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British Library Cataloguing in Publication Data
Audio engineering explained : professional audio recording.
1. Sound—Recording and reproducing. 2. Acoustical engineering.
I. Self, Douglas.
621.3828-dc22
Library of Congress Control Number: 2009933707
ISBN: 978-0-240-81273-1
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Preface
This is a collection of chapters from contemporary books on audio engineering. It covers a wide range
across a large and ever-increasing subject. Obviously a book of practical size cannot give a complete
account of even one area of audio technology, and that is not the intention here.
The aim is to give a good picture of the field by focusing on a number of audio topics without either
getting too specialized or making do with vague generalities. I have tried hard to make this a collection
which overall gives considerable insight into the whole subject of audio engineering, but which also
acts as a jumping-off point for further study. Most of the chapters here include comprehensive refer-
ences and extremely useful suggestions for further reading.
This book is divided into eight sections, which are as follows:
The first section, “Basics,” deals with the fundamental principles of audio, acoustics, and human hear-
ing. It comprises one chapter on the “Fundamentals of Audio and Acoustics,” by Pat Brown, which covers
the physical side of sound, such as decibels, frequency and wavelength, and a complementary chapter
by David Howard and Jamie Angus which gives a good introduction to the processes of human hear-
ing. This second chapter includes a wealth of material on the anatomy of the ear, the crucial function
of the basilar membrane, and critical band theory. This last topic may sound academically dry, but it
is in fact the foundation of audio compression systems such as MP3 which are so widely used today.
There is also useful information on the perception of loudness, which does not always work as you
might expect: if you want to double the loudness of a string section, you will have to hire ten times the
number of violinists.
vii
The second section of this book, “Microphones,” consists of two chapters taken from The Microphone
Book by John Eargle. The first chapter deals with the basic ways in which various types of sound field
are propagated and how they interact with a microphone. The second chapter focuses on the electrical
aspects of microphone operation, such as phantom powering, microphone transformers, mixing con-
sole microphone preamplifiers, microphone splitters, cable losses, radio microphones, and the latest
so-called “digital” microphones.
The third section “Preamplifiers, Mixers, and Interconnects” gets to grips with the electronics in sound
systems. It deals with what might be called the small-signal part of a system—the controlling electron-
ics, usually in the form of a mixing console, and the interconnections between the console and the
other components of the system. In Chapter 5, Francis Rumsey and Tim McCormick give a through
and comprehensive description of how mixing consoles work. I spent some years engaged in mixing
console design, as Chief Engineer at Soundcraft Electronics, and I can tell you that I think this chapter
is the best concise description of mixer operation that I have come across, and it was an easy deci-
sion to include it in this collection. The fact that one of the illustrations is of a console I had a hand
in designing (The Soundcraft Sapphyre) is, I hasten to add, merely a happy coincidence. Apart from
mixing console design as such, the chapter also deals with approaches to recording and mixing, and
console automation.
Compared with the complexities of mixing console design, Chapter 6 on interconnects, taken from
Handbook for Sound Engineers, and written by Glen Ballou, may sound uninspiring. It’s just a bit of
cable, isn’t it? No, it isn’t. It is no use constructing near-perfect boxes of electronics if the cable lead-
ing to them fatally compromises the quality of the signal before it even arrives there. Chapter 6 covers
loudspeaker cables, digital audio cables, and video cables, as well as ordinary audio connections.
The final chapter in this section, Chapter 7, is by Bill Whitlock, who is an acknowledged expert in the
fields of equipment grounding and interconnection. The technology of audio interconnection is not
only a good deal more complex than it appears, but can also present some fascinating (and in the
wrong circumstances thoroughly irritating) problems. The greatest danger to a signal in a cable is that
unwanted currents flowing down the ground connection, in the classic ground loop, cause hum and
Preface
noise to be added to the signal. There are several ways of preventing this, of which the best known is
the balanced interconnection, which uses two wires to carry the signal instead of one. It is now pos-
sible for the receiving equipment to respond to the difference between the two signal conductors and
(to a first approximation) ignore the unwanted voltage drop down the ground conductor. Like so many
technical matters, the principle is beautifully simple but the practicalities can get rather complicated;
this technology is thoroughly dealt with in Chapter 7 by an author who is a master of his subject.
