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AND NOISE CONTROL
Finn Jacobsen, Torben Poulsen, Jens Holger Rindel,
Anders Christian Gade and Mogens Ohlrich
Department of Electrical Engineering, Technical University of Denmark
November 2010 Note no 31200
ii
iii
CONTENTS
Page
1 An elementary introduction to acoustics .................................................................... 1
Finn Jacobsen
1.1 Introduction .................................................................................................................... 1
1.2 Fundamental acoustic concepts ...................................................................................... 1
1.2.1 Plane sound waves ............................................................................................. 4
1.2.2 Spherical sound waves ..................................................................................... 13
1.3 Acoustic measurements ................................................................................................ 15
1.3.1 Frequency analysis ........................................................................................... 15
1.3.2 Levels and decibels ......................................................................................... 18
1.3.3 Noise measurement techniques and instrumentation ....................................... 21
1.4 The concept of impedance ............................................................................................ 27
1.5 Sound energy, sound intensity, sound power and sound absorption ............................ 31
1.5.1 The energy in a sound field .............................................................................. 32
1.5.2 Sound absorption .............................................................................................. 35
1.6 Radiation of sound ....................................................................................................... 37
1.6.1 Point sources .................................................................................................... 37
1.6.2 Sound radiation from a circular piston in an infinite baffle ............................. 42
1.7 References .................................................................................................................... 49
1.8 Bibliography ................................................................................................................. 50
1.9 Appendix: Complex notation ....................................................................................... 51
2 Ear, Hearing and Speech ........................................................................................... 55
Torben Poulsen
2.1 Introduction .................................................................................................................. 55
2.2 The ear .......................................................................................................................... 55
2.2.1 The outer ear ..................................................................................................... 56
2.2.2 The middle ear .................................................................................................. 56
iv
2.2.3 The inner ear ..................................................................................................... 57
2.2.4 The frequency analyzer at the Basilar membrane ............................................ 59
2.3 Human hearing ............................................................................................................. 61
2.3.1 The hearing threshold ....................................................................................... 61
2.3.2 Audiogram ........................................................................................................ 62
2.3.3 Loudness level .................................................................................................. 63
2.4 Masking ........................................................................................................................ 64
2.4.1 Complete masking ............................................................................................ 65
2.4.2 Partial masking ................................................................................................. 67
2.4.3 Forward masking .............................................................................................. 67
2.4.4 Backward masking ........................................................................................... 67
2.5 Loudness ....................................................................................................................... 68
2.5.1 The loudness curve ........................................................................................... 68
2.5.2 Temporal integration ........................................................................................ 69
2.5.3 Measurement of loudness ................................................................................. 69
2.6 The auditory filters ....................................................................................................... 71
2.6.1 Critical bands .................................................................................................... 71
2.6.2 Equivalent rectangular bands ........................................................................... 72
2.7 Speech .......................................................................................................................... 73
2.7.1 Speech production ............................................................................................ 73
2.7.2 Speech spectrum, speech level ......................................................................... 74
2.7.3 Speech intelligibility ........................................................................................ 75
2.8 References .................................................................................................................... 78
3 An introduction to room acoustics ............................................................................ 81
Jens Holger Rindel
3.1 Sound waves in rooms .................................................................................................. 81
3.1.1 Standing waves in a rectangular room ............................................................. 81
3.1.2 Transfer function in a room .............................................................................. 83
3.1.3 Density of natural frequencies .......................................................................... 83
3.2 Statistical room acoustics ............................................................................................. 85
3.2.1 The diffuse sound field ..................................................................................... 85
v
3.2.2 Incident sound power on a surface ................................................................... 86
3.2.3 Equivalent absorption area ............................................................................... 87
3.2.4 Energy balance in a room ................................................................................. 87
3.2.5 Reverberation time. Sabine’s formula .............................................................. 88
3.2.6 Stationary sound field in a room. Reverberation distance ............................... 89
3.3 Geometrical room acoustics ......................................................................................... 92
3.3.1 Sound rays and a general reverberation formula .............................................. 92
3.3.2 Sound absorption in the air ............................................................................... 93
3.3.3 Sound reflections and image sources ............................................................... 94
3.3.4 Reflection density in a room ............................................................................ 95
3.4 Room acoustical design ................................................................................................ 96
3.4.1 Choice of room dimensions .............................................................................. 96
3.4.2 Reflection control ............................................................................................. 96
3.4.3 Calculation of reverberation time ..................................................................... 97
3.4.4 Reverberation time in nondiffuse rooms ......................................................... 98
3.4.5 Optimum reverberation time and acoustic regulation of rooms ....................... 99
3.4.6 Measurement of reverberation time ............................................................... 100
3.5 References .................................................................................................................. 101
4 Sound absorbers and their application in room design ........................................ 113
Anders Christian Gade
4.1 Introduction ................................................................................................................ 103
4.2 The room method for measurement of sound absorption .......................................... 103
4.3 Different types of sound absorbers ............................................................................. 104
4.3.1 Porous absorbers ............................................................................................ 105
4.3.2 Membrane absorbers ...................................................................................... 106
4.3.3 Resonator absorbers ....................................................................................... 108
4.4 Application of sound absorbers in room acoustic design ........................................... 109
4.5 References .................................................................................................................. 112
5 An introduction to sound insulation ....................................................................... 113
Jens Holger Rindel
5.1 The sound transmission loss ....................................................................................... 113
5.1.1 Definition ....................................................................................................... 113
vi
5.1.2 Sound insulation between two rooms ............................................................. 113
5.1.3 Measurement of sound insulation .................................................................. 114
5.1.4 Multielement partitions and apertures .......................................................... 114
5.2 Single leaf constructions ............................................................................................ 117
5.2.1 Sound transmission through a solid material ................................................. 117
5.2.2 The mass law .................................................................................................. 119
5.2.3 Sound insulation at random incidence ........................................................... 120
5.2.4 The critical frequency ..................................................................................... 121
5.2.5 A general model of sound insulation of single constructions ........................ 122
5.3 Double leaf constructions ........................................................................................... 123
5.3.1 Sound transmission through a double construction ........................................ 123
5.3.2 The massairmass resonance frequency ........................................................ 124
5.3.3 A general model of sound insulation of double constructions ....................... 124
5.4 Flanking transmission ................................................................................................ 126
5.5 Enclosures .................................................................................................................. 127
5.6 Impact sound insulation ............................................................................................. 127
5.7 Singlenumber rating of sound insulation .................................................................. 128
5.7.1 The weighted sound reduction index ............................................................. 128
5.7.2 The weighted impact sound pressure level .................................................... 129
5.8 Requirements for sound insulation ............................................................................. 131
5.9 References .................................................................................................................. 131
6 Mechanical vibration and structureborne sound .................................................. 133
Mogens Ohlrich
6.1 Introduction ................................................................................................................ 133
6.1.1 Sources of vibration ....................................................................................... 134
6.1.2 Measurement quantities .................................................................................. 134
