Professional Documents
Culture Documents
(Modern Acoustics and Signal Processing) Scott D. Snyder (Auth.) - Active Noise Control Primer-Springer-Verlag New York (2000) PDF
(Modern Acoustics and Signal Processing) Scott D. Snyder (Auth.) - Active Noise Control Primer-Springer-Verlag New York (2000) PDF
CONTROL PRIMER
AlP Series in
Modern Acoustics and Signal Processing
ROBERT T. BEYER, Series Editor-in-Chief
Physics Department, Brown University
EDITORIAL BOARD
YOICHI ANDO, Faculty of Engineering, Kobe University, Kobe, Japan
FLoYD DUNN, Bioacoustics Research Lab, University of Illinois,
Urbana , Illinois
JOHN ERDREICH, Ostergaard Associates, West Orange, New Jersey
CHRIS fuLLER, Department of Mechanical Engineering, Virginia Polytechnic
Institute, Blacksburg, Virginia
WILLIAM HARTMANN, Department of Physics, Michigan State University, East
Lansing, Michigan
IRA HIRSCH, Central Institute for the Deaf and the Department of Psychology,
Washington University, S1. Louis, Missouri
HERMAN MEDWIN, Naval Postgraduate School, Monterey, California
JOANNE L. MILLER, Department of Psychology, Northeastern University, Bos-
ton, Massachusetts
LARRY ROYSTER, Department of Mechanical and Aerospace Engineering,
North Carolina State University, Raleigh, North Carolina
JULIA DOSWELL ROYSTER, Environmental Noise Consultants, Raleigh, North
Carolina
WILLIAM A. VON WINKLE, New London, Connecticut
With 75 Illustrations
AlP , Springer
PRESS
Scott D. Synder
Department of Mechanical Engineering
University of Adelaide
Adelaide, South Australia 5005
Australia
Series Editor:
Robert T. Beyer
Physics Department
Brown University
Providence, RI 02912
USA
The use of general descriptive names, trade names, trademarks, etc., in this publication, even if
the former are not especially identified, is not ta be taken as a sign that such names, as under-
stood by the Trade Marks and Merchandise Marks Act, may accordingly be used freely by
anyone.
9 8 7 6 5 4 3 2 1
ISBN 978-1-4612-6437-8
To
Gill, Tom and Isaac
Series Preface
vii
viii Series Preface
member of the American Institute of Physics, and other related societies and
professional interest groups.
It is our hope that scientists and graduate students will find the books in this
series useful in their research, teaching, and studies. As James Russell Lowell
once wrote, "In creating, the only hard thing's to begin." This is such a begin-
ning.
Robert T. Beyer
Series Editor-in-Chief
Preface
Active noise control has become one of the most popular research topics in
the "engineering" domain, with hundreds of journal papers covering dozens of
associated topics reaching the academic press each year. However, despite this
research effort, the number of practical, commercial implementations reaching
the marketplace has been extremely slim. Apart from active headsets, and the
odd air conditioning and vehicle implementation, it is difficult to think of practi-
cal examples.
There are a large number of reasons for this lag between the commercial and
academic worlds. Active noise control systems are very complex, usually requir-
ing the designer to achieve some synergy between microelectronics, transducer
technology and physical acoustics; having the skills to do this requires signifi-
cant experience. Noise problems which are truly amenable to active control
solutions are not as widespread as many people think. I cannot, for example,
quiet your neighbour's dog, or stop traffic noise from entering the house built
next to a superhighway, or, in most cases, even provide a practical solution to
the problem of the noisy refrigerator. Even in instances where it does work, the
frequency range over which control can be achieved is usually quite limited. If
I dwell too long on all of these thoughts, I will be tempted to mutter a statement
along the lines "an expert in a useless field!"
However, having said all of these nasty things, I will say that when active
noise control works, it really works. There is almost a feeling of disbelief in the
audience when, for example, you reduce the level of the fundamental tone in a
commercial leaf vacuum by 30 dB, or the low frequency engine noise in a
vehicle cabin by a similar amount. The trick is to know when to apply the
technology, what problems are amenable.
This brings us to this book. This book grew out of a set of manuals and
papers a colleague of mine, George Vokalek, and I wrote to support an "active
control development kit." The aim of the kit was to provide the microelectronics
required for commercial designers to implement active noise control systems in
their various products. The problem was to give the designers some indication
of how active noise control actually (physically) worked, where it could be
IX
x Preface
applied, and what results could be reasonably expected without going into pages
of mathematical expressions. This book was my attempt at a solution. Since that
time, I have found it to be a useful introduction for new graduate students,
senior-level students undertaking active noise control projects, and secondary
and tertiary teachers looking for new ideas to aid the instruction of fundamental
physics. For those who are interested, there is an "experimental kit" which sup-
ports this text, available from the Michigan-based company Arbor Scientific:
www.arborsci.com.
In keeping with the aims, this book is short and descriptive, almost totally
without mathematical expressions. As the title indicates, it is meant to be a
"primer," an introductory text. It assumes that the reader has essentially no
knowledge of acoustics, signal processing, or noise control. Hopefully, after
reading the book, this will change.
Scott D. Snyder
Contents
1. Introduction ............................................................................... 1
Welcome to the World of Active Noise Control! ................................ I
Chapter Summary.. ..... ... ..... ...... ......... ....... ....... .... ... ... .... ... ..... ... ....... ... ... 2
Do I Have to Read the Whole Book? .... ....... ....... ....... ..... .......... ..... ..... 3
What Is Active Noise Control? ............................................................. 3
Adaptive Feedforward Active Control Noise ....................................... 4
Advanced Reading ......... ..... ............. .... ............ .... ...... ... .... ..... .... ....... ..... 5
Xl
xii Contents
• basic acoustics;
• basics of human perception of sound;
• sound power and related concepts;
• the fundamentals of passive noise control strategies for several classes of
problem (required for assessing whether passive control is a better option
than active control);
• the fundamentals of active noise control strategies for several classes of
problems;
• the basics of digital systems;
• the basics of adaptive controllers (to facilitate elementary operation and
tuning); and
• a more detailed description of the active noise control adaptive control
system (to facilitate "better" tuning).
Chapter Summary
Following this Introduction, the chapter contents are:
Chapter 2. Background: Fundamentals of Sound. This chapter provides a
brief discussion on the fundamental concepts which will be required knowl-
edge for understanding active noise control. Included in this discussion
are topics of frequency, waves, wavelengths, Fourier analysis, harmonic
signals, and human hearing.
Chapter 3. Fundamentals of Noise Control. This chapter provides a general
discussion on the topic of "noise control," both passive and active, from
the standpoint of the flow of energy.
Chapter 4. Free Space Noise Control. This chapter looks at the problem of
controlling noise radiating into a free space environment (basically, an
environment where there are no walls to impede the propagation of sound
waves). Both active and passive approaches are considered.
Chapter 5. Enclosed Space Noise Control. This chapter looks at the problem
of controlling unwanted sound fields in enclosed spaces (rooms, vehicles,
etc.). Again, both active and passive approaches are considered.
Chapter 6. Control of Sound Propagation in Ducts. This chapter looks at the
last general group of problems, those which involve sound propagating in
a duct (such as an air-conditioning system or car exhaust). As before, both
active and passive approaches are considered.
Chapter 7. Active Noise Controller Overview. This chapter provides an intro-
duction to adaptive feedforward active noise controllers, their operation,
and tuning. Included in this chapter is a basic discussion of digital systems
and their particular requirements, as well as a "heuristic" description of the
controller operation.
Chapter 8. Controller Fundamentals. This chapter provides a more detailed
description of the adaptive feedforward active noise controller, providing
Do I Have to Read the Whole Book? 3
1. Active noise control is only useful for certain types of problems. As will
be discussed, these are generally low-frequency problems, usually tonal,
with either simple or contained sound fields. It should be stated, how-
ever, that the sorts of problems which are amenable to active noise con-
trol are not uncommon in real life.
2. Active noise control is more complicated than passive noise control, in
that it involves the integration of electronics, transducers (loudspeakers,
microphones, etc.), and acoustics.
3. There are not a great number of noise control practitioners who have
experience in active noise control.
One of the aims of this book is to make active noise control more acces-
sible, enabling the technology to spread.
Fan noise
><><X
noise
\folse
.... .........-
~
....;..-
Reference Control Error
microphone source microphone
Control
system
FIGURE 1.1. Main components in a typical adaptive feedforward active noise control
system.
Advanced Reading 5
stream" of the control system, it can perform this required a priori sampling
of the noise.
A control system, which is responsible for taking the reference signal mea-
surement of the impending noise and calculating what is required to cancel
it. The control system must be fast enough to do this calculation before the
unwanted noise arrives.
A control source, which is used to generate the canceling sound field. In this
case, the control source is a loudspeaker.
An error microphone, which is used to sample what noise actually remains
after the cancellation operation. This measurement is referred to as the
error signal because it provides an indication of how much the controller
was "in error" when it derived the canceling waveform. If the calculation
was perfect, then the unwanted noise would have been completely can-
celed and the "error" would be zero.
While it is notionally straightforward to envisage the job of the control
system ("invert" the measured signal, so that it cancels the sound wave when
it arrives), the actual calculation procedure is quite difficult. This is because
the change in the noise disturbance as it propagates down the duct must be
accounted for, as must be change in the signal as it passes through the refer-
ence microphone and the change in the controller output as it passes through
an amplifier and loudspeaker. These various changes will themselves change
over time, as microphones age, as air speeds and temperatures change, and
even as fungus grows on the loudspeakers and microphones! Based upon
these considerations, it is apparent that the controller must be self-tuning, or
adaptive. That is, it must be able to adjust its calculation procedure to suit
the current environment in which it is operating. The error signal measure-
ment is actually used by the controller to do this adaptation, via the use of an
adaptive algorithm. The adaptive algorithm attempts to adjust the output
calculation such that the sound field measurement at the error microphone is
"0," meaning that there is complete cancellation at the error microphone.
In order to make an active noise control system work, or even to determine
if it is physically capable of solving a target problem, it is necessary to have
a good knowledge of the "acoustics" of the problem. Do not be misled by the
concepts of "adaptive control" and "an algorithm that tunes the controller to
get the best result." The controller can only work within the bounds defined
by the fundamental physics. It is therefore appropriate to begin our more
detailed discussions with chapters that address the fundamentals of acoustics
as required for active noise control.
Advanced Reading
If after reading the Primer you wish to consider active noise control in more
depth (equations and all), there are several books that delve into explicit
detail. These include:
6 1. Introduction
C.H. Hansen and S.D. Snyder (1997). Active Control of Noise and Vibration. E&FN
Spon: London.
C.R Fuller, SJ. Elliott, and P.A. Nelson (1996). Active Control of Vibration. Academic
Press: London.
S.M Kuo and D.R. Morgan (1996). Active Noise Control Systems. Wiley: New York.
P.A. Nelson and SJ. Elliott (l992).Active Control of Sound. Academic Press: London.
2
Background:
Fundamentals of Sound
What Is Sound?
Sound is the sensation produced at the ear by very small pressure fluctua-
tions present in the surrounding medium (which we will assume to be air).
This sensation is produced in response to the pressure fluctuation-in-
duced vibration of the ear drum. The fluctuations in the surrounding air
constitute a sound field. The pressure fluctuations themselves are usually
referred to as sound pressure or acoustic pressure.
We will discuss ways of quantifying a sound field, of putting a number
to "how big" it actually is, shortly. However, to arrive at some idea about
the size of the fluctuations, note that the sound field pressure fluctua-
tions are placed on top of (static) atmospheric pressure. This is similar to
saying that the ocean floor is continually subject to the "average," or
static, pressure from the weight of the ocean, along with the much smaller
pressure fluctuations which result from a change in ocean depth due to
the waves on the surface. If you were to fill a 1 liter softdrink bottle with
water and balance it on one finger, the pressure on your finger is similar in
magnitude to atmospheric pressure. In a healthy person, this pressure is
present on both sides of the ear drum. If you now take a standard hole-
punch and extract a small paper circle from a piece of photocopy paper,
the pressure caused by placing this paper circle on your finger is similar
in amplitude to the pressure fluctuations present during a loud conversa-
tion. The conclusions are that (1) sound pressure fluctuations are pretty
small, and (2) our ears are pretty sensitive to be able to detect the pressure
fluctuations.
You might now be wondering about the limits of human hearing. If you
were to take the above paper circle and cut it into something like 40,000
tiny pieces, a healthy young child would be able to detect a sound pres-
sure fluctuation that is similar in amplitude to the pressure caused by one
of these pieces placed on the end of your finger.
7
S. D. Snyder, Active Noise Control Primer
© Springer Science+Business Media New York 2000
8 2. Background: Fundamentals of Sound
What Is Noise?
Noise is unwanted sound. What constitutes noise is therefore subjective; what
is "music" to your ears might be "noise" to your parents, and what is an
inconsequential byproduct of one group's activities (such as a rock concert or
auto race) may be an outrageous production of noise to another group. One
type of sound which is generally agreed upon as being noise is that which
comes from industrial processes and, in particular, sound from machines:
punches, saws, sausage making machines, etc. However, even this may be
music to the factory owner's ears, as it may represent money being made.
Why is it important to do something to limit the unpleasantness of noise?
For centuries it has been recognized that noise is a hindrance to (or perhaps a
symptom of non-) peaceful coexistence between humans. In more modern
times, noise has been recognized as a serious health hazard, with significant
financial compensation paid to those whose hearing has been impaired by
their job or circumstances. This recognition, and the resultant penalties for
noncompliance with accepted standards, has provided the impetus for an
entire area of commercial activity and research: noise control. This text is
really a spin-off from this field.
The range and sensitivity of the ear led workers at the Bell Laboratories to
define a new unit to quantify acoustic pressure. The unit was originally called
the "Bel" in honor of their founder (Alexander Graham Bell), but was later
modified to become the decibel (deci = 10, so a decibel is equal to 10 Bels).
The decibel is commonly abbreviated as "dB." The decibel scale is a logarith-
mic scale. This means that the units increment proportional to the logarithm
of the quantity of interest. Sound pressure, denoted L p, in decibels is defined
by the relationship
Table 2.1 lists some common settings and scenarios for a range of sound
pressure levels.
Note that when expressed in the decibel scale, a sound pressure increment
of 20 dB is actually an increase in pressure amplitude by a factor of 10. For
example, a pressure amplitude of 1 Pa is 20 dB greater than a pressure ampli-
tude of 0.1 Pa. Does this mean that if the sound level is increased by 20 dB we
will perceive it to be ten times as loud? Or, if the sound level is decreased by
20 dB, will we perceive it to be one-tenth as loud? The answer is no. Tests on
humans have revealed the subjective assessment results in Table 2.2.
TABLE 2.1. Examples of typical settings for a range of sound pressure levels.
Sound
Pressure Subjective
Level (dB) Examples assessment
140 Artillery noise at gunner's position Pain
120 Rock concert (front rows), front of heavy
industrial presses
100 Industrial settings Very noisy
80 Shouting, next to a busy road Noisy
60 Speech levels, restaurants, shopping malls
40 Quiet residential Quiet
20 Recording studio Very quiet
o Threshold of hearing for a healthy young person
10 2. Background: Fundamentals of Sound
It is interesting to consider what the above results imply for the world of
advertising. A company might say that their new and improved product, for
example, a dishwasher, is so quiet because sound pressure has been reduced
by 50%. Well, a 50% reduction in sound pressure is a drop of 6 dB (try this
calculation, for example, for the change of 0.1 Pa to 0.05 Pa, and for 0.01 Pa
to 0.005 Pa). From the above table, we see that this reduction is barely percep-
tible. However, 50% certainly sounds nice on the surface. The moral of the
story is, beware of companies quoting sound reductions as percentages rather
than as decibels.
Sound Waves
We have already stated that a sound field is simply a group of small pressure
fluctuations. The next point to consider is, how did they get there? Further,
what happens to the pressure fluctuations as time progresses?
Sound pressure fluctuations are most commonly generated by something
which is vibrating. An example of this is a loudspeaker, where the sound is
"generated" by the vibrating loudspeaker cone. However, vibrations are not
the only way to generate the required pressure fluctuations. They may, for
example, be generated by some aerodynamic phenomena. This situation is
commonly encountered with large saw blades, where the forced motion of the
air around the teeth is responsible for significant noise generation.
Let us consider the case of a vibrating surface in a bit more detail, as it is
the simplest to visualize. As shown in Figure 2.1, if a vibrating surface placed
in air is at equilibrium (neither bent in or out, but rather in the middle at its
resting position) the air next to it will be at its equilibrium, atmospheric,
pressure. Suppose now the vibrating surface moves in an outward direction.
Initially the air immediately in front of it is not moving, and so is compressed
somewhat. The number of air molecules in front of the vibrating surface is the
same, but the volume available to them has decreased owing to the presence
of the outward-curved surface. The result: a pressure fluctuation is born! As
the vibrating surface reverses direction and heads inward, it passes once again
through equilibrium and then beyond. In the process, the air is "stretched"
(rarefaction, as the volume available to the air molecules is increased). An-
Sound Waves 11
other pressure fluctuation is born, this time being a small reduction in the
equilibrium value.
If we were to measure the pressure in front of the vibrating surface, say with
a microphone, we would see a series of equilibrium/compression/equilib-
riumlrarefaction levels, as shown in Figure 2.2. This looks very much like a
wave. In fact, this type of plot is referred to as a waveform.
Once the vibrating surface generates a pressure fluctuation, what happens
to it? Does it just sit there, waiting to be "undone" by subsequent panel
motion? The answer is no. The pressure fluctuation moves away from the
panel. This is analogous to what happens when you throw a stone into a pond.
The water in front of the stone is pushed in and then bulges out, and this
motion travels away from the point of contact as a wave.
This is precisely how sound pressure fluctuations move away from the
source: as waves.
M ure
3 3
FIGURE 2.2. Time history of pressure in front of the vibrating surface, showing alternat-
ing areas of (1) equilibrium, (2) compression, and (3) rarefaction.
12 2. Background: Fundamentals of Sound
While sound may travel as waves, the waves do not resemble the plot in Figure
2.2. That is, they do not look like a vibrating piece of guitar string. Rather than
moving as a bending wave (with displacement perpendicular to the direction of
travel), sound pressure fluctuations travel as longitudinal, or compression, waves.
This type of wave can also be seen, for example, in light springs which are shaken
on their ends (a "slinky"). If you could see an acoustic (sound) wave it would
look something like that which is shown in Figure 2.3, with areas of compression
and rarefaction moving away from the sound source.
It is intuitively obvious that sound pressure must travel as a compressional
wave; you cannot "bend" air, you can only compress and rarefy it. However, it
is worthwhile putting a more scientific rationale to this thought, as many
types of disturbances travel as waves and the rationale can be applied to all.
Consider a bending wave propagating along a piece of string, as shown in
Figure 2.4. How is it that the wave can travel? If we were able to look at little
"pieces" of the string, we would find that each piece is sliding against its
neighbor and so acting to push it up or down. This action is referred to as
shear. For a bending wave to travel, the medium in which it is traveling
(string, air, whatever) must be able to support shear stress. That is, the indi-
vidual medium elements must be able to slide against one another without
having the whole thing fall to pieces. Metal bars and pieces of string can
support shear stress, but air cannot. Technically speaking, the viscosity of
most gases and liquids is too low to support shear stress, and therefore bend-
ing waves cannot exist to any significant degree.
If bending waves cannot travel in air, then can compression waves (acous-
tic waves) travel in things like metal bars? Yes, they can. Practically anything
can be compressed to some degree, which is the requirement for propagation
of a compressional wave.
So, sound pressure fluctuations travel away from their source as waves. Is
there any particular speed with which they travel? The answer is yes, al-
though the speed will change with variations in temperature, pressure, and
constituents of the medium in which the wave will propagate. The speed of
FIGURE 2.3. Typical acoustic (compressional) wave, showing areas of compression and
rarefaction.
Sound Waves 13
Bending wave
in a string
Itl
string "elements"
FIGURE 2.4. A bending wave propagates by having one "element" slide up/down against
the next "element"; this is a "shear" motion.
You can also use the speed of sound to soothe frightened children in a
thunder storm. Once the lightning flashes, count the seconds until the thun-
der arrives. For approximately every 5 seconds counted, the storm is a further
mile away, so "no need to worry" (this strategy for soothing children can
backfire sometimes if the storm is close).
Frequency Analysis
If you are a witness to a robbery, and the police asked you to describe the
voice of the assailant, you would certainly include some mention of "pitch."
For example, you might say that he or she had a "deep, low" voice or a "high,
squeaky" voice. In fact, after the notions of "loud" and "soft," the description
of "high" or "low" pitch is probably the most common way to subjectively
qualify the characteristics of sound and to communicate "what something
sounds like."
What you are in fact doing when you describe a sound as high or low
pitched is making a statement about the frequency content of the sound field.
Frequency content is important for a whole host of reasons, ranging from how
a human perceives a given sound pressure level, to what equipment is re-
quired to reproduce sound, and to what techniques can be used to control
unwanted noise. It is therefore important that we be able to quantify fre-
quency, to assign it a number in the same way we did with sound pressure
amplitude so that we can communicate and work with the notion of "low
pitch/high pitch" in a scientific way.
So what actually is frequency? Frequency is a measure of how many times
an acoustic wave moves from compression to rarefaction and back again in a
given period of time (the standard time increment is 1 second).
Consider Figure 2.5. Here a sound wave is propagating past a given point,
and the sound pressure is being monitored with a microphone. In this in-
stance, the time between successive peaks is constant (we will discuss this
~
::l
CJ)
CJ)
~
a.. L...--+-+---t-"'T"\ \ _L..--\----?