The section on “Power Amplifiers” makes up the fourth part of this book, and I have contributed this
myself, having spent a good deal of time working on the apparently simple but, once again, deeply
complex and fascinating matter of turning small voltages and currents into large ones without distort-
ing them in the process. The three chapters are taken from my Audio Power Amplifier Design Handbook,
which is now available in a greatly extended fifth edition.
The first chapter taken from the Handbook explains the basic architecture of power amplifiers, and
shows how they are put into different classes such as Class-A, Class-B, and so on, depending on their
mode of operation, up to Class-S. Some less well-known types such as error-correcting and non-switch-
ing amplifiers are covered. The intelligent use of negative feedback is absolutely fundamental to power
amplifier design, and it is explained here.
The second chapter goes into more detail on power amplifier functioning, and shows how the appar-
ently hopelessly entangled sources of amplifier distortion can in fact be reduced to a few straightforward
mechanisms, most of which can be dealt with quite easily once they are properly identified. Topics such
as measuring open-loop gain are also covered.
In the third chapter, I explain the operation and design of Class-G amplifiers, which significantly
economize on power consumption and heat dissipation by running from low-voltage rails when repro-
ducing small amplitudes, and switching to high-voltage rails to accommodate the relatively infrequent
signal peaks. This technology is of increasing importance, as it has found its niche in subwoofers where
relatively large amplifier power outputs are required. When properly implemented—as described
here—it can give much better quality than the Class-D amplifiers which are sometimes used in this
viii
application.
The fifth section of this book “Loudspeakers” deals with the tricky business of turning electricity into
sound. It consists of four chapters, all taken from the book Loudspeakers by Philip Newell and Keith
Holland.
The first chapter sets out the most basic fundamentals of loudspeaker operation, covering acoustic
radiation and the load a loudspeaker presents to a power amplifier.
In the second chapter, diversity in loudspeaker design is explored, and the construction of both
moving-coil and electrostatic loudspeakers is gone into in more detail. The third chapter describes
loudspeaker cabinets, which are not just enclosures to keep the dust off, but an integral part of the
functioning of the loudspeaker.
Finally, the fourth chapter in this section deals with the crossover units that direct the appropriate fre-
quencies to the high-frequency and low-frequency units of a multi-way loudspeaker system. This alone
is a large and complex field of study.
The next section of this book, the sixth, addresses the vitally important field of “Digital Audio.” I am
particularly conscious that this is a huge and ever-growing field of knowledge, and one section of one
book can only give a taste of its extent. Digital audio entered most people’s lives with the introduc-
tion of the CD in 1982, and since then digital methods have extended from relatively simple storage
media to highly complex manipulations known as Digital Signal Processing or DSP. Very sophisticated
specialized processors have been developed to efficiently handle the enormous amounts of calculation
required.
Chapter 15, by Craig Richardson, gives a good account of the essential principles of digital audio, cov-
ering sampling theory, Z-transforms (which are not as scary as they sound), and the basics of choosing
a DSP chip.
Another aspect of digital technology is the use of the MIDI standard for controlling electronic musical
instruments. Back in the days when I had my own eight-track recording studio (a setup which dated
back to when eight tracks referred to tracks on magnetic tape, and which unfortunately succumbed to
Preface
sheer lack of time to make proper use of it) I made very extensive use of MIDI, and I have always been
impressed by the robustness of its technical specifications and its remarkable flexibility. In Chapter 16
David Huber gives a very good overview of MIDI technology, including MIDI sequencing software and
MIDI time-code.
The seventh section is “Acoustics and Sound Reinforcement”. The first two chapters are from Sound
System Engineering by Don and Carolyn Davis. The first chapter covers the complex subject of large-
room acoustics, dealing with reverberation, flutter-echoes, and the reverberant and direct sound fields.
The second chapter deals with the problem of equalizing the sound system and its associated acous-
tics to get the desired frequency response, describing the filters and equalizers that are used in this
endeavor, and how its success is evaluated by specialized measuring equipment.