6.1.3 Linear mechanical systems ............................................................................ 135
6.2 Simple mechanical resonators .................................................................................... 136
6.2.1 Equation of motion for simple resonator ........................................................ 137
6.2.2 Forced harmonic response of simple resonator .............................................. 138
6.2.3 Frequency response functions ........................................................................ 145
vii
6.2.4 Forced vibration caused by motion excitation ............................................... 147
6.3 Vibration and waves in continuous systems .............................................................. 148
6.3.1 Longitudinal waves ........................................................................................ 149
6.3.2 Shear waves .................................................................................................... 150
6.3.3 Bending waves ............................................................................................... 151
6.3.4 Input mobilities of infinite systems ................................................................ 152
6.4 Vibration isolation and power transmission ............................................................... 153
6.4.1 Estimation of spring stiffness and natural frequency ..................................... 154
6.4.2 Transmission of power in rigidly coupled systems ........................................ 155
6.4.3 Vibration isolated source ................................................................................ 156
6.4.4 Design considerations for resilient elements .................................................. 159
6.5 References .................................................................................................................. 162
List of symbols ....................................................................................................................... 163
Index… ................................................................................................................................... 167
viii
1
1 AN ELEMENTARY INTRODUCTION TO ACOUSTICS
Finn Jacobsen
1.1 INTRODUCTION
Acoustics is the science of sound, that is, wave motion in gases, liquids and solids,
and the effects of such wave motion. Thus the scope of acoustics ranges from fundamental
physical acoustics to, say, bioacoustics, psychoacoustics and music, and includes technical
fields such as transducer technology, sound recording and reproduction, design of theatres
and concert halls, and noise control.
The purpose of this chapter is to give an introduction to fundamental acoustic con
cepts, to the physical principles of acoustic wave motion, and to acoustic measurements.
1.2 FUNDAMENTAL ACOUSTIC CONCEPTS
One of the characteristics of fluids, that is, gases and liquids, is the lack of constraints
to deformation. Fluids are unable to transmit shearing forces, and therefore they react against
a change of shape only because of inertia. On the other hand a fluid reacts against a change in
its volume with a change of the pressure. Sound waves are compressional oscillatory distur
bances that propagate in a fluid. The waves involve molecules of the fluid moving back and
forth in the direction of propagation (with no net flow), accompanied by changes in the pres
sure, density and temperature; see figure 1.2.1. The sound pressure, that is, the difference be
tween the instantaneous value of the total pressure and the static pressure, is the quantity we
hear. It is also much easier to measure the sound pressure than, say, the density or tempera
ture fluctuations. Note that sound waves are longitudinal waves, unlike bending waves on a
beam or waves on a stretched string, which are transversal waves in which the particles move
back and forth in a direction perpendicular to the direction of propagation.
Figure 1.2.1 Fluid particles and compression and rarefaction in the propagating spherical sound field gener
ated by a pulsating sphere. (From ref. [1].)
2
In most cases the oscillatory changes undergone by the fluid are extremely small. One
can get an idea about the orders of magnitude of these changes by considering the variations
in air corresponding to a sound pressure level
1
of 120 dB, which is a very high sound pressure
level, close to the threshold of pain. At this level the fractional pressure variations (the sound
pressure relative to the static pressure) are about
4
10 2
÷
× , the fractional changes of the den
sity are about
4
10 4 . 1
÷
× , the oscillatory changes of the temperature are less than 0.02 °C, and
the particle velocity
2
is about 50 mm/s, which at 1000 Hz corresponds to a particle displace
ment of less than μm 8 . In fact at 1000 Hz the particle displacement at the threshold of hear
ing is less than the diameter of a hydrogen atom!
3
Sound waves exhibit a number of phenomena that are characteristics of waves; see
figure 1.2.2. Waves propagating in different directions interfere; waves will be reflected by a
rigid surface and more or less absorbed by a soft one; they will be scattered by small obsta
cles; because of diffraction there will only partly be shadow behind a screen; and if the me
dium is inhomogeneous for instance because of temperature gradients the waves will be re
fracted, which means that they change direction as they propagate. The speed with which
sound waves propagate in fluids is independent of the frequency, but other waves of interest
in acoustics, bending waves on plates and beams, for example, are dispersive, which means
that the speed of such waves depends on the frequency content of the waveform.
Figure 1.2.2 Various wave phenomena.
A mathematical description of the wave motion in a fluid can be obtained by combin
ing equations that express the facts that i) mass is conserved, ii) the local longitudinal force
caused by a difference in the local pressure is balanced by the inertia of the medium, and iii)
sound is very nearly an adiabatic phenomenon, that is, there is no flow of heat. The observa
tion that most acoustic phenomena involve perturbations that are several orders of magnitude
smaller than the equilibrium values of the medium makes it possible to simplify the mathe
matical description by neglecting higherorder terms. The result is the linearised wave equa
tion. This is a secondorder partial differential equation that, expressed in terms of the sound
1
See section 1.3.2 for a definition of the sound pressure level.
2
The concept of fluid particles refers to a macroscopic average, not to individual molecules; therefore
the particle velocity can be much less than the velocity of the molecules.
3
At these conditions the fractional pressure variations amount to about
10
2.5 10
÷
× . By comparison, a
change in altitude of one metre gives rise to a fractional change in the static pressure that is about 400000 times
larger, about 10
4
. Moreover, inside an aircraft at cruising height the static pressure is typically only 80% of the
static pressure at sea level. In short, the acoustic pressure fluctuations are extremely small compared with com
monly occurring static pressure variations.
3
pressure p, takes the form
2
2
2 2
2
2
2
2
2
1
t
p
c z
p
y
p
x
p
c
c
=
c
c
+
c
c
+
c
c
(1.2.1)
in a Cartesian coordinate system.
4
The physical unit of the sound pressure is pascal (1 Pa = 1
Nm
2
). As we shall see later the quantity
S
c K µ = (1.2.2a)
is the speed of sound. The quantity K
s
is the adiabatic bulk modulus, and ρ is the equilibrium
density of the medium. For gases, K
s
= γp
0
, where ( is the ratio of the specific heat at con
stant pressure to that at constant volume ( 1.401 for air) and p
0
is the static pressure (
101.3 kPa for air under normal ambient conditions). The adiabatic bulk modulus can also be
expressed in terms of the gas constant R ( 287 J·kg
1
K
1
for air), the absolute temperature T,
and the equilibrium density of the medium,
0
c p RT ¸ µ ¸ = = , (1.2.2b)
which shows that the equilibrium density of a gas can be written as
0
. p RT µ = (1.2.3)
At 293.15 K = 20°C the speed of sound in air is 343 m/s. Under normal ambient conditions
(20°C, 101.3 kPa) the density of air is 1.204 kgm
3
. Note that the speed of sound of a gas de
pends only on the temperature, not on the static pressure.
Adiabatic compression
Because the process is adiabatic, the fractional pressure variations in a small cavity driven by a vibrat
ing piston, say, a pistonphone for calibrating microphones, equal the fractional density variations multiplied by
the ratio of specific heats ¸ . The physical explanation for the ‘additional’ pressure is that the pressure in
crease/decrease caused by the reduced/expanded volume of the cavity is accompanied by an increase/decrease
of the temperature, which increases/reduces the pressure even further. The fractional variations in the density are
of course identical with the fractional change of the volume (except for the sign); therefore,
0
.
p V
p V
µ
¸ ¸
µ
A A
= = ÷
In section 1.4 we shall derive a relation between the volume velocity (= the volume displacement V A per unit
of time) and the resulting sound pressure.
Figure 1.2.3 A small cavity driven by a vibrating piston.
4
The lefthand side of eq. (1.2.1) is the Laplacian of the sound pressure, that is, the divergence of the
gradient. A negative value of this quantity at a certain point implies that the gradient converges towards the
point, indicating a high local value. The wave equation states that this high local pressure tends to decrease.
4
The linearity of eq. (1.2.1) is due to the absence of higherorder terms in p in combi
nation with the fact that
2 2
x c c and
2 2
t c c are linear operators.
5
This is an extremely impor
tant property. It implies that a sinusoidal source will generate a sound field in which the pres
sure at all positions varies sinusoidally. It also implies linear superposition: sound waves do
not interact, they simply pass through each other (see figure 1.2.5).
6
The diversity of possible sound fields is of course enormous, which leads to the con
clusion that we must supplement eq. (1.2.1) with some additional information about the
sources that generate the sound field, surfaces that reflect or absorb sound, objects that scatter
sound, etc. This information is known as the boundary conditions. The boundary conditions
are often expressed in terms of the particle velocity. For example, the normal component of
the particle velocity u is zero on a rigid surface. Therefore we need an additional equation
that relates the particle velocity to the sound pressure. This relation is known as Euler’s equa
tion of motion,
, p
t
µ
c
+ V =
c
u
0 (1.2.4)
which is simply Newton’s second law of motion for a fluid. The operator V is the gradient
(the spatial derivative ( z y x c c c c c c , , )). Note that the particle velocity is a vector, unlike
the sound pressure, which is a scalar.
Sound in liquids
The speed of sound is much higher in liquids than in gases. For example, the speed of sound in water is
about 1500 ms
1
. The density of liquids is also much higher; the density of water is about 1000 kgm
3
. Both the
density and the speed of sound depend on the static pressure and the temperature, and there are no simple gen
eral relations corresponding to eqs. (1.2.2b) and (1.2.3).
1.2.1 Plane sound waves
The plane wave is a central concept in acoustics. Plane waves are waves in which any
acoustic variable at a given time is a constant on any plane perpendicular to the direction of
propagation. Such waves can propagate in a duct. In a limited area at a distance far from a
source of sound in free space the curvature of the spherical wavefronts is negligible and the
waves can be regarded as locally plane.
Figure 1.2.4 The sound pressure in a plane wave of arbitrary waveform at two different instants of time.
5
This follows from the fact that
2 2 2 2 2 2
1 2 1 2
( ) . p p t p t p t c + c = c c + c c
6
At very high sound pressure levels, say at levels in excess of 140 dB, the linear approximation is no
longer adequate. This complicates the analysis enormously. Fortunately, we can safely assume linearity under
practically all circumstances encountered in daily life.
5
The plane wave is a solution to the onedimensional wave equation,
,
1
2
2
2 2
2
t
p
c x
p
c
c
=
c
c
(1.2.5)
cf. eq. (1.2.1). It is easy to show that the expression
), ( ) (
2 1
x ct f x ct f p + + ÷ = (1.2.6)
where f
1
and f
2
are arbitrary functions, is a solution to eq. (1.2.5), and it can be shown this is
the general solution. Since the argument of f
1
is constant if x increases as ct it follows that the
first term of this expression represents a wave that propagates undistorted and unattenuated in
the positive xdirection with constant speed, c, whereas the second term represents a similar
wave travelling in the opposite direction. See figures 1.2.4 and 1.2.5.
Figure 1.2.5 Two plane waves travelling in opposite directions are passing through each other.
The special case of a harmonic plane progressive wave is of great importance. Har
monic waves are generated by sinusoidal sources, for example a loudspeaker driven with a
pure tone. A harmonic plane wave propagating in the xdirection can be written
), sin( ) ( sin
1 1
¢ e ¢
e
+ ÷ =