Time
FIGURE 2.6. A more "typical" acoustic (sound) pressure measurement: what a mess!
point in more depth shortly). Over the course of I second, we find that there
are 137 transitions from compression to rarefaction and back again. There-
fore, the frequency of the acoustic wave is 137 Hertz (abbreviated Hz).' The
units of Hertz are also known as cycles per second. A wave with a frequency of
137 Hz will cycle between maximum/minimum/maximum amplitudes 137
times in I second.
This notion of measuring frequency by counting transitions between com-
pression/rarefaction/compression works when the acoustic wave is "nice and
smooth" (we will quantify this description shortly), as was the case in Figure
2.5. However, in the real world, it is rather unusual to encounter such a "pure"
acoustic wave. It is entirely possible, maybe even probable, that the time-
history of the sound pressure measurement will look something like that
which is shown in Figure 2.6. Subjectively, this is a mess! How do we assign
a frequency to this short of signal? The answer is, we don't. At least not a
single frequency. Rather than talk about the "frequency" of the acoustic wave,
we talk about the "frequency content." To expand upon this, it is necessary to
discuss the concept of a sine wave and the work of Fourier.
Sine Waves
The concept of a sine wave and the work of Fourier are absolute cornerstones
of scientific and engineering thought, and indeed of technology as we know
it. Practically any technical advancement, from the "simple" generation of
electricity to the extraordinary feats of space probes, would lack description
without them.
1 The units of Hertz were named in honor of German physicist Heinrich Rudolph Hertz,
who was the first person to produce electromagnetic waves artificially.
16 2. Background: Fundamentals of Sound
Consider the arrangement shown in Figure 2.7. Here we have a bar pinned
at one end and rotating in a circle. At any point in this rotation, we can
calculate the value of the sine of the angle between the horizontal and the
bar position. From simple trigonometry, the sine of the angle is the ratio of
the length of a line drawn straight up or down (perpendicular) between the
horizontal and the tip of the bar (denoted as x, or the "opposite"), and the
length of the bar itself (the "hypotenuse"). This value of sine will be positive
if the x line is pointing straight up and negative if the x line is pointing
straight down.
If we plot the value of the sine of the angle starting from an angle of
o degrees (when the bar is lying flat) and rotating completely around the
circle (through 360 degrees), we get a plot which looks something like that
which is shown in Figure 2.7. But we have encountered this shape before: it is
exactly the same shape as the plot of sound pressure amplitude for our "nice
and smooth" acoustic wave in Figure 2.5. Hence we call this "nice and smooth"
wave, the pressure amplitude of which varies over time in exactly the same
way as the sine of an angle varies as the bar is rotated, a sine wave. This sort
of wave is also referred to as tonal, or harmonic (although harmonic tends to
be more general, as we will discuss shortly).
Let us now expand our thought experiment to include some measure of
time. Suppose instead of plotting the sine of the angle for all bar positions in
the circle of rotation, we "spin" the bar at a certain speed, quantified by the
number of revolutions which take place in 1 second, and plot the sine of the
angle for all bar positions during that 1 second period. As shown in Figure
2.8, if the bar undergoes, say, four complete revolutions in 1 second, then we
expect to see the sine wave pattern repeated four times. Technically, the bar is
spinning at 4 Hz, or four cycles (complete revolutions) per second. If the bar
speeds up, to, say, eight revolutions per second then we expect to see eight
repetitions of the sine pattern over the 1 second interval; the bar is now
spinning at 8 Hz. The plots for 4 Hz and 8 Hz have the same general shape
:;
'jg ~h
V>R>J'
Q)
i
a. g> Angle
FIGURE 2.7. The plot of the sine of the angle between the rotating bar and the horizontal
edge.
Sine Waves 17
' slow" =
~ 4 Hz
II
Q)
Ol time (s)
C
tU
'5
Q)
~ c
'iii U5
8.
a.
t:: "fast" =
~________~______~ X
8Hz
FIGURE 2.8. Two plots of the sine of the angle for all bar positions over a 1 second period
for two different rotational speeds.
Phase =900
Peak amplitude
RMSamplitud
j
.l, Phase =180 0
t Phase =0 0
Phase =270 0
2 How can we have "negative pressure" in real life? Remember that the acoustic pres-
sure is simply a small amplitude perturbation of the background (atmospheric) pressure;
regions of negative acoustic pressure, that occur in the rarefaction part of the sound
field, are simply slight reductions in the atmospheric pressure and are still "positive" in
absolute terms.
Fourier Analysis 19
Fourier Analysis
As was mentioned, there are very few "pure" sinusoidal acoustic waves. So you
might be wondering, what is the use of this type of quantification?
The true value of the notion of a "sine wave" becomes apparent when consid-
ering the idea of a Fourier transform. Back in the nineteenth century, Joseph
Fourieil arrived at the conclusion that any steady-state waveform can be de-
scribed as the sum of a number of sine waves with differing amplitudes and
phases. So, for example, the complex waveform shown in Figure 2.10 is actually
FIGURE 2.10. A complex waveform that is the sum of two sine waves with different
amplitudes and phases.
31t is interesting to note that the early nineteenth century French mathematician J.B.
Fourier, the "discoverer" of the Fourier Transform, did not work with noise at all; he
was interested in the transfer of heat between objects.
20 2. Background: Fundamentals of Sound
the sum of two independent sine waves. More complex waveforms will be the
sum of even more sine waves. Even "random noise" can be described as the sum
of sine waves. The description of random noise is similar to that of white light, in
that it is the sum of "all" sine waves in the frequency range of interest.
This notion that any waveform can be described as the sum of a group of
sine waves is a powerful tool for studying and quantifying sound. It provides
us with a mathematical way of explaining a variety of phenomena, ranging
from why you can identify high- and low-pitched components in a complex
sound field (such as coming from an orchestra), to why a recording sounds
"tinny" when played through poor quality loudspeakers, and to why active
noise control works in some instances and not others (all of these will be
answered later in this book). In fact, sound fields are most commonly de-
scribed in terms of their spectrum, which is the variation in amplitude (and
possibly phase) of the components of the waveform, ordered in terms of (sine
wave) frequency (see Figure 2.11) . The analysis of a waveform in terms of its
constituent sine waves is referred to as spectral analysis,frequency analysis,
or Fourier analysis. 4 The sine waves which make up a given waveform are
referred to as the frequency components of the signal. For example, we might
say that a sound pressure field has "significant 120 Hz and 150 Hz compo-
nents," meaning that the set of sine waves that make up the measured wave-
form of the sound field include sine waves with frequencies of 120 Hz and
150 Hz, and that the amplitudes of these sine waves are relatively large.
Fourier not only arrived at the amazing conclusion that all waveforms can
be described by the sum of sine waves, but also developed a mathematical
way of working out what the frequencies, amplitudes, and phases of the sine
Q)
u
-HlHlfUI-Il-ft-il-'rit-tt-tthHIHt-~ time (s) .€
a.
E
<t:
Frequency
Measured waveform
Spectrum
(time series data)
4It is interesting to note that the human ear performs a sort of "physical" Fourier
analysis in its processing of sound, with different parts of the ear responding to the
different frequency components in the sound field.
Harmonics 21
waves are. This technique is known as the Fourier Transform. There is also an
"opposite" technique which is used to turn a group of sine waves into a
waveform. This latter technique is called the Inverse Fourier Transform.
While the Fourier transform has been a cornerstone technique in many
areas of engineering and mathematical science since its inception, the "prac-
tical" use of spectral analysis has truly blossomed in the last 30 years as a
result of the development of technique for fast calculation of the Fourier
transform: the Fast Fourier Transform, or FFT. 5 The FFT calculates the ampli-
tude and phase of frequency components which are evenly spaced across the
spectrum. For example, the FFT might calculate the amplitude and phase of
the 200 Hz, 202 Hz, 204 Hz, ". frequency components. What happens if the
actual frequency is not exactly one of those in the FFT calculation (such as
201 Hz for the example above)? Essentially, it is placed in the bins of the
closest frequency. The size of the bin for a given stated frequency encom-
passes the range of frequencies that lie from half-way between the bin of
interest and the one below it, to half-way between the bin of interest and the
one above it. For the example above, the 202 Hz bin encompasses frequency
components between 201 Hz and 203 Hz. The frequency which defines a
given bin lies in the center of the bin, and is referred to as a center frequency.
The results returned by the FFT for a given bin include contributions from all
frequencies which lie in the bin.
Harmonics
In the study of sound and vibration, and important concept is that of harmon-
ics. Harmonics are frequency components which are integer multiples of some
"fundamental" harmonic frequency. For example, if 100 Hz is the fundamen-
tal harmonic frequency, then 200 Hz is the second harmonic, 300 Hz is the
third harmonic, 400 Hz is the fourth harmonic, etc. If harmonics are plotted
in terms of their frequency, they appear as evenly spaced "spikes" as shown in
Figure 2.12.
If a waveform is periodic (that is, if it is made up of a pattern that repeats
itself every cycle), then the spectrum of the waveform will contain a series of
harmonics. The simplest example is that of a sine wave, that contains a single
harmonic (the fundamental). If the waveform is perfectly square, then it
contains all odd-numbered harmonics (1, 3, 5, etc.), as shown in Figure 2.13.
If the waveform is some other periodic shape, it will be constructed from
some other combination of harmonics. Note that the number of times the
pattern repeats itself every second will be the frequency of the fundamental
harmonic.
5The Fast Fourier Transform was first outlined in-depth only a quarter of a century
ago, in The Fast Fourier Transform, by E. Oran Brigham (Prentice-Hall, Englewood
Cliffs, NJ).
22 2. Background: Fundamentals of Sound
Frequency
Spectrum
OJ
"C
time (s) ~
a.
E
..... ..... «
"-- "-- I..-
Frequency
Measured waveform
Spectrum
(time series data)
/
A pressure pulse occurs
each time a blade passes
Measured waveform
(time series data)
that the frequency content (from the individual sound sources) is driving the
shape of the periodic waveform. What is harder to grasp, and what is far, far
more common, is for the shape of the waveform to drive the frequency con-
tent. In order for a given periodic waveform to exist, it must include multiple
harmonics. That is the only possible way to obtain a repeating shape on a
cycle-to-cycle basis. Additional sound sources are not required to produce
this shape. Consider, for example, a loudspeaker which is outputting a single
frequency. If you place your finger on the loudspeaker you will distort the
shape of the sound wave. You haven't introduced additional sound sources,
but you have altered the harmonic content of the sound field simply by alter-
ing the shape of the periodic waveform.
This may all sound like a "mathematical" concept aimed purely at describ-
ing the shape of a wave. What makes it all so important, though, is how
"physical" things respond when the wave impinges upon them. To a physical
object, the source of the wave is irrelevant. All that a physical object sees is
the multiple harmonically related frequencies which make up the wave. Your
ear is the classic example. If you put your finger on the loudspeaker and
distort the shape of the output sound wave, what do you hear? Usually, it is
"buzzing." You aren't physically driving the speaker at the frequency of the
buzzing. However, the buzzing will be a harmonic of the frequency at which
you are driving the loudspeaker, a harmonic frequency which is mathemati-
cally necessary to produce a wave with the distorted shape. If you change the
position where you touch the loudspeaker, and so change the distorted "shape"
of the output sound wave, then you will also change the harmonic content of
the signal. As a result, it will sound different.
The question of why physical objects respond to different (sine wave)
frequency components in different ways is beyond the scope of this text. It is
a fundamental component of the study of Mechanical Engineering. It will
suffice here to say "that is the way it is," and that everyone has a salient
example: their ear.
Returning to our discussion of sources of harmonics, fans are not the only
pieces of rotating machinery that produce harmonics. Other examples in-
clude pumps (for example, in a swimming pool), engines, aircraft engines
(propeller aircraft can be especially painful), compressors (think of the large
units used by road crews!), and in fact anything that rotates. To a listener, the
"low"- frequency components (we will classify these shortly) sound like a
"rumble," while the high-frequency components sound like a "rattle."
ure Cf!v?Y
coming out
the appliances may generate sound fields which are periodic, with frequency
components that are harmonics of the mains frequency (50 Hz or 60 Hz). This
is true for many low-cost electrical motors and fans. However, it is also true for
a large number of nonrotating devices such as transformers.
Transformers vibrate in response to changes in their internal magnetic
field. As this magnetic field is generated by the electricity entering the trans-
former, it is intuitively obvious that the vibrations must be, in some way,
related to the frequency of the electric current. They are, in fact, harmonically
related. The strength of the magnetic field increases and decreases in re-
sponse to the current "wave." It also changes direction, from "north" to "south"
(to adopt the terminology commonly used to describe the two ends of a bar
magnet). However, to the metal components of the transformer, whether the
field is north or south has no bearing upon the physical response. They only
see "strong" and "weak." (Recall that while north and south magnets attract,
and north and north magnets repel, both north and south magnets will attract
steel. The concept of "repelling" is only important if there are two magnetic
fields.) The result is that the fundamental frequency of vibration is twice the
frequency of the current. This can be either 120 Hz (in the United States) or
100 Hz (Australia). We call the pattern of the resultant vibration rectified.
That is, there is no "negative" component, as shown in Figure 2 .15.
As before, while being periodic, neither the vibration nor the resulting
sound pressure field is perfectly sinusoidal (it is rectified). Therefore, it con-
tains harmonics. Most people are familiar with the "buzz" caused by the
higher frequency harmonics in the sound pressure field generated by a trans-
former and/or power supply in an appliance.
stops," where the cone travels the maximum possible distance and is then
physically restrained, the waveform produced by the loudspeaker will be
distorted. This distortion results in the introduction of harmonics into the
signal. (The degree to which this happens when the loudspeaker is not driven
to its stops is often referred to as harmonic distortion by manufacturers. This
provides a measure of the quality of the sound production.) Most people
would be familiar with the "buzz" produced by a small or low-cost loud-
speaker which is driven to its limits by sound with a large "bass" component.
This is the generation of high-frequency harmonics. The generation of har-
monics and periodic sound fields will take on greater significance after we
discuss how human beings perceive sound.
Side Story. Have you ever noticed that a train appears to change frequency as
it comes toward you and eventually passes you? (Steam or diesel trains that
is. The trains which produce a loud harmonic sound field.) This is due to a
phenomenon on known as Doppler shift. As the train approaches, it is produc-
ing pressure fluctuations at a given rate (say, 30 fluctuations per second for
argument's sake). If the sound source was standing still, subsequent peaks
and troughs of the acoustic waves would arrive at fixed intervals of time (a
new peak would arrive every 1130 of a second). However, when the train in
coming toward you, it is decreasing the distance over which subsequent waves
have to travel. This means that subsequent waves will take less time to arrive.
The result is that new peaks arrive at time intervals that are shorter than 1130
of a second. Maybe, say, 32 waves arrive in the space of a second, rather than
30. To the human receiving the sound, it appears that the frequency is now 32
Hz, not the original 30 Hz. The frequency appears (and is, in fact) increasing!
The amount of increase depends upon the proportional change in distance
between you and the train, that occurs between the subsequent generation of
sound waves. This proportional distance change increases as the train gets
closer. The result is that the train noise increases in frequency as the train gets
closer. The converse is also true: the train noise decreases in frequency as the
train goes past and gets farther away. This change in frequency is the Doppler
shift.
This shift in frequency also applies to light (in fact, the concept of a Dop-
pler shift is more often applied to light than sound). Light can be split into a
spectrum in the same way as sound: red light is a "low"-frequency sine wave,
and blue light is a "high" -frequency sine wave. If light is being emitted by an
object that is moving away from you, then the light is made just a little bit
more "red" (the frequency drops just a little). Scientists can use this phenom-
enon to tell that the Universe is expanding. By looking at the spectra of light
from distant stars, and comparing it to the spectra of light from closer stars of
similar composition, it is apparent that distant stars have spectra with compo-
nents that are lower in frequency than components found in the spectra of
closer stars (the spectrum experiences a "red shift"). The conclusion is that
the stars are moving away from us and the Universe is expanding.
Human Perception of Sound 27
Side Story. An interesting phenomena occurs when two sine waves with al-
most the same frequency and same amplitude are put together. Over the course
of many cycles, the peaks of the two sine waves come together and draw apart.
As a result, the pressure amplitudes will add together, cancel out, add to-
gether, cancel out, setting up a low-frequency "beating" phenomena. As the
two frequencies get closer together the beating will slow, eventually stop-
ping when the frequencies are identical. This is a very convenient way to
match two frequencies, and is a phenomenon which is exploited by most
guitar players to tune their instruments.
6 It is interesting to note that this frequency of peak energy output is changing, particu-
larly for females. Regular surveys conclude that the female voice is becoming lower in
frequency, moving toward the frequency of the male voice. This trend poses interesting
questions for sociologists studying the "role" of women in society. It is well known that
babies prefer higher-pitched voices, while "power" figures in society are associated
with lower-pitched voices. Acoustics provides fuel for debate!
Human Perception of Sound 29
Humans are not, in general, good judges of pitch. Research has found that
humans tend to perceive frequencies below approximately 500 Hz to be higher
than they actually are, and frequencies above approximately 500 Hz to be
lower than they are. The difference between human perception and the actual
frequency decreases as the sound pressure level is increased.
It was mentioned that humans do not perceive frequency in a linear fashion.
For this reason, acoustics and noise control "professionals" tend to work with
sound split into frequency "bands" which are logarithmically related (as op-
posed to the linearly related bands of the Fast Fourier Transform, at, for example,
200 Hz, 202 Hz, 204 Hz, etc.) There are a variety of possible ways to construct
these bands. The industry standard bands, which are those agreed to by the Inter-
national Standards Organization, are based upon increments in frequency of 10",
where n is a single decimal number ranging from 1.4 to 4.3. The most common
band, called an octave band, ranges across three of the above increments (for
example, n = 1.4, 1.5, and 1.6 comprise the 31.5 Hz octave band). The next most
common band is the one-third octave band, which uses just one increment.
The preferred frequency bands in the range from 100 Hz to 1250 Hz are listed
in Table 2.3. Note that n defines the center (or middle) frequency of the band, and
that the upper and lower limits placed on the band lie half-way between adjacent
bands. Note also that the band details repeat themselves with a multiplication of
ten. For example, the limits on the 1000 Hz one-third octave band are ten times
those on the 100 Hz band. The limits on the 10,000 Hz one-third octave band
would be ten times those of the 1000 Hz band, and so on. Finally, octave bands
can be viewed as comprised of three one-third octave bands, as indicated above.
For example, the 125 Hz octave band comprises the 100 Hz, 125 Hz, and 160 Hz
one-third octave bands. The frequency limits of an octave band range from the
lower band limit on the lowest constituent one-third octave band and the upper
band limit on the highest constituent.
One of the most important phenomena in noise control is that, while hu-
man beings can detect frequencies in the range of 20 Hz to 20 kHz, they do
not perceive all frequencies equally. Even more importantly, the potential for
various frequency components to damage human hearing is not uniform across
the frequency range. These two facts inherently govern almost all areas of
noise control and noise control legislation.
What do we mean when we say "humans perceive different frequencies
differently"? Basically, if a sound pressure field has a given amplitude, say of
1 Pa (94 dB), then it will appearlouder to the listener if thefrequency is 1000
Hz than if the frequency is 100 Hz. Further, it will appear much louder if the
frequency is 1000 Hz than if the frequency is 30 Hz. The amplitudes are the
same, but the perception is different.
This difference in perception has lead to the development of a set of stan-
dard "weighting curves" that can be applied to a given sound pressure mea-
surement to make it better reflect human perception of the sound. These curves
have the somewhat dubious titles of "A" (for sound levels below 55 dB), "B"
(for sound levels in the range 55 dB-85 dB), and "e" (for sound levels above
85 dB). By far the most commonly encountered of these is the "A" curve, not
because most sound fields are below 55 dB, but because the A-weighting
curve most closely predicts the damage that can be caused to hearing by a
given sound field. That is, the potential for a sound field to damage human
hearing is a function of the frequency content of the field, and this function
closely matches the A-weighting curve.
A plot of the A-weighting curve is given in Figure 2.17, with some actual
numbers listed in Table 2.4. Note that the low frequencies are heavily dis-
counted in their contribution to the A-weighted average sound pressure level,
while the mid-range frequency components are left essentially unaltered. This
has important implications for active noise control, as will be discussed later.
How can the A-weighting curve be applied to a measurement? One way is
as depicted in Figure 2.18. First, break the measurement into one-third octave
bands, and then add or subtract the number of decibels shown in the weight-
ing curve on a band-by-band basis. For example, referring to Table 2.3, the
measurement in the 100 Hz one-third octave band would have 19.1 dB re-
moved from it, the measurement in the 125 Hz one-third octave band would
have 16.1 dB removed from it, etc. Finally, add the weighted levels together
to get a weighted RMS (average) level. The weighting average is denoted
with an "A"; for example, 85 dB(A). Professional measuring equipment does
this sort of procedure automatically.
o
.......
CD
"0 -10
-
'-'
....
o
()
co
-
-; -20 A-weighting curve
c
..c:
Ol
.Ci.i
S -30
-4 0 t.....L....J......J....J.....L....L.....J.......L....J'-'--'--'-J.....L...L...J....L...L....L.....J.--L-L.J.......J
has been known for years that smoking is damaging to an individual's health,
and yet most people know someone who has smoked heavily for decades with
little or no apparent bodily damage. Conversely, there are individuals who
apparently develop lung cancer in response to relatively small amounts of
passive smoking. Hearing damage due to environmental noise is much the
same. Criteria for "what is acceptable" in terms of an individual's exposure to
sound levels, and how to modify the allowable exposure time in response to
increasing or decreasing sound levels, are derived from empirical data. The
subjectivity that is inherent in the assessment of empirical data has meant
RMS value =
87 dB
Gl
"0/\
/1 :::l
.t:: _~ RMS value =
I....J
TIme
~ ~
<{
~Frequency> 84 dB(A)
"Raw" Apply
measurement A-weighting
that different criteria have been adopted in different countries. Noise control
professionals must be fluent in the regulations of their specific areas. The
specific values that will be given in this section will not be correct for all
countries. However, the general framework of the criteria and the "ballpark"
numbers will be globally applicable.