The final chapter in this section, from Sound Reproduction by Floyd Toole, covers the propagation of
sound in spaces, and how that interacts with human perception. The Precedence Effect, or Haas Effect,
in which delayed sounds are not heard as separate events but reinforce the initial sound, is a funda-
mental part of sound reproduction in a space, and is dealt with comprehensively here.
The final section “Recording Studios” is extracted from Philip Newell’s book Recording Studio Design.
The first chapter deals with the vital issue of sound isolation. The most sophisticated electronics in the
world is helpless against the sound of traffic, road-drills and worse infiltrating the recording studio
because the precautions taken against it were inadequate. Upgrading and correcting a poorly-designed
sound isolation scheme is almost certainly going to be expensive and disruptive to the business of
recording, and it is only sensible to carefully consider the initial plans before putting trowel to mortar.
This chapter provides an excellent basis for this sort of planning, dealing with the reflection, transmis-
sion and absorption of sound. The mass law, damping and decoupling, and floor, wall and ceiling iso-
lation are all dealt with in detail.
Having spent more time in recording studios than was probably good for me, I can vouch for the
fact that the studio environment is important if you want to get the best artistic results and a healthy
number of repeat bookings. The second chapter by Philip Newell discusses the environmental issues ix
of studio lighting, ventilation, decoration, and the provision of headphone foldback and clean mains
supplies.
This brief introduction can only give a taste of the varied and extensive information in this book. Each
chapter has its own preface which gives more detail on its contents.
Preface
This chapter begins our journey through the world of audio engineering by explaining some funda-
mental principles. Our hearing, like our other senses, works in a logarithmic fashion, which allows us
to cope with the enormous variation in sound levels that occur in the real world. Explanation starts
with the decibel, a unit whose logarithmic basis matches the response of our ears, and makes the plot-
ting of graphs and the doing of calculations much simpler and more intuitive than the alternative of
grappling with scientific notation and numbers wreathed in zeros.
From considering the mathematics of decibels (which is really very simple), Pat Brown moves on to
consider how they are used to measure absolute rather than just relative sound levels. Our ears respond
to sound pressure levels. Taking the quietest sound that can be detected, and calling it the Threshold
of Hearing, we can assign 0 decibels to this sound pressure level. This is invariably abbreviated to 0 dB.
The decibel scale can now be used to establish the loudness of common situations, such as a room
with quiet air-conditioning (about 30 dB), normal conversation (about 50 dB), and the traditional
jack-hammer at 1 meter (about 100 dB). The threshold of pain is usually defined as 130 dB, a level that
hopefully few of us will ever encounter; it is a sobering thought that hearing damage due to long-term
exposure can begin as low as 78 dB.
Our perception of frequency, i.e., pitch, is also logarithmic, which is why music has as its foundation
the octave, which stands for a doubling of frequency. When drawing graphs of quantities against fre-
quency, it is usual to use decades, or ratios of 10, instead of octaves, as this simplifies the maths. Pat
then relates frequency to wavelength and the speed of sound.
It is one of the most important features of human hearing that the frequency response of our ears
is not a flat line when plotted against sound pressure level. Worse than that, the deviation from flat
increases as the level falls. This disconcerting situation is summed up by the famous Fletcher and
Munson equal-loudness curves. This naturally complicates the business of measuring noise levels, and
so standard weighting curves have been defined that allow for this, the best-known being the A-weight-
ing curve, which is often used in the measurement of audio electronics. This curve has a pronounced
roll-off at the low-frequency end which takes into account the fact that our hearing is markedly less
sensitive at, say, 50 Hz, than it is at the reference frequency of 1 kHz. There is then a discussion of how
the ear integrates sound, one example being the Precedence effect, in which late-arriving reflections
within 40 milliseconds of the original sound reinforce its subjective volume rather than being heard as
separate events.
Pat Brown concludes this chapter with a look at some of the simplest cases of sound propagation: the
point source and the line source. These are of course mathematical abstractions rather than real sound
sources, but they are the basic foundation of more realistic and therefore more complex models, such
as the treatment of a loudspeaker cone as a rigid circular piston.
