or no people are familiar. If the
332
scene takes place in ancient Egypt or deep in the Amazon, the odds are
good that no audience members know what the environment repre-
sented truly sounds like. There are no audio recordings of ancient
Egypt, and few among us have been to the depths of the Amazon. In
these situations, the sound designer has the obligation to create a life-
like environment representative of the sounds in that unknown time or
location based on his or her assumptions of what we all think that
environment would sound like. In the deep jungle, one may use distant
screeches of chimpanzees, calls of exotic birds, or the growls of big cats.
Is that really what the deepest Amazon jungle sounds like? I dont know,
and your audience will not knoweither, so we simply trust our intuitions
and make our best assumptions. Regardless of whether we are working
from personal experience or our best guess in any given situation, the
keys to a successful sound cue in these situations are still believability in
its creation and subtlety in its delivery.
Semiotics
As sound designers we deal with a large variety of cues that are incor-
porated into productions in a variety of ways. In our desire to create
these cuessome of which fall upon the conscious mind and some of
which are subliminalthe eld of semiotics can be helpful in dening
and codifying our terms. Semiotics, as explained by Martin Esslin, is a
system of analyzing the theatre experience based on the signs given
from the stage. Although this system was originally designed to describe
the effects of visual cues on the audience, we can easily expand the def-
inition to include the audio experience.
Before we explore the system of semiotics, a tabular classication of
sound cues based on location and mood will be helpful in assessing the
various avenues that are available for the sound designer and the emo-
tions they will evoke (see Table 12.1).
Many of the examples given are specic to a particular setting and are
only samples of a sound that will evoke a particular emotion or location.
While some of these may seem trite and predictable, please keep in mind
that we are dealing with a base level, often subliminal reaction from your
audience. These cues work on that level. They evoke the desired emotion
or imply the desired surroundings. With the audience caught up in the
totality of the production, they will not be conscious of a trite but subtle
333
cue if the sound designer has done his job correctly. If, on the other hand,
the sound designer has created the cue well but plays it back too loud, it
will be too obvious and will be unlikely to evoke the desired effect.
Incorporating semiotics, when something onstage, such as a table, or a
sound cue, such as wind, is simply a table or wind, it is dened as an
icon. An icon is exactly what it appears to be. When the sign makes the
audience think of something else, it is an indexthe table is not just a
table, but representative of an ofce; the wind is not merely the wind,
but a sign of an incoming storm. An index takes a sign and points to
something else. If the table is representative of the disorder of the mind
Table 12.1 Identification of Location and Mood through Audio
Desired Identication Possible Cues
Location through sounds
of nature
Crickets for outdoors at night; rain/wind
to establish a storm
Location through sounds
of people
Several voices to establish presence of a
few people; many voices to create a crowd
Location through
machinery
Guns and bombs create a war location;
drill presses and table saws indicate a
factory; looms indicate a different type
of factory
Location through music A Russian country folk band will transport
the audience to Russia; Wagner will conjure
images of Germany; and America the
Beautiful will set the scene in the US
Mood: sadness A fog horn or bowed low strings, such as
whole notes; slow tempo on celli in a minor
key
Mood: loneliness A lone wolf howling or the same musical
piece as sadness on a single cello
Mood: love Birds singing or a sweet violin section
Mood: fear Thunder or tremolo strings
Mood: discomfort Fingernails on chalkboard
Mood: relaxation Ocean, rain, or other sounds rich in white
noise
Mood: anticipation Silence or high-frequency pizzicato strings
334
or the wind portends an impending disaster, it is a symbol. Sound cues,
such as props, can change meaning during a production, moving from
icon to index to symbol. The wind we discussed can begin as simply
wind, and then through the plot can move to index as we learn of a
storm coming in. It can then move to symbol as we realize the wind to
represent the general feeling of impending disaster. The sound cues of
the wind in the lm Key Largo move in exactly this fashion as the plot
unravels to reveal a forthcoming disaster, both through the storm and
in the hotel in which all the action occurs.
Semiotics also gives us a system for codifying the subtlety of a sound.
According to one theory in semiotics presented by Paul Nelsen, there
are three zones of reception, or three ways in which information and
cues will be received and processed by the audience. The rst zone is
focus, in which the audience receives the cue with cognitive aware-
ness; their senses are fully aware of the cue they are receiving. The
second zone is peripheral or para-cognitive, when the senses are
aware of the cue but not focused on it. The third zone is subliminal,
in which the audience is cerebrally unaware of the sound environ-
ment, but the senses are absorbing it. Much like the signs discussed
a moment ago, this is a dynamic and ever-changing system.
A cue can be peripheral and subliminal at the same time, and gradually
move up into focus, as in the movie Jaws, where the approach of the
shark is heralded at rst by a sound cue with an extremely low level,
sometimes buried deep in the underscoring, a subliminal cue. This
sound gradually moves into the peripheral, then into focus. By the
time the audience is focused on the sound and aware of it, the fear is
built up and the presence of the shark is anticipated. As we will see in a
moment, this anticipation builds tension tremendously. The threshold
between these zones is also dynamic, depending upon how absorbed
other senses are at the moment when a sound cue is called or dropped in.
If there is a great deal of visual movement in the lm or movement on
the stage and changing light cues, the threshold of a sound cue will rise.
Conversely, in a quiet and static scene, the threshold for a sound cue
can be quite low. As sound designers we must be aware of the total
production and how our cues t with other production elements; we
cannot simply make great cues and walk away.
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Emotion, Anticipation, Subtlety, and Continuity
Sound is an important factor in increasing the emotional response and
the anticipation of emotion in the audience. As mentioned before, in the
lm Jaws a low-frequency, repeating sound prepares the audience for
the sharks appearance. Due to its subtlety, the warning is primarily sub-
liminal at rst, and the subsequent appearance of the shark is much
more frightening as a result, as the sound cue moves to peripheral and
then into focus. This sound begins as an icon, exactly what it appears to
be, but later becomes a symbol that the shark is about to appear, as the
audience becomes accustomed to this sound forecasting disaster. If the
sound rst appeared at full volume, would the cue be as effective?
Another good example of sound enhancing the audiences experience
of fear is the lm Psycho, in which the high-pitched, repeating sound
during the shower scene heightens the audiences fear effectively.
An important concept in the sound cue in Jaws is the establishment of
continuity. The rst time the sound cue appears, followed by the shark,
the audience is surprised. Every time that sound is heard after that, the
audience anticipates another appearance of the shark and the fear that
accompanies its presence. The director can prepare the audience to be
Figure 12.2 The DFC Gemini digital console is tailor-made for post-production and
lm scoring. Photo courtesy of Neve.
terried by calling for this sound cue, and then show something other
than a shark. This causes comic relief, which only makes the audiences
experience of terror stronger when the shark nally does appear. This
continuity, establishing a pattern and continuing it throughout a pro-
duction, is an important tool in controlling the audiences mood and
336
manipulating the audiences many emotions to enable the director to
evoke precisely the reaction he or she desires.
Similarly, a sweet and mellow string section evokes feelings of love, a
high-tempo piece of music infers fast-paced action, and certain chord
structures will give the audience a feeling of resolution. Any of these
cues can be established early in a production and used again whenever
the designer wishes to evoke the same emotion, and any can be offered
and then withdrawn to put the audience through their emotional
paces. As with all sound cues, these must be used subtly to evoke
the greatest response. If the sweet string section is so loud that we
can barely hear the actors declare their love for one another, the emo-
tion will not be evoked successfully, and the cue will be a failure. Also,
if a sweet string section is used under fast-paced action, would the
designer evoke the desired emotion?
Subtlety must also be used with sound cues that reinforce plot points. If
the script calls for a car to pull up, the sound of a car pulling up should be
believable in character, location, and volume. Unless the script speci-
cally calls for a car that needs a mufer, the volume should be peripheral,
establishing the car in the distance and bringing it into focus as the car
gets closer. Always consider sound cues in the context of real-life situa-
tions: When a car pulls up to pick you up while you are in the midst of a
conversation, does that car drown out your conversation? Hopefully not.
The exception to this rule is when the script calls for a sound to be
appropriately loud to reinforce the plot. If the cue is an explosion
occurring onstage or in frame, it should be loud and in focus. In
these situations, especially if the explosion is unexpected, excessive
volume will increase the shock value of the cue, and shock may be
exactly what the director is looking for in your cue. Of course, this
cue should be coordinated to occur along with a lighting cue of a
quick, blindingly bright light. In virtually any lms starring Arnold
Schwarzenegger, Sylvester Stallone, or Jean-Claude Van Damme, this
can be seen and heard as helicopters explode, shells are red, and
whole cities are destroyed. These sounds are not subtle, but they are
effective. Due to the increase in volume, it becomes more important
that the sound is believable. If the script calls for an explosion that
levels a building, do not use the sound of a recracker. The audience
should feel that cue in their chest as well as hear it. Believability is still
337
our prime concern, even when the script encourages us to abandon
subtlety and move cues from peripheral to focus.
Another example of appropriate loud sound cues occurs when the
action takes place in a crowded bar. In the lm48 Hours, Eddie Murphy
and Nick Nolte are conversing in a loud bar, yelling to hear each other
over the very believable ambience. The same occurs in Stripes, with Bill
Murray and John Candy attempting to converse in a believably loud
bar. These are both examples of a sound cue in focus, acting as an
index to represent that the actors are in a bar.
Silence
Another potential choice in sound design is the use of no sound. Silence
can be an extremely effective dramatic tool, even at times when the
script or the director calls for music or a sound effect. If the climax
of a drama is reached through actors relating a tragic event, sometimes
the gravity of the events can be reinforced best through silence.
Consider what happens when there is silence in a room full of people.
People in the room shufe their feet. They cough or clear their throat.
They look around to see what is wrong. In a nutshell, they are uncom-
fortable as they wait for something to happen. They wonder what will
happen. What a valuable tool silence can be when the sound designer
wishes to make his audience feel discomfort or anticipation!
Use of silence requires a release of the sound designers ego, which
believes that every moment in a production should be reinforced with
sound. After all, anyone can create silence and inject it into the perfor-
mance. Nonetheless, this could be the best dramatic contribution to the
production. At times it is simply more appropriate to let an ominous
silence reinforce the mood than it is to create a large, imposing sound cue.
When sound dominates our other senses, the result can be unnatural,
detracting from the writers and the directors intent. If theatergoers
emerge from the theatre commenting on the sound, it was probably
overwhelming other elements of the play and therefore was inappropri-
ate. Even a compliment to a sound designer can be an indication that the
sound was out of balance with the other elements, because sound should
always create the necessary effect without being obvious. If it was
obvious enough to be noticed by anyone other than another audio pro-
fessional, even in a positive manner, it may have been too loud.
338
Methodology
The rst step in sound design may seem obvious. Read the script. Then
read it again. While reading the script the rst time, lift all audio cues,
including stage directions and notes that are usually either in parentheses
or italics, andbegin your initial cue sheet (see Figures 12.3 through 12.8).
Figure 12.3 The preliminary audio cue sheet from The Boys Next Door.
339
As you read the script, consider the emotions evokedboth your emo-
tions as you read and the emotions the director will wish to create and
enhance within the audience. These initial impressions will affect your
musical suggestions and the possible need for music or effects to estab-
lish or reinforce these emotions. Keep in mind that every play is
Figure 12.4 The nal cue sheet from The Boys Next Door. Notice the difference
between this and the sheet in Figure 12.3.
340
Figure 12.5 Preliminary audio cue sheet for On The Verge.
unique, and the emotions evoked by any event in any play must be
dealt with as an entirely new situation.