So what is an acceptable level of sound (noise)? Criteria used to answer
this are based on the following criteria:
1. the risk of hearing damage to an individual must be small;
2. the reduction in work efficiency due to background noise must be small;
3. speech is possible (where "necessary"); and
4. noise levels in the community must be acceptable.
A cynical person might call these the "motherhood statements of noise control."
(insofar as hearing damage goes-if you loose only one of ten toes, is this
acceptable?). Once the hearing loss is greater than 25 dB then the individual
has serious impairment. If the loss of hearing perception is in the range of 25
dB-92 dB, then the individual is said to be hearing impaired with the degree
of impairment equal to 1.5 percentage points for every decibel of perception
loss over the 25 dB threshold. If the hearing loss is over 92 dB, then the
individual is deaf.
So how much noise can a person be subject to over the course of his or her
working life and still hope to hear without impairment? Based upon the idea
that perception of speech is the most important criterion, then assessment of
human data suggests that exposure to 80 dB(A) for 8 hours per day, 250 days
per year, for a 40 year working life is an acceptable limit. However, most
countries have legal limits that are greater than this. Two common legal limits
are 90 dB(A) exposure for 8 hours per day, and 85 dB (A) exposure for 8 hours
per day. For the former limit, data suggests that 25% of the population will
suffer some form of permanent hearing damage after 30 years of daily expo-
sure, and 40% will develop permanent damage after 40 years. If the latter
limits were adopted, then the percentage of the population which would suf-
fer permanent hearing damage in each instance would drop by something like
10 percentage points.
What happens if you are forced to work in a noisy environment? The most
sensible thing to do is to wear hearing protection. However, if you must enter
a noisy environment, then the allowable exposure time must be reduced. The
generally adopted formulas are based upon the idea of cutting the exposure
time in half (daily or weekly) every time the sound level goes up by x dB(A),
where x is a country-specific or possibly industry-specific (such as the Navy)
number. Perhaps the most common number for x is 3 dB(A). This means that if
an individual was working in a country with a legal limit of 90 dB(A) expo-
sure and his or her working environment was actually 93 dB(A), they could
legally only work there without hearing protection for 4 hours per day instead
of 8, or perhaps 20 hours per week instead of 40. If the level was 96 dB(A),
then the time would be further halved to 2 hours per day or 10 hours per week.
A discussion of how to arrive at the number "3 dB" for the exposure time
halving criterion will be taken up in the next chapter.
Consider the dilemma of a nightclub bartender. It is not uncommon to have
sound levels of 105 dB(A)-110 dB(A) in the vicinity of a bar at a noisy
nightclub. Based upon the lower end of this range, the allowable time of
exposure should be halved five times (105-90 = 15, which is five times 3
dB(A) above the legal limit). This means that the bartender can legally work
for slightly more than 1 hour per week, and even then will stand a significant
risk of permanent hearing damage if the job becomes a life-long profession.
The bartender should be wearing hearing protection (ear muffs). However,
then he or she would not be able to hear the patrons ordering drinks. There
is the dilemma. (Most bartenders will tell you that they can't hear the
patrons anyway and usually lip-read drink orders. So the actual dilemma is
34 2. Background: Fundamentals of Sound
more likely to be based upon appearance, as a bartender with ear muffs might
"look silly.")
How much can hearing protection help? The ability of hearing protection,
such as ear muffs, to reduce the exposure of the ear to noise is very much
dependent upon the quality of the protection device and the quality of the fit.
Perhaps the most commonly encountered hearing protection devices are foam
ear inserts (which look like oversized cigarette butts). These are "squished"
and inserted into the ear, where they expand to fill the ear canal. These can
provide 10 dB(A) or more reduction of the sound levels at the ear drum.
However, the quality of the fit and the cleanliness of foam insert (which be-
comes dirty after one or two fittings) playa huge part in determining the
actual reduction. It is possible for the noise reduction to become 0 dB(A) in
some cases. This user-specific quality of foam inserts has seen them labeled
"not legally acceptable for hearing protection" in some countries.
High quality ear muffs can provide 20 dB(A) or more attenuation of the
sound pressure levels at the ear. Specifications are normally available from
the manufacturer.
levels inside areas of meetings and study (relatively "quiet" areas) are in the
range of 30-45 dB(A). Noisier areas, such as bus stations, may be slightly
higher than this, and quiet areas, such as hospitals, may be slightly lower.
Specification of the acceptable level of noise at the boundaries of domes-
tic dwellings is also dependent upon a number of factors, including time of
day, zoning of the dwelling, characteristics of the noise, etc. However, a level
of 40 dB(A) is a good starting point.
3
Fundamentals of Noise Control
Active and passive noise control are two approaches to a common problem: how to
get rid of unwanted noise. Active noise control aims to attenuate unwanted sound
by introducing an electronically generated "canceling" sound field. Passive noise
control aims to attenuate unwanted sound by modifying (structurally) the charac-
teristics of the environment in which the sound source operates. In many ways, the
two approaches are complementary, rather than alternative. To arrive at this conclu-
sion, it is necessary to investigate the conditions under which each noise control
technique performs well and the conditions under which they do not.
The purpose of this chapter is to provide some general background infor-
mation pertinent to the investigation of noise control system performance
and to understanding the physical mechanisms which are employed by a
given noise control technique. The investigation will concentrate on broad
"physical" characteristics related to the placement of components and will
not look at the questions relating to the electronic part of the active noise
control system. This latter topic will be the subject of two subsequent chap-
ters (7 and 8). We will begin by looking at the questions of "what are active
and passive noise control." We will then compare the techniques as applied to
common problems: control of sound radiating into free space, control of sound
propagating in a duct, and control of sound fields in an enclosed space.
It should be noted that it is not possible to provide anything more than a
superficial discussion of passive noise control in the space provided. Inter-
ested readers should consult a dedicated text for more thorough information. 1
I For example, D.A. Bies and C.R. Hansen (1996). Engineering Noise Control, 2nd ed.
36
S. D. Snyder, Active Noise Control Primer
© Springer Science+Business Media New York 2000
Prerequisite Discussion: Power and Impedance 37
watt bulb into the same socket, and draw out different amounts of energy?
After all, the socket is the same, as is the wiring and the electricity available
to the socket from the mains. Further, why is it we can plug a portable heater
into the same wall socket and draw out 20 times more electrical energy?
Again, the socket and connection are the same.
Consider another related question, which is easier to answer. Why is it that
we can take a garden hose, connect it to a tap, and get either a dribble of water
out of it or a raging gush? We have the same hose, the same tap, the same pipes
connected to the house, and yet different amounts of water flowing out of the
hose. The answer is simple: we turn on the tap a small amount for the dribble
or a large amount for the gush. In engineering terms, we could say we altered
the impedance of the system, which in turn altered the amount of flow.
In engineering and science, it is common to discuss phenomena (such as
acoustics, electricity, etc.) that involve the movement of something (sound
waves, electrons, etc.) in terms of three quantities: a quantity relating to a
potential difference, a quantity relating to flow, and a quantity relating to
impedance (or its inverse, admittance). A common potential difference quan-
tity is pressure: there is a difference in water pressure between the house pipes
on one side of the tap (high pressure) and the garden hose on the other side of
the tap (low pressure), and so there exists the potential to do some work (by
tapping into the flow of water as it moves from high pressure to low pressure
when we turn on the tap). Another common potential difference quantity is
voltage: there is a difference in voltage between wires on one side of the wall
socket (high voltage) and the light bulb on the other side of the socket (low
voltage), and so there exists the potential to do some work (by tapping into
the flow of electricity when we connect the light bulb). A third common
potential difference quantity, and one that will be important when discussing
one aspect of active noise control, is force. If you place your finger on an
object, such as a table or a loudspeaker cone, and push, then there is a differ-
ence between the force on one side of the object (where you are pushing) and
the other (where there is no pushing), and so there exists the potential to do
some work as the object moves.
Common flow quantities related to the above potential difference quanti-
ties are flow rate for the water flow (how many gallons or liters flow away from
the source per second), current for electricity (how many electrons flow away
from the source per second), and velocity for the force case.
What determines how much flow actually occurs? An impedance quantity
("impedance" from the word "impede," or to stop from moving). If a system
has a high impedance, then the flow is reduced. If a system has a low imped-
ance, then the flow is increased. For example, when the tap is opened only a
small amount then the impedance "seen" by the water in the pipe as it looks
at the tap is high. As a result, only a small amount of water is allowed to pass
over a given period of time. As the tap is opened further the impedance is
reduced. More water is now allowed to pass, until we reach the point where we
have a gush.
38 3. Fundamentals of Noise Control
In the case of electricity, as the supply of electrons "looks" out the wall
socket and possibly down a wire (say, leading to the light bulb), it sees some
resistance to movement. In the case of the 25 watt bulb, the resistance is high
and so relatively few electrons can pass through the wire in a given period of
time. In the case of the 2000 watt portable heater the resistance is low and so
a relatively large number of electrons can move down the wire in a given
period of time.
In the case of the force input, we can talk about mechanical impedance.
The mechanical impedance of, for example, a table top (which moves a rela-
tively small amount when you push it) is greater than the mechanical imped-
ance of a loudspeaker cone (which moves a larger amount).
In all of these cases, we can write a mathematical relationship
leaving the fan. For a given acoustic pressure (corresponding to a given fan
rotational speed), if the acoustic impedance is low then the flow of acoustic
waves away from the fan will be high. However, if the acoustic impedance is
high then the flow of acoustic waves away from the fan will be low.
An example of a constant volume velocity source is a loudspeaker. Re-
gardless of what is in front of it (within reason), it will continue to displace a
constant volume of air. This is analogous to saying you are going to make
your way through the jungle at a constant speed regardless of what is in front
of you. Pretend you are tied to a rope, being pulled along at a constant speed.
If the impedance is low, just a little light grass, the pressure you must exert on.
the environment to get through at the given speed is relatively small. If,
however, the impedance is large (vines, tress, etc.), then you must exert a
much greater pressure to keep moving. This is the same for the sound pressure
field leaving a loudspeaker. For a given volume velocity (speaker cone dis-
placement), if the acoustic impedance is high then the sound pressure levels
will also be high. However, if the acoustic impedance is low then the sound
pressure levels will also be low.
While it is useful to be able to relate the three quantities pressure/volume
velocity/impedance using the mathematical relationship stated in the previ-
ous paragraph, more important relationships from the noise control perspec-
tive are those that relate the above quantities to power. Power is the flow of
energy per unit of time. It is the quantity which "does something," which acts
to change the environmental circumstances in which we find ourselves. Power
is most commonly defined in units of watts (W), which describe the flow of
energy in Joules per second. For example, a 25 watt light bulb has 25 Joules
of energy flowing through and away from it every second, whereas a 100 watt
light bulb has 100 Joules of energy flowing through and away from it every
second. Note that in the case of a light bulb, the majority of this energy flow
is commonly in the form of heat, not light.
Power is normally defined by the mathematical relationship
lady, if the sound power doubles then the decibel measure increases by 3 dB,
not 6 dB, as in the case of sound pressure.
Decibel units for sound power are derived from a base measurement of 10-12
watts (that is, 0.000000000001 watts!). This amount is given a value of 0 dB.
Therefore, acoustic power W in decibels is calculated by
sound pressure (as sound power is related to sound pressure squared). When
you plot points of sound pressure level versus hearing loss for a given expo-
sure time the data should reflect one of these "laws" (risk of hearing loss
doubles with every 3 dB or 6 dB increment). This will provide the answer to
the original question of what is responsible for hearing loss.
Unfortunately, data relating biological damage to some environmental factor
never plots a neat straight line. In the case of noise levels and hearing loss, there
is so much data scatter that drawing a line through the points has some degree of
subjectivity. The present accepted line of thought (generally) is that power dam-
ages hearing, and so the line should indicate that the risk of hearing damage
doubles every time the background pressure level increases by 3 dB, not 6 dB
(although this view is being challenged in the academic publications). Note that
this is also the conservative result, which is safer for workers.
What difference does it all make? If the base figure for worker occupa-
tional health and safety legislation is, say, 90 dB(A) for 8 hours exposure
every working day, then how long can a worker stay in an area with noise
levels above this? For example, if a worker is situated in an environment of 96
dB(A) (without hearing protection), how long can they work there? If sound
pressure was thought to be responsible for hearing damage, then the exposure
time should be halved for every 6 dB increase in level. The conclusion is that
the worker can remain in the 96 dB(A) environment for 4 hours (8 hours/2).
However, if sound power is taken to be responsible for hearing damage, then
the exposure time should be halved for every 3 dB increase in level. The
conclusion is that the worker can remain in the 96 dB(A) environment for 2
hours (8 hours/2 (to 93 dB(A»/2 (to 96 dB (A» =2 hours). Most countries and!
or organizations have legislation that sits somewhere between the 3 dB and 6
dB rules.
bor from the noise. What have you done? You have redirected the acoustic
energy flow away from your neighbor's house. Where does the energy go? A
small amount of it will be absorbed when the acoustic waves hit the wall, but
in the absence of an explicit surface treatment on the wall most of the energy
will be reflected back in your direction. The end result: twice as much energy
is flowing onto your side. You have just doubled the sound pressure levels in
your yard! The price of luxury.
The first noise control category of interest here is acoustic radiation into free
space. What is acoustic radiation, and what is free space? Acoustic radiation
is simply the generation of sound waves by a source. Acoustic radiation is a
term commonly used amongst noise control practitioners. The term "free space,"
or "free field," refers to the environment in which the sound source is operat-
ing. Free space means that there is nothing to reflect the sound back; it can
travel away forever. This is as opposed to acoustic radiation into an enclosed
space, like a room. The free space environment is also referred to as "anechoic"
(literally, no echo). If you are working in the field of noise control, then you
may further subdivide free space into spaces where unreflected acoustic ra-
diation is possible in only some directions, like a "half-space" (for example,
if the sound source is sitting on the floor, it can only radiate up, not down).
This might also be referred to as "semianechoic." However, for our purposes
of qualitative results examination, we will not be particularly fussed. We will
simply assume that the sound can travel away without interference.
Having defined free space, there is a variety of common acoustic radiation
problems of this type which immediately come to mind: electrical transformer
substations, swimming pool pumps, the Rolling Stones playing at an outdoor
stadium (remember, noise is in the ear of the beholder), traffic on a busy road
next to a residential area, etc. What can be done to fix these problems, to
reduce the noise levels experienced by some human observer?
Side Story. I was once doing some consulting work for an automobile manu-
facturer, looking at problems of noise in the Press Shop. The Press Shop is
basically a huge warehouse-like building filled with machines that punch
components out of rolls of sheet metal, up to 6 mm (quarter inch) thick. The
average noise level in walking around the plant was of the order of 110
dB(A), which is loud. Near a machine, however, it could peak at around 130
dB(A), which is really loud. What I remember most about the job was the staff
coffee area, which was a large fenced-off area in the middle of the shop. The
fence was the common mesh-stuff that is seen everywhere, around schools,
playgrounds, etc. To reduce the noise in the coffee area, someone had decided
to turn the fence into an acoustic barrier. The staff apparently rallied around
the idea, and decided to do this conversion themselves (I imagine there was
no money available from management, although I never asked). In looking
around for a material which they could obtain cheaply, with a contribution
from everyone, they decided on: egg cartons. I was dumbfounded at the sight:
thousands of egg cartons, each lovingly tied to the fence. I was told that the
staff were very proud of this - and that the coffee break was much quieter now.
Unfortunately, I fear that the construction would have been useless insofar
as an acoustic barrier goes. The underlying assumption of an acoustic barrier
is that sound cannot pass through the barrier; it can only go around. Aside
from the fence being too low, the egg cartons would not have been able to
impede the low-frequency noise that dominated the environment (discussed
in a moment). I didn't have the heart to tell them that, apart from being a
source of morale improvement for the staff, they had erected art, not function.
Upon later consideration, the comment of a "quieter coffee period" puzzled
me. Knowing that the plant did not possess acoustic measuring equipment
(that's why I was there), the result must have been subjective. Knowing how
much attenuation is required before a difference is really noticeable (dis-
cussed in the previous chapter), I am left with several possible conclusions:
1. Mind over matter; the importance of worker happiness in any environ-
ment can never be underestimated.
2. The staff were all deaf.
3. All of the machines were turned off during coffee-break time.
4. The egg cartons were in fact made of lead, and so my hypothesis about
them not being able to impede the sound was incorrect.
Assuming that the barrier is of sufficient surface density, the sound field on
the "shadow" side of the barrier (the side without the noise source) will be
due entirely to sound field diffraction over the top, and possibly around the
edges if the wall is short. Diffraction refers to the deviation of a wave which
occurs when the wave encounters an object. The most important parameter in
determining the sound levels from the diffracted wave is (obviously) the
height of the wall relative to the height of the sound source and the wave-
length of the sound. Technically, this is normally quantified in noise control
work by a Fresnel number, which relates the difference between the length of
48 4. Free Space Noise Control
a direct path from source to observer to the length of the path over the wall
relative to the wavelength of sound. Other important factors include the abil-
ity ofthe ground to absorb sound (as sound can travel from the source, reflect
off the ground, hit the top of the wall, and diffract around), the ability of the
wall to absorb sound, the coherence of the sound, and the thickness of the
barrier. While calculation of the exact attenuation levels on the shadow side
is complicated and will vary from installation to installation, a reasonable
sized wall usually provides something of the order of 10 dB of attenuation.
Attenuation levels of 20 dB or more are almost impossible with a simple
barrier.
Consider now the technique of building an enclosure around a sound
source. As mentioned, this is a technique that aims to provide global sound
attenuation by reducing the flow of energy into the acoustic field. The first
question to ask is, if a sound source is enclosure by an air-tight structure, and
therefore the sound waves cannot escape, how does the sound get out? The
answer is vibration: the sound field shakes the walls of the enclosure, and the
vibrating walls reradiate the acoustic field. While this might be taken for
granted, it is interesting to ponder this point for a while. The amount of
energy flowing out of a sound source and into a sound field is incredibly
small, often less than the energy consumed by a single light bulb. And yet,
this sound field can shake massively thick walls and generate noise on the
other side.
Given that it is the vibration of the enclosure walls that is responsible for
sound generation, it is intuitive that "how difficult" a given wall is to shake,
quantified by its mechanical input impedance, will play an important role in
determining how effective the enclosure is at reducing noise. Generally speak-
ing, a mechanical "thing," be it a wall, structure, whatever, is easier to shake
at low frequencies than at high frequencies. This leads to a general conclu-
sion: enclosures do a better job at providing attenuation at high frequencies
than at low frequencies. This is a point that most people would already be
aware of, based upon their personal experience. However, we will state this
explicitly as it will be important when we come to compare active and pas-
sive noise control.
Just stating that "enclosures work better at high frequencies" tends to
overly simplify the situation. Shown in Figure 4.1 is the typical shape of a
"transmission loss" (T.L.) curve for a panel as used in the wall of an enclosure.
Transmission loss is the negative of the fraction of the energy in the incident
acoustic field which is reradiated by the panel. The greater the transmission
loss, the less the fraction of energy that is transmitted. In terms of decibels,
transmission loss =noise inside - noise outside.
There are two important points to note regarding the plot shown in Figure 4.1.
The first is the large dip in the curve at the first resonance frequency of the panel.
What is the resonance frequency, and why should the curve dip there? A reso-
nance frequency is a frequency at which the mechanical component wants to
Passive Noise Control Approaches 49
TL controlled TL controlled
en by panel stiffness by panel damping
~
c:
o
·iii
en
·E Coincidence frequency
en
c:
(1j TL controlled
t= by panel mass
First panel
resonance frequency
Frequency (Hz)
vibrate. It is the frequency where the energy contained within the structure is
most easily exchanged between potential and kinetic. Most people will have
experienced resonant behavior at some time. A common example is a plucked
guitar string, which vibrates predominantly at its first resonance frequency.
At resonance, the input impedance of the structure is very small; the struc-
ture almost "wants" to move. Therefore, there is a lot of movement for little
energy input. A lot of movement means a lot of sound being generated on the
other side. That is why there is a dip in the curve at this point.
can vary markedly between materials. For a concrete wall, it may be a few tens of
Hertz, while for a lead curtain it may be tens of thousands of Hertz.
Given this variation amongst materials, is there any common factor that
produces a panel with a high transmission loss? There is: surface density. The
greater the surface density, the better the material. Lead, for example, is an
excellent material to make enclosures from (acoustically, that is; there are
other obvious health problems which can arise from constructing a "lead
wall"). A common commercial material is vinyl impregnated with minerals to
increase the surface density. The worst enclosure materials are those which
are light and stiff. Table 4.1 lists the measured transmission loss values in dB
for several materials.
Side Story. I was once investigating whether active noise control could be
applied to a small medical appliance. This appliance had an air compressor
inside, which produced a sound field with a fundamental frequency of 25 Hz
and every harmonic imaginable. The sound used to drive the patients crazy
after several days. The manufacturer had gone to great lengths to mechani-
cally isolate the compressor from the case with rather expensive silicon mounts.