.

\

+ ÷ = kx t p x ct
c
p p (1.2.7)
where 2πf e = is the angular (or radian) frequency and c k e = is the (angular) wavenum
ber. The quantity p
1
is known as the amplitude of the wave, and φ is a phase angle (the arbi
trary value of the phase angle of the wave at the origin of the coordinate system at t = 0). At
any position in this sound field the sound pressure varies sinusoidally with the angular fre
quency ω, and at any fixed time the sound pressure varies sinusoidally with x with the spatial
period
2π 2π
.
c c
f k
ì
e
= = = (1.2.8)
The quantity λ is the wavelength, which is defined as the distance travelled by the wave in
one cycle. Note that the wavelength is inversely proportional to the frequency. At 1000 Hz
6
the wavelength in air is about 34 cm. In rough numbers the audible frequency range goes
from 20 Hz to 20 kHz, which leads to the conclusion that acousticians are faced with wave
lengths (in air) in the range from 17 m at the lowest audible frequency to 17 mm at the high
est audible frequency. Since the efficiency of a radiator of sound or the effect of an obstacle
on the sound field depends very much on its size expressed in terms of the acoustic wave
length, it can be realised that the wide frequency range is one of the challenges in acoustics. It
simplifies the analysis enormously if the wavelength is very long or very short compared with
typical dimensions.
Figure 1.2.6 The sound pressure in a plane harmonic wave at two different instants of time.
Sound fields are often studied frequency by frequency. As already mentioned, linear
ity implies that a sinusoidal source with the frequency ω will generate a sound field that var
ies harmonically with this frequency at all positions.
7
Since the frequency is given, all that
remains to be determined is the amplitude and phase at all positions. This leads to the intro
duction of the complex exponential representation, where the sound pressure is written as a
complex function of the position multiplied with a complex exponential. The former function
takes account of the amplitude and phase, and the latter describes the time dependence. Thus
at any given position the sound pressure can be written as a complex function of the form
8
) j( j j j
e e e e ˆ
¢ e e ¢ e +
= = =
t t t
A A A p (1.2.9)
(where φ is the phase of the complex amplitude A), and the real, physical, timevarying sound
pressure is the real part of the complex pressure,
{ } { }
j( )
ˆ Re Re e cos( ).
t
p p A A t
e ¢
e ¢
+
= = = + (1.2.10)
Since the entire sound field varies as e
jωt
, the operator t c c can be replaced by jω
(because the derivative of e
jωt
with respect to time is jωe
jωt
),
9
and the operator
2 2
t c c can be
replaced by ω
2
. It follows that Euler’s equation of motion can now be written
, ˆ ˆ j 0 u = V + p eµ (1.2.11)
and the wave equation can be simplified to
7
If the source emitted any other signal than a sinusoidal the waveform would in the general case
change with the position in the sound field, because the various frequency components would change amplitude
and phase relative to each other. This explains the usefulness of harmonic analysis.
8
Throughout this note complex variables representing harmonic signals are indicated by carets.
9
The sign of the argument of the exponential is just a convention. The e
jωt
convention is common in
electrical engineering, in audio and in related areas of acoustics. The alternative convention e
jωt
is favoured by
mathematicians, physicists and acousticians concerned with outdoor sound propagation. With the alternative
sign convention t c c should obviously be replaced by jω. Mathematicians and physicists also tend to prefer the
symbol ‘i’ rather than ‘j’ for the imaginary unit.
7
, 0 ˆ
ˆ ˆ ˆ
2
2
2
2
2
2
2
= +
c
c
+
c
c
+
c
c
p k
z
p
y
p
x
p
(1.2.12)
which is known as the Helmholtz equation. See the Appendix (section 1.9) for further details
about complex representation of harmonic signals. We note that the use of complex notation
is mathematically very convenient, which will become apparent later.
Written with complex notation the equation for a plane wave that propagates in the x
direction becomes
. e ˆ
) j(
i
kx t
p p
÷
=
e
(1.2.13)
Equation (1.2.11) shows that the particle velocity is proportional to the gradient of the pres
sure. It follows that the particle velocity in the plane propagating wave given by eq. (1.2.13)
is
.
ˆ
e e
ˆ
j
1
ˆ
) j( i ) j(
i
c
p
c
p
p
k
x
p
u
kx t kx t
x
µ µ eµ eµ
e e
= = =
c
c
÷ =
÷ ÷
(1.2.14)
Thus the sound pressure and the particle velocity are in phase in a plane propagating wave
(see also figure 1.2.10), and the ratio of the sound pressure to the particle velocity is ρc, the
characteristic impedance of the medium. As the name implies, this quantity describes an im
portant acoustic property of the fluid, as will become apparent later. The characteristic im
pedance of air at 20°C and 101.3 kPa is about 413 kg·m
2
s
1
.
Figure 1.2.7 A semiinfinite tube driven by a piston.
Example 1.2.1
An semiinfinite tube is driven by a piston with the vibrational velocity
j
e
t
U
e
as shown in figure 1.2.7.
Because the tube is infinite there is no reflected wave, so the sound field can be written
j( ) j( ) i
i
ˆ ˆ ( ) e , ( ) e .
t kx t kx
x
p
p x p u x
c
e e
µ
÷ ÷
= =
The boundary condition at the piston implies that the particle velocity equals the velocity of the piston:
j j i
ˆ (0) e e
t t
x
p
u U
c
e e
µ
= = .
It follows that the sound pressure generated by the piston is
j( )
ˆ ( ) e .
t kx
p x U c
e
µ
÷
=
The general solution to the onedimensional Helmholtz equation is
j( ) j( )
i r
ˆ e e ,
t kx t kx
p p p
e e ÷ +
= + (1.2.15)
8
which can be identified as the sum of a wave that travels in the positive xdirection and a
wave that travels in the opposite direction (cf. eq. (1.2.6)). The corresponding expression for
the particle velocity becomes, from eq. (1.2.11),
. e e
e e
ˆ
j
1
ˆ
) j( r ) j( i
) j(
r
) j(
i
kx t kx t
kx t kx t
x
c
p
c
p
p
k
p
k
x
p
u
+ ÷
+ ÷
÷ =
÷ =
c
c
÷ =
e e
e e
µ µ
eµ eµ eµ
(1.2.16)
It can be seen that whereas c u p
x
µ ˆ ˆ = in a plane wave that propagates in the positive x
direction, the sign is the opposite, that is, c u p
x
µ ˆ ˆ ÷ = , in a plane wave that propagates in the
negative xdirection. The reason for the change in the sign is that the particle velocity is a
vector, unlike the sound pressure, so
x
uˆ is a vector component. It is also worth noting that the
general relation between the sound pressure and the particle velocity in this interference field
is far more complicated than in a plane propagating wave.
Figure 1.2.8 Instantaneous sound pressure in a wave that is reflected from a rigid surface at different instants of
time. (Adapted from ref. [2].)
A plane wave that impinges on a plane rigid surface perpendicular to the direction of
propagation will be reflected. This phenomenon is illustrated in figure 1.2.8, which shows
how an incident transient disturbance is reflected. Note that the normal component of the
gradient of the pressure is identically zero on the surface for all values of t. This is a conse
quence of the fact that the boundary condition at the surface implies that the particle velocity
must equal zero here, cf. eq. (1.2.4).
However, it is easier to analyse the phenomenon assuming harmonic waves. In this
case the sound field is given by the general expressions (1.2.15) and (1.2.16), and our task is
to determine the relation between p
i
and p
r
from the boundary condition at the surface, say at
x = 0. As mentioned, the rigid surface implies that the particle velocity must be zero here,
which with eq. (1.2.16) leads to the conclusion that p
i
= p
r
, so the reflected wave has the
same amplitude as the incident wave. Equation (1.2.15) now becomes
( ) ( ) , e cos 2 e e e e e ˆ
j
i
j j j
i
) j( ) j(
i
t t kx kx kx t kx t
kx p p p p
e e e e
= + = + =
÷ + ÷
(1.2.17)
and eq. (1.2.16) becomes
9
. e sin
2
j ˆ
j i t
x
kx
c
p
u
e
µ
÷ = (1.2.18)
Note that the amplitude of the sound pressure is doubled on the surface (cf. figure
1.2.8). Note also the nodal
10
planes where the sound pressure is zero at x =  λ/4, x =  3λ/4,
etc., and the planes where the particle velocity is zero at x =  λ/2, x =  λ, etc. The interfer
ence of the two plane waves travelling in opposite directions has produced a standing wave
pattern, shown in figure 1.2.9.
The physical explanation of the fact that the sound pressure is identically zero at a dis
tance of a quarter of a wavelength from the reflecting plane is that the incident wave must
travel a distance of half a wavelength before it returns to the same point; accordingly the in
cident and reflected waves are in antiphase (that is, 180° out of phase), and since they have
the same amplitude they cancel each other. This phenomenon is called destructive interfer
ence. At a distance of half a wavelength from the reflecting plane the incident wave must
travel one wavelength before it returns to the same point. Accordingly, the sound pressure is
doubled here (constructive interference). The corresponding pattern for the particle velocity
is different because the particle velocity is a vector.
Another interesting observation from eqs. (1.2.17) and (1.2.18) is that the resulting
sound pressure and particle velocity signals at any position are 90° out of phase (otherwise
expressed, if the sound pressure as a function of time is a cosine then the particle velocity is a
sine). As we shall see later this indicates that there is no net flow of sound energy towards the
rigid surface. See also figure 1.2.10.
Figure 1.2.9 Standing wave pattern caused by reflection from a rigid surface at x = 0; amplitudes of the sound
pressure and the particle velocity.
Example 1.2.2
The standing wave phenomenon can be observed in a tube terminated by a rigid cap. When the length
of the tube, l, equals an oddnumbered multiple of a quarter of a wavelength the sound pressure is zero at the
input, which means that it would take very little force to drive a piston here. This is an example of an acoustic
resonance. In this case it occurs at the frequency
0
,
4
c
f
l
=
10
A node on, say, a vibrating string is a point that does not move, and an antinode is a point with
maximum displacement. By analogy, points in a standing wave at which the sound pressure is identically zero
are called pressure nodes. In this case the pressure nodes coincide with velocity antinodes.
10
and at oddnumbered multiples of this frequency, 3f
0
, 5f
0
, 7f
0
, etc. Note that the resonances are harmonically
related. This means that if some mechanism excites the tube the result will be a musical sound with the funda
mental frequency f and overtones corresponding to oddnumbered harmonics.
11
Brass and woodwind instruments are based on standing waves in tubes. For example, closed organ
pipes are tubes closed at one end and driven at the other, open end, and such pipes have only oddnumbered
harmonics. See also example 1.4.4.
The ratio of p
r
to p
i
is the (complex) reflection factor R. The amplitude of this quantity
describes how well the reflecting surface reflects sound. In the case of a rigid plane R = 1, as
we have seen, which implies perfect reflection with no phase shift. However, in the general
case of a more or less absorbing surface R will be complex and its magnitude less than unity
(R ≤ 1), indicating partial reflection with a phase shift at the reflection plane.
If we introduce the reflection factor in eq. (1.2.15) it becomes
( )
j( ) j( )
i
ˆ e e
t kx t kx
p p R
e e ÷ +
= + , (1.2.19)
from which it can be seen that the amplitude of the sound pressure varies with the position in
the sound field. When the two terms in the parenthesis are in phase the sound pressure ampli
tude assumes its maximum value,
( )
max i
1 p p R = + , (1.2.20a)
and when they are in antiphase the sound pressure amplitude assumes the minimum value
( )
min i
1 p p R = ÷ . (1.2.20b)
The ratio of p
max
to p
min
is called the standing wave ratio,
max
min
1
.
1
R
p
s
p R
+
= =
÷
(1.2.21)
From eq. (1.2.21) it follows that
,
1
1
+
÷
=
s
s
R (1.2.22)
which leads to the conclusion that it is possible to determine the acoustic properties of a ma
terial by exposing it to normal sound incidence and measuring the standing wave ratio in the
resulting interference field. See also chapter 1.5.
Figure 1.2.10 shows the instantaneous sound pressure and particle velocity at two dif
ferent instants of time in a tube that is terminated by a material that does not reflect sound at
all (case (a)), by a soft material that partly absorbs the incident sound wave (case (b)), and by
a rigid material that gives perfect reflection (case (c)).
11
A musical (or complex) tone is not a pure (sinusoidal) tone but a periodic signal, usually consisting
of the fundamental and a number of its harmonics, also called partials. These pure tones occur at multiples of the
fundamental frequency. The n’th harmonic (or partial) is also called the (n1)’th overtone, and the fundamental
is the first harmonic. The relative position of a tone on a musical scale is called the pitch [2]. The pitch of a mu
sical tone essentially corresponds to its fundamental frequency, which is also the distance between two adjacent
harmonic components. However, pitch is a subjective phenomenon and not completely equivalent to frequency.
We tend to determine the pitch on the basis of the spacing between the harmonic components, and thus we can
detect the pitch of a musical tone even if the fundamental is missing.
11
Figure 1.2.10 Spatial distributions of instantaneous sound pressure and particle velocity at two different in
stants of time. (a) Case with no reflection (R = 0); (b) case with partial reflection from a soft surface; (c) case
with perfect reflection from a rigid surface (R = 1). (From ref. [3].)
Sound transmission between fluids
When a sound wave in one fluid is incident on the boundary of another fluid, say, a sound wave in air
is incident on the surface of water, it will be partly reflected and partly transmitted. For simplicity let us assume
that a plane wave in fluid 1 strikes the surface of fluid 2 at normal incidence as shown in figure 1.2.11. Antici
pating a reflected wave we can write
j( ) j( )
1 i r
ˆ e e
t kx t kx
p p p
e e ÷ +
= +
for fluid 1, and
j( )
2 t
ˆ e
t kx
p p
e ÷
=
12
for fluid 2. There are two boundary conditions at the interface: the sound pressure must be the same in fluid 1
and in fluid 2 (otherwise there would be a net force), and the particle velocity must be the same in fluid 1 and in
fluid 2 (otherwise the fluids would not remain in contact). It follows that
i r t
p p p + = and
t i r
1 1 2 2
.
p p p
c c µ µ
÷
=
Combining these equations gives
r 2 2 1 1
i 2 2 1 1
,
p c c
R
p c c
µ µ
µ µ
÷
= =
+
which shows that the wave is almost fully reflected in phase ( 1 R ) if ρ
2
c
2
>> ρ
1
c
1
, almost fully reflected in
antiphase ( 1 R ÷ ) if ρ
2
c
2
<< ρ
1
c
1
, and not reflected at all if ρ
2
c
2
= ρ
1
c
1
, irrespective of the individual properties
of c
1
, c
2
, ρ
1
and ρ
2
.
Figure 1.2.11 Reflection and transmission of a plane wave incident on the interface between two fluids.
Figure 1.2.12 Reflection of a pressure wave at the interface between a medium of high characteristic impedance
and a medium of low characteristic impedance. (Adapted from ref. [2].)
Figure 1.2.13 Standing wave pattern in a medium of high characteristic impedance caused by caused by reflec
tion from a medium of low characteristic impedance; amplitudes of the sound pressure and the particle velocity.
13
Because of the significant difference between the characteristic impedances of air and water (the ratio is about 1:
3600) a sound wave in air that strikes a surface of water at normal incidence is almost completely reflected, and
so is a sound wave that strikes the airwater interface from the water, but in the latter case the phase of the re
flected wave is reversed, as shown in figure 1.2.12. Compare figures 1.2.8 and 1.2.12, and figures 1.2.9 and
1.2.13.
1.2.2 Spherical sound waves
The wave equation can be expressed in other coordinate systems than the Cartesian. If
sound is generated by a source in an environment without reflections (which is usually re
ferred to as a free field) it will generally be more useful to express the wave equation in a
spherical coordinate system (r, θ, φ). The resulting equation is more complicated than eq.
(1.2.1). However, if the source under study is spherically symmetric there can be no angular
dependence, and the equation becomes quite simple,
12
2 2
2 2 2
2 1
.
p p p
r r r c t
c c c
+ =
c c c
(1.2.23a)
If we rewrite in the form
,
) ( 1 ) (
2
2
2 2
2
t
rp
c r
rp
c
c
=
c
c
(1.2.23b)
it becomes apparent that this equation is identical in form with the onedimensional wave
equation, eq. (1.2.5), although p has been replaced by rp. (It is easy to get from eq. (1.2.23b)
to eq. (1.2.23a); it is more difficult the other way.) It follows that the general solution to eq.
(1.2.23) can be written
), ( ) (
2 1
r ct f r ct f rp + + ÷ = (1.2.24a)
12
This can be seen as follows. Since the sound pressure depends only on r we have
,
p p r
x r x
c c c
=
c c c
which, with
2 2 2
, r x y z = + +
becomes
.
p x p
x r r
c c
=
c c
Similar considerations leads to the following expression for the secondorder derivative,
2 2 2 2 2
2 2 2 3
1 1 1 1 1
.
p p p p x p p x p x p
x
r r x r r r r r r r r r r r x r r r
c c c c c c c c c c    
= + = + = + ÷
 