Sometimes the musical choices will seem obvious; if the play is The
Inspector General, set in a rural Russian village, Russian folk music
341
Figure 12.6 Interim audio cue sheet for On The Verge. Notice the changes as the work
progresses.
could be an obvious choice. If the play was written by Tennessee
Williams, dark, steamy southern music may seem obvious. The obvi-
ous choice is not necessarily wrong; however, the sound designers job
is not always to provide the obvious. I once executed a sound design
342
Figure 12.7 Final cue sheet for On The Verge. This cue sheet could be used by the
board operator through the run of the show.
for The Boys Next Door, a play about four developmentally disabled
roommates, and the script suggested Cole Porter to me for no partic-
ular reason. I gave the director a cassette with several Cole Porter
songs, and he thought it was the perfect choice for his approach to
343
Figure 12.8 An interim cue sheet from The Inspector General. Cue sheets do not need
to be fancy to be effective. By the time you are a few weeks into a production, it is
entirely possible that you too will have grease from potato chips or coffee stains on
your cue sheets. In the event that your director objects to food in the theatre during
rehearsal, it may be better to redo your cue sheets at the point when they get this
messy. One director almost red me for eating onions on my sandwich during lunch
an hour earlier.
344
this play. The design was a success, even though the music selected was
not the obvious choice.
Attention to Detail
This idea of not settling for the obvious carries over into the sound
designers ability to help establish or reinforce either the location or
the mood. If the characters are strolling to a lake, the sound designer
may suggest the subtle sound of bullfrogs in the distance, moving from
subliminal to peripheral, perhaps even into focus as the scene pro-
gresses. If the characters are talking about the awful rainstorm outside,
a good sound designer does not need stage directions that say sound
of rain to know that sound effect would be appropriate. Nonetheless,
a good sound designer always reads between the lines to nd the cues
that could enhance the production.
There are many cues that can be included successfully that are not called
for in the script. While dealing with location, these will be found in the
movement of the script. When locations are named at the beginning of a
scene, or when a scene moves into another location through the move-
ment of the characters, the sound designer must be focused on what
aural experience is necessary to establish or reinforce the new location.
Incorporating Subtlety and Subliminal Cues
Analyzing the subtlety of the script becomes even more important when
establishing or reinforcing a mood, such as anticipation. This must be
done subtly and carefully. While the proper chord played on low string
instruments, such as celli and double bass, repeated at a slow, regular
interval will surely inspire anticipation in the audience, if this sound is
too much in focus or played at too fast an interval, it will surely fail. At
best, the audience will subconsciously feel uncomfortable; at worst, the
audience will be distracted to the point where the sound is camp, a fatal
error in drama. In the instances of comedy or farce, camp may be the
goal; in drama it will invariably subtract from the overall project.
Often these emotional cues can be done subliminally. It is not neces-
sary for the audience to consciously hear the low, repeating sound for
345
it to have the desired effect. It can be played low enough that the audi-
ence is impacted by the sound without hearing it consciously. Simi-
larly, if the designer wishes the audience to experience the same
discomfort as the characters portrayed, a high frequency played at
an amplitude too low to hear will have the desired effect. It can some-
times be very effective to position these sounds such that they move
from subliminal to focus at the proper time. If the low, repeating
sound were inaudible at rst, creating the sense of anticipation, and
then it were brought into focus at the moment when anticipation
turned to fear, the effect would be to set up and reinforce both emo-
tions. Returning to the example of Jaws, most viewers can repeat the
sound that occurs once the shark is seen, but only the most astute
observers are aware that the same sound was present on a subliminal
level for several minutes before the sharks appearance, when the
anticipation of the fear was occurring.
Incorporating Silence for Anticipation
As mentioned earlier, another strong subliminal emotional cue is the
proper use of silence. Silence creates anticipation. It makes the audience
uncomfortable to hear only the sounds of their fellow audience mem-
bers shufing their feet and waiting for something to happen. This
essence of anticipation can be very effective in setting the emotional
table, as the audience waits impatiently for the next event. Whatever
emotion is evoked after a silence, it will be that much stronger due to
the anticipation and discomfort experienced by the audience.
As noted in the previous section, an overpowering sound or effect at
the wrong moment can negatively impact the entire production, which
should encourage the sound designer to consider silence in these most
critical scenes. While reading the script, the sound designer must stay
focused enough to nd the proper places for effects to create moods,
keeping silence among his options as he considers the potential down-
side of any other more obvious effects.
The Challenge of Ambience
One of the most difcult and common sound cues to incorporate is
ambient sound. While creating a believable bar, party, or city street
is relatively easy, some difculty may arise when we attempt to
346
establish our playback level. The level must be loud enough to be in
focus to establish the location, yet it must somehow never distract from
the actors lines. Ambient cues tend to have an excess of 1 kHzboth
natural ambiences, such as rain, water owing in a stream, or wind,
and human scenes, such as bars or parties. This is equally problematic
in nature cues rich in white noise and when the ambient sound includes
human voices, because the ambient voices may mask the actors voices.
If the designer simply lowers the overall ambient level, it may be too
subliminal to be convincing. The best solution to this problem is often
found through curing the masking with equalization, by lowering the
frequencies at which the voice normally occurs (1 kHz to 3 kHz) in the
ambient track. This essentially creates a notch in the ambient tracks
frequency, which the actors voices dovetail into very comfortably,
making their voices audible above the din. Another effective method
is to use the ambient cue at a loud level at the beginning of the scene,
and then lower it slowly, subtly into the periphery as the scene pro-
gresses. If this is done correctly, the lowering of ambience is not notice-
able to the audience and it opens up timbral space for the actors
voices.
More subtle types of ambient sound, such as crickets or the occasional
bullfrog establishing an outdoors, rural scene, can also borrow from
the subtlety of the ne line between the peripheral and the subliminal.
Crickets can be audible at rst to establish the location, and then fade
slowly to affect only the subconscious. The effect will still be present,
but only subconsciously. Conversely, the bullfrogs can begin sublimi-
nally to set the emotional table for the scene at the lake, as the actors
stroll in that direction. The audience will be aware that the action is
moving to the lake, though not consciously. As the action progresses,
the bullfrogs can be brought slowly through the threshold to periph-
eral, conrming the location effectively and involving the audience in
the movement.
Establishing Location Characteristics
Another type of ambient sound occurs when the sound designer needs
to establish the size or character of the location. If a scene takes place
in an abandoned warehouse, the designer may choose to mike the
actors and process that signal through a reverb unit with a long
347
decay time. If the action is occurring in a cave, the same processor set
on a short decay time with a repeating delay (feedback) will create a
believable location. In a recent production of Hecuba in a small theatre
where actors would normally not use microphones, the director chose
to mike the actors and send the signal to monitors in the rear of the
house, pointed toward the back wall. The effect was subtle but suc-
cessful, as the ambience replicated that of the large, outdoor amphi-
theaters in Greece in which the original production was staged. As
with every other effect, subtlety is critical when dealing with this
type of ambient sound. While the desire of the designer may not be
to point the monitors directly at a rear wall, often offsetting ll mon-
itors by degrees can help diffuse the sound, softening it and helping
add to the overall subtlety.
The Second Reading
When the sound designer has read the script, lifted both the obvious
and subtle cues, and jotted down some ideas about music for pre-
show, post-show, intermission, and scene transitions, it is time to
read the script again. On the second read through, look for cues that
were missed the rst time. Consider more subtle approaches. Fine-tune
your impressions and emotional decisions regarding effects and ambi-
ence. This is as important as the rst time you read the script, because
the increased familiarity will give rise to more ideasand often better
thought-out ideasthan the rst read. The second read also gives the
Figure 12.9 The Marantz PMD670 C Professional Solid State Personal Recorder is a
great option for gathering ambient sound and other eld recording. Photo courtesy
of Marantz.
348
sound designer the opportunity to reconsider his or her original ideas
and see whether they hold up.
After reading the script twice, begin searching for the necessary cues
and music and head for the production studio. Assemble your cues
and audition music that you believe will be appropriate. Record
these cues and music and play them for the director, keeping in
mind that the director has the nal say. If the director has a different
opinion about the production, the director is right. The director under-
stands the unique direction of this production better than the sound
designer and is responsible for an overall vision of the production. If
another direction is preferred for certain effects or music, you must
listen carefully to what the director suggests and the direction he or
she gives you, and then fulll that direction. Remember, sound is
never the star; its only one component of many involved in the total
production, and the director has the best overview of all the design
elements.
A valuable option for the sound designer, when possible, is to add the
sound cues as rehearsal progresses, rather than waiting for tech load-
in. Adding these cues as the production develops will help others in the
production in establishing their directions and moods, and it will help
the designer to ensure that the cues selected are working as planned.
Throughout this process the sound designer must be in constant touch
with the director and, in theatre situations, the stage manager. The
director is often extremely busy at this point, coordinating everything
from blocking to props. Often the stage manager or assistant director
can be your best friend at this point; along with being grateful for the
added interest in his position and the increase in responsibility, he can
act as an effective liaison between the designer and the director, help-
ing incorporate minor changes as things settle in.
Selecting a Playback Format
After the sound cues and music have been selected and approved, the
sound designer must select the best format for playback. There are
many in theatre: hard drives, reel-to-reel, DAT, mini-disc, or CD, to
name a few. Some of these formats are quite dated, but if the theatre
you have agreed to work in only has a DAT machine for playback, you
349
had better know its shortcomings. Each format offers advantages and
disadvantages.
DAT became a popular choice a few years ago, due to the digital qual-
ity and good indexing, allowing the board operator to move quickly to
the next cue simply by selecting the correct index number. The prob-
lem with DATs is that they do not play back consistently after you
press the play button. There is often a delay of up to two seconds
after you hit the play button. This could leave the actors in a most
uncomfortable position if a time-critical cue, such as a gunshot, is
needed. If an actor pulls the trigger, the stage manager calls the cue,
the board operator hits the button, and it takes two seconds before the
shot is heard, that will be a very long two seconds for the actor being
shot and waiting to fall. DATs should therefore only be used if they are
the only option in a particular situation, and even then they should
only be used for soft cues, those that are not time critical.
Using a hard drive or CD solves all of these problems, and as a result
CDs have become standard for theatre cues. The sound quality is
excellent, the indexing is intuitive, they are simple to assemble and
record, and they play promptly when you press play.
Another option, though even more dated than the DATs, is a mini-disc
recorder. This format allows the sound designer to record cues on a
digital disc, similar to a CD but smaller, and to re cues efciently,
although mini-disc recorders are becoming rarer due to their low mem-
ory capacity compared to CDs.
Reel-to-reel machines are still a favorite choice for sound cues among
some sound designers and board operators. Cues can be assembled on
Figure 12.10 The Denon C550R Professional CD + CD-R/RW Recorder Combo-Deck is
ideal for both recording cues and playback during a production. Photo courtesy of
Marantz.
350
one reel, or one reel per act, with white leader between each cue. Sound
quality is not digital, but it is excellent nonetheless. Indexing is easy,
because at the end of each cue the board operator simply advances to
the end of the next leader. And they re promptly and consistently.
Sometimes with low-budget productions, the equipment available lim-
its the sound designer. Any combination of formats can be incorpo-
rated, as long as you recognize the limitations of each. These days,
with CD burners and CD players available at such reasonable prices,
many designers and board ops choose not to consider any other option.
Incorporating Room Characteristics
and Existing Equipment
When you have determined your playback format, assess the house.
Examine the existing monitors, amps, the console, and signal-processing
gear. Will the monitors and amps be adequate for the needs of the pro-
duction or will you need a ll monitor or two? Does the console have
enough modules for all microphones, machines, and effects returns?
Do you have all the processing gear required? Do you know what
types of microphones you will need? Are they available or will you
need to rent? Is there a budget for gear rental?
Also assess the house itself. Is the oor carpeted or bare? Is there
acoustic tile on the ceiling? Are the seats soft or hard? These factors
will affect both the level needed to establish your cues and the overall
balance of sound in the house. A reective house will require less level
Figure 12.11 The Marantz CDR420 Portable CD Recorder can record your cues and
effects and play them back. Photo courtesy of Marantz.
351
and will be more of a chore to balance, ensuring the level is even
throughout the house. An absorptive house will require more level,
but will be easier to balance.
Assessing Available Equipment
Assuming the house has monitors and amps, review your cues and
see whether they will sufce. If there is a cluster overhead, make
sure the sound is even; if not, you will need to add ll monitors to
balance the sound. If there are cues that need to come from a specic
location, such as horses pulling up and stopping offstage, stage right,
you may need to add a ll in that location for that cue. Whenever you
add a ll monitor, an amp must be added for that monitor, and a bus
or send from the console must be dedicated to it. If the actors are per-
forming some sound cues live offstage, you may need to give them a
separate microphone for those cues.