He had also called in an "expert" to advise on materials to put on the walls of
the cabinet enclosure, to improve the transmission loss. The expert recom-
mended a custom foam product, which, by coincidence, he manufactured.
What was the problem? The foam had negligible surface density; it was essen-
tially a % inch thick sponge. As such, it was completely useless in providing
low-frequency attenuation of the noise. The unit could have stood a few other
minor improvements. However, the overwhelming problem was the poor se-
lection of material for providing transmission loss at low frequencies.
Active Control Approaches 51
Up to now, we have been content with the idea that active noise control works
through sound field cancellation. If we have an unwanted sound field, and we
are able to produce a second sound field with loudspeakers that is equal in
amplitude but opposite in phase, then the two will add together to be "zero,"
or quiet. This is a perfectly legitimate way to think of the technique, both
mathematically and intuitively.
How can we now fit this line of thought into our general framework of
noise control, which says we are either reducing or redirecting acoustic en-
ergy flow? The redirecting option, which provides local attenuation at the
1 Global is normally used to mean that the sound field at all locations is reduced.
However, we will use a slightly more lenient definition: global will be taken to mean
that, on average (over all locations), the sound level is reduced. This definition means
that it is possible to have some locations which actually experience an increase in sound
level when the active control system is switched on, although on average the sound
levels go down.
52 4. Free Space Noise Control
defined by the product of volume velocity and pressure, if the (total) pressure in
front of one sound source can be made negative with respect to the displacement
of air, then the acoustic power output will be negative and the sound source will
be absorbing. This requires that we have compression (positive pressure) when
the loudspeaker cone is moving in (negative volume displacement), and rarefac-
tion (negative pressure) when the loudspeaker cone is moving out (positive vol-
ume displacement). When does the energy go? Recall that the actual amount of
energy associated with an acoustic wave is very small, typically, a fraction of a
watt. Therefore, the amount of energy being absorbed is also very small. Physi-
cally, the absorbed energy ends up helping to actually move the cone against the
mechanical impedance associated with the diaphragm and suspension, the mag-
netic stiffness and damping, etc.
In general, for free-space acoustic radiation problems the absorption idea
is a poor one. If you follow the analysis through you find that in the process
of absorbing power the control source often induces more energy flow out of
the source of unwanted noise than it absorbs. The end result is often an
increase in energy flow, not a decrease.
Based upon our earlier discussion of the mechanisms that are responsible for
global sound attenuation with active control, we can construct a list of vari-
ables which will influence the performance of an active noise control system.
This list can be arranged in a hierarchical fashion, with the entries general-
ized into four categories. Doing this, the performance of an active noise con-
trol system is dependent upon:
1. Control source (loudspeaker) arrangement; this sets an upper limit on
how much global sound attenuation can be achieved.
2. Error sensor (microphone) placement; this determines how close to the
upper limit (set by the control source arrangement) the given system can
possibly come.
3. Reference signal quality; the coherence between the reference signal and
the error signal, that is inherently the coherence between the canceling
sound field and the unwanted sound field, sets a limit upon the perfor-
mance of the "electronic" part of the control system. The coherence must
be very high for high levels of sound cancellation.
4. Quality of the controller software; this determines how much cancella-
tion at the error microphones actually occurs, given the constraints placed
by signal coherence.
unwanted noise for a reduction in total energy flow into the acoustic environ-
ment to be possible. Reducing the total acoustic power output is desirable, as
it is the only way to achieve global sound attenuation.
The question which immediately follows is, how close? Before we can quan-
tify this requirement we must decide upon the units to measure distance. Intu-
itively, the control source must be able to duplicate the shape of the sound field
generated by the primary noise source, as defined by the amplitude of the sound
pressure at all points. In this way, the phase can simply be inverted and cancella-
tion will be possible at most points in space. Referring to Figure 4.3, the sound
field patterns of two sources quickly become "different" as the noise sources are
separated. Note that the rings in the sketch in Figure 4.3 are meant to indicate
successive wavefronts as they leave the sound source, with the rings representing
crests and troughs. Based upon this, it is more correct to say that the shapes of the
sound field patterns begin to differ as the sources separate relative to the length
between crests and troughs. Inherently, we are quantifying the effect of separa-
tion distance relative to the wavelength of the sound, which was defined in
Chapter 2 as the distance between successive sound wave crests. Wavelength is a
quantity that changes with frequency, or is frequency dependent: for a given
frequency of sound, the wavelength is approximately:
FIGURE 4.3. The sound fields of two sources quickly differ as the separation distance
moves from "small" (less than one-fifth of a wavelength) to one full wavelength.
Active Control Approaches 57
- 20
-
a:l
"0
c:
0
ca 15
:;:::;
::J
c:
Q)
:=
....ca 10
~
a.
u
:;:::;
5
(/)
::J
0
«
U 0
0.00 0.25 0.50 0.75 1.00
Separation distance (wavelengths)
FIGURE 4.4. Maximum possible acoustic power attenuation for two small sound sources,
plotted as a function of separation distance between them.
~
1 foot (343 mm)
Separation
moderate attenuation for the next couple of harmonics, a little attenuation for
the next couple, and then no attenuation for the high harmonics. In other
words, active noise control is useful at low frequencies, and useless at higher
frequencies.
The above result is a good example of why active and passive noise con-
trol are complementary: passive noise control works best at high frequencies,
while active noise control works best at low frequencies. An astute vehicle
manufacturer, particularly a heavy vehicle manufacturer whose exhaust noise
spectrum contains a lot of low-frequency harmonics, might attempt to com-
bine active and passive noise control for the best result at the best size and
best price. Remember, low-frequency passive noise control equates to mass,
size, and bulk, and so if the responsibilities of low-frequency sound attenua-
tion can be removed from the passive techniques there is tremendous poten-
tial for cost savings.
Question 4: What is the relationship between the primary noise source and
control source output?
We have mentioned previously that the primary noise source and the control
source must be coherent for cancellation to occur. Every primary source-
generated wave peak must be paired to a control source-generated wave trough,
etc. However, it is worthwhile having a closer look at the relationship be-
tween the primary source and control source outputs.
In our discussion about how active noise control can bring about an over-
all reduction in energy flow into the acoustic field, we mentioned that the
acoustic pressure at the primary source could be canceled by the control
source output, and vice versa. However, completely canceling the sound pres-
sure at both sources simultaneously is not really possible in the free space
environment. This is due to the "spherical spreading" phenomena of waves as
they move away from a source. Sound waves spread out in a circle as the move
away from a source, in the same way waves in a pond move away from the
point of impact when a stone is thrown in. It is useful to think of each wavefront
as having a set amount of energy, which must be distributed over the entire
wavefront. As the wavefront expands in diameter when moving away from the
source, the energy, and hence displacement amplitude, at any given point
will decrease. As the distance from the source increases (the radius of the
circle), the length of the wavefront (the circumference of the circle) will in-
crease proportionally. Therefore, we might expect the amplitude of the wave
to decrease proportional to the radius; this is exactly the case. For example,
the sound pressure amplitude at 1 meter from a sound source is twice (6 dB
more than) the amplitude at 2 meters from the source.
Consider now the case where there are two sound sources operating in free
space, where the aim is to adjust the sources so as to minimize the radiated
acoustic power. If the acoustic pressure in front of source 1 is x dB, then for
this pressure to be canceled by source 2 the acoustic pressure in front of
Active Control Approaches 59
source 2 will have to be greater than x dB. This is because as the sound wave
travels from source 2 to source 1, its amplitude will decrease by an amount
proportional to the distance separating the two sources. However, if source 2
is adjusted to this level, then the sound pressure amplitude at source 2 from
the source 1 acoustic wave will not be sufficient to cancel completely the
source 2 sound pressure. Unless the two sources are on top of each other, it
will only ever be possible to cancel completely the sound pressure at one
source. The other source pressure will only be canceled partially.
There is a general result which states that the best global result,2 resulting
from the maximum reduction in total sound power, will occur when the con-
trol source amplitude is adjusted such that the sound pressure in front of the
control source is completely canceled. In this way, the control source acous-
tic power output will be zero, and the output from the primary (unwanted
noise) source will be reduced to some degree. Shown in Figure 4.6 is the
relationship between the primary and control source outputs for the two-
source active control arrangement considered previously in Figure 4.4. Note
that the amplitude of the control source output drops as the separation dis-
tance increases. This is in response to the drop in the primary source-gener-
ated sound pressure amplitude arriving at the control source location (which
must be canceled) which accompanies an increase in separation distance.
You may ask, what happens if the primary and control source amplitudes are
equal? Will the result be much different? The answer is yes. As before, as the
separation distance increases the acoustic power reduction reduces. However,
e co
-~
-E
§ ._ 0.0 f---------~-------"7i
00:
-0.4 L -_ _---L-_ _-----L_ _ _--'-----_ _----'
FIGURE 4.6. Ratio of optimum control source output (volume velocity) to the primary
source output (volume velocity) for total acoustic power reduction.
this time, as the separation distance increases, the reduction is not limited at 0
dB. As the separation distance approaches one-half wavelength and beyond, the
total acoustic power output increases over the original case. The two sources
look just like that: two sources, operating independently in space. Needless to
say, this is a poor result to obtain in practice.
The results presented previously are based upon the notion of minimizing the
total acoustic power radiated into space. The problem is, in practice, we are
not usually in a position to measure total radiated acoustic power and pro-
vide this quantity to our adaptive control system with the aim of minimizing
it (we would like to, if it was possible). Instead, active noise control systems
are most often implemented with microphones as sensors, and microphones
measure pressure, not power. Typically, we will place a microphone some-
where in space, and ask our adaptive control system (politely, of course) to
adjust the output of the control loudspeaker until the sound pressure mea-
sured at the microphone position is minimized. Does this "approximation" to
what we would really like have any influence upon system performance?
The answer is yes. Illustrated in Figure 4.7 is a plot of the reduction in total
radiated sound power for the case of one small sound source being used to
control the output of another small source radiating in free space, with the
two sources separated by one-tenth of a wavelength. From the previously
discussed acoustic power considerations, the maximum possible amount of
acoustic power attenuation that can be achieved by this arrangement of sources
is approximately 10 dB. From the plot, note that at some microphone loca-
(i)
~ 0.50
]j
~
~ 0.00
i~
-0.50
FIGURE 4.7.Acoustic power attenuation as a function of error sensor placement, for two
small sound sources separated by one-tenth wavelength (maximum possible acoustic
power attenuation = 10 dB).
Active Control Approaches 61
tions (near a line running between the sources) this maximum result is pos-
sible. However, in some locations, the result is actually an increase in total
power output. If the microphone is located close to the primary noise source,
this can be a significant increase.
This result is somewhat disheartening, as we often do not know where the
best microphone location is a priori. The practitioner can perform some ana-
lytical modeling only, but this is normally beyond the scope of the "suck it
and see" projects that represent a relatively high proportion of active noise
control trials. It is easier to simply move the microphone around the place and
see what happens. Alternatively, additional microphones can be added if the
scope of the electronics permits.
A few rules of thumb which can be used to guide microphone placement
include:
1. Never locate the microphone too close to the primary noise source. This
inevitably leads to a control source output that is too large, and often
controller saturation in response to trying to satisfy the requirement of
cancellation at the microphone location.
2. Do not locate the microphone too close to the control source if global
sound attenuation is desired. If the microphone is too close to the control
source, then the control source output will be too low to provide cancel-
lation away from the microphone position.
3. If multiple error microphones are used, avoid too much symmetry in place-
ment. Often a random placement will work best.
4. The sensitivity of microphone placement tends to reduce as the control
and primary source separation distance reduces.
Our previous discussion has indicated that as the separation distance be-
tween the sound sources increases toward one-half wavelength and beyond,
the amount of global sound attenuation that is possible becomes zero. How-
ever, local sound cancellation is still possible. This may be of interest, for
example, if the noise source is a large machine and the aim is to cancel the
noise only at the machine operator's location. The question which follows is,
if local cancellation is targeted, over how large an area (around the error
microphone) will the effect prevail?
The answer to this question is dependent upon a number of variables,
including the characteristics of the radiation pattern of the primary noise
source which are often not known a priori. However, as a conservative limit,
consider the case where the sound field is diffuse, which means that the sound
waves can be coming from any angle. While this characteristic is not often
associated with a free space radiation problem, the result does provide us with
a worse-case result for tonal noise radiation. For this case, cancellation will
be limited to a small sphere around the error microphone, with a radius that is
62 4. Free Space Noise Control
Noise levels inside aircraft, even modern jet aircraft, can lead to passenger dis-
comfort after a period of time. Active noise control systems in many different
forms are being developed and implemented as a way of reducing this noise. One
suggested implementation is to include active noise control in the passenger
headrests. If, for argument's sake, it is decided that the zone of quiet was to have
a minimum radius around the noise source of 100 millimeter (four inches), then
the limit on frequency where this active noise control system would be of use is
Active Control Approaches 63
around the 500 Hz mark. In practice, this can be expanded somewhat using
multiple microphones, but the upper frequency limit is still under 1000 Hz.
Fortunately, for an aircraft (unlike an ambulance), this is still a useful result. The
difference between the ambulance and the aircraft is that passengers in an aircraft
sit still, and so placing a small zone of silence around the ear is a possibility; the
same is not true for an emergency worker.
FIGURE 4.8. A given spot on the reference signal for a periodic noise source can be used
to predict the sound field at practically any point in time. Therefore, the control system
can be noncausal, where a given point in the reference signal can be used to calculate the
canceling signal for any point in the sound field . Precise matching in time is not
required.
64 4. Free Space Noise Control
FIGURE 4.9. A given spot on the reference signal for a nonperiodic noise source can only
be used to predict the sound field at one point in time. Therefore, the control system can
be causal, with each point in the reference signal used to calculate the canceling signal
for the matching point in the sound field.
field and reference signal is exclusive: only one section of the reference sig-
nal is responsible for, or correlated with, any given section of the sound field.
Therefore, for active noise control to cancel out a given section of a sound
field, the corresponding section of the reference signal must be measured and
processed, and the canceling signal fed out at precisely the right time. The
pairing of reference signal and unwanted noise is critical. Such a control
arrangement is termed causal. The reference signal must be the precise cause
of the section of sound field that is canceled out based upon its acquisition
by the controller.
The major problem with implementing a causal active noise control sys-
tem for free space radiation problems is time delays. It takes a finite period of
time for a signal to pass through a digital control system. There are delays
associated with input and output filtering, sampling, and calculation of re-
sults, as discussed in Chapters 7 and 8. There are also delays associated with
driving a loudspeaker, as it takes a finite amount of time for the loudspeaker
to produce sound once an electrical signal is fed to the loudspeaker. These
delays are dependent upon a number of physical variables, such as loud-
speaker size and filter cutoff frequency, and are frequency dependent (differ-
ent frequency components take different amounts of time to pass through the
control system). However, typical values of delay are of the order of several
milliseconds or more. On the surface, this sort of time delay may sound trivial.
However, recall that sound waves travel through space at 343 meters per
second. During the delay of several milliseconds, the sound wave would have
traveled 1 meter, 2 meters, or even more. What does this mean physically? To
implement a causal active noise control system, a reference signal measure-
ment must be taken from the target noise disturbance several milliseconds
before it arrives, in order to accommodate the delays in the electronics and
Active Control Approaches 65
Side Story. One nonperiodic free space acoustic radiation problem that is
possibly amenable to solution through the use of active noise control is found
in outdoor rock concerts. Several years ago, I was contacted by the local
organizer for the Australian leg of a Rolling Stones world tour. The Rolling
Stones were in the process of booking large outdoor concert venues, such as
football stadiums, and many city councils were placing restrictions on noise
levels. In particular, councils were concerned about the bass noise. At previ-
ous concerts of this nature, residents several kilometers away had complained
about this. The organizers' question was this: would it be possible to cancel
the bass noise (say, frequencies under 50 Hz) which is propagating away from
the stadium only? Of course, it wouldn't be acceptable to globally cancel
bass noise, including inside the stadium. As it turns out, it is possible to get a
reference signal from the bass several milliseconds before it comes out of the
loudspeaker, by tapping into the input to the sound mixer. The problem then
becomes one of making canceling sound sources which are directional, throw-
ing sound away from the stadium and not in. This would potentially cancel
the noise outside the stadium only. This too is possible in theory, by properly
designing trumpet-like horns on the front of the speakers and using a little
66 4. Free Space Noise Control
The second group of noise control problems that we will briefly discuss here
concerns the control of sound in enclosed spaces. Examples in this group
include noise in rooms, noise in vehicle cabins, and noise inside aircraft. If
the concept of control is taken in the most general way, it can also mean
preferential modification of the acoustic environment in places like concert
halls and video conferencing rooms.
Before discussing the control of sound in an enclosed space, it will be
worthwhile discussing a few physical phenomena associated with enclosed
sound fields.
the noise source (say, a motor, for argument's sake), and then turn off the
source and replay the tape through a loudspeaker in the same location and at
the same volume as the sound source. Assuming that the loudspeaker cabinet
is not moving wildly, the noise can now only propagate from the source to the
enclosed space via an airborne path; the vibration option has been elimi-
nated by turning off the original noise source. The noise which now exists in
the enclosed space is the airborne component, and the difference between
these acoustic levels and those which exist when the real source is operating
are due to the structure-borne path.
acoustics testing facility called an anechoic room. The idea in these rooms is
to absorb the entire incident acoustic wave. To do this, large wedges of foam
or other absorbing material are mounted on the walls. The resulting high-tech
appearance makes these rooms common candidates for background settings
in loudspeaker brochures. Amongst other things, the wedge shape provides a
greatly increased surface area for the sound wave to be absorbed over versus
a flat wall, which in turn greatly enhances the absorption.
1.0
0.5.
~ O·
-----,.- - 0. 5 ;
- 1.0 i
25 20 .
15 10 .
Enclosed space
5
FIGURE 5.1. Sound field in an enclosed space, showing a standing wave pattern (dark
areas are locations of high-pressure amplitude, light areas are locations of low-pressure
amplitude). Note that there are three one-half wavelengths displayed in the above mode.
which is shown in Figure 5.1. This is the pattern of a standing wave, a wave
pattern that does not propagate away but "stands" in one place. This is also
one of the mode shapes of the room.
There are two comments which must be made concerning this idea.
1. This perfect fit of the sound wave into the enclosure to form the specific
mode shape shown will only occur at one frequency. For the case of an
enclosure that appears rectangular from end to end, this frequency is
where the wall-to-wall length is equal to one and one-half acoustic wave-
lengths. This frequency is one of the resonance frequencies of the en-
closed space. Aresonance frequency is also called a natural frequency, as
it is a frequency which naturally fits with the enclosed space.
2. The pattern shown in the figure is not the only one which perfectly fits
into the enclosure. If the end-to-end distance of the enclosure is the same
as any integer number of half-wavelengths the pattern will fit perfectly.
The frequencies which correspond to these wavelengths are all resonance
frequencies of the enclosure.
If we examine the standing wave pattern we will see areas where the pres-
sure peaks, and other areas where the pressure is essentially zero. The areas of
peak pressure are referred to as antinodes, while the areas of zero pressure are
referred to as nodes. In a standing wave, these areas of peak and zero pressure
will remain in the same position in the room for all time. This means that if
you walk around in the room while it is resonating at one of its natural fre-
quencies you will find areas where it is very loud (the antinodes), and areas
where it is dead quiet (the nodes).
Side Story. If you want to experience a mode first-hand, then next time you
are in a toilet cubicle, shower enclosure, or some other small enclosed space
that is reasonably free of soft hanging materials, try humming a single tone.
Start humming at as Iowa frequency as you can manage and gradually in-
crease the frequency. At some point you should notice the sound level in-
crease dramatically-you have just hit a modal resonance frequency.
How Does the Sound Field Arrange Itself? 71
2Technically, the total amount of energy in the system is the sum of two components:
potential energy, associated with pressure, and kinetic energy, associated with velocity. As
one decreases the other increases, such that the total remains the same. Interestingly, this is
what vibration is, a swap of the total energy between the potential and kinetic states.
How Does the Sound Field Arrange Itself? 73
343 Hz. The third resonance frequency occurs when three half-wavelengths
fit into the 1 meter space (that is, the wavelength is 0.667 meters): this is
(343/0.667) = 514.5 Hz. Each subsequent axial mode resonance frequency
will be found at an increment of 171.5 Hz. Limiting consideration to axial
modes, if the frequency response of the enclosed space were plotted (say,
pressure amplitude at one of the walls versus frequency), we would see an
even distribution of pressure peaks at each resonance frequency. Physically,
if you were to stand at the wall of the enclosed space, generate a sine wave in
the space and vary the frequency, the amplitude would appear to get louder as
the frequency approached one of the resonance frequencies. The sound pres-
sure would peak right at resonance and then get quieter as the frequency
increased beyond the resonance frequency. This amplitude up/amplitude
down phenomenon would repeat itself as each resonance frequency was ap-
proached. For the example above, this is every 171.5 Hz.
The relationship between axial mode resonance frequencies is straightfor-
ward. Life becomes more complicated when considering tangential, or two-di-
mensional, modes. Using the idea of a wave fitting into the space, in a
rectangular enclosure the dimension "seen" by the wave that defines a tan-
gential mode resonance frequency is really the length of the two sides consid-
ered in parallel. This is somewhat hard to visualize, but relatively straight-
forward to derive mathematically. The antinode/node pattern of a tangential
mode will divide the two sides into sections, like a patchwork tablecloth.