c c c c c c c c c c
\ . \ .
Combining eq. (1.2.1) with this expression and the corresponding relations for y and z finally yields eq.
(1.2.23a):
2 2 2 2 2 2 2 2 2 2 2 2
2 2 2 2 2 3 2 2 2
3 2 1
.
p p p p x y z p x y z p p p p
r r r r r x y z r r r r c t
c c c c + + c + + c c c c
+ + = + ÷ = + =
c c c c c c c c c
14
that is
( ), ) ( ) (
1
2 1
r ct f r ct f
r
p + + ÷ = (1.2.24b)
where f
1
and f
2
are arbitrary functions. The first term is wave that travels outwards, away
from the source (cf. the first term of eq. (1.2.6)). Note that the shape of the wave is preserved.
However, the sound pressure is seen to decrease in inverse proportion to the distance. This is
the inverse distance law.
13
The second term represents a converging wave, that is, a spherical
wave travelling inwards. In principle such a wave could be generated by a reflecting spherical
surface centred at the source, but that is a rare phenomenon indeed. Accordingly we will ig
nore the second term when we study sound radiation in chapter 1.6.
Figure 1.2.14 (a) Measurement far from a spherical source in free space; (b) measurement close to a spherical
source. ––, Instantaneous sound pressure;   , instantaneous particle velocity multiplied by Dc. (From ref. [4].)
A harmonic spherical wave is a solution to the Helmholtz equation
. 0 ˆ
) ˆ (
2
2
2
= +
c
c
p r k
r
p r
(1.2.25)
Expressed in the complex notation the diverging wave can be written
.
e
ˆ
) j(
r
A p
kr t ÷
=
e
(1.2.26)
13
The inverse distance law is also known as the inverse square law because the sound intensity is in
versely proportional to the square of the distance to the source. See chapters 1.5 and 1.6.
15
The particle velocity component in the radial direction can be calculated from eq. (1.2.11),
.
j
1
1
ˆ
j
1
1
e ˆ
j
1
ˆ
) j(


.

\

+ =


.

\

+ =
c
c
÷ =
÷
kr c
p
kr r c
A
r
p
u
kr t
r
µ µ eµ
e
(1.2.27)
Because of the spherical symmetry there are no components in the other directions. Note that
far
14
from the source the sound pressure and the particle velocity are in phase and their ratio
equals the characteristic impedance of the medium, just as in a plane wave. On the other
hand, when kr << 1 the particle velocity is larger than c p µ ˆ and the sound pressure and the
particle velocity are almost in quadrature, that is, 90° out of phase. These are near field char
acteristics, and such a sound field is also known as a reactive field. See figure 1.2.14.
1.3 ACOUSTIC MEASUREMENTS
The most important measure of sound is the rms sound pressure,
15
defined as
½
2 2
rms
0
1
( ) lim ( )d .
T
T
p p t p t t
T
÷·
 