Assessing the console is fairly simple. Count the outputs from your
machines, microphones, and processing gear. If the total exceeds the
number of modules, you need to upgrade.
Processing gear is also fairly simple. If you have determined that you
require a reverb unit, a delay unit, and a noise gate, make sure they are
installedandreadytogobefore your rst techrehearsal, because process-
ing will always affect the levels and often the timbre. In any case, in
theatre applications the house monitors will have limiters for speaker
protection. Always check that the threshold is set high, so your cues
are not unexpectedly compressed.
Speaker Considerations
In the event that the house does not have adequate monitors, there are
several considerations. First of all, observe whether the existing mon-
itors will ll all of your needs. Suppose there is a cluster of speakers
hung above the audience in the house. If the only sound cues needed
are pre-show and post-show music, this cluster will probably be ade-
quate, but what will happen if you have a cue of a car pulling up, stage
right. Will the cluster sufce, convincing the audience that the car is
pulling up stage right, when the sound of the car is emanating from
352
above? Not likely. What if the script calls for the sound of helicopters,
and the only speakers in the house are mounted on either side of the
stage? Will these speakers reproduce a believable helicopter?
In these cases, as in many others, a speaker or speakers will need to be
added. Our rst consideration when discussing speakers in a theatre is,
therefore, location. Standard options for locations are in arrays or
clusters overhead lling the house, on either side of the stage also
lling the house, and offstage or hidden in sets, creating the illusion
of the sound emanating from behind the stage. While these are stan-
dard locations, there are other less conventional locations for speakers.
Suppose the script calls for a radio playing onstage during a scene.
How would we place our speaker in this situation? If we placed the
cue in the overhead cluster, it would not be believable coming from
above. If we placed the sound in the speakers on either side of the
stage, it would sound too present, too much in the audiences face.
A speaker placed in the set, behind the radio, would work, but lets
consider another more creative possibility. Suppose we were to wire
an actual radio speaker, in the radio, to an amplier. We could send
the cue through the amp, to the radio speaker onstage. This level of
realism would produce the most believable cue.
There are other unconventional choices in speaker placement that are
very effective. In the Establishing Location Characteristics section
earlier in this chapter, we discussed a production of Hecuba in which
the director chose to mike the actors and send the signal to speakers
placed in the rear of the house, facing the back wall. This may not have
been a conventional approach, but it certainly was a success.
As with every aspect of audio, consider the environment and act accord-
ingly. Settling for the conventional may not always be the correct choice
to elicit the desired emotion or location. Consider alternative speaker
locations to create more believable cues.
Choosing and Placing Microphones
Once you are comfortable with all the other equipment considerations,
you can start to place your microphones if the size of the theatre war-
rants it. The most popular and commonly used microphones in theatre
353
applications are wireless lavalieres, although shotguns and pressure
zone microphones may also be considered.
If one or more of the principals needs his or her voice amplied, wire-
less lavalieres can be mounted in the costume to pick up the principals
voice and little else (see Figures 12.12 through 12.14).
While there is no formal etiquette regarding the application of lava-
lieres, if an actor is shy about having you reach into his costume to
mount the microphone, be sure to warn him before you do. Be wary
when placing lavalieres; they will not operate properly if they are faced
too far away from the sound source or if they are placed directly upon
the actors larynx. Another potential problem with lavalieres is the dif-
ference between actors who are picked up by lavalieres and those who
are not. Actors wearing lavalieres will sound both louder and more
Figure 12.12 Lavalieres, like this classic Electro-Voice RE85, are industry-standard in
theatre applications. Photo courtesy of Stan Coutant.
354
Figure 12.14 The MKE102 Omni Lavaliere Microphone is ideal for speech. Photo cour-
tesy of Sennheiser.
Figure 12.13 Lavalieres offer a variety of mounting options, making them easy to hide
in costumes or wigs. Photo courtesy of Sennheiser.
355
present than those without; as such, if any one principal wears a lava-
liere, all principals must wear lavalieres.
When the stage needs to be blanketed with microphones, such as when
a chorus needs to be amplied or when the director shuns the use of
lavalieres, shotguns can be hung from the pipes above the stage or
mounted offstage pointed toward center stage. Pressure zone micro-
phones can also be useful in this situation, mounted on the oor, as
long as they can be located where none of the actors will step on them.
Often a combination of two or three pressure zone microphones
mounted downstage along with two or three shotguns hung from
above can effectively blanket the stage (see Figure 12.15).
Cue Sheets
When the cues are in place and the microphones have been mounted
and tested, you are ready for a sound tech rehearsal. As you go through
your cues, mark initial levels and settings on your preliminary cue
sheet. Try sounds at different levels and listen from different sections
of the house to ensure that the levels work for you and that the house
is balanced throughout. When these levels are established, you have an
interim cue sheet. Have your board operator work off the interim cue
sheet during the remaining rehearsals as you ne-tune the levels,
effects, and equalization. Always be sensitive to the needs of the
Floor Floor
Downstage
Floor
Shotgun Shotgun Shotgun Shotgun
Figure 12.15 A stage can be blanketed effectively with a combination of shotguns
hung from the pipes for the upstage area and boundary or pressure zone microphones
taped to the oor for the downstage area.
356
director; check frequently that the cues are as effective for the director
as they are for you. As you near opening night, your ne-tuned interim
cue sheet will become your nal cue sheet. Make extra copies of this
sheet; store one in a safe place, give one to the stage manager, and give
one to the board operator. This cue sheet should be ne for the run of
the show, although you may need to tweak it a bit as the production
runs.
As with all aspects in audio, the top priority should be using your ears.
Train yourself to understand what you are hearing, and learn how to
implement that information with your own style. In any audio situa-
tion, the best advice anyone can give you is to use your ears, trust your
ears, pay attention, and do your best to anticipate situations. This
combination, along with a pile of luck and a dollop of talent, will vir-
tually ensure your success.
Additional Reading
2004.
Anderton, Craig. The Digital Delay Handbook, rev. ed. Woodstock,
NY: Beekman Publishers, 1990.
Arnott, Peter D. Public and Performance in the Greek Theatre.
London: Routledge, 2005.
Aston, Elaine and George Savona. Theatre as a Sign-System: A Semi-
otics of Text and Performance. London: Routledge, 1991.
Bartlett, Bruce and Jenny Bartlett. Practical Recording Techniques.
Carmel, IN: Sams Publishing, 1992.
357
McGraw-Hill, 1988.
Camras, Marvin. Magnetic Recording Handbook. New York: Van
Nostrand Reinhold, 1988.
Carlin, Sr., Dan. Music in Film and Video Productions. Boston: Focal
Press, 1991.
Collison, David. Stage Sound. Hollywood: Quite Specic Media
Group, 1982.
Deutsch, Diana. The Psychology of Music, 2nd ed. Orlando, FL:
Academic Press, 1998.
Springer, 2005.
Esslin, Martin. The Field Of Drama. New York: Methuen, 1987.
Giannetti, Louis. Understanding Movies, 11th ed. Englewood Cliffs:
Gillette, J. Michael. Theatrical Design and Production, 5th ed. New
York: McGraw-Hill, 2004.
Hagen, Earle. Scoring for Films. Sherman Oaks, CA: Alfred, 1989.
Huber, David Miles. Audio Production Techniques for Video.
Burlington, MA: Butterworth-Heinemann, 1992.
358
CA: Alfred, 1989.
Publications, 1993.
Nelson, Mico. The Cutting Edge of Audio Production and Audio
Post-Production: Theory, Equipment, and Techniques. Boston: Focal
Press, 1995.
Press, 1989.
ing, 1991.
Taplin, Oliver. Greek Tragedy in Action, 2nd ed. Oxford: Routledge, 2002.
Taplin, Oliver. The Stagecraft of Aeschylus: The Dramatic Use of
Exits and Entrances in Greek Tragedy. Oxford: Oxford University
Press, 2001.
359
Walne, Graham. Sound for the Theatre. London: A&C Black, 1990.
Press, 2000.
Sams, 1989.
360
Appendix A
BIBLIOGRAPHY
2004.
Altman, Rick, ed. Sound Theory/Sound Practice. New York:
Routledge, 1992.
Anderton, Craig. The Digital Delay Handbook, rev. ed. London:
Music Sales Corp., 1985.
Arnott, Peter D. Public and Performance in the Greek Theatre.
London: Routledge, 2005.
Aston, Elaine and George Savona. Theatre as a Sign-System: A Semi-
otics of Text and Performance. London: Routledge, 1991.
Backus, John. The Acoustical Foundations of Music, 2nd ed. New
York: W. W. Norton, 1977.
361
McGraw-Hill, 1988.
Borwick, John. Microphones: Technology and Technique. London:
Focal Press, 1990. Excellent guide to microphone usage.
Burroughs, Lou. Microphones: Design and Application. Plainview
NY: Sagamore, 1974.
Campbell, Murray and Clive Greated. The Musicians Guide to
Acoustics. London: Oxford University Press, 2001.
Springer, 2001.
Carlin, Sr., Dan. Music in Film and Video Productions. Boston: Focal
Press, 1991.
Clifford, Martin. Microphones, 3rd ed. Blue Ridge Summit, PA: Tab,
1986.
Collison, David. Stage Sound. Hollywood: Quite Specic Media
Group, 1982.
Deutsch, Diana. The Psychology of Music, 2nd ed. Orlando, FL:
Academic Press, 1998.
Springer, 2005.
Eisenberg, Evan. The Recording Angel: Explorations in Phonography.
New York: McGrawHill, 1986.
Esslin, Martin. The Field Of Drama. New York: Methuen, 1987.
Press, 1993.
362
Giannetti, Louis. Understanding Movies, 11th ed. Englewood Cliffs,
NJ: Prentice-Hall, 2007.
Gillette, J. Michael. Theatrical Design and Production, 5th ed. New
York: McGraw-Hill, 2004.
Hagen, Earle. Scoring for Films. Sherman Oaks, CA: Alfred, 1989.
Hausman, Carl, Philip Benoit, Frank Messere, and Lewis B. ODon-
nell. Modern Radio Production: Production, Programming, and Per-
formance, 6th ed. Belmont, CA: Wadsworth, 2003.
Huber, David Miles. Audio Production Techniques for Video.
Burlington, MA: Butterworth-Heinemann, 1992.
CA: Alfred, 1989.
Hutchins, Carleen Maley. The Physics of Music. San Francisco: W. H.
Freeman, 1978.
Jones, Steve. Rock Formation: Music, Technology, and Mass Com-
munication. Newbury Park, CA: Sage, 1992.
Focal Press, 2002.
Keene, Sherman. Practical Techniques for the Recording Engineer,
3rd ed. Torrance, CA: Mix Books, 1989.
Keith, Michael. Radio Production: The Art and Science. Boston: Focal
Press, 1990.
Appendi x A
363
Lehrman, Paul. The Insider Audio Bathroom Reader. Thomson
Course Technology PTR, 2006.
Publications, 1993.
McLeish, Robert. The Technique of Radio Production: A Manual for
Broadcasters, 2nd ed. Boston: Focal Press, 1988.
Martin, George. All You Need Is Ears: The Inside Personal Story of
the Genius Who Created the Beatles. New York: St. Martins Grifn,
1994.
Miller, T. Music in Advertising. New York: Amsco, 1985.
Nelson, Mico. The Cutting Edge of Audio Production and Audio Post-
Production: Theory, Equipment, and Techniques. Boston: Focal
Press, 1995.
Olson, Harry. Music, Physics and Engineering. NewYork: Dover, 1967.
Press, 1989.
Pierce, John. The Science of Musical Sound, Revised ed. New York:
W. H. Freeman & Company, 1992.
ing, 1991.
Recording Industry Sourcebook. Los Angeles: Recording Industry
Sourcebook, Annual.
364
Robair, Gino. Making the Ultimate Demo. Boston: ArtistPro, 2000.