This will produce a modal pattern which appears as a regular arrangement of
peaks and troughs in the enclosed space, as shown in Figure 5.2. There will be
1.0
0.5
0
-0.5
-1.0
25
25
FIGURE 5.2. Typical tangential mode pressure distribution (in an enclosed space); the
displayed mode forms a 3 x 2 pattern of nodal "areas."
74 5. Enclosed Space Noise Control
Consider the first of the two approaches. Methods that are employed to
"stop the noise from getting there" vary depending upon whether the noise
disturbance is structure-borne or airborne. If the disturbance is structure-borne,
then the simplest approach is to isolate mechanically the source using some
form of elastic isolator (such as a piece of rubber). Most machinery is mounted
using isolators; look at how a car engine or refrigerator compressor is mounted
for example. It is also important to isolate the lines, wires, etc., going to a
machine, as they are also sources of vibration.
Side Story. Vibration isolation is beyond the scope of this book. However, it
is worth commenting on the last sentence in the above paragraph. The num-
ber of noise problems that have been created by overlooking the isolation of
lines and wires, going to and coming from a machine, is incredible. Two
examples which immediately come to mind are:
1. A name-brand refrigerator manufacturer had the compressor beautifully
isolated, and then simply bolted the heat exchanging coils coming from
the compressor to the back of the refrigerator. As a result, the refrigerator
cabinet shook and radiated more noise than the compressor.
2. A medical equipment manufacturer spent a small fortune on custom sili-
con isolators for a small compressor inside a piece of equipment, and
then tied all of the lines to and from the compressor to the outer case. As
a result, the case shook and made a lot of noise.
Of course, there are still a large number of manufacturers who will not spend
the extra 2 cents on rubber grommets or a piece of foam to isolate moving
components from the equipment structure. I would estimate that 95% of the
products we have examined for the possible installation of active noise con-
trol, which usually involves noise from a motor or other rotating machine,
have had their problems solved by a piece of rubber! This is the original 2
cent solution (plus my consulting fee, of course).
If the disturbance is airborne, then the first port of call is to make sure that
all cracks and openings are sealed. Even a small gap can lead to a large noise
problem.
If all openings are sealed and all machinery is isolated but the enclosed
space is still noisy, then the remaining approach to keeping the noise out is
to reduce the vibration of the walls of the enclosure. The most common re-
medial approach here is to cover the walls with some form of vibration-damp-
ing material, something which is soft, heavy, and has a reasonably high
viscosity. The bitumen-based undercoating material in automobiles is a good
example of this. This material is applied to reduce vibration, which inher-
ently reduces noise. The classic example of vibration-damping material not
being there is an aircraft. In a propeller aircraft, all of the openings to the
outside are certainly sealed. Still, it can be very noisy inside. However, it is
not practical to smother the plane in tons of undercoating, for obvious rea-
sons of weight.
76 5. Enclosed Space Noise Control
enclosed spaces. The most important indicator relates to the modal (or other-
wise) response of the enclosure at the frequencies of interest.
We have discussed already that the physical mechanism which provides a
reduction in energy flow in an active noise control implementation is a change
in at least one of the parameters in the set of pressure/volume velocity/imped-
ance. When this happens, and global sound attenuation is achieved, a by-
product result is that the sound field produced by the canceling sources has
the same amplitude as, and opposite phase to, the original unwanted sound
field. For the previously considered case of active noise control in free space,
we found that this required the two sound sources (primary and control) to be
in close proximity.
Let us apply this line of thought of duplicating sound field shapes to the
enclosed space noise control problem. At low frequencies, the response of the
enclosed space is dominated by the response at one or more resonance fre-
quencies. We have seen that these are very discrete entities at the low-end of
the frequency response spectrum of the enclosed space. Each of these reso-
nance frequencies is associated with a mode of the enclosed space, as we have
also discussed. The important fact about this modal type of response is that it
is essentially independent of the position of the source of noise in the en-
closed space. Unless you happen to be extremely unlucky and have the noise
source located exactly at a node of the mode, you can drive the mode with a
sound source at almost any location. This is extremely advantageous for
active noise control. It means that if the source of unwanted noise is eliciting
a modal response from the enclosed space, then the canceling source can
elicit the same form of response from a large variety of locations in the enclo-
sure. That is, when the response is modal the canceling source does not have
to be located close to the source of unwanted noise. This is quite different
from the free space radiation case.
Consider now what happens at high frequencies. If the response of the
enclosed space to a noise source is a diffuse sound field, then essentially
there is no structure to the sound field. A given point in space is equally
likely to be struck by a sound wave from any given direction. In other words,
the response is more or less random in nature; sound waves are running amuck
in the enclosed space! How can the canceling sound source possibly repro-
duce this form of sound field? The only way is if the source of unwanted noise
is small and the canceling sound source is placed next to it. This is more or
less the same as the free space radiation criterion, where "close" means a
small fraction of a wavelength. We have already decided that such a criterion
is impractical at mid- and high-frequencies.
We can therefore conclude that active noise control in enclosed spaces has
the potential to provide global sound attenuation at low frequencies, where
the response is dominated by a few modes. There is almost no potential for
global sound attenuation with active methods at mid- and high-frequencies.
What is a low frequency? Based upon our previous discussion of modes, and
78 5. Enclosed Space Noise Control
how a sound wave must fit into the enclosure for a mode to be resonant, we
can say that a low frequency is one where the size of the wavelength of sound
is of the order of the largest dimension of the enclosure. If one dimension is
disproportionately large, as is the case with a tube, then this criterion can be
relaxed to some degree.
Summarizing, in an enclosed space the location of the control source(s) is
secondary in importance to the form of the response of the enclosed space (modal
versus diffuse) at the frequency of interest insofar as determining the potential of
active noise control. The frequency must be low and the response modal for
active noise control to be effective at providing global sound attenuation.
Side Story. The requirement of a modal response for active noise control to be
effective in enclosed space problems is one of the unfortunate limiting fac-
tors in the field. There are a number of potential applications which are not
physically (practically) possible because of it, including:
It was mentioned earlier that one of the problems which has driven the field of
active noise control is noise inside propeller-driver aircraft. While there are nu-
merous technical difficulties related to implementation, the concept appears, at
least physically possible as the fundamental frequency of excitation by the pro-
pellers is low and so the response inside the target planes can still be classified as
modal. Further, if the fundamental bladepass frequency was reduced in level it
would make a huge difference in many propeller-driver aircraft.
It would also be nice to apply active control technology to problems of
noise inside jet aircraft, as these make up the bulk of the passenger-carrying
fleets. One of the physical problems with this, however, is that the frequency
content of the noise is higher than that of propeller aircraft, and so the re-
sponse inside the plane can not be classified as modal. The noise is also
largely nontonal, arising from the flow of air around the fuselage. Without a
modal response, it is practically impossible for active noise control to pro-
duce a global result.
As an alternative, at least one manufacturer is now putting active noise
control into passenger headrests (for first class passengers, of course). This
active control implementation specifically targets local, rather than global,
attenuation. Basically, it is working within the physical constraints placed on
active control by the acoustic response of the aircraft interior.
Side Story. When an enclosed space is sealed, such as the case of the aircraft
interior, the noise must enter the space via vibration of the enclosure walls.
Therefore, rather than attack the noise problem once it enters the space, it is
sometimes possible to actively cancel out the vibration on the walls (essen-
tially by shaking the walls out-of-phase). For low-frequency problems, espe-
cially those where the vibration travels along compact structural components
such as struts, this can be quite an effective approach.
The last point to mention in this brief discussion on active noise control in
enclosed spaces relates to a question which has been given a lot of attention
by researchers in recent years: how many canceling sources and error micro-
80 5. Enclosed Space Noise Control
phones are required to provide good sound attenuation? To some degree the
answer to this question is application specific. However, an upper limit on the
minimum number of sources required for global attenuation is equal to the
number of modes which are excited in the target frequency band. If there are
two modes resonant then two canceling sources are required, etc. For error
microphones, it is often the case that the more error microphones there are,
the better the result. To a point. If an extremely large number of error micro-
phones are used then there are problems with the control system implementa-
tion, as will be discussed in later chapters.
6
Control of Sound Propagation
in Ducts
The final group of noise control problems which we will briefly discuss
here concerns the control of sound propagation in ducts. Ducts can be
viewed as enclosures where one dimension is very long, often terminating
into open space. Common examples of a duct include the airways used in
central heating and cooling systems, and any piping system (including
vehicle exhaust systems). Another example of a duct which may not be so
obvious is a long hallway connecting two adjacent rooms or halls. The
essential acoustic ingredient for a duct is that sound waves be constrained
in two dimensions while being allowed to travel more-or-less freely in the
third. For this reason, ducts are often referred to in acoustics literature as
waveguides, where the constraining walls guide the travel of sound waves
in the third dimension.
Once again, before we examine passive and active noise control approaches
to attenuating sound propagation in ducts, we will examine important char-
acteristics of the sound field in ducts. Once a set of characteristics is estab-
lished it is straightforward to understand and optimize the physical
mechanisms behind the noise control approaches.
sound waves will simply travel away and there will be no resonance re-
sponse associated with this side of the duct.
This intuitive model of sound fields in ducts is essentially correct. There
are, however, some details which must be added to provide a complete
description.
Modes in Ducts
Ducts do have a modal form of response. However, unlike modes in an en-
closed space, duct modes are restricted to being one- or two-dimensional
(associated with the cross section). Duct modes also propagate, or move
down the duct.
The fundamental, or lowest frequency, mode in a duct is the plane wave
mode. Referring to Figure 6.1, a plane wave has a uniform sound pressure
distribution in the duct cross section, and has the same waveform of pressure
distribution down the duct as an acoustic wave in free space. A plane wave
does not have a "true" resonance response associated with cross section. In
theory, the plane wave mode has a resonance frequency of 0 Hz, being an
axial mode with an infinite length in one direction.
All modes in the duct other than the plane wave mode are referred to as
higher-order modes. These modes have an enclosure-like modal pressure
distribution in the cross section.
It was mentioned that duct modes will travel, or propagate, down a duct.
We can view the acoustic energy that is flowing down a duct to be divided
up amongst the traveling modes; each mode carries a bit of the total energy.
One of the most important results in duct acoustics is that a duct mode can
only travel if the frequency of sound is greater than or equal to what we have
been thinking of as the resonance frequency of the mode. Because of this
result, ducts modes are usually referred to as having a cut-on frequency,
rather than a resonance frequency. At all frequencies above this one, the
duct will be "cut-on" and physically allowed to carry acoustic energy down
Sound source
FIGURE 6.1. Plane wave sound propagation in a duct. Note that the pressure distribu-
tion is uniform in all duct cross sections, and that the pressure peaks and troughs
travel down the duct at the speed of sound.
Impedance in Ducts 83
the duct. If the frequency is below a cut-on of a given duct mode the mode is
said to be "cut-off," or evanescent.
The fact that duct modes can only travel and carry energy at frequencies
above their cut-on frequency has significant implications for noise control
approaches. For example, if the duct cross section is small compared to the
frequency of interest, and so none of the higher-order modes are cut-on,
then essentially all of the acoustic energy flowing down the duct will be
carried by the plane wave mode. Therefore, any noise control approach we
may wish to adopt must specifically target the plane wave mode. Higher-
order modes do not explicitly have to be targeted for attenuation.
The process of cutting-on a mode is not binary. It is not the case at 1 Hz
below the cut-on that the mode is doing nothing, and at 1 Hz above the cut-on it
is carrying a large amount of acoustic energy. As mentioned, when the frequency
is below the cut-on of a mode the response of the mode will decay with increas-
ing distance from the source. As the frequency approaches cut-on, the rate of
decay decreases. To illustrate the mode is like a bit of chewing gum stuck to the
bottom of your shoe. While it will stay attached to the location of the noise
source, it can be "stretched" in the same way chewing gum will stretch when
you try to pull it off your shoe. The closer the noise source frequency is to cut-
on, the longer it will stretch. At cut-on, a bit of the mode is finally allowed to
break away, just like a bit of chewing gum coming off your shoe. The entire
mode does not suddenly propagate down the duct, just a bit of it. As the fre-
quency increases beyond the cut-on, more and more bits of the mode can propa-
gate, and so the acoustic energy carried by the mode increases as the frequency
increases beyond the cut-on frequency. Therefore, when assessing how impor-
tant a duct mode is in carrying the portion of the total acoustic energy, and so
assessing how much attention should be paid to it in the noise control process, it
is not enough to know simply whether the mode is cut-on. When it cut on is also
important.
Impedance in Ducts
When a sound waves travels in free space, it sees only one impedance. Tech-
nically, this is the specific acoustic impedance, defined by the ratio of (pres-
sure)/(particle velocity), where particle velocity is the velocity of a small
bit of the continuous medium in which the acoustic wave flows. Specific
acoustic impedance for a given medium is defined by the product: (speed of
sound in the medium) x (density of the medium). If a duct is infinitely long,
and so the acoustic wave never reached the end, then the impedance the
1 A cynical person might ask, how many infinitely long ducts are there in the world?
None, I suppose. However, if an acoustic wave is completely absorbed before it
reaches the end of the duct, then the duct appears to be infinitely long to the wave as
it travels along; the end is never reached. There are many examples of this.
84 6. Control of Sound Propagation in Ducts
wave would see in the duct would be the same as the free space impedance. 1
However, most real ducts are very much finite in length. When the wave
hits the end of the duct, it will usually hit an impedance change in moving
from the duct to the open environment into which the duct exhausts. This
impedance change will cause some of the traveling wave to be reflected and
move back upstream. This scenario has obvious implications for acoustic
energy flow and noise control approaches.
Calculation of the actual impedance seen by an acoustic wave as it looks
down a duct with some specific end conditions is complicated, and beyond
the scope of this book. However, a few general comments can be made.
1. Impedance is a complex number quantity. The real part of the impedance,
which is the resistance, is associated with acoustic energy flow. The imagi-
nary part of the impedance, which is the reactance, is not associated with
acoustic energy flow. Therefore, to reduce acoustic energy flow, it is neces-
sary to reduce the real part of the impedance only. If the end conditions of
the duct were such that there was no real part to the impedance, then the
acoustic power output of the duct would be zero. Achieving this is often the
aim of vehicle muffler design.
2. For a duct exhausting into free space, the resistance is dependent upon a
number of quantities. These include: duct cross-sectional area, duct pe-
rimeter, duct length, characteristics of the medium (such as air), fre-
quency of sound, and the velocity of air flowing down the tube.
3. It is possible to have a given end arrangement such that the impedance
of the duct is the same as that of the free space environment. In this
case, there will be no reflection of the sound waves back upstream. Wave
reflection requires an impedance change.
work over a wider frequency range and are cheaper and simpler to build.
We will first consider reactive techniques for passive noise control, and re-
strict ourselves to three common devices: the side-branch resonator (which in-
cludes the Helmholtz resonator), the expansion chamber, and the Helmholtz
filter.
Side-branch Resonator
A side-branch resonator is a useful device for attenuating pure tone sound
propagation in a duct, such as might arise from a fan. A side-branch resona-
tor is basically a tuned piping arrangement placed off the main piping run,
as shown in Figure 6.2. The tuned arrangement is commonly a sealed sec-
tion of pipe with a length equal to one-quarter the wavelength of the target
frequency (a "quarter-wave stub"), or a "volume" section connected to the
pipe via a smaller orifice section (a "Helmholtz resonator," which is shown
in the diagrams of Figure 6.2). The aim of the side-branch resonator is to
essentially offer the sound wave a parallel alternative to the main piping
run, a parallel that is designed to have negligible impedance at the target
frequency. This is analogous to running through the jungle and being of-
fered an alternative, parallel path of an open field (which would you take?).
The state of negligible impedance occurs at the resonance of the side-piping
section, which is why it is called a resonator. If the resonator is perfect and
so the impedance is zero, then all of the acoustic energy will flow into the
resonator and none will continue down the duct.
For the Helmholtz resonator sketched in the figure, the resonance fre-
quency is defined by
Sound source
f )
Direction of sound propagation
where c is the speed of sound (343 meters per second, nominally), A is the
cross-sectional area of the pipe connecting the volume to the pipe, L is the
length of the connecting pipe, and V is the volume of the enclosed space
behind. It should be noted that it is difficult to manufacture a "perfect" side-
branch resonator, one which will resonate at some specified frequency when
it is placed in the duct. It is common to include some way of adjusting the
system for in situ tuning.
In order for the resonator to be most effective in providing sound attenu-
ation, it should be placed at a point in the duct where the acoustic pressure is
maximum. Intuitively, the pressure will then be most sensitive to an imped-
ance change. This is commonly an odd multiple of one-quarter wavelengths
downstream from the noise source (fan, etc.).
Side Story. Side-branch resonators appear in many places. One clever appli-
cation was in a recent model station wagon. When the wagon was first built,
the designers found that there was an axial acoustic mode running across the
back seat which had a resonance frequency that was the same as the fre-
quency emitted when air entered the intake manifold (part of the engine)
when the throttle was partially open. The result was that if the car was driven
under certain conditions the occupants in the back seat were subjected to
unacceptably high levels of noise, as the axial mode resonance frequency
"latched on" to the noise of air entering the engine. To fix the problem, the
designers included a Helmholtz resonator section in the plastic ductwork
which guides air into the air filter. The resonator essentially removed the
unwanted frequency as the acoustic waves traveled along the air inlet duet-
ing, and so the back seat resonance noise problem was solved!
Expansion Chamber
Any reader who has experience with a two-stroke motor cycle will know
immediately what an expansion chamber is. An expansion chamber is basi-
cally a relatively large opening in the piping system, as shown in the dia-
Helmholtz Filter 87
Sound source
Expansion chamber
gram of Figure 6.3. As inferred by the first sentence, they are perhaps the
most common form of muffler fitted on two-stroke engines.
Expansion chambers are able to provide sound attenuation over a wider
range of frequencies than the previously discussed side-branch resonator.
Referring to the figure, the ability of the expansion chamber to provide sound
attenuation at low frequencies is limited by the resonance of the volume/
exhaust tube combination. This resonance frequency can be calculated us-
ing the same equation as given for calculating the resonance frequency of
the Helmholtz resonator, where the speed of sound c is often higher as the
medium is hot, high-speed exhaust gas. At the resonance frequency, the ex-
pansion chamber actually amplifies the noise, rather than attenuating it. Above
the chamber/tailpipe resonance frequency, attenuation will be provided up
to the point where resonances of the actual piping system come into play.
Helmholtz Filter
An extension of the expansion chamber idea is the low-pass, or Helmholtz,
filter shown in the diagram of Figure 6.4. This device is commonly used to
suppress pressure fluctuations in flowing gas. Qualitatively, the performance
of the system can be viewed as a magnification of the expansion chamber
88 6. Control of Sound Propagation in Ducts
Sound source
Noise
source
+
Direction of sound propagation
Control
system
FIGURE 6.5. Feedforward active noise control system for attenuating sound propaga-
tion in a duct.
90 6. Control of Sound Propagation in Ducts
Note for future reference. While referring to the figure, it is worth noting
that another important consideration is the amount of feedback that occurs
between the control source and reference microphone. This feedback can
lead to the same howling effect as occurs when a microphone is moved to-
ward a loudspeaker in a public address system. Small amounts of feedback
can be compensated for in the controller design. Discussion of this design
consideration will be left for the last chapter.
We will discuss each of the important criteria outlined above separately.
measuring sound fields in wind tunnels, and if designed correctly can yield a
reasonable measurement of the sound field, even in high-speed air flow envi-
ronments.
Antiturbulence microphone probes commonly consist of a long tube
plugged at the upstream end and having the microphone mounted at the down-
stream end, as shown in Figure 6.6. In the tube is a slit, that is covered with
a porous cloth material. The idea behind the arrangement makes use of the
fact that the sound waves and the wind responsible for the unwanted wind
noise travel at greatly different speeds. Sound entering the microphone tube
through the slit will tend to amplify itself via additions of more sound from
the outside as it travels along the tube. The wind noise, however, will not.
The wind outside the tube will not travel as fast as the wind noise inside the
tube, and so the wind noise entering at one point in the tube will not be
related to the wind noise entering at another point in the tube. The end result
is that the sound field component of the measurement increases in propor-
tion to the wind noise point. Technically, the signal-to-noise ratio improves.
A similar situation holds for the error microphone signal. The controller
uses the error microphone signal to tune its internal calculation process.
Basically, the controller is looking for any residual component in the sound
field measurement which is related to the information it was provided with
by the reference signal. Technically, it is looking for any correlation be-
tween the reference signal and error signal. If the wind noise dominates the
error signal, then when the controller "compares" the reference and error
signals it will think that all of the referenced sound field has been removed.
It cannot find any trace of it in the measurement, as it has been swamped by
the wind noise. Therefore, for the control system to properly tune itself, the
error signal measurement must provide information about the sound field
with as little corruption from wind noise as possible.
able canceling output, and actually turn this into sound coming from the
loudspeaker, or control source. This time period is often referred to as the
system's group delay. In a digital system, this time period can be dominated
by the components used to actually get the signals into and out of the micro-
processor (the analog to digital and digital to analog conversion process,
which will be discussed in the next chapter). It will also take a period of
time for the loudspeaker to generate sound once it has received an electrical
signal. All together, the required time period between receiving a reference
signal and outputting a canceling signal is usually a period of milliseconds:
as low as a couple of milliseconds for an active noise control system target-
ing moderate frequencies (say, up to 1000 Hz) up to as long as several tens
of milliseconds for very low-frequency systems. Typically, the time period
will be between 5-10 milliseconds for industrial duct noise problems.