= =

\ .
}
(1.3.1)
However, as we shall see, a frequency weighting filter
16
is usually applied to the signal before
the rms value is determined. Quite often such a single value does not give sufficient informa
tion about the nature of the sound, and therefore the rms sound pressure is determined in fre
quency bands. The resulting sound pressures are practically always compressed logarithmi
cally and presented in decibels.
Example 1.3.1
The fact that sin
2
ωt = ½ (1  cos2ωt) and thus has a time average of ½ leads to the conclusion that the
rms value of a sinusoidal signal with the amplitude A is / 2 A .
1.3.1 Frequency analysis
Single frequency sound is useful for analysing acoustic phenomena, but most sounds
encountered in practice have ‘broadband’ characteristics, which means that they cover a wide
frequency range. If the sound is more or less steady, it will practically always be more useful
to analyse it in the frequency domain than to look at the sound pressure as a function of time.
Frequency (or spectral) analysis of a signal involves decomposing the signal into its
spectral components. This analysis can be carried out by means of digital analysers that em
ploy the discrete Fourier transform (‘FFT analysers’). This topic is outside the scope of this
note, but see, e.g., refs. [5, 6]. Alternatively, the signal can be passed through a number of
14
In acoustics, dimensions are measured in terms of the wavelength, so that ‘far from’ means that r >>
λ (or kr >> 1), just as ‘near’ means that r << λ (or kr << 1). The dimensionless quantity kr is known as the
Helmholtz number.
15
Root mean square value, usually abbreviated rms. This is the square root of the mean square value,
which is the time average of the squared signal.
16
A filter is a device that modifies a signal by attenuating some of its frequency components.
16
contiguous analogue or digital bandpass filters
17
with different centre frequencies, a ‘filter
bank.’ The filters can have the same bandwidth or they can have constant relative bandwidth,
which means that the bandwidth is a certain percentage of the centre frequency. Constant
relative bandwidth corresponds to uniform resolution on a logarithmic frequency scale. Such
a scale is in much better agreement with the subjective pitch of musical sounds than a linear
scale, and therefore frequencies are often represented on a logarithmic scale in acoustics, and
frequency analysis is often carried out with constant percentage filters. The most common
filters in acoustics are octave band filters and onethird octave band filters.
Figure 1.3.1 The keyboard of a small piano. The white keys from C to B correspond to the seven notes of the C
major scale. (Adapted from ref. [7].)
An octave
18
is a frequency ratio of 2:1, a fundamental unit in musical scales. Accord
ingly, the lower limiting frequency of an octave band is half the upper frequency limit, and
the centre frequency is the geometric mean, that is,
, , 2 , 2
u l c c
½
u
½
c l
f f f f f f f = = = (1.3.2a, 1.3.2b, 1.3.2c)
where f
c
is the centre frequency. In a similar manner a onethird octave
19
band is a band for
which f
u
= 2
⅓
f
l
, and
, , 2 , 2
u l c c u c l
6
1
6
1
f f f f f f f = = = (1.3.3a, 1.3.3b, 1.3.3c)
Since
10 3
2 1024 10 = it follows that
10 3
2 10 and
1 3 1 10
2 10 , that is, ten onethird
octaves very nearly make a decade, and a onethird octave is almost identical with one tenth
of a decade. Table 1.3.1 gives the nominal centre frequencies of standardised octave and one
third octave band filters.
20
As mentioned earlier, the human ear may respond to frequencies in
the range from 20 Hz to 20 kHz, that is, a range of three decades, ten octaves or thirty one
third octaves.
17
An ideal bandpass filter would allow frequency components in the passband to pass unattenuated,
but would completely remove frequency components outside the passband. Real filters have, of course, a certain
passband ripple and a finite stopband attenuation.
18
Musical tones an octave apart sound very similar. The diatonic scale contains seven notes per octave
corresponding to the white keys on a piano keyboard; see figure 1.3.1. Thus an octave spans eight notes, say,
from C to C'; hence the name octave (from Latin octo: eight).
19
A semitone is one twelfth of an octave on the equally tempered scale (a frequency ratio of 2
1/12
:1).
Since 2
⅓
= 2
4/12
it can be seen that a onethird octave is identical with four semitones or a major third (e.g. from
C to E, cf. figure 1.3.1). Accordingly, onethird octave band filters are called Terzfilters in German.
20
Round numbers are convenient. The standardised nominal centre frequencies are based on the fact
that the series 1.25, 1.6, 2, 2.5, 3.15, 4, 5, 6.3, 8, 10 is in reasonable agreement with 10
n/10
, with n = 1, 2,....., 10.
17
Table 1.3.1 Standardised onethird octave and octave (bold characters) band centre frequencies (in hertz).
20 25 31.5 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1000
1250 1600 2000 2500 3150 4000 5000 6300 8000 10000 12500 16000 20000
An important property of the mean square value of a signal is that it can be partitioned
into frequency bands. This means that if we analyse a signal in, say, onethird octave bands,
the sum of the mean square values of the filtered signals equals the mean square value of the
unfiltered signal. The reason is that products of different frequency components average to
zero, so that all cross terms vanish; the different frequency components are uncorrelated sig
nals. This can be illustrated by analysing a sum of two pure tones with different frequencies,
. 2 / ) (
sin sin 2 sin sin ) sin sin (
2 2
2 1 2
2 2
1
2 2 2
2 1
B A
t t AB t B t A t B t A
+ =
+ + = + e e e e e e
(1.3.4)
Note that the mean square values of the two signals are added unless ω
1
= ω
2
. The validity of
this rule is not restricted to pure tones of different frequency; the mean square value of any
stationary signal equals the sum of mean square values of its frequency components, which
can be determined with a parallel bank of contiguous filters. Thus
2 2
rms rms,
,
i
i
p p =
¿
(1.3.5)
where p
rms,i
is the rms value of the output of the i’th filter. Equation (1.3.5) is known as
Parseval’s formula.
Random noise
Many generators of sound produce noise rather than pure tones. Whereas pure tones and other periodic
signals are deterministic, noise is a stochastic or random phenomenon. Stationary noise is a stochastic signal
with statistical properties that do not change with time.
White noise is stationary noise with a flat power spectral density, that is, constant mean square value
per hertz. The term white noise is an analogy to white light. When white noise is passed through a bandpass
filter, the mean square of the output signal is proportional to the bandwidth of the filter. It follows that when
white noise is analysed with constant percentage filters, the mean square of the output is proportional to the cen
tre frequency of the filter. For example, if white noise is analysed with a bank of octave band filters, the mean
square values of the output signals of two adjacent filters differ by a factor of two.
Pink noise is stationary noise with constant mean square value in bands with constant relative width,
e.g., octave bands. Thus compared with white noise low frequencies are emphasised; hence the name pink noise,
which is an analogy to an optical phenomenon. It follows that the mean square value of a given pink noise sig
nal in octave bands is three times larger than the mean square value of the noise in onethird octave bands.
Example 1.3.2
The fact that noise, unlike periodic signals, has a finite power spectral density (mean square value per
hertz) implies that one can detect a pure tone in noise irrespective of the signaltonoise ratio by analysing with
sufficiently fine spectral resolution: As the bandwidth is reduced, less and less noise passes through the filter,
and the tone will emerge. Compared with filter bank analysers FFT analysers have the advantage that the spec
tral resolution can be varied over a wide range [6]; therefore FFT analysers are particular suitable for detecting
tones in noise.
When several independent sources of noise are present at the same time the mean
square sound pressures generated by the individual sources are additive. This is due to the
18
fact that independent sources generate uncorrelated signals, that is, signals whose product av
erage to zero; therefore the cross terms vanish:
. ) ( ) ( ) ( ) ( 2 ) ( ) ( )) ( ) ( (
2
2
2
1 2 1
2
2
2
1
2
2 1
t p t p t p t p t p t p t p t p + = + + = + (1.3.6)
It follows that
¿
=
i
i
p p .
2
rms,
2
tot rms,
(1.3.7)
Note the similarity between eqs. (1.3.5) and (1.3.7). It is of enormous practical importance
that the mean square values of uncorrelated signals are additive, because signals generated by
different mechanisms are invariably uncorrelated. Almost all signals that occur in real life are
mutually uncorrelated.
Example 1.3.3
Equation (1.3.7) leads to the conclusion that the mean square pressure generated by a crowd of noisy
people in a room is proportional to the number of people. Thus the rms value of the sound pressure in the room
is proportional to the square root of the number of people.
Example 1.3.4
Consider the case where the rms sound pressure generated by a source of noise is to be measured in the
presence of background noise that cannot be turned off. It follows from eq. (1.3.7) that it is possible to correct
the measurement for the influence of the stationary background noise; one simply subtracts the mean square
value of the background noise from the total mean square pressure. For this to work in practice the background
noise must not be too strong, though, and it is absolutely necessary that it is completely stationary.
1.3.2 Levels and decibels
The human auditory system can cope with sound pressure variations over a range of
more than a million times. Because of this wide range, the sound pressure and other acoustic
quantities are usually measured on a logarithmic scale. An additional reason is that the sub
jective impression of how loud noise sounds correlates much better with a logarithmic meas
ure of the sound pressure than with the sound pressure itself. The unit is the decibel,
21
abbre
viated dB, which is a relative measure, requiring a reference quantity. The results are called
levels. The sound pressure level (sometimes abbreviated SPL) is defined as
, log 20 log 10
ref
rms
10 2
ref
2
rms
10
p
p
p
p
L
p
= = (1.3.8)
where p
ref
is the reference sound pressure, and log
10
is the base 10 logarithm, henceforth writ
ten log. The reference sound pressure is 20 μPa for sound in air, corresponding roughly to the
lowest audible sound at 1 kHz.
22
Some typical sound pressure levels are given in figure 1.3.2.
21
As the name implies, the decibel is one tenth of a bel. However, the bel is rarely used today. The use
of decibels rather than bels is probably due to the fact that most sound pressure levels encountered in practice
take values between 10 and 120 when measured in decibels, as can be seen in figure 1.3.2. Another reason might
be that to be audible, the change of the level of a given (broadband) sound must be of the order of one decibel.
22
For sound in other fluids than atmospheric air (water, for example) the reference sound pressure is 1
:Pa. To avoid possible confusion it may be advisable to state the reference sound pressure explicitly, e.g., ‘the
sound pressure level is 77 dB re 20 μPa.’
19
Figure 1.3.2 Typical sound pressure levels. (Source: Brüel & Kjær.)
20
The fact that the mean square sound pressures of independent sources are additive (cf.
eq. (1.3.7)) leads to the conclusion that the levels of such sources are combined as follows:
,
0.1
,tot
10log 10 .
p i
L
p
i
L
 