Rossing, Thomas. The Science of Sound, 3rd ed. Reading, MA:
Addison-Wesley, 2001.
Rumsey, Francis. Stereo Sound for Television. London: Focal Press,
1989.
Taplin, Oliver. Greek Tragedy in Action 2nd ed. Oxford: Routledge,
2002.
Taplin, Oliver. The Stagecraft of Aeschylus: The Dramatic Use of Exits
and Entrances in Greek Tragedy. Oxford: Oxford University Press, 2001.
Walne, Graham. Sound for the Theatre. London: A&C Black, 1990.
Press, 2000.
Winckel, Fritz. Music, Sound and Sensation: A Modern Exposition.
New York: Dover, 1967.
Woodward, Walt. An Insiders Guide to Advertising Music. New
York: Art Direction, 1982.
Sams, 1989.
Appendi x A
365
Appendix B
GLOSSARY
absorption coefcient. A ratio that compares the amount of energy that
is absorbed to the amount of energy that is reected by a given surface.
ADSR. Attack, decay, sustain, releasethe components of the enve-
lope, or duration of a sound event.
amplitude. The quantitative size of a sound wave, which creates our
perception of loudness.
antinodes. Fixed high-pressure locations within an enclosure or room
that, along with nodes, form a stationary waveform called a standing
wave.
attack. Howa sound event begins, a component of envelope, or duration.
audible bandwidth. The range of the frequency spectrum that humans
can hear, approximately 20 Hz to 20 kHz.
auto panner. A signal-processing device that varies the output signal
between the left and right sides of the stereo bus.
aux send master. A master trim that controls the overall level output
by the summing network of a specic auxiliary send.
aux sends. See auxiliary sends.
auxiliary sends. Additional feeds fromeach module that allowfor simul-
taneous multiple mixes. Commonly referred to as aux sends or simply
sends, they are commonly used for monitor or cue mixes and effects
sends.
367
bandpass lter. A signal processor that eliminates all frequency com-
ponents of a sound above and below selected frequencies.
bias head. The rst head in the tape path, the bias head, or erase head,
erases tape by returning magnetic elds to a neutral or random posi-
tion. This also places the molecules in an excited state, providing supe-
rior signal-to-noise ratio when recording.
bidirectional. A polar pattern for microphones in which they are most
sensitive directly in front of and behind a microphone and least sensi-
tive to the sides.
binaural hearing. Hearing through two ears, which allows us to per-
ceive and localize sounds by hearing in three dimensions. Binaural
hearing has three componentsinteraural intensity, interaural arrival
time, and physiology.
bookshelf speakers. Monitors used in the studio that mimic common
household speakers, used by engineers while mixing to ensure that
their mix will be effective in the consumer market.
boundary microphones. Flat, metal plate microphones are usually
attached to stiff sound-reecting surfaces, such as walls, oors, or desk-
tops. Also known as oor mics or pressure zone microphones, they are
commonly used in theatre or for ambient sound gathering.
bus. A send of any type that contains a summing network on a console.
The most common types are the multitrack busses, aux sends, and the
stereo bus.
bus faders. A master trim that controls the overall output of a specic
bus.
bussing. The process of sending a signal into a bus; usually associated
with multitrack bussing.
cannon plug. See XLR.
capacitance. The ability of a condenser to store a charge and release it
at predetermined intervals. This is the electronic theory behind con-
denser microphones.
368
capstan. The capstan, along with the pinch roller, initiates and main-
tains the motion of the tape transport during play and record func-
tions, controlling the tape speed.
capstan motor. The motor that controls the capstan, and therefore the
tape speed.
carbon microphones. The oldest of microphones, carbon microphones
were used in telephones.
cardioid. A heart-shaped polar pattern for microphones that is most
sensitive at the front, least sensitive in the rear, and gradually less sen-
sitive down the sides as one moves toward the rear.
channel assignment switching matrix. A combination of push buttons
and pan pots that determines where a signal in the channel fader will
be routed, usually onto a track of the multitrack recorder or into the
stereo bus. Also known as multitrack busses.
channel faders. A separate gain control for each I/O module of the
console that commonly leads to the multitrack recorder, the stereo
bus, or the speakers in the theatre.
chorusing. A combination of pitch shifting and short delays used to
make an individual voice or other input signal sound doubled, as if
more that one instrument is present.
clip-on microphones. Microphones typically used where an invisible
microphone is needed. Also known as lavalieres; applications include
television newscasters and body microphones on theatrical performers.
clipping. See distortion.
close-miking. When a microphone is placed close to a sound source,
the acoustic phenomena caused by the surrounding environment have
little or no effect on the signal captured. Close-miked signals lack the
natural ambience of an environment, and as a result they can sound
unnatural and one-dimensional. To a great extent the development
of signal-processing equipment is a result of close-miking technique
and multitrack recording, something of an effort to recreate an
environment.
Appendi x B
369
complex wave. A wave containing harmonics and overtones, which can
be viewed through Fourier analysis as a combination of sine waves.
compression. When molecules move toward each other within a medium
after the force of a sound event has momentarily displaced them.
compressor. A device that decreases the dynamic range of program
material.
condenser microphones. Microphones that work on the principle of
variable capacitance, generally accepted as the highest quality and
most expensive microphone type.
console. The heart of the control room, the device through which all
signals pass and are routed.
constructive interference. Constructive interference occurs when two
sound waves combine and the result, referred to as the sum wave, is
an increase in amplitude of the sound waves.
contact microphone. A small, clip-on microphone, somewhere in
design between a lavaliere and a guitar pickup.
control room. The heart of the production facility. Along with housing
the console, tape recorders, signal-processing gear, and the monitor
system, it is the location where signal routing is determined.
cross talk. Information from a track playing back on an adjacent head.
crossover frequency. The dividing point between frequency bands, deter-
mined by the frequency dividing network, or crossover, in a speaker.
crossovers. A frequency dividing network, directing frequencies to
specic speakers within a monitor.
cue mix. A mix used by musicians to monitor themselves, other musi-
cians, and/or sounds already on tape; often a separate mix.
cue sheet. In recording, a running list of the beginning and end of each
take, or attempt to record a piece, including spaces for start and end
times, title, take number, code (such as CT for complete take, FS for
false start, or PB for playback), and comments. In theatre, a numbered
list of all cues used in a production.
370
cycle. One oscillation of a waveform, comprised of one complete com-
pression and one complete rarefaction. One cycle per second equals
one Hertz.
de-esser. A combination effect made up of an equalizer and a com-
pressor.
decay. The decrease in volume of a sound after the stimulus has been
removed; a component of envelope.
decibel. A ratio describing the difference between one power and
another or one sound pressure level and another.
delay. A single, discreet reection of a sound; the ears rst clue about
the size and shape of a room.
destructive interference. Occurs when two sound waves combine and
the result is a decrease in the amplitude of the sound wave.
DI. See direct box.
diaphragm. A thin, exible membrane under tension in microphones,
similar to a drum skin or the paper in a kazoo, which vibrates in response
to the changes in atmospheric pressure caused by the compression and
rarefaction of molecules of a sound wave.
diffraction. The property of low-frequency sound waves that allows
them to bend around corners more readily than high frequencies.
diffusion. The spreading out of a sound. Due to the physical properties
of lower frequencies having larger wavelengths, low frequencies dif-
fuse while high frequencies are directional.
digital-to-analog conversion. The process of converting a signal from
digital to analog.
direct box. A step-down transformer used to change line-level, high
impedance signals to mic-level low impedance signals.
directionality. See polar pattern.
distortion. The equivalent in equipment of the threshold of pain in
humans; unpleasant or unwanted sound caused by excessive amplitude.
Appendi x B
371
dropouts. In editing, when the level literally drops out momentarily at
the edit point. In tape, when the magnetic particles held into place by
the binder fall or drop off of the tape, which leaves an area that cannot
hold any audio information.
dub. (n.) A tape copy. (v.) To make a tape copy.
dump mode. Amethod used in editing to eliminate large sections of tape.
duration. Or envelope. The volume shape of a sound over time, or the
lengths of time of the components of any sound; how much time passes
as the sound begins, continues, and ends.
dynamic microphones. Microphones that work on the principle of
inductance, in which electric current is created, or induced, by a wire
or any conductor as it senses movement within a magnetic eld. There
are two types of dynamic microphones, moving coil and ribbon.
dynamic range. The span of volume that the human ear can perceive,
ranging from the threshold of hearingthe softest sound the ear can
hear or the minimum energy required for the average person to expe-
rience the sensation of soundto the threshold of painthe point at
which sound intensity causes pain in the average listener.
early reection. The rst few reections to arrive at the listeners ears
just after the direct sound, which can often be discerned as discrete
from the reverberation.
echo. A discrete individual reection, indicating a large space.
edit mode. See dump mode.
editing. The removal of unwanted noise and reordering of recorded
material.
effect loop. Used when a signal needs to be split into a processed and
unprocessed signal. The processor is inserted at the end of an auxiliary
send, allowing signals from any I/O module to be processed. The output
of the processor is recombined with the dry signal at the master fader by
using either an effects return or an available line input.
372
envelope. Or duration. The volume shape of a sound over time, or the
lengths of time of the components of any sound; how much time passes
as the sound begins, continues, and ends.
equalization. Frequency-selective amplication.
erase head. See bias head.
expander. Anamplier whoseoutput level decreases byapreprogrammed
amount when an input signal falls below a user-dened threshold.
expansion ratio. In an expander or noise gate, a comparison between the
input and output signals once the input signal falls below the threshold.
external triggering. Dropping the level of a track being modied in an
expander or gate when the keying or triggering signal is not occurring.
fader. A sliding control over a potentiometer or voltage-controlled
amplier; commonly found in audio consoles.
anger. A signal-processing device that splits an input signal and then
recombines it with a dynamically changing phase relationship, causing
a sweeping sound.
oor microphones. See boundary microphones.
Fourier analysis. A graph that plots amplitude versus frequency of the
component frequencies. Fourier analysis is a concept in which complex
waveforms are viewed as a combination of many sine waves.
frequencies. The measurement of the speed at which a periodic or
repeating waveform oscillates. Responsible for the pitch of a sound.
frequency dividing network. See crossovers.
full normal. Connected to the input side of a patch bay, full normals
disconnect the bridging wire when a patch cord is inserted. This is nec-
essary because input signals cannot be combined without a summing
network.
full track. A mono tape format in which the entire tape is used as one
track.
Appendi x B
373
fundamental frequency. The frequency within a complex wave that is
most responsible for the sounds pitch. Usually the lowest and loudest
frequency in a complex waveform.
glitch. A pop or an electronic jolt. In razorblade editing, usually the
result of a blade containing a magnetic charge, or an old blade that
results in a jagged cut.
graphic equalizer. A processor that changes the harmonic content of a
sound, giving a graphic representation of the change.
guard band. The space between tracks and on the edges of magnetic
tape. Reduces crosstalk and edge damage.
half normal. The patch point for the output side of equipment, allows
the signal to ow through the bridging wire whether or not a patch
cord is inserted.
harmonic content. Whole-number multiples of a fundamental fre-
quency. The timbre of a sound is a direct result of its harmonic content.
harmonics. Simple waves of varying frequencies and amplitudes, each
representing one component of a complex waveform.
harmonizer. Extreme pitch shifters that are also used where pitch shift-
ing will create a very unusual and mechanical sound.
head lifters. A component of tape machines that engages during fast
wind, moving the tape back off the heads.
Hertz. A measure of frequency; one Hertz (Hz) equals one cycle per
second.
high impedance. A line-level signal, generally between 30 dB and
0 dB.
high-pass lter. A lter that affects only low frequencies, allowing high
frequencies to pass unaffected.
hyper-cardioid. A polar pattern for microphones used to describe the
directionality of shotguns.