For active noise control systems targeting the entire sound field spec-
trum, and not simply the harmonic components associated with, say, the fan
rotation at the end of the duct, the control system must be causal. This means
that the separation between control source and reference signal must be long
enough to give the control system the time it needs to output a signal before
the (referenced) sound field arrives. Sound travels roughly 1 meter every 3
milliseconds, so if the group delay is 9 milliseconds the control source and
reference sensor must be separated by at least 3 meters, or approximately 10
feet. A safety margin should be added to this, so a better separation distance
would be 4 meters.
discrete modes responsible for the majority of the unwanted noise. This same
idea holds true for noise control in ducts: active control is most effective
when only one mode (the plane wave mode) is cut-on. In fact, it can be said
that active control will be largely ineffective at frequencies where there are
multiple higher-order modes cut-on. This result is opposite to that of passive
noise control, where passive control is much more effective at controlling
high-order mode sound propagation than plane wave mode sound propaga-
tion. The two techniques can therefore be used together to get the best re-
sult: active control for low frequencies (where the plane wave mode
dominates), and passive control for mid and high frequencies.
Suppose, then, that you are interested in controlling noise inside an in-
dustrial exhaust stack which is several meters in diameter, and that you are
interested in frequencies up to 200 Hz. Your preliminary calculations show
that higher-order modes will start cutting-on at around 50 Hz. What can you
do? The common solution to this problem is to put a splitter in the duct. A
splitter consists of one or more partitions placed in the duct to effectively
turn it into several smaller ducts. The partitions must be long enough to fit
the reference sensor, control source, and error sensor within the section, as
well as provide a few meters at the front to accommodate the sound field
transition from the higher-order mode to plane wave sound propagation. By
splitting the duct, it is possible to force the sound to propagate in the plane
wave mode in the smaller duct sections and to achieve good control using
active methods.
Side Story. Duct splitters can be used in a clever way to provide active
control-like sound cancellation with electronics. Consider the curved duct
section shown in Figure 6.7. If the duct section is split in half, then the
effective length of the top section is longer than the effective length of the
__- - - - - Splitter
tFIOW
FIGURE 6.7. Flow splitter in a curved duct section, that presents a propagating sound
field with two different paths of travel (with two different path lengths).
94 6. Control of Sound Propagation in Ducts
bottom section. Suppose now that a pure tone sound field is propagating
down the duct. When the sound field enters the duct section, the sound waves
are in phase. However, because the top section is longer than the bottom, the
sound waves will no longer be in phase at the exit of the pipe. If the duct
work is designed correctly, the exiting waves will be exactly out of phase
and cancellation will occur-active noise control without electronics! This
technique has been used effectively for problems such as exhaust ducts for
cement kilns, where the disturbance is tonal (from the exhaust fans) but the
environment inside the duct is so terrible that even the most robust loud-
speakers and microphones would have a short life span. Metal partitions last
a lot longer!
7
Active Noise Controller Overview
on how to tune a controller, or just want to know what a controller is, then this
chapter may be all that you need to read. Chapter 8 will consider the control-
ler in more depth.
Analog ~
Storage ~
Magnetic
Electrical
~
~
Numbers ~
Computer
Digital
Chip
Storage
FIGURE 7.1 . Analog answering machines utilize a continuous magnetic signal in their
storage system, whereas digital answering machines store discrete numbers on a com-
puter chip.
98 7. Active Noise Controller Overview
Voltage Signal
~t*-(~i
10 11 12
;>.time
(milliseconds)
Time
10.1
10.2
10.3
Computer
chip
FIGURE 7.2. The numbers stored in a digital system correspond to the amplitude of the
voltage input signal at discrete, evenly spaced, moments in time.
Computer
chip Voltage signal
0.92 -D.4
~
Hold and
smooth
FIGURE 7.3. The digital (computer chip) output is turned into discrete voltages at discrete
moments in time. These voltage levels are held constant over the sample period, and the
resulting signal smoothed before output to the loudspeaker.
Digital System Requirements 99
number of samples taken per second (evenly distributed over the second).
Sample rate may be given the units of Hertz for convenience, even though
there are no "cycles" involved. For example, if the sample rate is labeled 1000
Hz then 1000 samples are taken every second, corresponding to a sample
being taken every 0.001 second, or 1 millisecond.
What should the sample rate be for a given implementation? In active
noise control, the answer to this is entirely dependent upon your target fre-
quency. Suppose, for example, the unwanted noise is a tone (sine wave) at 50
Hz. Theoretically, it is possible to measure and control this frequency with a
sample rate which is twice the target frequency, or 100 Hz. However, this is
not an advisable choice, for a number of reasons. Basically, this is the theo-
retically extreme case, which is not always practically achievable. A much
better choice is ten times the target frequency, which in this example is 500
Hz or 500 samples per second.
While experience has shown that a sample rate of ten times the target
frequency is about optimal, an adaptive feedforward active noise control
system will work satisfactorily with sample rates that are something like 3-50
times the target frequency. The precise range is dependent upon a number of
hardware and software factors, but 3-50 is a good starting point. For the 50 Hz
tonal problem in the example above, this means we could expect reasonable
performance from a system with a sample rate somewhere in the range of 150
Hz-2500 Hz. If it is any slower or faster it will begin to have a telling effect
upon both sound attenuation and controller speed. Note that if there are a
number of discrete target frequencies, or perhaps a frequency range, then
ideally the frequencies should fit within the range defined by one-third to
one-fiftieth of the sample rate.
The second important digital parameter, the accuracy of the sample, needs
to be considered in two parts: the accuracy with which the sample is actually
measured, and the accuracy with which it is recorded and manipulated in the
calculation process by the microprocessor.
The first part, the accuracy with which the measurement is taken, is set by the
analog to digital converter (ADC). This is either a specific piece of hardware on
the circuit board (a microchip dedicated to the process of measuring the input
voltage signal) or possibly a specific hardware section of a single special-pur-
pose microprocessor. The accuracy of the measurement is normally quantified in
terms of a number of "bits." The question that follows is, what is a "bit"?
Normally, a digital number has a binary representation, expressed as a
number of bits. Each bit has two possible values: 0 or 1. A number of bits are
used together to store a range of numbers. If, for example, the digital number
is "3 bits" wide, then there are three bits available for number representation,
each with a possible value of 1 or O. The representation of a number by bits is
referred to as a binary representation. The binary form of a number is written
as X 2X l X o for a three-bit system, where Xl is the value (0 or 1) of bit 2, Xl is the
value of bit 1, and Xo is the value of bit O. One common way of storing num-
bers in a three-bit format would be as shown in Table 7.1.
100 7. Active Noise Controller Overview
system is just a glorified box of switches, each with two states: on (1), or
off (0). There is no underlying intelligence; this is provided by the pro-
grammer and/or designer who decides what to do with the switches.
Analog to digital converters are classified by the number of bits they use to
represent a quantity. For example, a 12-bit ADC converter will represent a sampled
physical system variable as a set of 12 bits, each with a value of 0 or 1. This
means that there are 2 12, or 4096, possible numbers that can be used to repre-
sent the amplitude of the voltage signal. Common accuracies of ADC used in
active noise control are (at present) 12 bit, 14 bit, and 16 bit. In general, the
more bits, the better the system.
As a digital system user, it is important to recognize that it is not enough to
simply have the bits available to you; you have to use them. The 2" numbers
that are available to you are spread evenly over the input range of the system.
This range must be adjusted to match the actual voltage range you are inter-
ested in (or vice versa). Suppose, for example, that the bipolar input voltage
range of the digital system is -10 volts to + 10 volts, and you are feeding it
with a microphone signal that has a voltage range of -50 millivolts to +50
millivolts (not an uncommon output voltage range for small lapel micro-
phones). You will only ever be using a small subset of the digital numbers that
are available to you, as the higher numbered bits that represent larger voltage
values will always be equal to O. This will greatly diminish the performance
of your system. As illustrated in Figure 7.4, the solution is to supply some
gain, or amplification, to your input signal so that it takes up as much of the
-10/+ 10 voltage range of the hardware as possible.
As mentioned, the accuracy of the measurement by the ADC is only one
aspect of the system accuracy. The other is the accuracy of the data storage
and manipulation, or the internal accuracy, of the microprocessor. When as-
sessing the internal accuracy of the microprocessor there are two aspects that
must be considered: the number of bits used, and the format of the number
representation (fixed or floating point). The "number of bits used" aspect
follows the same line of thought as outlined in the ADC discussion. The
microprocessors used in active noise control, at present, are typically l6-bit,
102 7. Active Noise Controller Overview
I....L._~-----' To microprocessor
FIGURE 7.4. Gain (amplification) should be applied to input signals to ensure that the
entire ADC input voltage range is used .
32-bit, or 64-bit devices. In general, the more bits, the better (and the more
expensive, of course).
The number format aspect of accuracy refers to how the concept of a "deci-
mal point" is handled by the microprocessor. Effectively, afixed-point micro-
processor assumes that all numbers have the same (fixed) decimal point
location. When combined with the previous discussion of accuracy (being
limited by the number of bits) this means, for example, that the number 0.12345
might be stored as 0.12345, but the number 0.000012345 might be stored as
0.00001. The extra information is lost to the system. Referring to Figure 7.5,
this is referred to as quantization error.
Ajloating-point microprocessor represents each number by two pieces of
information: the actual data, without all of the preceding zeros, and the loca-
tion of the decimal point. Therefore, with a floating point system the numbers
Fixed accuracy
of digital system
( )
)I I~ Lost data
0.43212 ~ (quantization error)
Actual amplitude at sample
time = 0.432129856432 volts
I Number used by
)
microprocessor
Mathematical operation Fixed accuracy
)E>
0.43216 x 0 2. 1734 of digital system
t I( 0.09392 : 56544
Lost data
(quantization error)
I
L -_ _ __ _ _ _ _ _-?)
Actual result = 0.0939256544
)
Number used by
microprocessor
FIGURE 7.5. The finite accuracy of digital systems means that the measurements and
calculations are truncated, with some data lost. The lost data is referred to as the quan-
tization error.
Digital System Requirements 103
Control system
FIGURE 7.6. The control system must do the "opposite" of the acoustic (primary) system
in order to generate a canceling sound field. In other words, the control system must
model the acoustic system and invert the phase of the output.
Controller Output (Digital Control Filter) Requirements 105
Reference
signal
Digital filter J -- - - '
~ does this
FIGURE 7.7. The generation of the appropriate canceling sound field from the reference
signal input is performed by a digitalfilter.
cients, and adding the products to produce a new output sample.! This idea is
illustrated in Figure 7.8. The filter coefficients are simply a set of numbers,
and so this multiplication and addition process is the same as an elementary
school mathematics problem. From this description, it is intuitive that there
are three parameters that are important in determining the performance of the
digital filtering process:
1. the values of the filter coefficients;
2. the "form" of the digital filter; and
3. the length of (number of samples and coefficients in) the digital filter.
The tuning of the first of these parameters, the values of the filter coefficients,
is largely the responsibility of an adaptive algorithm. This is discussed later
in this chapter.
To explain the impact that the second of these parameters, the "form," or
type, of digital filter, has upon performance, consider the following: The
calculation process which is running on the microprocessor is simply a set of
mathematical operations (multiplications and additions) that must mirror what
happens in the real world. For example, if in the real world (the acoustic
system) the reference signal is altered in amplitude by 3.2 dB, then the num-
bers that are output from the mathematical operations must also have altered
in value by a factor of 3.2 dB. In the real world, a sound pressure wave may
travel forward forever, as is the case for radiation into free space. Alterna-
1 It should be noted that digital filtering is discussed in more detail in the next chapter.
106 7. Active Noise Controller Overview
F--+~ X weight 1
\--- +> x weight 2
~--+~ x weight 3
x weight 4 Sum of
~ x weight 5 products
> x weight 6
,r---r-> X weight 7
f---~ X weight 8
~-~ x weight 9
Digital filter
FIGURE 7.8. The digital filter calculation involves multiplying samples of the input
reference signal (and possibly the output signal, in the case of an IIR filter) by a set of
weights, or filter coefficient, and outputting the sum of the products.
tively, the sound pressure wave may reflect off boundaries and/or take the
form of a modal resonance. Each of these characteristics must be mirrored in
the mathematical operations undertaken by the digital filter.
How can the mathematics model sound wave reflections? By including
past values of calculated outputs as well as signal inputs in the calculation
process. This type of arrangement can be referred to as a "feedback loop" in
the digital filter, where outputs are fed back into the calculation process.
Digital filters that include a feedback loop are called Infinite Impulse Re-
sponse (UR) filters. Digital filters that do not include a feedback loop are
called Finite Impulse Response (FIR) filters.
It is intuitively sensible that if there are reflections of the sound pressure
waves in the target acoustic environment, then an IIR filter is a more appropri-
ate choice for the controller output calculation mathematics than an FIR
filter. Reflections can be obvious, as in systems that exhibit modal response
characteristics. Reflections can also be subtle, such as when there is feedback
from the control source to the reference microphone. To the controller, this
latter case has the same appearance as a wave measured by the reference
microphone being reflected back off a wall. If there are no reflections, then an
FIR filter is an appropriate choice.
There is much more to consider in the selection of an FIR or IIR filter than
just "reflections." However, it is a good intuitive starting point. More de-
tailed discussion can be found in the next chapter.
Adaptive Algorithm Requirements 107
The final parameter which influences the performance of the digital filter-
ing process is the filter length. Filter length refers to the number of samples,
and hence weight coefficients, used in the calculation process. 2 The longer
the filter, the more accurate the calculation process. To a point. If the filters
are too long, the adaptive algorithm (described in the next section) is likely
to be slow in tuning the weighs, or not will not converge at all, meaning that
the weight coefficient values will be (very) suboptimal.
What is a suitable filter length? This is dependent upon the frequency
characteristics of the reference signal, and hence the active noise control
target. If the target is a single tone, then an FIR filter of 4-20 taps will usually
work well. If there are multiple tones/harmonics, then 4-20 taps per tone/
harmonic is a good starting point. The actual number required for a given
level of performance is greatly influence by the system sample rate. The more
appropriate the choice of sample rate, the less the required number of weights.
If there are resonances in the target frequency band, then 4-20 taps in both
the feedforward and feedback paths of an IIR filter is a good starting point. In
general, an IIR filter will require less taps for a given level of performance
than an FIR filter.
There is one more point which should be made here. It must be recognized
that the digital filtering process takes a finite amount of time to produce an
output. When this is added to the finite amount of time that is required for the
analog-to-digital and digital-to-analog conversion process, as well as the
finite time that is required for a loudspeaker to produce sound after receiving
an electrical input, it can take anywhere from a few to a few tens of millisec-
onds between sampling the reference signal and outputting a canceling sound
wave. In this amount oftime, a sound wave can travel between 1-10 meters.
The location of the reference microphone relative to the control loudspeakers
must accommodate this delay if the system is to be causal.
2 The weights are sometimes referred to as taps, as outlined in the next chapter, and so
§
Wind
FIGURE 7.9. The operation of the adaptive algorithm is analogous to running down a hill
while looking through a periscope. The optimum filter weights are at the bottom of the
hill; this is where the algorithm wants to go.
Adaptive Algorithm Requirements 109
is the optimum set of digital filter weight values, the "holy grail" of the
algorithm. As you move up the hill and away from the optimum values your
performance will decrease.
The parameters and quantities used by the gradient descent algorithm have
a number of effects, analogous to:
1. setting a pace at which you run down the algorithm;
2. blocking the wind that is trying to blow you back up the hill;
3. determining whether you have enough energy to actually get down the
hill, and/or whether you are so "hyper" that you are unable to stop at the
bottom; and
4. determining the direction which your viewing periscope is pointing (down
the hill as desired, or completely in the wrong direction, back up the
hill).
The parameters that influence effects 1-3 can be lumped together, while
the parameters that influence effect (4) should be discussed separately. Con-
sider effect 1, the pace at which you run down the hill. If you try to run too fast
you will become "unstable" and fall over. If you proceed down the hill too
slowly you take an exorbitant amount of time to reach the optimum values. If
the bottom of the hill "moves," which is what happens to the active control
gradient descent algorithm when environmental conditions change, you never
seem to catch up.
The algorithm parameter that is chiefly responsible for the speed of run-
ning down the hill is the convergence coefficient, also referred to as the algo-
rithm step size. This is arguably the single most important parameter in the
algorithm, and usually requires some form of manual adjustment. If the value
is too small the progress down the hill is too slow. If the value is too large the
progress is too fast and the algorithm becomes unstable. 3
At this point you may ask, why not simply make the convergence coeffi-
cient small? Does it really matter if the adaptive algorithm takes a few sec-
onds to converge? There is a problem associated with making the convergence
coefficient too small, a problem that arises from the finite-precision digital
environment in which the algorithm is implemented. If the convergence coef-
ficient is too small then the calculation process will actually stop prema-
turely, as the finite precision of the digital environment will essentially treat
small numbers in the calculation of the slope of the hill as '0.' The algorithm
will think it has reached the bottom. This relates to effect 3 in the list above;
the algorithm "runs out of energy" before it reaches the bottom.
3You may ask, how do you know if the algorithm becomes unstable? The answer is,
"you will know." When the algorithm becomes unstable the digital filter weight coeffi-
cients typically become very large, too large for the finite bounds placed upon the
calculation process by the digital environment. This is referred to as calculation, or filter
weight, saturation. The end result is that the controller output sounds something like a
jet engine exhaust at 10 feet away!
110 7. Active Noise Controller Overview
This effect may sound funny, but it is quite real. If you take a stable adap-
tive algorithm implemented in an active noise control system and increase
the convergence coefficient, you can often hear a marked improvement in the
performance. If you then decrease the convergence coefficient the increased
performance will go away.
The converse effect happens when the convergence coefficient is too large
(but still stable). The adaptive algorithm will have so much energy that it is
unable to stop at the bottom, rather running down and up, down and up, in a
steady and stable fashion. The end result here is that the performance of the
control system is never quite as good as it could be.
Putting together the above factors, the effect that the convergence coeffi-
cient value has upon algorithm performance takes a shape that looks some-
thing like that which is shown in Figure 7.10. In this figure, performance is
quantified by the final mean square error value, which is basically the aver-
age amplitude of the squared value of the error signal (the measured sound
pressure in active noise control).
You might be wondering, what is the optimum value of the convergence
coefficient and how can it be calculated? The optimum value is unfortunately
application specific, being dependent upon a number of factors: signal pow-
ers, loudspeaker characteristics, characteristics of the response of the acous-
tic system, etc . This is why some manual adjustment of the convergence
coefficient is usually required.
It is worth mentioning here that in an active noise control implementation
there is one specific factor that greatly limits the stability of the calculation
process and hence the maximum usable value of the convergence coefficient:
the time delay between calculating a new set of weight values and seeing the
effects of the new weight values registered in the error signal. This time delay
comes about from the finite time it takes for a signal to travel out of the
......
e......
Q)
~
C1l
::l
0-
en
c
C1l
Q)
E
(U Optimum convergence
c ' / coefficient value
i..L L -________________ ~----~
Convergence coefficient
FIGURE 7.10. The value of the convergence coefficient used by the adaptive algorithm
has a significant impact upon the final performance (cancellation) of the system.
Adaptive Algorithm Requirements III
4 The adaptive algorithm uses the slope, or "gradient," of the "hill" to assess the direc-
tion in which it should travel (down the hill, hence the name "gradient descent algo-
rithm"). When the bottom is reached, the slope should become "0," and ideally the
algorithm will stop.
112 7. Active Noise Controller Overview
be identified, or modeled, by the controller. It is often the case that the can-
cellation path must be continually modeled, as it will change with changing
temperature, air flow, and even sound field frequency content. This modeling
process often involves inserting a small amount of random noise into the
canceling signal, as will be outlined in the next chapter.
The quality of the model of the cancellation path determines the "direc-
tion of the periscope." If the phase estimate of the model is completely wrong,
the periscope will be pointing up the hill instead of down. As a result, the
algorithm will go the wrong way, leading the weight value saturation and
instability. If the model is perfect, the periscope will point directly down the
hill and the algorithm will run in the intended direction. Fortunately, the
model does not have to be perfect for the system to function. It simply has to
point more-or-Iess down the hill (in theory, simply below the horizontal line
which splits "down" from "up"; in practice, a little bit better than this).
8
Controller Fundamentals
FIGURE 8.1. The "physical control system" consists of the microphones and loudspeak-
ers that actually produce and measure the sound field. The "electronic control system"
performs the calculations required to generate the canceling sound field.
1 For example, your bicycle is a system, and your foot can provide excitation, when you
2 In the block diagram of Figure 8.3, the circle with the Greek letter sigma (L) inside
represents a summation of signals (physically, this is the same as two sound fields
"adding"). If a negative sign appears next to one of the inputs to the summing process,
it means that the input is subtracted from the total, rather than added.
116 8. Controller Fundamentals
I
"Error" =
example, if our toaster were fitted with a closed loop control system to ensure
correct darkness, the measured darkness of the toast would be compared to
the desired darkness and the control system action (to pop up or not to pop
up) would be based upon the results of this comparison.
In active noise control we are not, in general, interested in toasting an
object, or for that matter moving or altering the equilibrium state of a system.
We are principally interested in disturbance attenuation. For us, then, the
measured system output is an acoustical disturbance (an unwanted sound
field), and the desired system output is normally zero (quiet). Therefore, the
typical feedback control structure used in active control is as shown in Figure
8.4, where the system output is used to derive the control input. The system
output may, for example, be the system response as measured by a micro-
phone. This output is sometimes referred to as an "error signal," as it repre-
sents the "error" between the desired response (a response of magnitude zero)
and the actual system response.