=

\ .
¿
(1.3.9)
Another consequence of eq. (1.3.7) is that one can correct a measurement of the sound
pressure level generated by a source for the influence of steady background noise as follows:
( )
,tot ,background
0.1 0.1
,source
10log 10 10
p p
L L
p
L = ÷ . (1.3.10)
This corresponds to subtracting the mean square sound pressure of the background noise
from the total mean square sound pressure as described in example 1.3.5. However, since all
measurements are subject to random errors, the result of the correction will be reliable only if
the background level is at least, say, 3 dB below the total sound pressure level. If the back
ground noise is more than 10 dB below the total level the correction is less than 0.5 dB.
Example 1.3.5
Expressed in terms of sound pressure levels the inverse distance law states that the level decreases by 6
dB when the distance to the source is doubled.
Example 1.3.6
When each of two independent sources in the absence of the other generates a sound pressure level of
70 dB at a certain point, the resulting sound pressure level is 73 dB (not 140 dB!), because 10log 2 3 . If one
source creates a sound pressure level of 65 dB and the other a sound pressure level of 59 dB, the total level is
6.5 5.9
10log(10 10 ) 66 dB + .
Example 1.3.7
Say the task is to determine the sound pressure level generated by a source in background noise with a
level of 59 dB. If the total sound pressure level is 66 dB, it follows from eq. (1.3.10) that the source would have
produced a sound pressure level of
6.6 5.9
10log(10 10 ) 65 dB ÷ in the absence of the background noise.
Example 1.3.8
When two sinusoidal sources emit pure tones of the same frequency they create an interference field,
and depending on the phase difference the total sound pressure amplitude at a given position will assume a value
between the sum of the two amplitudes and the difference:
A B
j j j j
e e e e .
t t
A B A B A B A B A B
¢ ¢ e e
÷ s + = + = + s +
For example, if two pure tone sources of the same frequency each generates a sound pressure level of 70 dB in
the absence of the other source then the total sound pressure level can be anywhere between 76 dB (constructive
interference) and  ∞ dB (destructive interference). Note that eqs. (1.3.7) and (1.3.9) do not apply in this case
because the signals are not uncorrelated. See also figure 1.9.2 in the Appendix.
Other firstorder acoustic quantities, for example the particle velocity, are also often
measured on a logarithmic scale. The reference velocity is 1 nm/s = 10
9
m/s.
23
This reference
is also used in measurements of the vibratory velocities of vibrating structures.
The acoustic secondorder quantities sound intensity and sound power, defined in
chapter 1.5, are also measured on a logarithmic scale. The sound intensity level is
23
The prefix n (for ‘nano’) represents a factor of 10
9
.
21
, log 10
ref
I
I
L
I
= (1.3.11)
where I is the intensity and I
ref
= 1 pWm
2
= 10
12
Wm
2
,
24
and the sound power level is
, log 10
ref
a
P
P
L
W
= (1.3.12)
where P
a
is the sound power and P
ref
= 1 pW. Note than levels of linear quantities (pressure,
particle velocity) are defined as twenty times the logarithm of the ratio of the rms value to a
reference value, whereas levels of secondorder (quadratic) quantities are defined as ten times
the logarithm, in agreement with the fact that if the linear quantities are doubled then quanti
ties of second order are quadrupled.
Example 1.3.9
It follows from the constant spectral density of white noise that when such a signal is analysed in one
third octave bands, the level increases 1 dB from one band to the next
1 3
(10log(2 ) 1dB) .
1.3.3 Noise measurement techniques and instrumentation
A sound level meter is an instrument designed to measure sound pressure levels. To
day such instruments can be anything from fairly simple devices with analogue filters and
detectors and a moving coil meter to advanced digital analysers. Figure 1.3.3 shows a block
diagram of a simple sound level meter. The microphone converts the sound pressure to an
electrical signal, which is amplified and passes through various filters. After this the signal is
squared and averaged with a detector, and the result is finally converted to decibels and
shown on a display. In the following a very brief description of such an instrument will be
given; see e.g. refs. [8, 9] for further details.
The most commonly used microphones for this purpose are condenser microphones,
which are more stable and accurate than other types. The diaphragm of a condenser micro
phone is a very thin, highly tensioned foil. Inside the housing of the microphone cartridge is
the other part of the capacitor, the back plate, placed very close to the diaphragm (see figure
1.3.4). The capacitor is electrically charged, either by an external voltage on the back plate or
(in case of prepolarised electret microphones) by properties of the diaphragm or the back
plate. When the diaphragm moves in response to the sound pressure, the capacitance changes,
and this produces an electrical voltage proportional to the instantaneous sound pressure.
Figure 1.3.3 A sound level meter. (From ref. [10].)
24
The prefix p (for ‘pico’) represents a factor of 10
12
.
22
Figure 1.3.4 A condenser microphone. (From ref. [11].)
Figure 1.3.5 The ‘freefield correction’ of a typical measurement microphone for sound coming from various
directions. The freefield correction is the fractional increase of the sound pressure (usually expressed in dB)
caused by the presence of the microphone in the sound field. (From ref. [11].)
Figure 1.3.6 Freefield response of a microphone of the ‘freefield’ type at axial incidence. (From ref. [11].)
The microphone should be as small as possible so as not to disturb the sound field.
However, this is in conflict with the requirement of a high sensitivity and a low inherent
noise level, and typical measurement microphones are ‘½inch’ microphones with a diameter
23
of about 13 mm. At low frequencies, say below 1 kHz, such a microphone is much smaller
than the wavelength and does not disturb the sound field appreciably. In this frequency range
the microphone is omnidirectional as of course it should be since the sound pressure is a sca
lar and has no direction. However, from a few kilohertz and upwards the size of the micro
phone is no longer negligible compared with the wavelength, and therefore it is no longer
omnidirectional, which means that its response varies with the nature of the sound field; see
figure 1.3.5.
One can design condenser microphones to have a flat response in as wide a frequency
range as possible under specified sound field conditions. For example, ‘freefield’ micro
phones are designed to have a flat response for axial incidence (see figure 1.3.6), and such
microphones should therefore be pointed towards the source. ‘Randomincidence’ micro
phones are designed for measurements in a diffuse sound field where sound is arriving from
all directions, and ‘pressure’ microphones are intended for measurements in small cavities.
The sensitivity of the human auditory system varies significantly with the frequency
in a way that changes with the level. In particular the human ear is, at low levels, much less
sensitive to low frequencies than to medium frequencies. This is the background for the stan
dardised frequency weighting filters shown in figure 1.3.7. The original intention was to
simulate a human ear at various levels, but it has long ago been realised that the human audi
tory system is far more complicated than implied by such simple weighting curves, and B
and Dweighting filters are little used today. On the other hand the Aweighted sound pres
sure level is the most widely used singlevalue measure of sound, because the Aweighted
sound pressure level correlates in general much better with the subjective effect of noise than
measurements of the sound pressure level with a flat frequency response. Cweighting, which
is essentially flat in the audible frequency range, is sometimes used in combination with A
weighting, because a large difference between the Aweighted level and the Cweighted level
is a clear indication of a prominent content of low frequency noise. The results of measure
ments of the A and Cweighted sound pressure level are denoted L
A
and L
C
respectively, and
the unit is dB.
25
If no weighting filter is applied, the level is sometimes denoted L
Z
.
Figure 1.3.7 Standardised frequency weighting curves. (From ref. [8].)
25
In practice the unit is often written dB (A) and dB (C), respectively.
24
Table 1.3.2 The response of standard A and Cweighting filters in onethird octave bands.
Centre frequency (Hz) Aweighting (dB) Cweighting (dB)
8
10
12.5
16
20
25
31.5
40
50
63
80
100
125
160
200
250
315
400
500
630
800
1000
1250
1600
2000
2500
3150
4000
5000
6300
8000
10000
12500
16000
20000
77.8
70.4
63.4
56.7
50.5
44.7
39.4
34.6
30.2
26.2
22.5
19.1
16.1
13.4
10.9
8.6
6.6
4.8
3.2
1.9
0.8
0.0
0.6
1.0
1.2
1.3
1.2
1.0
0.5
0.1
1.1
2.5
4.3
6.6
9.3
20.0
14.3
11.2
8.5
6.2
4.4
3.0
2.0
1.3
0.8
0.5
0.3
0.2
0.1
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.1
0.2
0.3
0.5
0.8
1.3
2.0
3.0
4.4
6.2
8.5
11.2
In the measurement instrument the frequency weighting filter is followed by a squar
ing device, a lowpass filter that smooths out the instantaneous fluctuations, and a logarithmic
converter. The lowpass filter corresponds to applying a time weighting function. The most
common time weighting in sound level meters is exponential, which implies that the squared
signal is smoothed with a decaying exponential so that recent data are given more weight
than older data:
2 ( ) / 2
ref
1
( ) 10log ( ) e d .
t
t u
p
L t p u u p
t
t
÷ ÷
÷·
 
 
=
 
\ .
\ .
}
(1.3.13)
Two values of the time constant τ are standardised: S (for ‘slow’) corresponds to a time con
stant of 1 s, and F (for ‘fast’) is exponential averaging with a time constant of 125 ms.
The alternative to exponential averaging is linear (or integrating) averaging, in which
all the sound is weighted uniformly during the integration. The equivalent sound pressure
level is defined as
25
2
1
2 2
eq ref
2 1
1
10log ( )d .
t
t
L p t t p
t t
 
 
=



÷
\ .
\ .
}
(1.3.14)
Measurements of random noise with a finite integration time are subject to random
errors that depend on the bandwidth of the signal and on the integration time. It can be shown
that the variance of the measurement result is inversely proportional to the product of the
bandwidth and the integration time [6].
26
As can be seen by comparing with eqs. (1.3.1) and (1.3.8), the equivalent sound pres
sure level is just the sound pressure level corresponding to the rms sound pressure determined
with a specified integration period. The Aweighted equivalent sound pressure level L
Aeq
is
the level corresponding to a similar time integral of the Aweighted instantaneous sound
pressure. Sometimes the quantity is written L
Aeq,T
where T is the integration time.
Whereas exponential averaging corresponds to a running average and thus gives a
(smoothed) measure of the sound at any instant of time, the equivalent sound pressure level
(with or without Aweighting) can be used for characterising the total effect of fluctuating
noise, for example noise from road traffic. Typical values of T are 30 s for measurement of
noise from technical installations, 8 h for noise in a working environment and 24 h for traffic
noise.
Sometimes it is useful to analyse noise signals in onethird octave bands, cf. section
1.3.1. From eq. (1.3.5) it can be seen that the total sound pressure level can be calculated
from the levels in the individual onethird octave bands, L
i
, as follows:
0.1
Z
10log 10 .
i
L
i
L
 
=

\ .
¿
(1.3.15)
In a similar manner one can calculate the Aweighted sound pressure level from the onethird
octave band values and the attenuation data given in table 1.3.2,
, 10 log 10
) ( 1 . 0
A

.