374
I/O module. See input/output module.
icon. In the language of semiotics, a sign that is exactly what it appears
to be.
index. In the language of semiotics, a sign that points to something else.
inductance. The theory by which dynamic microphones work, in which
electric current is created, or induced, by a wire or any conductor as it
senses movement within a magnetic eld.
input mode. Aconsole mode inwhichmicrophone pre-amps feedchannel
faders and line pre-amps feed monitors. Used commonly for recording.
input/output module. A console module capable of handling both
input and output signals simultaneously and discretely.
interaural arrival time. Perceiving the location of a sound by the dif-
ference in time of arrival at each ear.
interaural intensity. Perceiving the location of a sound by the differ-
ence in loudness at arrival at each ear.
isolation. Separating sounds from each other in different rooms, or
within one room.
keying. Triggering a noise gate or expander to allow signal to pass
unaffected.
lavalieres. Clip-on microphones commonly used in theatre and television.
leader tape. White or yellow tape that cannot be recorded upon; used
to mark locations within a reel of audio tape.
leveling amplier. See compressor.
limiter. A device that decreases the dynamic range of program material
by a greater than 10:1 compression ratio.
line input. A console input designed to accommodate line-level signals.
line level. The typical level of signals from electronic instruments and
recorder outputs, 30 dB to 0 dB.
Appendi x B
375
line pre-amps. A passive attenuator designed to boost a line-level signal
to the consoles standard operating level.
line trim. A potentiometer that controls the level of a line input.
location. One of the ve perceptions of sound establishing distance and
direction.
longitudinal compression waves. When waves propagate in the same
direction as the displacement of the molecules in the medium. Sound
waves are always longitudinal compression waves.
loop insertion. When a processor is inserted at the end of an auxiliary
send, allowing signals from any I/O module to be processed. The out-
put of the processor is recombined with the dry signal at the master
fader by using either an effects return or an available line input.
loudness. One of the ve perceptions of sound; the perception of
amplitude.
low impedance. A mic-level signal, generally between 65 dB and
30 dB.
low-pass lters. A lter that affects only high frequencies, allowing low
frequencies to pass unaffected.
magnetic tape. Commonly used with analog recording devices and
sometimes with digital recorders. Stores the audio information con-
verted at the record heads to magnetic information.
masking. One sound blocking another through loudness, pitch, or
location.
master. A passive attenuator or voltage controlled amplier that con-
trols the output of any bus.
master fader. A fader that controls the overall output of the console.
master section. The part of the console that contains the master fader,
monitor source selection switch, monitor pot, aux send masters, and
aux returns, among other specialized features, depending on the
console.
376
medium. A space with molecules in sufcient quantities for sound
waves to propagate, such as air or water. One of the three minimum
requirements for a sound event.
mic level. The typical level of signals from microphones, 65 dB to
30 dB.
mic pre-amps. An amplier connected to the mic trim, a passive
attenuator, designed to boost a mic-level signal to the consoles stan-
dard operating level.
mic trim. A potentiometer that controls the output of a mic preamp.
microphone. A transducer that converts acoustic energy to electricity.
mid-range driver. A speaker that is responsible for middle frequencies.
mixing console. A device responsible for the processing and routing of
many signals; the center of any production room. The console is the
link between all devices in the facility.
monitor. (n.) Aspeaker or groupof speakers inone cabinet. (v.) Tolisten.
monitor fader. A fader that feeds the monitor bus. Depending on the
consoles mode, this bus can either feed the stereo bus or can be routed
elsewhere.
monitor modes. The various modes of a console, including input, mix,
and mixdown/overdub.
monitor pot. A passive attenuator that adjusts the control room
monitor volume.
monitor source selector switch. Part of the consoles master section;
allows the engineer to select which of the busses or machines will be
monitored.
mono. A format requiring only one track.
moving coil microphones. Microphones, such as dynamic micro-
phones, that work on the principle of inductance.
mult. A patch bay option that allows one output signal to be sent to
many locations.
Appendi x B
377
multitrack. A recording device that allows recording on more than one
track, either simultaneously or subsequently.
multitrack busses. Sends used to access individual tracks on a multi-
track recorder; also used in mixing as additional aux sends.
nodes. Fixed low-pressure locations within an enclosure, or room, that,
alongwithanti-nodes, forma stationary waveformcalledastandingwave.
noise. Undesirable sound.
noise oor. The ambient noise present in all devices.
noise gate. An amplier whose output level decreases by a greater than
10:1 expansion ratio when an input signal falls below a user-dened
threshold.
non-harmonically related overtones. Overtones that are not whole-
number multiples of the fundamental frequency. Non-harmonic over-
tones are responsible for white noise.
nondirectional. A description of microphones with an omnidirectional
polar pattern, equally sensitive in all directions.
normalling. The process of returning all console controls and other
equipment to their null points.
normals. See full normal.
observer. One of the three minimum requirements for a sound event to
occur.
octaves. A tonal relationship between sounds with a 2:1 frequency
ratio.
omnidirectional. A polar pattern for microphones in which they are
equally sensitive in all directions.
outboard gear. Signal-processing equipment that is not located within
the console.
overdub. Adding new tracks to existing tracks on a multitrack recorder.
378
overtones. Whole-number multiples of a fundamental frequency. The
timbre of a sound is a direct result of its overtones, also known as its
harmonic content.
pan pot. See panoramic potentiometer.
panning. The process of placing a sound from left to right in the stereo
bus. Also used in multitrack bussing to place a sound between two
tracks.
panoramic potentiometer. A dual passive attenuator that continuously
varies a single signal between two or more output busses. Used when
placing a sound from left to right in the stereo bus and when including
a signal in two tracks on a multitrack.
parametric equalizers. A frequency-selective amplier that allows con-
trol over the bandwidth of the frequencies.
passive attenuators. Resistors controlled by a potentiometer. When set
for maximum level, generally turned all the way to the right, the trim
control is adding minimum resistance to a circuit, allowing the maxi-
mum signal to pass. When set to the left, the trim control is reducing
the level of the signal.
patch bay. Access points that interrupt the normal signal ow through
a module and give access to the inputs and outputs of every device in a
production facility. A patch bay will provide access to signal ow
between normally interconnected devices. The purpose of the patch
bay is to allow for the rerouting of the normal signal ow and the
insertion of additional devices.
peak meter. Used to monitor the amount of electricity passing through
equipment. Specially calibrated volt meters are inserted in the signal
ow within the recorder or console. Peak meters are calibrated to
respond faster than the human ear does, showing transients.
peaking equalizer. A frequency-selective amplier featuring a xed
bandwidth.
period. The inverse of frequency, seconds per cycle.
Appendi x B
379
peripheral. The second zone in semiotics, when the senses are aware of
a sign but not focused on it.
phantom power. The electrical charge (+48vdc) held by the capacitor
within the condenser microphones diaphragm supplied directly from
the audio console. Phantom power has no effect on the audio signal
passing through.
phase. The phase relationship between two sound waves is determined
by a comparison between the point of compression or rarefaction at
which each waveform is in its period when they meet.
phase shifters. A signal processor that splits an input signal and then
recombines it with a dynamically changing phase relationship, causing
a sweeping or swishing sound.
phase shifting network. Creates a cardioid polar pattern in moving coil
microphones by causing sound waves approaching the microphone
from the rear to impact on both sides of the diaphragm. These sound
waves will have their phase reversed on either side of the diaphragm
and will therefore be greatly reduced in intensity.
pickup pattern. See polar pattern.
pinch roller. Initiates and maintains the motion of the tape during play
and record functions, controlling the tape speed, along with the cap-
stan. When play is pressed, the pinch roller presses against the capstan,
causing the tape that is in between the capstan and the pinch roller to
be pulled along.
pitch. One of the ve perceptions of sound; our perception of frequency.
plate reverb. A form of classic analog signal processing where a thin,
metal sheet is mounted under tension in a box. When signal is fed to
the plate, the waves travel through the plate, bouncing back from the
sides, simulating the way that the waves bounce through a room. By
mounting a pickup on the far end of the plate, the resulting signal will
be analogous to the same wave traveling through a room.
playback head. A transducer found in recording devices that converts
magnetic information previously stored on tape into electricity.
380
polar pattern. A map of a microphones directional sensitivity, graph-
ically depicting the way a microphone will respond depending upon
the position of the sound source relative to the microphones position.
post-fader send. A send into an auxiliary summing network whose
level will be unaffected by movement of the corresponding fader.
potentiometer. See passive attenuators.
pre-amp. See mic pre-amps or line pre-amps.
pre-delay. Used in signal processing as an emulation of early reections.
print through. When audio information passes through the backing of
the tape, creating a faint imprint on the next ring.
production rooms. A space designed to capture, route, process, and
store audio information.
pulse waves. Complex waves that are very rich in harmonics, with all
harmonics equal in amplitude to the fundamental frequency.
Q control. In parametric equalization, the control that modies the
bandwidth of frequencies affected.
quarter track. A consumer format featuring quarter-inch tape with
four tracks, two in each direction.
range. The amount of level reduction, in decibels, that will be applied
to the output signal once the input signal falls below the threshold on
an expander.
rarefaction. When molecules move away from each other within a
medium after the force of a sound event has momentarily displaced
them.
record head. A transducer that converts electricity to magnetic energy,
to be stored on tape or disc.
redundancy. A feature of I/O modules in consoles where if one module
is understood, they can all be understood.
reections. See early reection.
Appendi x B
381
release. A component of envelope, release is the eventual cessation of a
sound.
reverberation. A series of random, indistinguishable reections, grow-
ing denser and diminishing over time.
ribbon microphones. A type of dynamic microphone with a thin, metal-
lic ribbon for a diaphragm.
rocking. A technique used in editing to move the tape back and forth
across the heads.
route. To send or bus a signal.
RT-60. Reverb time minus 60, or the time required for reverberation
to reduce by 60 dB.
saturation. Excessive level on tape.
sawtooth waves. A complex wave that contains all harmonics at rela-
tively high amplitudes relative to the fundamental frequency.
semiotics. A system of codifying signs received from various media.
send. A bus; a method of routing signal.
send masters. A passive attenuator that controls the overall output of
a send.
shelving equalizer. A frequency-selective amplier that affects all
frequencies above or below a user-selected or preset frequency by an
equal amount.
shotgun microphones. Microphones withtight polar patterns, commonly
used in theatre, lm, sporting events, and surveillance.
signal ow. The chosen path for a signal to follow.
signal processing. Effecting a signal.
signal-to-noise ratio. The relationship between desirable signal and unde-
sirable signal.
sine wave. A simple wave or pure tone, devoid of harmonics.
382
slope. A choice between a linear or exponential (logarithmic) rate
applied in an expander; a choice between expansion proceeding at a
constant or accelerating pace.
speaker. A transducer that converts electricity to acoustic energy.
splicing. The removal of unwanted noise and reordering of recorded
material.
square waves. A complex wave that contains only odd harmonics at
high amplitudes in relation to the fundamental frequency.
standing wave. When some of the energy of a sound wave travels back
along the same path as the direct energy, which happens between parallel
surfaces, it will interfere with the incoming sound wave, causing increases
and decreases in amplitude, depending on the phase relationship between
the two waves. Frequencies with wavelengths that are whole-number
multiples or subdivisions of the distance between the surfaces off which
they are reecting will interfere with each other in such a way as to cause
a standing wave.
stereo. A format that requires two tracks, typically bussed to a left
speaker and a right speaker.
stereo bus. The sends from the console that feed the mix machines and
studio monitors.
stereo master. A passive attenuator that controls the output of the
stereo bus.
sub-faders. Also known as bus faders, they alter the level of a group of
faders as they feed the stereo bus, but will not alter the level of a signal
routed to a post-fader send.
sub-masters. Along with altering the level of a group of faders as they
feed the stereo bus, they alter the level of a signal routed to a post-fader
send.
sub-woofer. A speaker that is responsible for reproducing the lowest
frequencies.
subliminal. The third zone of reception in semiotics, in which the audi-
ence is cerebrally unaware of a sign that is subconsciously understood.