In the implementation of active control systems it will, in many instances, be
possible to obtain some a priori measure of the impending disturbance input,
referred to in active control literature as a reference signal. An example of this
occurs when the disturbance propagates along a duct, where it is possible to
obtain an upstream measurement. A second example is where the source of the
disturbance (the primary source) is rotating machinery, the disturbance is peri-
C on t ro I Output
... Control input
, Target ,
I
system system
"Error" =
residual output from
system (such as noise)
Figure 8.4. Closed loop control system as often constructed in active noise control
implementations.
General Control System Outlines and Definitions 117
Output "Error" =
Disturb ance
input ,..
Target . L
, ) residual signal from
system disturbance input
Control
Refere nce input
signal , Control
system
Control
Reference
Control input
signal
system
f ./\V/\V/\
Fan noise
source
Cancellation Residual
~~wanted
noise
",olse
~
ditions and transducer wear. Based upon these factors, it becomes apparent
that the control system must be adaptive. That is, it must continually tune
itself to provide the optimal result. To facilitate this, a measure of the residual
sound field, an error signal, is provided to the control system via an "error
microphone." An adaptive algorithm is normally implemented as part of the
control system that continually alters the characteristics of the controller so
as to minimize the disturbance at this sensor.
Summarizing, a standard feedforward active noise control system has four
basic components: a reference sensor (microphone) to provide a measure of
the impending disturbance; a control source (loudspeaker) to introduce the
controlling or canceling disturbance into the acoustic environment; an error
sensor (microphone) to provide a measure of the residual acoustic field after
the introduction of the controlling disturbance; and an electronic controller,
which uses the reference signal to derive a control signal that will minimize
the acoustic field at the error sensor.
Side Note. As mentioned, adaptive feedforward control systems are most com-
monly used for a number of types of active noise control system implementa-
tion. One area where they are not generally used, however, is in active hearing
protection (active headsets). In these implementations, the problem of group
delays in digital electronic control systems (discussed next) make adaptive
feedforward implementation essentially impossible. Instead, a simple analog
(not digital) feedback control system is used. These are cheap, low power, and
have virtually no group delay. They cannot, however, adapt themselves to a
changing environment. Fortunately, the characteristics of a low-frequency
sound field inside a headset cup do not change very much once a good seal
between cup and head is established.
must be able to produce an acoustic field with the same spatial characteristics as
the unwanted sound, a criterion that generally dictates that the loudspeakers be
placed in close proximity to the source of the unwanted noise. For example, if the
source of the unwanted disturbance radiates the same acoustic field in all direc-
tions, and the control source is also uni-directional, then, to achieve a reduction
of 10 dB in radiated power, the two acoustic sources must be no greater than one-
tenth of an acoustic wavelength apart. Noting that the acoustic wavelength for a
given frequency is approximately equal to 343 (meters per second) divided by
the frequency (in Hz), then to control 100 Hz sound (wavelength approximately
3.4 meters) the sources cannot be separated by more than 340 millimeters. For
sound at 400 Hz, this allowable separation distance is in the order of 80 millime-
ters. Again, this is provided that the control source can mirror the radiation pat-
tern of the primary noise source, a feat not easily achieved when the radiation is
from a complicated piece of machinery. These physical limitations are usually
the critical determinant when assessing what noise problems are amenable to
active control.
The disturbance most amenable to active control is a periodic one, where
the sound is characterized by discrete harmonically related tones. Periodic
sound fields are usually straightforward to "predict" for feedforward control
implementation, as there will be a constant relationship between the sound at
one point in the acoustic environment and the sound at some other point. As
an alterative to a microphone, a tachometer could be used to provide a refer-
ence signal for a periodic disturbance, as there will also be a constant rela-
tionship between the shaft rotation and the acoustic field.
Periodic disturbance problems provide other benefits when implementing
the electronic control system, one of which is that the control system does not
have to be causal. Basically, this means that it is not necessary to get the
"timing" between the reference signal and the control input correct for the
system to function effectively. The characteristics of periodic sound do not
change with time, and so a measurement of the disturbance at some instant in
time can be used to predict the impending disturbance in 1 millisecond, or in
1 second, or even in 1 minute without any problems. If the disturbance is
random noise, however, the system must be causal. This means that the timing
between the reference signal and control input must be correct. The measure-
ment of the disturbance at some instant in time will only predict the impend-
ing disturbance at one other instant in time. For example, a measurement of
random noise at an upstream point in an air-handling duct may be used to
predict the noise at some downstream point in 3 milliseconds time if the two
points are separated by approximately 1 meter. However, the measurement
cannot be used to predict the noise at the downstream point in, say, 10 milli-
seconds time.
Combining the results of the last two paragraphs, the most viable targets
for active noise control are arguably those where the disturbance is periodic
and of low frequency. This is important to bear in mind in subsequent
discussions.
Background 121
Background
What distinguishes digital systems from their counterparts (analog systems)
is how digital systems work with quantities. For a digital system to do some
task, it must be presented with a set of discrete parameters, in response to
which it will produce one or more discrete outputs. This is similar to the way
in which human beings perform mathematical tasks. If, for example, you are
asked to do some addition, you will expect to receive a set of discrete num-
bers (23, 37, 12, etc.) which are to be added; this is your input. In response,
you will produce a discrete output (the answer is "72").
The alternative to a digital, or discrete, approach is an analog, or continu-
ous, approach. Returning to the addition example, consider the piping ar-
rangement shown in Figure 8.8. Here two smaller pipes are feeding into one
larger pipe. This can be viewed as an arrangement for performing addition:
the flow through the larger pipe is the sum of the flows from the two smaller
pipes. However, the pipe does not work with discrete quantities: it does not
take one discrete "chunk" of liquid from pipe A, one discrete chunk of liquid
from pipe B, add then together and deliver one discrete "result" chunk to
main pipe C. Rather, the addition process is continuously happening. An
engineer might say the process is continuous in time.
At the physical level, active noise control is a continuous process: two
sound fields are continuously adding together in space to provide cancella-
tion. However, digital systems cannot perform continuous operations; they
can only work with discrete numbers (2 x 3 = 6, 1 + 2 = 3, etc.). How can we
integrate these two different modes of operation? The answer is: sampling.
Flow into
pipe A
Flow into
pipe B
FIGURE 8.8. Example of a continuous (analog) addition: the flow from two smaller pipes
entering a larger pipe.
122 8. Controller Fundamentals
Consider the problem shown in Figure 8.9, where we want to add two waves
together using a digital (discrete) approach. To perform this task in an approxi-
mate way, we could sample the waves at certain points in time, obtaining discrete
values of the amplitude of the waves at those precise instants in times. At any
point we could add together these sampled values, and produce an output which
describes the sum of the waves at that one instant. Taken together, the discrete
results yield a skeleton of the continuous (desired) result.
If we wanted to move from the skeletal result shown in Figure 8.9 to one
that more closely resembles the desired continuous outcome, what would we
need to do? Two steps are required: (1) somehow the discrete results must be
joined together in time, and (2) the resulting edges must be smoothed off.
This is, in fact, what occurs in a digital control system implementation.
Consider step 1 first. Perhaps the most ideal way to connect the skeletal results
from the digital process would be to "connect the dots," to draw a straight line
between subsequent results. There is a problem with implementing this idea,
though. If you are at one instant in time and calculate a result, you cannot draw a
straight line to the next result until you calculate it, which is at some future
instant in time. That is, to connect the dots as shown, you would have to be able
to predict the future; not even a digital system can do that!
A compromise position is to hold the current result as an output until a new
result is calculated, at which time it replaces the previous one as the output.
The result is a stepped' output, as shown in Figure 8.10.
The stepped output shown in Figure 8.10 resembles the continuous output
more closely than did the skeletal result. We can now further improve things
by implementing step 2: smooth off the edges.
How is it possible to smooth off the edges? To get an answer to this, we
must go back to our friend Fourier. We know from Fourier that any waveform
can be considered as the sum of a number of sine waves (frequency compo-
Discrete samples
of waveform A I ;>
+ +
Discrete samples
of waveform B
Discrete result:
A+B
Discrete result:
Discrete result
with data held
constant between
samples:
FIGURE 8.10. Discrete (digital) addition of two waves from Figure 8.9, with the result
held constant between samples.
FIGURE 8.11. A low pass filter will remove the edges from the stepped (discrete value)
waveform.
124 8. Controller Fundamentals
So, to summarize, we can use a discrete number system to add the two
continuous waveforms using the following methodology:
1. sample that input data;
2. perform the discrete operation;
3. output and hold the result; and
4. low pass filter the result to smooth the edges.
These are exactly the same steps that are taken when using a digital electronic
system to produce some continuous result, with one addition:
Step 0: In a "real" implementation, there is an additional step that must top
the list above. This is low pass filtering of the input data, for reasons of
aliasing to be discussed later in this chapter. Further, in a real system the
input sampling process often involves some form of comparison operation
between the measured signal and a set of reference voltage levels to assign a
value to the sample (described later). During this time the input must remain
constant. For this reason, step 1 is usually augmented with a "sample and
hold" operation, which is responsible for sampling the continuous data and
holding the value constant while electronic hardware figures out what the
numerical value is.
Note that in some instances the output side of the procedure, steps 3 and 4,
might not be required. This is commonly the case in monitoring systems,
where the aim is to collect samples of continuous data only (for example, the
temperature of a liquid in a chemical process). Also, "step 2: perform the
operation," may be long-winded and distributed. For example, in digital com-
munications, the input speech may be sampled on one end of the world,
relayed to another part of the world as a set of discrete numbers, and then
output on a loudspeaker at the receiver's end.
Step I on the list above, which entails sampling a continuous time signal
and deriving a discrete value, is usually referred to as analog-to-digital con-
version. Similarly, step 4 on the list above, the output of a discrete result, is
usually referred to as digital-to-analog conversion. In an electronic system,
these processes are usually handled by dedicated microchips: an analog-to-
digital converter, or ADC, for the input and a digital-to-analog converter, or
DAC, for the output.
Fan noise
+f\J\NOise
source
Unwanted
"><XX
Cancellation
~
Residual
noise
~--------------------~.. m~----__~--
Reference Control Error
microphone source microphone
....--.. %. . .- -.
Anti-alias Smoothing Anti-alias
filter filter filter
Microprocessor
FIGURE8.12. Adaptive feedforward active noise control system with the required digital
system components shown in block form.
As mentioned, ADCs and DACs provide an interface between the real (continu-
ous) world and the world of a digital system. ADCs take some physical variable,
usually an electrical voltage, and convert it to numbers that are sent to the digital
system. Referring to Figure 8.13, these numbers usually arrive at intervals of some
fixed time period, called the sample period. The numbers arriving from the ADC
are usually representative of the value of the signal at the start of the sample period,
as the data input to the ADC is normally sampled and then held constant during
the conversion process to enable an accurate conversion (discussed further shortly).
Commonly, the sampling period Ts is implicitly referred to by a sample rate1" which
is the number of samples taken in 1 second; Ts = 1/j s' Thus, the ADC provides
discrete time samples of a physical variable which is continuous in time. The entire
system, consisting of both continuous and discrete time signals, is referred to as
a sampled data system.
The digital signal coming from the ADC is quantized in level. This simply
means that the stream of numbers sent to the digital control system has some
finite number of digits, hence finite accuracy. For example, the continuous
signal may have a value of 23.00012735, but the digital system is only accu-
rate to four decimal points, so it sees the value as 23.0001. Referring to Figure
8.13, it can be seen that, as a result, while the value of the analog signal fed
into the ADC is increased in a continuous nature, the output is increased by
126 8. Controller Fundamentals
Value
discrete increments given by the quantum size. Normally, the digital signal
has a binary representation, expressed as a number of bits, each with a state of
oor 1 (binary signal representation was discussed in more depth in the previ-
ous chapter). This leads to ADCs being classified by the number of bits they
use to represent a quantity. For example, a 16-bit ADC will represent a sampled
physical system variable as a set of 16 bits, each with a value of 0 or 1. It
follows that the accuracy of the digital representation of the analog (continu-
ous) value is limited by the quantum size, given by
The "dual" of the ADC, the DAC, works in an opposite fashion to the ADC
in that it provides a continuous output signal in response to an input stream
of numbers. This continuous output is achieved using the sample and hold
circuit, normally incorporated "on-chip." This circuit is designed to progres-
sively extrapolate the output signal between successive samples in some
prescribed manner, most often simply holding the output voltage constant
between successive samples. This type of extrapolation process is referred to
as a zero-order hold. By incorporating a zero-order hold circuit, the output of
the DAC is continuous in time but cantiest in level. To smooth out this pat-
tern, a low pass smoothing filter, or reconstruction filter, is placed at the
output of the DAC and sample and hold circuitry as shown in Figure 8.12.
r
input output ~ in put
Digital
~ filter
I
Cancellation
path modeler ~
Adaptive I
~
algorithm I'
actually input the canceling disturbance into the acoustic system. The con-
trol signal is derived in response to the reference signal via a digital filtering
operation. As will be described shortly, the digital filter takes discrete samples
of current and past reference inputs and possibly filter outputs, multiplies
them by a set of coefficients or weights, and adds the products to produce an
output sample. The values of the filter weights determine the relationship
between the reference signal and control signal. For an active noise control
system this means that, given some reference signal, the derived control sig-
nal is a function of the digital filter weights.
The adaptive algorithm component of the controller is responsible for tuning
the digital filter weights such that the derived control signal provides the opti-
mum level of disturbance attenuation. To do this, the adaptive algorithm requires
a measurement of three items. The fIrst of these is one or more error signals, that
are measurements of the residual sound field that exists after the introduction of
the canceling sound fIeld. These measurements are taken at locations where the
unwanted noise is to be minimized. In an active noise control system, these
measurements are typically provided by one or more microphones.
The second quantity the algorithm must have is a measurement of the signals
upon which the control signal calculation has been based. This at least entails
taking copies of the reference signal samples, and sometimes copies of the actual
control output samples.
The third quantity required by the adaptive algorithm is not actually a
signal, but rather an effect. The adaptive algorithm requires a knowledge of
what will happen to the control signal between its calculation in the digital
filtering operation and its appearance in the error signal measurement. Such a
knowledge is quantified technically as a transfer function, which defines in
numbers, as a function of frequency, the relationship between the control
signal and the error signal. In active noise control work, the transfer function
between the control signal and error signal is often referred to as the cancel-
lation path transfer function.
The third component of the adaptive feedforward controller shown in Figure
8.14 is the cancellation path transfer function modeler. As might be guessed, this
component is responsible for obtaining a measurement of the cancellation path
transfer function for use by the adaptive algorithm component of the system.
A more detailed discussion of these three adaptive feedforward control
system components follows.
Structural/acoustic system
transfer function
Primary
disturbance Error
Control
Frequency input
~1~J
Reference
signal n~ Frequency
FIGURE 8.15. The control system transfer function should have the same amplitude, but
inverted phase, as the target structural/acoustic system.
celing sound field is generated. In essence, the response of the controller must be
a mirror image of the response of the acoustic system to the reference signal: the
amplitude of the response must be the same, but the phase inverted. So, for ex-
ample, if the reference signal input to the acoustic system produces an output at
the loudspeaker location of "23," then the reference signal input to the controller
must produce an output at the loudspeaker location of "-23" for cancellation to
occur. In engineering, the relationship between the signal going into a system
and the signal coming out of a system is referred to as a transfer function (specifi-
cally, a transfer function is the ratio of (output)/(input), which usually varies with
changing frequency of the signal). So, we can say that the controller calculation
process must mirror the transfer function of the acoustic system (to the reference
signal). "Component I" in our discussion of the controller is responsible for this
calculation procedure.
So, a system transfer function defines the relationship between the signal
coming into a system and the signal coming out of the system. This relation-
ship is usually frequency dependent. Consider, for example, what happens
when an input voltage is sent to a small loudspeaker. If a low-frequency
signal (say, 30 Hz) is fed into the loudspeaker, the output sound field will be
very small in amplitude. If a high-frequency signal (say, 3000 Hz) is fed into
the loudspeaker, the output sound field amplitude will be much larger. These
130 8. Controller Fundamentals
characteristics for the entire range of frequencies are the frequency response
characteristics of the loudspeaker, that are quantified by a set of numbers in a
transfer function.
Calculation of the control canceling signal in an adaptive feedforward
active noise control system is accomplished by digitally filtering the refer-
ence signal input. A digital filter is a mathematical structure, or series of
mathematical operations (specifically, multiplications and additions), that
can mimic some desired transfer function. In active noise control, the desired
transfer function would be that which transforms the reference signal into a
control signal that provides the maximum level of disturbance attenuation.
That is, the desired transfer function is the mirror image of the acoustic trans-
fer function of the system targeted for active noise control.
The transfer function model provided by a digital filter is referred to as a
discrete transfer function, as it is calculated using a series of multiplications
and additions with discrete samples of the reference signal (it is implemented
digitally). This is as opposed to the continuous transfer function of the acous-
tic system it is mimicking.
The transfer function of a system reflects its inner workings. For example, if
our system was an amplifier which increased the size of the signal by a factor of
10, then the transfer function of the amplifier at all frequencies would be "10";
this result reflects the inner workings of the system. For physical systems in
general (responding to heat transfer, fluid flow, vibration, etc.), and acoustic
systems in particular, the inner workings are described by calculus, using differ-
ential equations. These are relatively complex mathematical expressions, ex-
pressions that should be well known to all engineering students.
The discrete transfer functions that digital filters implement have their
roots in finite difference equation approximations of differential equations.
Finite difference equations provide a simple way to solve approximately a
differential equation, using simple multiplication and addition operations
performed on a number of discrete signal samples. As such, the basic building
blocks of digital filters are the same as those of finite difference equations:
multiplication and addition operations performed on discrete samples of the
input and output signals.
Hint. In engineering, the symbol Z-l is used to denote a unit time delay (delay of
one sample period). Representation of the unit time delay is shown in Figure
8.16, where the input x(k) is the value of some sampled signal at sample time k,
and the output x(k-l) is the value of the signal at the previous sample time (k-l).
In its most general form, the current output value y(k) of a digital filter is
equal to the weighted sum of present and past inputs and past outputs, de-
fined by the expression:
In this expression, the a and b terms that multiply the signal samples are the filter
coefficients or weights. Derivation of the filter output is therefore constructed
from a series of multiplications (signal sample values times filter weights) and
additions (of the products). In engineering, this is referred to as a convolution
operation, or a multiply/accumulate (MAC) operation. Figure 8.17 contains a
"sketch" of the above equation. This is, in fact, a sketch of a "direct form" digital
filter. While there is a variety of ways to structure the mathematical operations
that define a digital filter, the direct form filter is the simplest and most common
as it directly reflects the underlying mathematical expression.
The discrete transfer function associated with the above filter output equa-
tion can be summarized as
where z-x relates the filter coefficient to what it multiplies: the signal samples
that were taken x times ago.
In the diagram of Figure 8.17, as each new input sample x(k) arrives, the
previous input samples are shifted by one position. The pipeline which con-
tains the data samples is sometimes referred to as a delay line or delay chain.
Once a new input sample has been received, and the old input and output
samples have been shifted one position in the delay line, the filter output is
Output
Sam pled
in put Feedforward Out put
transfer function
bo+bl Z-1+b2Z-2+ ... ~
---
Feedback
transfer function ~
a 1z - 1+ a 2 z - 2+ ...
FIGURE 8.18. Direct-form digital filter split into feedforward and feedback components.
Side Note. It is worth noting that FIR filters are also referred to in the engi-
neering literature as nonrecursive filters, all-zero filters (as there are no poles,
or denominator terms, in the transfer function, but there are zeros, or numera-
tor terms), moving average (MA) filters, or simply tapped delay lines or trans-
versal filters. Similarly, IIR filters are also referred to as recursive filters ,
pole-zero filters, and autorecursive moving average (ARM A) filters.
means obvious. Jumping ahead slightly, it was mentioned in the previous chapter
that the adaptive algorithms used to tune the digital filters in an active noise
controller are gradient descent algorithms. These algorithms rely on the charac-
teristics of the error criterion to achieve satisfactory results: if there is a single
(global) optimum set of digital filter weight values then gradient descent algo-
rithms work well. However, if there is a number oflocally optimum sets of weight
values the algorithm can become trapped in a local optimum, and the weights
will not converge to the globally optimum values. When FIR filters are used,
there is always a single (global) optimum set of weight values. When IIR filters
are used, there can be several (local) optimum sets of weight values. Therefore,
gradient descent adaptive algorithms do not always provide the best possible
result when used with IIR filters.
Given the above outlined characteristics, it is possible to put forward a few
guidelines for selection of the correct filter for a given problem. First, it
should be stated that, where possible, use of an FIR filter is arguably a better
option than use of an IIR filter. This is due to the inherent stability and
algorithm behavior associated with FIR filters. FIR filters are ideally suited to
tonal noise problems, where the reference signal is one or more sinusoids
(probably the most common reference signal in active noise control work),
and implementations where the control signal does not in any way corrupt the
reference signal.
IIR filters are better suited to broadband work, where the target is a wide
range of frequency values. This is especially true where the target system has
resonances in the referenced frequency band or where the phase speed is not
constant (such as higher-order modes propagating in air-handling ducts). A
second situation where IIR filters are the preferred options are in systems
where there is feedback from the control source to the reference sensor. This
can occur, for example, when implementing active noise control in air-han-
dling ducts where the reference signal is provided by a microphone in the
duct. Once again, if there is feedback in the physical system, the mathematic
model (the digital filter) should also include it. IIR filters are the best option
for this.
ments of very large weight values. This is especially true if the sampling rate
is significantly greater (say, more than 20 times) than the reference sine wave
frequency. It is better to use an FIR filter with 4-20 taps for a sine wave
reference signal to avoid these very large weight values (move toward the
higher number of taps as the target disturbance becomes lower in frequency
relative to the sample rate). If multiple sine waves are present in the reference
signal, then 4-20 taps per tonal component is a useful starting point. If the
result is unsatisfactory then usually the number of taps should be increased.