\

=
¿
+
i
K L
i i
L (1.3.16)
where K
i
is the relative response of the Aweighting filter (in dB) in the i’th band, given in
table 1.3.2.
Example 1.3.10
A source gives rise to the following onethird octave band values of the sound pressure level at a cer
tain point,
Centre frequency (Hz) Sound pressure level (dB)
315
400
500
630
800
52
68
76
71
54
26
In the literature reference is sometimes made to the equivalent integration time of exponential de
tectors. This is two times the time constant (e.g. 250 ms for ‘F’), because a measurement of random noise with
an exponential detector with a time constant of τ has the same statistical uncertainty as a measurement with lin
ear averaging over a period of 2τ [9].
26
and less than 50 dB in all the other bands. It follows that
( )
5.2 6.8 7.6 7.1 5.4
Z
10log 10 10 10 10 10 77.7 dB, L + + + +
and
( )
(5.2 0.66) (6.8 0.48) (7.6 0.32) (7.1 0.19) (5.4 0.08)
A
10log 10 10 10 10 10 74.7 dB. L
÷ ÷ ÷ ÷ ÷
+ + + +
Noise that changes its level in a regular manner is called intermittent noise. Such
noise could for example be generated by machinery that operates in cycles. If the noise oc
curs at several steady levels, the equivalent sound pressure level can be calculated from the
formula
0.1
eq,
10log 10 .
i
L
i
T
i
t
L
T
 
=

\ .
¿
(1.3.17)
This corresponds to adding the mean square values with a weighting that reflects the relative
duration of each level.
Example 1.3.11
The Aweighted sound pressure level at a given position in an industrial hall changes periodically be
tween 84 dB in intervals of 15 minutes, 95 dB in intervals of 5 minutes and 71 dB in intervals of 20 minutes.
From eq. (1.3.17) it follows that the equivalent sound pressure level over a working day is
8.4 9.5 7.1
Aeq
15 5 20
10log 10 10 10 87.0 dB.
40 40 40
L
 
= + +

\ .
Most sound level meters have also a peak detector for determining the highest abso
lute value of the instantaneous sound pressure (without filters and without time weighting),
p
peak
. The peak level is calculated from this value and eq. (1.3.8) in the usual manner, that is,
peak
ref
20log .
p
p
L
p
= (1.3.18)
Example 1.3.12
The crest factor of a signal is the ratio of its peak value to the rms value (sometimes expressed in dB).
From example 3.1 it follows that the crest factor of a pure tone signal is 2 or 3 dB.
The sound exposure level (sometimes abbreviated SEL) is closely related to L
Aeq
, but
instead of dividing the time integral of the squared Aweighted instantaneous sound pressure
by the actual integration time one divides by t
0
= 1 s. Thus the sound exposure level is a
measure of the total energy
27
of the noise, normalised to 1 s:
27
In signal analysis it is customary to use the term ‘energy’ in the sense of the integral of the square of
a signal, without regard to its units. This should not be confused with the potential energy density of the sound
field introduced in chapter 1.5.
27
2
1
2 2
AE A ref
0
1
10log ( )
t
t
L p t dt p
t
 
 
=
 

\ .
\ .
}
(1.3.19)
This quantity is used for measuring the total energy of a ‘noise event’ (say, a hammer blow or
the take off of an aircraft), independently of its duration. Evidently the measurement interval
should encompass the entire event.
Example 1.3.13
It is clear from eqs. (1.3.14) and (1.3.19) that L
Aeq,T
of a noise event of finite duration decreases with
the logarithm of T if the T exceeds its duration:
2 2
Aeq, A ref AE
0
1
10log ( )d 10log .
T
T
L p t t p L
T t
·
÷·
   
= = ÷
 
\ .
\ .
}
Example 1.3.14
If n identical noise events each with a sound exposure level of L
AE
occur within a period of T (e.g., one
working day) then the Aweighted equivalent sound level is
Aeq, AE
0
10log 10log ,
T
T
L L n
t
= + ÷
because the integrals of the squared signals are additive; cf. eq. (1.3.7).
28
1.4 THE CONCEPT OF IMPEDANCE
By definition an impedance is the ratio of the complex amplitudes of two signals rep
resenting cause and effect, for example the ratio of an AC voltage across a part of an electric
circuit to the corresponding current, the ratio of a mechanical force to the resulting vibra
tional velocity, or the ratio of the sound pressure to the particle velocity. The term has been
coined from the verb ‘impede’ (obstruct, hinder), indicating that it is a measure of the opposi
tion to the flow of current etc. The reciprocal of the impedance is the admittance, coined from
the verb ‘admit’ and indicating lack of such opposition. Note that these concepts require
complex representation of harmonic signals; it makes no sense to divide, say, the instantane
ous sound pressure with the instantaneous particle velocity. There is no simple way of de
scribing properties corresponding to a complex value of the impedance without the use of
complex notation.
The mechanical impedance is perhaps simpler to understand than the other impedance
concepts, since it is intuitively clear that it takes a certain vibratory force to generate me
chanical vibrations. The mechanical impedance of a structure at a given point is the ratio of
the complex amplitude of a harmonic point force acting on the structure to the complex am
plitude of the resulting vibratory velocity at the same point,
29
28
Strictly speaking this requires that the instantaneous product of the ‘event’ and any of its time shif
ted versions time average to zero. In practice this will always be the case.
29
Note that the sign of the imaginary part of the impedance changes if the e
iωt
convention is used in
stead of the e
jωt
convention. Cf. footnote no 9 on p. 6.
28
.
ˆ
ˆ
m
v
F
Z = (1.4.1a)
The unit is kg/s. The mechanical admittance is the reciprocal of the mechanical impedance,
.
ˆ
ˆ
m
F
v
Y = (1.4.1b)
This quantity is also known as the mobility. The unit is s/kg.
Example 1.4.1
It takes a force of F = a·M to set a mass M into the acceleration a (Newton’s second law of motion);
therefore the mechanical impedance of the mass is
m
ˆ ˆ
j .
ˆ ˆ j
F F
Z M
v a
e
e
= = =
Example 1.4.2
It takes a force of F = ξK to stretch a spring with the stiffness K a length of ξ (Hooke’s law); therefore
the mechanical impedance of the spring is
m
ˆ ˆ
.
ˆ
ˆ j
j
F F K
Z
v e
eç
= = =
Figure 1.4.1 A mass hanging from a spring.
Example 1.4.3
A simple mechanical oscillator consists of a mass M suspended from a spring with a stiffness constant
of K, as sketched in figure 1.4.1. In order to set the mass into vibrations one will have to move the mass and
displace the spring from its equilibrium value. It follows that the mechanical impedance of this system is the
sum of the impedance of the mass and the impedance of the spring,
( )
( )
2
m 0
j j j 1 ,
j
K K
Z M M M e e e e e
e e
 