Appendi x B
383
summing network. Located in every bus, individual signals are reduced
by 3 dB, then combined. Since 3 dB represents an approximate halving
of the signal, when they are combined the end result is an output level
consistent with the original input levels.
super-cardioid. A tight polar pattern, one of the patterns used in shot-
gun microphones.
sustain. Acomponent of envelope, sustain is the continuation of a sound.
sweetening. See overdub.
symbol. In semiotics, a symbol is a sign that represents something
totally different than what it appears to be.
sync mode. A recorder mode in which some information is being
played back while other information is being recorded, both tasks per-
formed by the record head.
tails out. A method of storing tape that protects the audio information
and reduces print through.
take sheet. See cue sheet.
tape release form. A form commonly used in studios before a tape will
be released. Used for tracking purposes.
tension idler. A multipurpose tape guide that acts as an on/off switch.
threshold. In a compressor or expander, the user-selected level mea-
sured in volts or decibels at which a change in level will begin.
threshold of hearing. The softest sound we can hear; the bottom of the
dynamic range in humans.
threshold of pain. The loudest sound we can hear without pain; the top
of the dynamic range in humans.
timbre. One of the ve perceptions of sound, timbre is our perception
of harmonic content.
track. A memory location on tape.
track sheet. A way of cataloguing the track locations on which instru-
ments are recorded.
384
transducer. A device that converts one form of energy to another.
transients. Instantaneous peaks in amplitude.
transverse wave. A wave that propagates perpendicularly to the ori-
ginal displacement.
triangle waves. A complex wave that contains only odd harmonics at
very low amplitudes relative to the fundamental frequency.
trim pot. See mic trim, line trim.
tweeter. A speaker that is responsible only for reproducing high
frequencies.
two mix. See stereo bus.
ultra-cardioid. A tight polar pattern, one of the patterns used in shot-
gun microphones.
unidirectional. A polar pattern for microphones that are not equally
sensitive in all directions.
volume control. See monitor pot.
V.U. meter. Used to monitor the amounts of electricity being passed
through equipment. Specially calibrated volt meters are inserted in the
signal ow within the recorder or console. V.U. meters are calibrated
to respond in a similar fashion to the human ear.
wavelength. The physical distance required for one cycle of a sound
wave to complete.
white noise. Any and all frequencies occurring randomly.
woofer. The speaker that is responsible for reproducing low frequencies.
work order. Studio paperwork that contains pertinent information,
including the names of the client, producer, and artist; the time the
session begins and ends; spaces for purchase order numbers and job
numbers; setup information for the assistant, including the instrumen-
tation and the types of mics requested by the engineer; and billing
information.
Appendi x B
385
wow. A clearly audible and sudden dip in the frequency of the
recorded material, usually the result of tape stretching.
XLR. A three-pin, barrel-shaped connector commonly used for micro-
phones and balanced lines.
386
Index
A
absorbed sound waves, 3839
absorption coefcients, 39
acoustic energy, 6061
acoustic guitars, microphones for, 156
acoustics. See also performance areas
of control room, 6869
synthesizers/drum machines and, 230
ADATs, 9192
aesthetics, 68
application of, 812
audio technology and, 1315
dened, 89
of mixing, 291294
of preamps, 67, 105
AIFF format, 194
air, density of, 2223
AKG
414 microphones, 142143
D12 microphones, 134
D12E microphones, 135
alcohol use, 312
Alcorn McBride Digital Binloop, 198
Alesis ADAT, 91
algorithms for analog-to-digital conversion
(A-D), 189
aliasing, 192193
Allen, Woody, 330
AM radio signals, 4344
ambience in sound design, 346348
ampliers, 180184. See also preamps
coupled devices, 181182
distortion in, 183184
frequency response of, 183
speakers and, 179180
wattage ratings, 182
amplitude
and moving coil speakers, 165166
perception of, 1314, 2427
standing waves and, 48
AMS RMX-16 digital reverb, 194
analog signals, 235
analog-to-digital (A/D) conversions, 56,
59, 189
deterioration of signal and, 236
antenna for wireless microphones, 148149
anti-aliasing lters, 192
anticipation
silence for, 346
sound design for, 336338
antinodes, 48
Ashley Powerex 6250 Integrated Amplier,
181182
assistant engineers, 298, 306310
anticipating situations, 324
best assistant, tips for, 322326
blame, taking, 326
condence, projection of, 324325
menial jobs, attitude about, 325326
people skills, 325
take sheets, 316318
track sheets, 318320
trouble reports, 320322
work orders, 313315
assistant managers, 302
attack, decay, sustain, and release (ADSR),
3536
compressors and, 239240
audible bandwidth, 30
audio consoles. See consoles
Audio-Technica condenser microphones,
142
Auratone sound cubes, 174
auto panners, 257
auxiliary sends (aux sends). See sends
Avantone sound cubes, 174
Avid software, 200
B
bandpass lters, 245, 248
bandwidth and sampling rates, 190191
bass drum
microphones, 134
mixing, 288
387
bass guitars, compressors for, 238239
The Beatles, 230
believability of cues, 331333
Bell, Alexander Graham, 62, 164
Bell Laboratories
equalizers from, 245
Nyquist Theorem, 191
Bergman, Ingmar, 330
Berliner, Emile, 62
bias head, 82
bias tone, 82
bidirectional microphones, 149
binaural hearing, 3637
binaural time of incidence, 37
birdies/birdsinging, 193
bit quantization, 195196
bit rate, 195196
blame, taking, 326
bookshelf speakers, 173
boomy, dened, 249250
boost controls in equalizers, 245246
boundary microphones, 144145
boxy, dened, 250251
The Boys Next Door cue sheets, 339340,
343, 345
brass
condenser microphones for, 142
microphone placement for, 157
broadcast
production facilities, 54
Bryston B-100 amplier, 181, 183
bus faders, 118
in mixing, 277
busses. See sends
BWF (broadcast wave) format, 194
C
Candy, John, 338
Cannon plugs, 106107
capacitance, 138139
condenser microphones, 138143
capstan-less transport systems, 7879
capstan motors, 7778
carbon microphones, 130
cardioid microphones, 149
phase shifting network and, 150
sensitivity of, 151
Carnegie Hall, 228
CDs
burners/players, 351
digital audio tape (DATs) and, 91
routing for playing, 78
sampling rate of, 190191
cellos. See strings
Center for Media Arts track sheet, 319
channel assignment switching matrix,
113114, 276277
channel faders, 110
pan pots and, 114
as passive attenuators, 110111
chief engineers, 298, 302303
china markers/china whites, 211
chorusing, 244
cleaning tape heads, 209210
clients
dealing with, 301302
studio etiquette and, 311
Clinton Recordings Studio A, 66, 67
clipping, 27
close-miking, 5, 228
ambience, creating, 255256
combining amps. See summing networks
compression
in digital audio, 193194
ratio, 239
state of, 1819
compressors, 237240
patch bays for, 71, 7374
signal ow and, 259
computers. See also digital audio
editing, 220221
recording with, 65
concert halls, early recordings in, 38, 228
for brass, 157
directionality of, 151152
for percussion instruments, 158
for woodwinds, 157
condenser speakers, 165, 167
congas, microphones for, 158
consoles, 6263. See also equalizers; faders;
I/O modules; preamps; summing
networks
with dedicated monitor sections, 123
impedance, 106112
input signal, ow of, 124
line level signals, 104
master section of, 278281
mic level signals, 104
monitor modules, 102
monitor signal, ow of, 124
normalling, 281283
in production room, 6971
redundancy in, 70, 100101
sound designers assessing, 352
trim pots (potentiometers), 107108
types of, 102
versatility of, 71, 100102
388
constructive interference, 4546
contact microphones, 144
continuity and sound design, 336338
control room, 6869
Coppola, Francis Ford, 330
Crescent Moon studios, 299
Crest Audio Century VX mixing console, 63
crossovers, 171
crosstalk, 88
SMPTE and, 199
Crown
GLM-100 lavaliere microphones, 144
PCG boundary microphones, 145
PZM boundary microphones, 145
SASS boundary microphones, 145
Cubase
Advanced Music Production System, 214
compression, 238
HALion One, 258
Mixer, 269
SX4 Screenshots, 199
cue sheets, 316318, 339344, 356357
cut controls in equalizers, 245246
cymbals in mixing, 287
D
dbx 160 compressor, 232
de-essers, 245
decay time, 229
decibels (dBs), 2425
in ampliers, 184
value, derivation of, 2526
delay
digital delay, 230
in sound design, 352
tape delay, 229230
Denon C550R Professional CD+CD-R/RW
Combo-Deck, 350
density and speed of sound, 2224
Derek & the Dominos, 320321
destructive interference, 4546
DFC Gemini digital console, 336
diaphragms, 130131
in condenser microphones, 139140
in dynamic microphones, 134135
in ribbon microphones, 136137
diffraction, 44
diffusion, 44
digital audio, 187201
aliasing, 191193
compression in, 193194
editing, 218222
error correction in, 193194
MIDI, 196197
quantization distortion, 192193
sampling rates, 189191
SMPTE (Society of Motion Picture
Television Engineers), 198200
Digital Audio Stationary Head (DASH), 91
digital audio tape (DATs), 63, 9091
ADATs, 9192
Digital Audio Stationary Head C
(DASH), 91
digital delay, 230
digital processors, 232233
digital reel-to-reel, 63
digital-to-analog (D/A) conversions, 56, 59
deterioration of signal and, 236
digital workstations, 54
direct box (DI), 107
for electric bass, 155156
in I/O modules, 123
direct insertion, 261
direct reections, 3940
direction, perception of, 38
directional microphones, 149
directionality of sound waves, 4344
distortion, 27
in ampliers, 183184
mixing and, 268
quantization distortion, 191193
signal ow and, 5
of speakers, 177180
dithering, 193
double walls
for control room, 69
in performance area, 66
dropouts, 93, 218
in razorblade editing, 212
drug use, 312
drum machines and acoustics, 230
drums
mixing and, 286287
placement of microphones on, 154155
tracks, 121
dubbing, 207
dump mode, 214215
duration, 3536
signal processors and, 254255
dynamic microphones, 131138
for drums, 155
dynamic processors, 237243
compressors, 237240
expanders, 240242
limiting, 240
noise gates, 242243
I ndex
389
dynamic range, 27
expanders and, 240242
E
ear
Fletcher Munson equal loudness
contours, 4243
physiology of, 3738
ear canal, 3738
early reections, 3940, 230
EAW
KF730 compact line array, 177
KF760/KF761 line arrays, 176
echo, 40, 257
edit mode, 214215
edit rooms, 54
EDITall Splicing Block, 212
editing, 205223
cleaning tape heads, 209210
complex edits, methodology
of, 214218
digital editing, 218222
from downbeat to downbeat, 216
dropouts, 218
dump mode, 214215
edit mode, 214215
rst cut, locating, 210
glitches, 218
leader tape, 217218
marking the tape, 210211
musical sections, eliminating, 215
objectives of, 207208
problems in, 218
razorblade editing, 206207, 211212
with digital audio, 219220
rocking the tape, 208209
for simple edit, 210
shuttling tape, 8081, 208
simple edits, methodology
of, 209213
speech, 214215
splicing/editing tape, 213
tails out storage, 217
terminology, 208209
efciency of speakers, 177178
eight track format, 89
elasticity and speed of sound, 2224
electret capacitor lavalieres, 144
electric bass, direct box (DI) for, 155156
electric guitars. See guitars
electricity
acoustic energy into, 6061
and magnetism, 132
Electro-Voice
CO94 lavaliere microphones, 144