For broadband reference signals, where the system is targeting a wide fre-
quency range, the question of tap numbers is more complex. If an IIR filter is
being used in a system designed to attenuate a resonant response, then a good
starting point is 4-10 taps per resonance peak in both the feedforward and
feedback weight banks. However, the number of taps can increase to several
hundred or more for applications such as broadband control of noise in an air
duct. Unfortunately, selecting the number of taps to use is largely a matter of
experience and trial and error.
Side Note. A squared error criterion, such as the squared value of the acoustic
pressure at the error microphone location, is used because if minimization of
the unsquared error signal was the control object, a very large negative error
signal would result. This is clearly undesirable.
To understand how a gradient descent algorithm works, consider the error
surface shown in Figure 8.19. This is the typical shape ofthe plot of the mean
square value of a single error signal as a function of filter weights for a two-
tap FIR filter. The error surface shape has the appearance of a "bowl", and is
technically a hyper-paraboloid. There is a single combination of weight val-
ues that will minimize the error criterion; these values are located at the
bottom of the bowl. The task set for the adaptive algorithm is to modify the
Controller Component 2: The Adaptive Algorithm 137
FIGURE 8.19. Typical error surface ("bowl"), that is a plot of the mean square value of the
error input as a function of two digital filter weights.
filter weight values to arrive at this optimum set, thereby minimizing the
error criterion.
To obtain an intuitive derivation of a gradient descent algorithm for calculat-
ing the optimum weight coefficients of the FIR filter, consider what would hap-
pen if the error criterion bowl was constructed and a ball was placed at some point
on its edge, as shown in Figure 8.20. When released, the ball would roll down the
sides of the bowl, eventually coming to rest (after some oscillation) at the bottom.
This is exactly what we would like our algorithm to do to find the optimum set of
filter weights. When first released, the ball will roll in the direction of maximum
negative change in the slope, or gradient, of the error surface. If we examine the
position of the ball at discrete moments in time as it descends, we would find that
its new position is equal to its old position (one discrete moment ago) plus some
distance down the negative gradient of the bowl.
As with the digital filter, what we want to do is put together a mathematic
expression that can be implemented on a microprocessor to mimic the real world.
The characteristics of the "ball and bowl" are somewhat formalized in a
gradient descent algorithm. This type of algorithm attempts to arrive at a
calculation of the optimum set of filter weights (at the bottom of the bowl) by
adding to the present estimate of the optimum weight coefficient vector a
portion of the negative gradient of the error surface at the location defined by
this estimate. In this way, the current value of the mean square error descends
down the sides of the error bowl, eventually arriving at the bottom. This is the
location corresponding to the optimum weight coefficients.
Mathematically, this notion, that defines a generic gradient descent algo-
rithm, can be expressed as
where Aw is the gradient of the error surface at the location given by the
current weight coefficient vector and 11 is a positive number that defines the
portion of the negative gradient to be added, referred to as the convergence
coefficient.
The question now arises, how is the gradient of the error surface at the
location of the current weight values calculated? It is worthwhile doing some
simple mathematics to answer this question. If consideration is limited to a
single error sensor system for simplicity, then at any instant in time k the error
signal e(k) is a function of two components: a component p(k) due to the
unwanted disturbance and a component s(k) generated by the active noise or
vibration control system. As acoustic and structural systems are linear in the
normal operating range, the error signal is the sum of these two components
Note that these two components are actually the signals which are output
from the error sensor measurement system when either the unwanted noise
source or the active control system are operating alone.
As mentioned, the error criterion is the squared value of the error signal. Math-
ematically, the gradient is calculated by differentiating this error criterion with
respect to the filter weights. For a single error sensor system, noting that the
unwanted disturbance component p(k) of the error signal is not a function of the
digital filter weights, this differentiation produces the following expression:
Cancellation path
FIGURE 8.21. Cancellation path in an adaptive feedforward active noise control system.
140 8. Controller Fundamentals
term in the gradient) is equal to the change in the control signal (filter output)
that accompanies a change in filter weights modified by the cancellation path
transfer function. As a simple illustrative example, suppose that the cancellation
path was an amplification by a factor of 2. Any change in the control signal would
be seen as a similar change, with twice the amplitude, in the control source
component of the error signal. Therefore, for a control signal y(k), the partial
derivative as(k)/aw(k) would be equal to (2 X ay(k)law(k)).
In practice, the cancellation path transfer function is unlikely to be simply
a gain. Rather, the change in amplitude and phase that it describes is usually
frequency dependent, and can vary dramatically over the operating frequency
range of the system. This is especially true if the target structural and/or
acoustic system has resonances in the operating frequency range.
Calculation of the partial derivative ay(k)/aw(k) is relatively straightfor-
ward. From the previous description of the digital filter, the filter output y(k)
is the product of two components: the filter weights w(k) and the signal samples
in the filter delay lines. Therefore, the partial derivative ay(k)/aw(k) is simply
equal to the values of the signal samples in the delay lines (the partial deriva-
tive for each individual filter weight is equal to the value of the signal sample
at the point in the delay line where the weight is situated). It follows that the
partial derivative as(k)/aw(k) used in the gradient calculation is equal to the
signal samples in the filter delay chain modified by the cancellation path
transfer function. This modification can be viewed as a filtering operation,
where the signal samples are filtered by (a model of) the cancellation path
transfer function to produce the partial derivative used in the gradient calcu-
lation. This filtered set of signals is then multiplied by the error signal to
produce the gradient estimate used to modify the current weight values in
such a way that the levels of disturbance attenuation are improved.
This filtering of the signal samples in the process of deriving the gradient
estimate is what differentiates the active noise control implementation of
adaptive filtering from the more common implementations, such as those
used in telephone echo cancellation. In the common implementations there is
no cancellation path, and so the raw signal samples in the digital filter are
used in the gradient calculation. This need to filter the signal samples, to
derive the gradient in the active noise and vibration control implementation,
has led to adaptive algorithm names such as the "filtered-x LMS algorithm,"
that is the active noise and vibration control version of the standard "LMS
algorithm."
dom noise, than it is when the reference signal is sinusoidal. Also, when
the reference signal consists of one or more sinusoids, the maximum stable
value of the convergence coefficient decreases markedly if the sample
rate is many times greater than the frequencies of the sinusoids (say, more
than 50 times faster). In other words, systems with low-frequency refer-
ence signals require smaller values of convergence coefficient than simi-
lar implementations with higher-frequency reference signals.
3. If any gains in the system are increased the convergence coefficient value
should be decreased. If the system gains are reduced the convergence
coefficient can be increased.
4. If more control sources or error sensors are added to the system, the value
of the convergence coefficient should be decreased. Conversely, if the
number of control sources or error sensors is reduced, the convergence
coefficient can be increased.
5. If the separation distance between the control source( s) and error sensor( s)
is increased, so that the time required for a signal to propagate from
source to sensor is increased, the value of convergence coefficient should
be decreased. Conversely, if the distance between source(s) and sensor(s)
is decreased, a larger value of convergence coefficient can be used.
6. If the size of the digital filter(s) used to generate the control signal(s) is
increased in size, then the value of the convergence coefficient should be
reduced. Conversely, if the size of the filter is reduced, the convergence
coefficient can be increased.
7. The maximum stable value of the convergence coefficient for FIR digital
filter implementations is often greater than the maximum stable value of
the convergence coefficient for IIR digital filter implementations.
explanation of the effect that this has upon the long-term operation can be made
with reference to Figure 8.22. Illustrated in this figure is a typical two-dimen-
sional plot of the mean square value of the error criterion (mean square error, or
MSE) as a function of two weights in an FIR filter, that is essentially how the bowl
in Figure 8.19 would appear if the viewer was on top looking down. There is a
number of combinations of weight values that will result in the same mean square
value of the error criterion. These combinations form a set of concentric contours
centered around the optimum combination of filter weights. During long-term
operation of the adaptive algorithm, quantization errors cause the values of the
filter weights to increase in magnitUde, analogous to a build-up of energy. How-
ever, this increase in value is not evident to the outside viewer, as the mean square
value of the error signal is unchanged. Rather, the combination of weights moves
along a contour of a constant mean square value of the error criterion. Eventually
the values of the weights become larger than the maximum value allowed by the
digital system, and the filter calculations begin to overflow. Only at this point
does the outside viewer realize that something has gone terribly wrong.
Fortunately, it is relatively simple to fix this problem using what is referred
to as tap leakage. Tap leakage works by removing a small portion of the
current weight values with each new weight calculation. In this way the build-
up of energy that arises from quantization errors is avoided and long-term
stable operation of the adaptive algorithm is possible. When tap leakage is
implemented, the gradient descent algorithm is modified to
-
..c
OJ
Contours of
constant MSE
£
convergence
th
Weight Wo
FIGURE 8.22. Quantization errors will add "energy" to the adaptive algorithm, eventually
driving it unstable (saturate the filter weights) if some precautions are not taken .
Controller Component 2: The Adaptive Algorithm 145
50.---------------------------,
1"'..j(-----
40 ~/'
Case 2 instability
c: Case 1 instability
-o 30
a:l
:l
-co 20
c:
Q)
Case 1:
"C
c: Adaptation every cycle
:l
o Case 2:
C/) Adaptation every 3 cycles
10 Case 3:
Adaptation every 5 cycles
O~----~ __~____~____~____ -J
FIGURE 8.23. A plot of the sound attenuation versus the convergence coefficient for a
typical active noise control implementation. Note that the "faster" algorithm implemen-
tations are not always amenable to the "optimum" convergence coefficient value, due to
premature instability arising from long time delays in the cancellation path.
the time delay in the cancellation path. By effectively reducing this time delay
(as measured in numbers of weight updates that occur in the time delay period),
the algorithm is stabilized and the convergence coefficient can be increased.
One additional point that is of interest here is that slowing down the adap-
tive algorithm calculation rate need not slow down the actual time it takes
for the algorithm to converge to a set of weight values that provide a given
level of sound attenuation. If a larger convergence coefficient is used in the
stabilized implementations, it will counteract the effect of not updating the
weights as frequently as is physically possible. An example of this is shown in
Figure 8.24, where the error signal response after a restart is shown for two
different adaptation rates and convergence coefficient values (the best result
for each case is shown). Note that each algorithm converges at more-or-Iess
the same actual rate (in seconds).
...ro
..........
Adapt every cycle
Q) 0.4 Convergence coefficient = 200
c
-::::
----
~ 0.0
c
C)
e...
(f)
-0.4
UJ
o 1 2 3
Time (seconds)
....
..........
Adapt every 10 cycles
ro
Q) 0.4 Convergence coefficient = 1000
c
-::::.-
----
cti 0.0
c
C)
e
(f)
-0.4
....
w
Time (seconds)
FIGURE 8.24. Initial convergence of the adaptive feedforward active noise control system
for different adaptation rates and convergence coefficient values.
ous section on the adaptive algorithm, knowledge of how the control signal
is altered between its output from the digital filter and appearance in the error
signal is required to calculate the gradient used in the adaptive algorithm.
This knowledge takes the form of a model of the transfer function, the deriva-
tion of which is the job of this part of the control system.
Before progressing further, it should be noted that the general operation of
formulating a mathematical model of a physical transfer function is referred
to as system identification. This terminology will be used frequently in the
subsequent discussion.
The gradient descent adaptive algorithm used to adjust the weights of the
digital filter model of the cancellation path transfer function is slightly dif-
ferent than the gradient descent algorithm described previously for adjusting
the weights of the control signal-generating digital filter. Referring to Figure
8.25, the difference occurs because there is no transfer function in the can-
cellation path with the system identification arrangement. The control signal
148 8. Controller Fundamentals
Cancellation Path
ouct
rvv
~~ Speaker Microphone
Modeling
s ignal
(noise, etc.)
"- +
Model - L
')
LMS error
~
Algorithm
FIGURE8.25. Arrangement for modeling the cancellation path in an active noise control
implementation.
generated by the digital filter model is used directly in the calculation of the
error signal, without having to first pass through loudspeakers, microphones,
filters, etc. This means that the signal samples in the digital filter can be used
directly in the gradient calculation, rather than having to be filtered through
a model of the cancellation path transfer function. That is, the algorithm
implementation has the standard adaptive filtering form, as found in applica-
tions such as telephone echo cancellation.
Before discussing the system identification procedure further, it is worth-
while considering an important question: how accurate does the model of the
cancellation path transfer function have to be for the control filter adaptive
algorithm (described in the previous section) to function correctly? Fortu-
nately, it does not have to be exact. Errors in the estimate of the amplitude of
the transfer function have the simple effect of reducing the maximum stable
value of the control filter adaptive algorithm convergence coefficient in an
inverse proportional manner (if the gain estimate is too high, the maximum
stable value of the convergence coefficient is reduced; if the gain estimate is
too low, the maximum stable value of the convergence coefficient is increased).
However, the most important parameter is the phase of the transfer function.
Theoretically, for a single control source, single error sensor system, it is
possible to have stable operation of the control filter adaptive algorithm
Controller Component 3: Cancellation Path Modeler 149
provided that the phase response of the model is within ±90° of the actual
phase response for the frequency components being targeted for active noise
or vibration control. In practice, it is better to restrict errors in the estimate of
the phase response to be less than half (45°) of this value. For systems with
multiple control sources and multiple error sensors, the robustness of the
system increases. In this case, it is possible to have stable operation of the
control filter adaptive algorithm provided that the total error in the phase
response of the models between a given control source and each error sensor (the
sum of the errors in each model) is less than ±N x 90°, where N is the number of
error sensors in the system (this is actually an approximate value; the actual value
is dependent upon the characteristics of the transfer functions themselves).
The characteristics of the cancellation path transfer function are usually
not constant, but rather are slowly time varying. In some instances this varia-
tion can be extremely slow, such as due to the change in loudspeaker re-
sponse due to mold growth! In other instances, the variation can be more
rapid, such as due to the change in flow rate in an air-handling duct. In either
case it is necessary to update the model of the cancellation path transfer
function to account for these changes. In most practical active noise control
systems, modeling of the cancellation path transfer function is done in paral-
lel to the adaptation of the control filter weights. To do this, the signal used in
the modeling procedure, injected into both the cancellation path and the
transfer function model as shown in Figure 8.25, must be included in the
output from the digital control filter.
The most common way of conducting this "on-line" modeling, in both
active noise control and other (general) adaptive control implementations. is
to inject random noise into both the system to be modeled and the model
itself. The advantage of random noise is that it is uncorrelated with the other
disturbances in the system to be modeled, which in this case includes the
unwanted noise and control disturbances. This reduces the chance of bias in
the model (although some bias can still occur, as will be discussed shortly).
While this effectively means adding an additional, uncontrollable distur-
bance into the system targeted for active noise or vibration control, the am-
plitude of the modeling disturbance can usually be quite small and still
produce an adequate model (say, 30 dB below the peak signal levels in the
external environment).
There is a second approach to modeling that can be useful, and is some-
times used, in some circumstances. This approach uses the actual control
signal as the modeling disturbance, injecting it into the model of the cancel-
lation path transfer function (it is already being injected into the cancellation
path itself). Intuitively, this would appear to be a very risky way of conduct-
ing system identification, as the control signal is correlated with disturbances
in the external environment. This can potentially lead to such high degrees of
signal bias that the control filter adaptive algorithm becomes unstable. How-
ever, there are means of accounting for the correlated environment which
make this approach useful for tonal disturbances.
150 8. Controller Fundamentals
Note. The procedure used for identification of the cancellation path transfer
function is similar to the procedure used for tuning the control filter weights.
In both instances, an adaptive algorithm is used to modify the weights of a
digital filter so as to minimize some error criterion. When adjusting the con-
trol filter weights, this error criterion is minimization of the sum of the squared
error signal amplitudes. When modeling the cancellation path transfer func-
tion, the error criterion is minimization of the squared value of the prediction
error, which is the difference between the model output and the system out-
put. Because of the similarities between the two procedures, and the similar-
ity of parameters and effects, many of the parameter details discussed in the
previous section are mirrored here and so will not be discussed.
Side Note. Modeling of the cancellation path transfer function entails direct-
ing the output of a modeling signal into both the cancellation path and its
model, and the use of an adaptive algorithm to minimize the difference be-
tween the system (cancellation path) and model outputs. If the modeling
disturbance is injected into an environment which has "auto-correlated" noise,
noise with strong periodic characteristics, such as systems with tonal distur-
bances, or systems with strong resonances, the model can become biased or
even completely wrong. This can result in system instability. One way to get
around this problem is to use extended system identification procedures.
Consult a more "in-depth" active noise control text for more information.
Sinusoid 2 ~
data. Aliasing can have a significant detrimental effect upon control system
performance if there are substantial levels of high-frequency data, f> f s/2,
that are allowed to alias onto the low-frequency data, f < f,l2. To combat this
problem, antialiasing filters are placed in front of the analog-to-digital con-
verter (ADC). These are low-pass filters that remove frequency components
greater than half the sample frequency, f > f,l2, from the input spectrum.
In theory, then, the lower bound on the sample rate for a given problem is
twice the highest frequency of interest. However, actually implementing a
system with this sample rate is not advisable. First, while it is theoretically
possible to reconstruct a harmonic signal sampled at twice its frequency, the
filter required to do so is of infinite length, and not bounded input, bounded
output (BIBO) stable. By this, we mean that for a finite input there is not
necessarily a finite output. Second, there is no margin for error in the upper
frequency limit. Any slight change in the upper frequency results in aliasing.
Third, practical antialiasing filters do not have perfectly square pass/stop
characteristics, but rather have some finite transition band from pass (fre-
quencies below half the sample rate that are allowed through the filter) to
stop (frequency components above half the sample rate that are filtered out).
Therefore, expecting the antialiasing filter to pass all frequencies up to half
the sample rate, while stopping all frequencies above half the sample rate, is
optimistic.
To paint a qualitative picture of what happens if the sample rate is too low,
consider the problem of sampling the "step response" of a system that has a
resonance at 1 Hz (that is, we are looking at the response of a system that is
dominated by a frequency of 1 Hz). As shown in Figure 8.27, if the step
response is sampled at 2 Hz the characteristics are indistinguishable to the
viewer. Sampled at 5 Hz, the characteristics begin to appear. At 10 Hz, the step
is apparent. In fact, if the samples are connected by straight lines, the recon-
struction of the step is in error by less than 4%. Intuitively, then, we can
postulate that the filtering exercise, which is analogous to the reconstruction
of a signal, becomes "easier: as the sample rate increases.
There is, however, a limit to this process for an adaptive control system. At
high sample rates, tens or even hundreds of times the disturbance frequency,
152 8. Controller Fundamentals
x X
-
Q)
Ol
CIl
g
Sample rate =2 Hz
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1.0
Time (8)
X X X
X X
-
Q)
Ol
CIl
g
Sample rate =5 Hz
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1.0
Time (8)
X X X X X X X X
X
Q)
Ol X
$
g
Sample rate = 10Hz
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1.0
Time (8)
there are problems with numerical accuracy in the digital environment. Per-
haps more seriously, there are problems in the convergence behavior of the
adaptive algorithm.
The upper bound on sample rate selection for a given problem is usually
determined by adaptive algorithm performance characteristics. When discussing
adaptive algorithm performance, the focus is usually on two competing factors:
algorithm stability and speed of adaptation. The key factor that influences these
characteristics is the adaptive algorithm convergence coefficient. Algorithm sta-
Controller Component 3: Cancellation Path Modeler 153
-3 -2 -1 o 2
Weight Wo
steep sides and some very shallow sides). With such an error surface, the
convergence coefficient must be very small to maintain stability due to the
steep sides. However, with such a small convergence coefficient the speed of
adaptation will be very poor due to the shallow sides. Conversely, the best
error surface for algorithm convergence is one that is perfectly symmetric.
Here the steep and shallow sides, and hence speed and stability, are perfectly
in balance.
The reason for this discussion is that sample rate largely determines the
degree to which the error surface bowl is squashed. In particular, as the sample
rate increases relative to the target frequency of excitation, the steep and
shallow sides of the error surface become more and more disparate. If the
target frequency is heavily oversampled (say, 50 or more samples per excita-
tion frequency cycle), then the error surface will have some very steep sides
and some very shallow sides. In this case, the adaptive algorithm performance
will be very poor. The convergence coefficient will have to be very small, and
the speed of convergence will be very slow. If the degree of oversampling is
several hundred or more, attempts at control are largely futile.
(see the control filter section of this chapter for a discussion of recommended
tap numbers). Third, as could be deduced from the step response example,
having only four samples per cycle may lead to accuracy problems.
The optimum sample rate is a compromise between fast and slow. Both of
these extremes lead to problems with adaptive algorithm convergence and
stability, and to problems with numerical accuracy. The "optimum" sample
rate compromise is often cited as ten times the frequency of interest. In
practice, this sample rate provides for rapid convergence of the adaptive algo-
rithm and reasonable levels of stability. In implementing active control sys-
tems we often find that for a given sample rate f. the system will work
reasonably well from frequencies approaching f/100 to frequencies up to f/
3. On the low end of the scale, adaptation of the controller with frequencies
below 0)/100 is often (extremely) slow, and not particularly stable. While this
is sometimes improved by increasing the length of the digital control filter,
the only real solution is a reduction in sample rate. On the high end of the
scale, the adaptive algorithm appears ineffective with excitation frequencies
above 0)s/3.
Index
157
158 Index