1
cHaPter
fundamentals of audio
and acoustics
Handbook for Sound engineers by Pat Brown
IntroductIon
Many people get involved in the audio trade prior to experiencing technical training. Those serious
about practicing audio dig in to the books later to learn the physical principles underlying their craft.
This chapter is devoted to establishing a baseline of information that will prove invaluable to anyone
working in the audio field.
Numerous tools exist for those who work on sound systems. The most important are the mathematical
tools. Their application is independent of the type of system or its use, plus, they are timeless and not
subject to obsolescence like audio products. Of course, one must always balance the mathematical
approach with real-world experience to gain an understanding of the shortcomings and limitations of
the formulas. Once the basics have been mastered, sound system work becomes largely intuitive.
Audio practitioners must have a general understanding of many subjects. The information in this chap-
ter has been carefully selected to give the reader the big picture of what is important in sound systems.
In this initial treatment of each subject, the language of mathematics has been kept to a minimum,
opting instead for word explanations of the theories and concepts. This provides a solid foundation
for further study of any of the subjects. Considering the almost endless number of topics that could
be included here, I selected the following based on my own experience as a sound practitioner and
instructor. They are:
1. The Decibel and Levels.
2. Frequency and Wavelength.
. The Principle of Superposition.
4. Ohm’s Law and the Power Equation.
. Impedance, Resistance, and Reactance.
6. Introduction to Human Hearing.
7. Monitoring Audio Program Material.
8. Sound Radiation Principles.
9. Wave Interference.
A basic understanding in these areas will provide the foundation for further study in areas that are of
particular interest to the reader. Most of the ideas and principles in this chapter have existed for many
years. While I haven’t quoted any of the references verbatim, they get full credit for the bulk of the
information presented here.
Audio Engineering Explained
Copyright © 22000190 by Elsevier Inc. All rights of reproduction in any form reserved.
Part I
Basics
tHe decIBel
Perhaps the most useful tool ever created for audio practitioners is the decibel (dB). It allows changes
in system parameters such as power, voltage, or distance to be related to level changes heard by a lis-
tener. In short, the decibel is a way to express “how much” in a way that is relevant to the human per-
ception of loudness. We will not track its long evolution or specific origins here. Like most audio tools,
it has been modified many times to stay current with the technological practices of the day. Excellent
resources are available for that information. What follows is a short study on how to use the decibel for
general audio work.
Most of us tend to consider physical variables in linear terms. For instance, twice as much of a quantity
produces twice the end result. Twice as much sand produces twice as much concrete. Twice as much
flour produces twice as much bread. This linear relationship does not hold true for the human sense of
hearing. Using that logic, twice the amplifier power should sound twice as loud. Unfortunately, this is
not true.
Perceived changes in the loudness and frequency of sound are based on the percentage change
from some initial condition. This means that audio people are concerned with ratios. A given ratio
always produces the same result. Subjective testing has shown that the power applied to a loud-
speaker must be increased by about 26% to produce an audible change. Thus a ratio of 1.26:1 pro-
duces the minimum audible change, regardless of the initial power quantity. If the initial amount
of power is 1 watt, then an increase to 1.26 watts (W) will produce a “just audible” increase. If the
initial quantity is 100 W, then 126 W will be required to produce a just audible increase. A number
scale can be linear with values like 1, 2, 3, 4, 5, etc. A number scale can be proportional with values
like 1, 10, 100, 1000, etc. A scale that is calibrated proportionally is called a logarithmic scale. In
fact, logarithm means “proportional numbers.” For simplicity, base 10 logarithms are used for audio
work. Using amplifier power as an example, changes in level are determined by finding the ratio
of change in the parameter of interest (e.g., wattage) and taking the base 10 logarithm. The result-
ant number is the level change between the two wattages expressed in Bels. The base 10 logarithm
6 is determined using a look-up table or scientific calculator. The log conversion accomplishes two
things:
1. It puts the ratio on a proportional number scale that better correlates with human hearing.
2. It allows very large numbers to be expressed in a more compact form, Figure 1.1.
The final step in the decibel conversion is to scale the Bel quantity by a factor of 10. This step converts
Bels to decibels and completes the conversion process, Figure 1.2. The decibel scale is more resolute
than the Bel scale.