RE20 microphones, 134
RE85 lavaliere microphone, 354
REV wireless microphone system, 148
electronic patching, 7677
electrostatic speakers, 165, 167
elevation and speed of sound, 23
EMT 140 analog plate reverb, 229
enclosures for speakers, 174
energy
acoustic energy, 6061
in production rooms, 60
envelope. See duration
environment, sound and, 12, 3842
EQ IN switch, 115116
equalization (EQ). See also equalizers
in mixing, 289290
personal aesthetic and, 11
subtlety in, 252
subtracting EQ, 290
equalizers, 115117, 244245
boost controls, 245246
boosting frequency on, 252253
cut controls, 245246
de-essers, 245
graphic equalizers, 248
monitor EQ, 117
parametric equalizers, 245
peaking equalizers, 245246
Q controls, 246
shelving equalizers, 247
in signal path, 275276
equipment trouble reports, 320322
erase head, 82
error correction in digital audio, 193194
ESI M8U 8-In/8-Out USB-to-MIDI
interface, 197
Esslin, Martin, 333
etiquette in recording studio, 311313
Eventide
Clockworks 2826 Omnipressor
compressor, 239
H949 Harmonizer, 243
expanders, 237, 240242
external triggering/keying, 242243
F
faders, 108112. See also channel faders
gain-staging with, 273
in graphic equalizers, 248
in I/O modules, 122123
normalling, 281282
390
summing networks and, 117
Voltage Controlled Ampliers (VCAs)
and, 108, 110
Fairchild 670 compressor, 232
feedback and tweeters, 170
lm, 65. See also sound design
Avid/Pro Tools software, 200
context, creating sound in, 12
shotgun microphones in, 146
lters
anti-aliasing lters, 192
bandpass lters, 245, 248
high-pass lters, 247
low-pass lters, 192, 247
angers, 258259
Fletcher Munson Equal Loudness Contours,
13, 4243
bookshelf speakers and, 173
oating rooms. See suspended/oating
rooms
oor microphones, 145
FM radio signals, 4344
Focal SM8 near-eld monitor, 164
focus zone, 335
foley rooms, 54
force, 18, 1920
molecules and, 21
formulas
frequency, calculation of, 28
for harmonics in triangle wave, 33
period, calculation of, 29
wavelength, calculation of, 29
48 Hours, 338
four track, 8990
Fourier analysis, 31
freelance engineers, 304305
frequency. See also sampling
dened, 2728
direction, impact on, 38
duration and, 35
formula for calculating, 28
fundamental frequencies, 3132
low frequencies, spreading of, 4445
masking and, 47
period and, 29
and shotgun microphones, 145146
sine waves of, 31
types of, 31
frequency dividing networks, 171
frequency response, 130
of ampliers, 183
of condenser microphones, 141142
of speakers, 177180
in subwoofers, 179
frequency-selective ampliers, 115
full normals. See normals
full track, 89
fundamental frequencies, 3132
G
gain reduction, 239
gain-staging, 273
gating, 242243. See also noise gates
external triggering/keying, 242243
general managers, 300
glassy, dened, 251252
glitches, 218
in razorblade editing, 212
gophers. See interns
graphic equalizers, 248
grease pencils, 211
green room, 54
guitars
acoustic guitars, microphones for, 156
compressors for, 238239
mixing, 288
H
half normals, 7475
patch bays wired as, 76
half track, 89
handheld wireless microphones, 149
hard drives, 5, 63
routing for recording to, 56, 59
hard panning, 288
hardness of surface and sound, 39
harmonics, 3132
pulse waves, 35
square wave, 35
in triangle wave, 33
harmonizers, 243244
head assembly, 8183
cleaning, 209210
head lifters, 81
headphones, 285
headroom, 27
headset microphones, wireless, 149
heavy metal, listening to, 271272
Hecuba, 348, 353
Heil Sound microphones, 134136
Hendrix, Jimi, 230
Hertz (Hz), 28
sampling rates in, 189190
I ndex
391
HHB
Communications Portadat, 90
MDP500 Portable MiniDisc
Recorder, 196
high impedance signals, 106
high-pass lters, 247
hi-hats in I/O modules, 122123
Hitchcock, Alfred, 330
home stereo systems, 171172
home studios, 54
humidity and magnetic tape storage, 94
hyper-super-cardioid microphones, 149
I
I/O modules, 102, 273278
working with, 120125
icons, 334335
Imaginary Road Studios, 303
impedance, 106112
low-impedance signals, 106
speakers and, 179180
indexing, 334335
on reel-to-reel, 350351
inductance and dynamic microphones, 131
input list, 120121
with track assignments, 121
input/output modules (I/O modules). See
I/O modules
input signal ow, 124
The Inspector General, cue sheets for,
341, 344
insert send/insert return ow, 259
The Insider Audio Bathroom Reader
(Lehrman), 325
insults in studio, 311
interaural arrival time, 37
interaural intensity, 3637
interns, 310
blame, taking, 326
menial jobs, attitude about, 325326
Iron Maiden concert, 176
J
jacks
for patch bays, 7374
1/4-inch jacks, 106107
Jaws, sound cues in, 335336
job descriptions, 298306
K
Key Largo, sound cues in, 335
keying with gate, 242243
kick drums, microphones for, 155
L
lapel microphones, 143
lavalieres, 143144
mounting options, 354356
for theatre sound, 354356
Layla (Derek & the Dominos), 321
leader tape, 217218
Lehrman, Paul, 325
leveling ampliers, 239
Lexicon 224 digital reverb
(reverberation), 233
light and magnetic tape storage, 94
lighting cues, 337
limiters/limiting, 237, 240241
line arrays, 175177
line trims, 107108, 109, 274
listening professionally, 270272
live-to-two-track recording, 56, 58
location. See also reverb (reverberation)
characteristics, establishing, 347348
echo and, 257
for guitars, 288
masking and, 47
panning, 257
perception of sound and, 3638
signal processors and, 255257
sound cues establishing, 331334
locked up SMPTE machines, 198199
longitudinal waves, 1819
speed of, 2224
loop insertion, 261
lossless compression, 193194
lossy compression, 193194
loudness. See also dynamic processors;
Fletcher Munson Equal Loudness
Contours
interaural intensity and, 3637
sound cues and, 337338
wattage ratings and, 182
low frequencies, spreading of, 4445
low-impedance signals, 106
low-pass lters, 192, 247
M
Mackie
CR series console, 103
SR 24-4 console, 63
magnetic tape, 8792
binding solution, 8788
characteristics of, 9294
drop out, 93
eight-track format, 89
392
four-track format, 8990
full track, 89
guard band, 88
half track, 89
print through, 94
sixteen-track format, 90
storage of, 9394
stretching, 93
tails out storage, 94, 217
tracks on, 88
magnetism, 6061
electricity and, 132
maintenance engineers, 298, 302, 305306
mallets, microphones for, 158
managers, 301302
Marantz PMD670 C Professional Solid
State Personal Recorder, 348
marimbas, microphones for, 158
masking, 4647
in mixing, 277
troubleshooting, 253254
master fader/master trim, 118119, 279
masters, take and track sheets with, 319
MEG. See musikelectronic geithain (MEG)
metallic, dened, 251252
Meyer Sound monitors, 165
mic pres. See preamps
mic trims, 107109, 274
microphones, 129160. See also cardioid
microphones; close-miking; con-
denser microphones; diaphragms;
lavalieres; polar patterns;
preamps; ribbon microphones;
shotgun microphones
boundary microphones, 144145
consoles, signals to, 104
critical listening and, 152153
durability of, 132133
dynamic microphones, 131138
in early recording, 227228
high impedance mics, 106
in I/O modules, 122
for kick drums, 155
line input-selector switch, 273
placement of, 153158
pressure zone microphones, 354, 356
sound designers and, 353356
sound pressure level (SPL) and, 132134
as transducers, 6162
tube microphones, 142, 234235
wireless systems, 147149
mid-range speakers, 169170
MIDI, 196197
mini-discs, sound design for, 349350
mixing, 267296. See also consoles; I/O
modules
aesthetics of, 291294
bottom-up mixing, 286287
equalization (EQ) and, 289290
good mix, elements of, 268272
methodology of, 286291
personal taste and, 268
processors in, 259262
remixers, 287
speakers, understanding, 283286
sphere for, 291294
starting a mix, 286
MKE 102 Omni Lavaliere
Microphone, 335
molecules
force and, 21
and sound, 1820
vibration of, 2122
monitor EQ, 117
monitor modules, 102
monitor pots, 110112
track assignments with, 122
monitor source selector switch, 119,
279280
monitor speakers. See speakers
mono level signals, 104
mood, sound cues establishing, 334
movies. See lm
moving coil microphones, 131132
lavalieres, 144
moving coil speakers, 165166
MPET-2 layer 3 compression, 194
MP3 compression, 194
multiprocessors, 197
multitrack analog devices, 63
multitrack bussing, 113114, 276277
multitrack recording, 5, 5657
ADATs, 9192
erase head/bias head in, 82
signal processing and, 228
mults (multiples), 76
Munson, Fletcher, 13. See also Fletcher
Munson Equal Loudness Contours
Murphy, Eddie, 338
Murray, Bill, 338
musikelectronic geithain (MEG)
Basis 3 subwoofer, 179
RL 901k Studio Reference Monitor,
162, 179, 180
mylar strips, 87
I ndex
393
N
National Edison Studios
track sheet, 318
work order, 314
Nelsen, Paul, 335
Neumann
KM series microphones, 142
M 149 microphones, 141
U 67 microphones, 140
U 87 microphones, 142
Neve consoles, 104
88R mixing console, 102
8068 mixing console, 310
8078 mixing console, 284
night managers, 302
nodes, 48
noise. See also signal-to-noise (S/N) ratio;
white noise
in ampliers, 184
dealing with, 65
dithering and, 193
mixing and, 268
shotgun microphones and, 146
tape speed and, 7980
noise oor, 27
noise gates, 237, 242243
external keying of, 242
Nolte, Nick, 338
non-harmonically related overtones,
31, 33
non-lossy compression, 193194
nondirectional microphones, 149
normalling, 95, 281283
normals, 7475. See also half normals
breaking on input signal, 75
use of term, 7576
Nuendo
Finalizer, 241
Media Production System, 206
Mixer, 271
O
octaves, 30
harmonics and, 32
omnidirectional microphones, 149
On The Verge, cue sheets for, 341343
opera, listening to, 272
oscillations of sound waves, 2728
Otari
pinch-rollerless transport systems, 7879
tape transport controls, 80
outboard equalizers, 6
outer ear, 3738
overheads, microphones for, 155
overtones, 31, 3233
owners of studios, 298300
P
pan pots, 114115
in mixing, 277, 287
panning, 257
with drums, 287
hard panning, 288
in mixing, 277, 292
panoramic potentiometers, 114115
papery, dened, 251252
parallel surfaces
non-controlled environment, recording
in, 67
parametric equalizers, 245
passive attenuators, 108
channel faders as, 110111
patch bays, 7177. See also normals
electronic patching, 7677
jacks for, 7374
mults (multiples), 76
patch cords, 7677
Paul, Les, 230
peak-hold meters, 86
peak meters, 8586
peaking equalizers, 245246
people skills, 325
perception
environment and, 3842
human perception of sound, 2438
and location, 3638
of pitch, 2730
in signal processing, 236237
sound as, 2022
of timbre, 3135
percussion instruments, microphones
for, 158
performance areas, 6567
double walls, 66
isolation in, 65
patch bays, 7177
reective surfaces in, 67
suspended/oating rooms, 6667
period, calculation of, 2930
peripheral zone, 335
personal aesthetic, 11
phantom power, 140141
phase relationship, 46
394
phase shifters, 258259
for network, 150
physiology of ear, 3738
pianos
in I/O modules, 123
pickup patterns. See polar patterns
pinch rollers, 78
cleaning, 209210
pinna, 3738
pitch, 244. See also frequency
fundamental frequency and, 3132
harmonizers, 243244
perception of, 2730
signal processors affecting, 243244
plate reverb, 228229
playback head, 6162
polar patterns, 149152
of boundary microphones, 145
of shotgun microphones, 145
Porter, Cole, 343, 345
post-fader sends, 112113
post-production facilities, 54
powering up/down production room,
9495
pre-fader sends, 112113
preamps, 103106
aesthetics of, 67, 105
character of sound and, 104105
coupled with ampliers, 181
in I/O modules, 122123
impedance, 106112
line input preamps, 105106
tube preamps, 234235
pressure zone microphones, 354, 356
Prince, 244
print through, 94
Pro Tools, 200
HD 7, 221
LE 7, 219
M-Powered 7, 293
production room, 7, 5398. See also
performance areas; recording
studios; speakers
components of, 6069
consoles in, 6971
control room, 6869
dened, 54
performance space, 6567
procedures, 9495
storage devices in, 6364
tape transports, 7780
types of, 54
Psycho, sound in, 336
pulse waves, 3435
Pultec
EQH-2 equalizer, 231
EQP-1 equalizer, 231
punchy, dened, 250
Q
quantization
distortion, 192193
1/4-inch jacks, 106107
R
radio stations, 54
production room functions, 56
routing signal in, 8, 11
range of expanders, 241242
rarefaction, state of, 1819
razorblade editing. See editing
RCA
44BX microphones, 137138
77DX microphones, 137
Real Audio lossy compression, 194
reception, zones of, 335
record head, 6162
recording studios, 54. See also assistant
engineers; interns; production
room
chief engineers, 298, 302303
as decisions by, 300
entry-level positions, 306310
etiquette, 311313
freelance engineers, 304305
job descriptions, 298306
maintenance engineers, 305306
managers, 301302
operations, 313322
owners, 298300
routing in, 7, 10
sound designers, 305
staff engineers, 303304
take sheets, 316318
tape release forms, 315316
track sheets, 318321
trouble reports, 320322
voice-over studios, 54
work orders, 313316
redundancy in consoles, 70, 100101
reel size controls, 79
reel-to-reel
digital reel-to-reel, 63
reected sound waves, 3839
I ndex
395
reections, 3940
early reection of sound, 3940, 230
reverberation and, 4042
refracted sound waves, 3839
rejection of sound, 130
remote recording, 56
resistance, 106
respecting artists, 271272
reverb (reverberation), 4042, 255256
auto-panning with, 257
with drums, 287
in mixing, 292
personal aesthetic and, 11
plate reverb, 228229
reections and, 4042
RT-60s for, 42, 256
spring reverbs, 228
ribbon microphones, 131132, 136138
for percussion instruments, 158
for woodwinds, 157
ribbon speakers, 165, 166167
rocking the tape. See editing
Roland DM-800, 92
room mics, 144
rooms. See also production room; sus-
pended/oating rooms
characteristics in sound, 351352
control room, 6869
green room, 54
reections, 40
tracking rooms, 66
rotary pots, 108
routing. See signal routing
Royce Hall line array, 176
RT-60s, 42, 256
rumble, dened, 249
S
sampling
bandwidth and, 190191
editing and, 206
quality of sample, 188
rates, 189191
satellite uplinks, 56
saturation, 27
sawtooth waves, 3334
diagram of, 34
Schwarzenegger, Arnold, 337
scripts. See sound design
Seismic Audio Titanium Horn Tweeter, 170
self-powered speakers, 182
semiotics, 333335
send masters, 118
sends, 112114, 277278
multitrack busses, 113114
pan pots and, 114115
Sennheiser
816 shotgun microphones, 146147
MD421 microphones, 133
MKE system lavalieres, 144
shakers, microphones for, 158
shelving equalizers, 247
shimmery, dened, 252
shotgun microphones, 145147
for theatre sound, 354, 356
Shure
55 series microphones, 131
SM57/SM58 microphones, 133134
VP88 stereo middle-side
microphone, 154
wireless microphone system, 148
shuttling tape, 8081, 208
sibilant range, de-essers and, 245
Sigma Sound Studios, 285
signal ow, 45, 259262. See also patch
bays; signal routing
in consoles, 71
diagram of, 56
example of, 78
in live sound, 55
signal path, 7. See also patch bays
signal processing, 225264. See also close-
miking; dynamic processors;
equalization (EQ); reverb
(reverberation)
classications of, 236237
combining digital and analog, 233234
control room, gear in, 68
digital technology, 233236
direct insertion, 261
duration and, 254255
envelope and, 254255
angers, 258259
history of, 227233
and location, 255257
loop insertion, 261
phase shifters, 258259
pitch and, 243244
tape delay, 229230
terminology of, 248253
timbre and, 244254
tube processors, 230232
396
signal routing
in consoles, 71
diagram, 6
explanation of, 7
in live-to-two-track recording, 56, 58
in mixing, 276
signal-to-noise (S/N) ratio, 27
in ampliers, 184
silence, use of, 338, 346
sine waves, 33
diagram of, 34
of frequencies, 31
sixteen-track format, 90
skins, microphones for, 158
Skywalker
Scoring, 306
Sound, 102
slope of expanders, 241242
SMPTE (Society of Motion Picture
Television Engineers), 198200
solid state electronics, 231232
Solid State Logic (SSL), 104, 113, 233234
6000 consoles, 246
aux sends from console, 278
equalizer section, 116
equalizer section, I/O module, 275
I/O module, 273
master section of console, 279
monitor source selections, 280
9000 console, 101, 308
shelving EQ control from, 247
in Studio Davout, 308
Sony
ECM 66/77 lavaliere microphones, 144
PCM-800, 92
3324 Digital Multitrack Recorder, 220
Sorcerer Sound, 310
sound cubes, 174
sound cues
classication of, 333334
location and, 331333
loudness and, 337338
sound design, 329360
ambience in, 346347
available equipment, assessing, 352
believability of cues, 331333
cue sheets, 356357
detail, attention to, 345
emotion and, 336338
location, establishing, 347348
methodology of, 339345
microphones, consideration of, 353356
playback format, selecting, 349351
room characteristics, incorporating,
351352
second reading of script, 348349
semiotics, 333335
silence, use of, 338, 346
speaker considerations, 352353
subliminal cues, 345346
subtlety in, 345346
sound designers, 305
sound pressure level (SPL), 2526
microphone durability and, 132134
of moving coil speakers, 165
of speakers, 178
sound reinforcement, 7, 10
and production room, 54, 56
signal ow in, 55
sound stage, 6566
sound waves, 1820
absorbed waves, 3839
behavior of, 4348
complex waveforms, 31
constructive interference, 4546
destructive interference, 4546
diffraction, 44
diffusion of, 44
echoes, 40
longitudinal waves, 1819
masking, 4647
oscillations, 2728
phase relationship, 46
reected waves, 3839
refracted waves, 3839
speed of, 2224
standing waves, 4748
Soundcraft I/O module, 274
speaker array, 5, 78
speakers, 5, 7, 10. See also ampliers
accuracy and clarity of, 178
bookshelf speakers, 173
character of, 283286
compressors with, 239
crossovers, 171
different types, mixing with, 284285
distortion of, 177180
efciency of, 177178
enclosures, 174
frequency response, 177180
importance of, 162163
line arrays, 175177
mid-range speakers, 169170
in mixing, 283286
moving coil speakers, 165166
music for judging, 285
recorder monitor modes, 8384
self-powered speakers, 182
signal ow, 124
I ndex
397
speakers (continued)
sound cubes, 174
sound pressure level (SPL) of, 178
studio monitors, 171173
in telephones, 163164
theory of, 163165
as transducer, 61
tweeters, 170
voice coil in, 164165
volume control, 279280
woofers/subwoofers, 167168
special effects, 230231
speed of sound, 2224
Spielberg, Steven, 330
splicing, 207208, 213
sporting events, shotgun microphones in, 146
spring reverbs, 228
square waves, 3435
SSL. See Solid State Logic (SSL)
staff engineers, 303304
stage sound. See theatre
Stallone, Sylvester, 337
stand-alone synchronizers, 199
standard operating level (SOL), 27
standing waves, 4748
microphone placement and, 155
stereo bus, 114, 279
stereo recording, 228
storage
devices, 6364
of magnetic tape, 9394
stretching, 93
strings
mixing for, 290291
Stripes, 338
Studer
A827 multitrack recorder, 64
D827 multitrack recorder, 64
pinch roller tape transports, 79
tape heads, 81
Vista 5 digital console, 189
Studio Davout, 308
studio monitors, 171173
studios. See recording studios
Studios Guillaume Tell, 301
sub-masters, 277
sub-mixes, 112
subliminal cues, 345346
subliminal zone, 335
subtlety in sound design, 336338,
345346
subwoofers, 167168
frequency response in, 179
summing networks, 74, 117119, 277278
master fader/master trim, 118119
super-cardioid microphones, 149
Superior Line Source (SLS)
8290 ribbon speaker, 167
LS8695AX line source array
column, 169
LS8800 source array module, 168
RLA/2s, 176
supply motors, 7778
surfaces and sounds, 3839
surveillance, shotgun microphones in, 146
suspended/oating rooms
for control room, 69
for performance areas, 6667
synthesizers and acoustics, 230
T
tail leadering, 207
tails out storage, 94, 217
take sheets, 309, 316318
take-up motors, 7778
tambourines, microphones for, 158
tape delay, 229230
tape guides, 79
tape recorders, 5, 6364
tape release forms, 315316
tape transports, 7780
capstan-less transport systems, 7879
edit mode, 81
fast forward control, 81
head assembly, 8183
head lifters, 81
input mode, 83
monitor modes, 8384
motors on, 7778
play control, 81
playback head, 83
ready/safe switch, 81
record head, 8283
reel size control, 79
reproduce mode, 8384
rewind control, 81
shuttling tape, 8081
speed controls, 7980
stop control, 81
sync mode, 84
tension idlers, 79
transport controls, 8081
Tascam DA-38, 91
technology and aesthetics, 1315
398
telephones
components of, 163164
equalizers and, 245
microphones in, 62, 130
ring sounds, 332
speakers in, 163164
Teletronix
LA-2 tube limiter, 231
LA-3A limited, 232
television, lavalieres for, 143144
temperature
for magnetic tape storage, 94
speed of sound and, 23
tension idlers, 79
Texel FMR-100 wireless microphone
system, 147
theatre, 54. See also sound design
boundary microphones in, 145
context, creating sound in, 12
lavalieres, 143144
shotgun microphones in, 146
signal ow in, 55
wireless microphone systems, 147
theatre booths, 54
threshold
for compressors, 239
for expanders, 241
of hearing, 27
of pain, 27
tie lines, 71
timbre. See also equalization (EQ)
Fourier analysis, 31
masking and, 47
perception of, 3135
signal processors affecting, 244254
tom toms, 287
total harmonic distortion (THD), 184
toys, microphones for, 158
track sheets, 309, 318321
archiving, 320321
tracking, processors in, 259261
tracking rooms, 66
tracks, 88
four track, 8990
input list with track assignments, 121
two-track format, 89
transducers, 60
examples of, 6162
transients, 36
compressors evening, 238239
ribbon microphones and, 137
V.U. meters and, 86
transport controls, 8081
triangle waves, 3334
triggering with gate, 242243
trim pots (potentiometers), 107108
trouble reports, 302, 320322
tube microphones, 142, 234235
tube preamps, 234235
tube processors, 230232
turntables, 105
tweeters, 166167, 170
two-track format, 89
tympanis, microphones for, 158
U
ultra-cardioid microphones, 149
unidirectional microphones, 149
unity gain, 27
Universal Audio 2-610 Tube
Preamplier, 231
Urei
813 studio monitors, 172
Teletronix LA-3A limited, 232
V
Van Damme, Jean-Claude, 337
VCR setup, 7, 910
vibes, microphones for, 158
vibration of molecules, 2122
Video Post & Transfer control room, 68
violins. See strings
vocal booths, 6567
vocals
compressors and masking, 238
harmonizers for, 243244
mixing for, 289
voice-over studios, 54
Voltage Controlled Ampliers (VCAs),
108, 110
voltmeters, 8587. See also V.U. meters
volume control for speakers, 279280
V.U. meters, 8487
transients on, 36
0 dB V.U., 27
W
wattage ratings, 182
WAV format, 194
waveforms, 1820. See also sound waves
WaveLab 6 audio editing/mastering soft-
ware, 216
wavelength, formula for calculating, 29
I ndex
399
Webster Universitys take sheet, 317
Welles, Orson, 330
white noise, 33
gating and, 242243
quantization distortion and, 193
Williams, Tennessee, 342
wireless microphone systems, 147149
woodwinds
woofers, 166, 167168
work orders, 302, 313316
wow phenomenon, 93
X
XLR plugs, 106107
xylophones, microphones for, 158
Y
Yamaha SPX9011 Digital Multi-Effects
Processor, 244
Z
0 dB V.U., 27
zones of reception, 335
400

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