The decibel is always a power-related ratio. Electrical and acoustical power changes can be converted
exactly in the manner described. Quantities that are not powers must be made proportional to
power—a relationship established by the power equation:
E2
W (1.1)
R
where,
W is power in watts,
E is voltage in volts,
R is resistance in ohms.
This requires voltage, distance, and pressure to be squared prior to taking the ratio. Some practitioners
prefer to omit the squaring of the initial quantities and simply change the log multiplier from 10 to 20.
This produces the same end result.
Figure 1.3 provides a list of some dB changes along with the ratio of voltage, pressure, distance, and
power required to produce the indicated dB change. It is a worthwhile endeavor to memorize the
changes indicated in bold type and be able to recognize them by listening.
A decibel conversion requires two quantities that are in the same unit, i.e., watts, volts, meters, feet. The
unit cancels during the initial division process, leaving the ratio between the two quantities. For this
cHaPter 1
Fundamentals of Audio and Acoustics
Linear scale
01 k 10 k 20 k 30 k 40 k 50 k
Log scale
1 10 100 1 K 10 K 100 K
fIgure 1.1
A logarithmic scale has its increments marked by a fixed ratio, in this case 10 to 1, forming a more compact representation than a
linear scale. (Courtesy Syn-Aud-Con.)
7
fIgure 1.2
The steps to performing a decibel conversion are outlined. (Courtesy Syn-Aud-Con.)
reason, the decibel is without dimension and is therefore technically not a unit in the classical sense. If
two arbitrary quantities of the same unit are compared, the result is a relative level change. If a stand-
ard reference quantity is used in the denominator of the ratio, the result is an absolute level and the
unit is dB relative to the original unit. Relative levels are useful for live work. Absolute levels are use-
ful for equipment specifications and calibration. Figure 1.4 lists some references used for determining
absolute levels.
The decibel was originally used in impedance-matched interfaces and always with a power reference.
Power requires knowledge of the resistance that a voltage is developed across. If the resistance value
is fixed, changes in applied voltage can be expressed in dB, since the power developed will be directly
proportional to the applied voltage. In modern sound systems, few device interfaces are impedance
matched. They are actually mismatched to optimize the voltage transfer between components. While
the same impedance does not exist at each device interface, the same impedance condition may. If
a minimum 1:10 ratio exists between the output impedance and input impedance, then the voltage
transfer is essentially independent of the actual output or input impedance values. Such an interface
is termed constant voltage, and the signal source is said to be operating open circuit or un-terminated. In
constant voltage interfaces, open circuit conditions are assumed when using the decibel. This means
that the level change at the output of the system is caused by changing the voltage somewhere in the
processing chain and is dependent on the voltage change only, not the resistance that it is developed
across or the power transfer. Since open-circuit conditions exist almost universally in modern analog
systems, the practice of using the decibel with a voltage reference is widespread and well accepted.
One of the major utilities of the decibel is that it provides a common denominator when considering
level changes that occur due to voltage changes at various points in the signal chain. By using the deci-
bel, changes in sound level at a listener position can be determined from changes in the output voltage
Part I
Basics
fIgure 1.
8 Some important decibel changes and the ratios of power, voltage, pressure, and distance that produce them. (Courtesy
Syn-Aud-Con.)
fIgure 1.4
Some common decibel references used by the audio industry.
of any device ahead of the loudspeaker. For instance, a doubling of the microphone output voltage
produces a 6 dB increase in output level from the microphone, mixer, signal processor, power amplifier,
and ultimately the sound level at the listener. This relationship assumes linear operating conditions
in each device. The 6 dB increase in level from the microphone could be caused by the talker speaking
6 dB louder or by simply reducing the miking distance by one-half (a 2:1 distance ratio). The level con-
trols on audio devices are normally calibrated in relative dB. Moving a fader by 6 dB causes the output
voltage of the device (and system) to increase by a factor of 2 and the output power from the device
(and system) to be increased by a factor of four.