= + = ÷ = ÷

\ .
where
0
K M e =
is the angular resonance frequency. Note that the impedance is zero at the resonance, indicating that even a very
small harmonic force at this frequency will generate an infinite velocity. In practice there will always be some
losses, of course, so the impedance is very small but not zero at the resonance frequency.
The acoustic impedance is associated with average properties on a surface. This quan
tity is mainly used under conditions where the sound pressure is more or less constant on the
surface. It is defined as the complex ratio of the average sound pressure to the volume veloc
ity, which is the surface integral of the normal component of the particle velocity,
29
ˆ ˆ d ,
S
q = ·
}
u S (1.4.2)
where S is the surface area. Thus the acoustic impedance is
a av
ˆ ˆ Z p q = . (1.4.3)
The unit is kgm
4
s
1
. Since the total force acting on the surface equals the product of the aver
age sound pressure and the area, and since
n
u S q ˆ ˆ = if the velocity is uniform, it can be seen
that there is a simple relation between the two impedance concepts under such conditions:
.
2
a m
S Z Z = (1.4.4)
This equation makes it possible to calculate the force it would take to drive a massless piston
with the velocity
n
uˆ . In other words, the acoustic impedance describes the load on a (real or
fictive) piston caused by the medium. If the piston is real, the impedance is called the radia
tion impedance. This quantity is used for describing the load on, for example, a loudspeaker
membrane caused by the motion of the medium.
30
The concept of acoustic impedance is essentially associated with approximate low
frequency models. For example, it is a very good approximation to assume that the sound
field in a tube is onedimensional when the wavelength is long compared with the cross
sectional dimensions of the tube. Under such conditions the sound field can be described by
eqs. (1.2.15) and (1.2.16), and a tube of a given length behaves as an acoustic twoport.
31
It is
possible to calculate the transmission of sound through complicated systems of pipes using
fairly simple considerations based the assumption of continuity of the sound pressure and the
volume velocity at each junction [12].
32
The acoustic impedance is also useful in studying the
properties of acoustic transducers. Such transducers are usually much smaller than the wave
length in a significant part of the frequency range. This makes it possible to employ socalled
lumped parameter models where the system is described by an analogous electrical circuit
composed of simple lumped element, inductors, resistors and capacitors, representing masses,
losses and springs [13, 14]. Finally it should be mentioned that the acoustic impedance can be
used for describing the acoustic properties of materials exposed to normal sound incidence.
33
30
The load of the medium on a vibrating piston can be described either in terms of the acoustic radia
tion impedance (the ratio of the sound pressure to the volume velocity) or the mechanical radiation impedance
(the ratio of the force to the velocity).
31
‘Twoport’ is a term from electric circuit theory denoting a network with two terminals. Such a net
work is completely described by the relations between four quantities, the voltage and current at the input ter
minal and the voltage and current at the output terminal. By analogy, an acoustic twoport is completely de
scribed by the relations between the sound pressures and the volume velocities at the two terminals. In case of a
cylindrical tube such relations can easily be derived from eqs. (1.2.15) and (1.2.16) [12].
32
Such systems act as acoustic filters. Silencers (or mufflers) are composed of coupled tubes.
33
In the general case we need to describe the properties of acoustic materials with the local ratio of the
sound pressure on the surface to the resulting vibrational velocity. In most literature this quantity, which is used
mainly in theoretical work, is called the specific acoustic impedance. In many practical applications the proper
ties of acoustic materials are described in terms of absorption coefficients (or absorption factors), assuming ei
ther normal or diffuse sound incidence (see chapter 1.5). It is possible to calculate the absorption coefficient of a
material from its specific acoustic impedance, but not the impedance from the absorption coefficient.
30
Example 1.4.4
The acoustic input impedance of a tube terminated by a rigid cap can be deduced from eqs. (1.2.17) and
(1.2.18) (with x = l),
a
j cot ,
c
Z kl
S
µ
= ÷
where l is the length of the tube and S is its crosssectional area. Note that the impedance goes to infinity when l
equals a multiple of half a wavelength, indicating that it would take an infinitely large force to drive a piston at
the inlet of the tube at these frequencies (see figure 1.4.2). Conversely, the impedance is zero when l equals an
oddnumbered multiple of a quarter of a wavelength; at these frequencies the sound pressure on a vibrating pis
ton at the inlet of the tube would vanish. Cf. example 1.2.2.
Figure 1.4.2 The acoustic input impedance of a tube terminated rigidly.
At low frequencies the acoustic impedance of the rigidly terminated tube analysed in
example 1.4.4 can be simplified. The factor cotkl approaches 1/kl, and the acoustic imped
ance becomes
2
a
j ,
j
c c
Z
Slk V
µ µ
e
÷ = (1.4.5)
where V = Sl is the volume of the tube, indicating that the air in the tube acts as a spring.
Thus the acoustic impedance of a cavity much smaller than the wavelength is springlike,
with a stiffness that is inversely proportional to the volume and independent of the shape of
the cavity. Since, from eq. (1.2.2b),
2
0
, c p µ ¸ = (1.4.6)
it can be seen that the acoustic impedance of a cavity at low frequencies also can be written
0
a
,
j
p
Z
V
¸
e
= (1.4.7)
in agreement with the considerations on p. 3.
Example 1.4.5
A Helmholtz resonator is the acoustic analogue to the simple mechanical oscillator described in exam
ple 1.4.3; see figure 1.4.3. The dimensions of the cavity are much smaller that the wavelength; therefore it be
haves as a spring with the acoustic impedance
31
2
a
,
j
c
Z
V
µ
e
=
where V is the volume; cf. eq. (1.4.5). The air in the neck moves back and forth uniformly as if it were incom
pressible; therefore the air in the neck behaves as a lumped mass with the mechanical impedance
m eff
j , Z Sl eµ =
where l
eff
is the effective length and S is the crosssectional area of the neck. (The effective length of the neck is
somewhat longer than the physical length, because some of the air just outside the neck is moving along with
the air in the neck.) The corresponding acoustic impedance follows from eq. (1.4.4):
eff
a
j l
Z
S
eµ
= .
By analogy with example 1.4.3 we conclude that the angular resonance frequency is
0
eff
.
S
c
Vl
e =
Note that the resonance frequency is independent of the density of the medium.
It is intuitively clear that a larger volume or a longer neck would correspond to a lower frequency, but
it is perhaps less obvious that a smaller neck area gives a lower frequency.
Figure 1.4.3 A Helmholtz resonator.
Yet another impedance concept, the characteristic impedance, has already been intro
duced. As we have seen in section 1.2.1, the complex ratio of the sound pressure to the parti
cle velocity in a plane propagating wave equals the characteristic impedance of the medium
(cf. eq. (1.2.14)), and it approximates this value in a free field far from the source (cf. eq.
(1.2.27)). Thus, the characteristic impedance describes a property of the medium, as we have
seen on p. 12. The unit is kgm
2
s
1
.
1.5 SOUND ENERGY, SOUND INTENSITY, SOUND POWER AND SOUND AB
SORPTION
The most important quantity for describing a sound field is the sound pressure. How
ever, sources of sound emit sound power, and sound fields are also energy fields in which
potential and kinetic energies are generated, transmitted and dissipated. Some typical sound
power levels are given in table 1.5.1.
It is apparent that the radiated sound power is a negligible part of the energy conver
sion of almost any source. However, energy considerations are nevertheless of great practical
importance in acoustics. The usefulness is due to the fact that a statistical approach where the
energy of the sound field is considered turns out to give very useful approximations in room
acoustics and in noise control. In fact determining the sound power of sources is a central
32
point in noise control engineering. The value and relevance of knowing the sound power ra
diated by a source is due to the fact that this quantity is largely independent of the surround
ings of the source in the audible frequency range.
Table 1.5.1 Typical sound power levels.
Aircraft turbojet engine 10 kW 160 dB
Gas turbine (1 MW) 32 W 135 dB
Small airplane 5 W 127 dB
Tractor (150 hp) 100 mW 110 dB
Large electric motor (0.5 MW) 10 mW 100 dB
Vacuum cleaner 100 μW 80 dB
Office machine 32 μW 75 dB
Speech 10 μW 70 dB
Whisper 10 nW 40 dB
1.5.1 The energy in a sound field
It can be shown that the instantaneous potential energy density at a given position in a
sound field (the potential sound energy per unit volume) is given by the expression
( )
( )
.
2
2
2
pot
c
t p
t w
µ
= (1.5.1)
This quantity describes the local energy stored per unit volume of the medium because of the
compression or rarefaction; the phenomenon is analogous to the potential energy stored in a
compressed or elongated spring, and the derivation is similar.
The instantaneous kinetic energy density at a given position in a sound field (the ki
netic energy per unit volume) is
( ) ( )
2
kin
1
2
w t u t µ = . (1.5.2)
This quantity describes the energy per unit volume at the given position represented by the
mass of the particles of the medium moving with the velocity u. This corresponds to the ki
netic energy of a moving mass, and the derivation is similar.
The instantaneous sound intensity at a given position is the product of the instantane
ous sound pressure and the instantaneous particle velocity,
( ) ( ) ( ) t p t t = I u . (1.5.3)
This quantity, which is a vector, expresses the magnitude and direction of the instantaneous
flow of sound energy per unit area at the given position, or the work done by the sound wave
per unit area of an imaginary surface perpendicular to the vector.
33
Energy conservation
By combining the fundamental equations that govern a sound field (the conservation of mass, the rela
tion between the sound pressure and density changes, and Euler’s equation of motion), one can derive the equa
tion
( )
( ) ,
w t
t
t
c
V· = ÷
c
I
where ( ) t V· I is the divergence of the instantaneous sound intensity and w(t) is the sum of the potential and
kinetic energy densities. This is the equation of conservation of sound energy, which expresses the simple fact
that the rate of change of the total sound energy at a given point in a sound field is equal to the flow of converg
ing sound energy; if the sound energy density at the point increases there must be a net flow of energy towards
the point, and if it decreases there must be net flow of energy diverging away from the point.
The global version of this equation is obtained using Gauss’s theorem,
34
( )
( )
( )d ( ) d (t)d ,
V S V
E t
t V t w V
t t
c c
V· = · = ÷ = ÷
c c
} } }
I I S
where S is the area of an arbitrary, closed surface, V is the volume inside the surface, and E(t) is the total instan
taneous sound energy within the surface. This equation shows that the rate of change of the total sound energy
within a closed surface is identical with the surface integral of the normal component of the instantaneous sound
intensity.
In practice the timeaveraged energy densities,
,
2
1
,
2
2
rms kin 2
2
rms
pot
u w
c
p
w µ
µ
= = (1.5.4a, 1.5.4b)
are more important than the instantaneous quantities, and the timeaveraged sound intensity
(which is usually referred to just as the ‘sound intensity’),
( ) ( ) ( ), t p t t = = I I u (1.5.5)
is more important than the instantaneous intensity I(t). Energy conservation considerations
lead to the conclusion that the integral of the normal component of the sound intensity over a
closed surface is zero,
d 0
S
· =
}
I S (1.5.6)
in any sound field unless there is generation or dissipation of sound power within the surface
S. If, on the other hand, the surface encloses a source the integral equals the radiated sound
power of the source, irrespective of the presence of other sources of noise outside the surface:
a
d
S
P · =
}
I S (1.5.7)
Often we will be concerned with harmonic signals and make use of complex notation,
as in chapters 1.2 and 1.4. Expressed in the complex notation eqs. (1.5.4) and (1.5.5) become
2
2
pot kin
2
ˆ
1
ˆ , ,
4 4
p
w w u
c
µ
µ
= = (1.5.8a, 1.5.8b)
34
According to Gauss’s theorem the volume integral of the divergence of a vector equals the cor
responding surface integral of the (outward pointing) normal component of the vector.
34
{ }
*
1
ˆ ˆ Re
2
p = I u . (1.5.9)
(Note that the two complex exponentials describing the time dependence of the sound pres
sure and the particle velocity cancel each other because one of them is conjugated; see the
Appendix.) The component of the sound intensity in the xdirection is
{ }
*
1
ˆ ˆ Re
2
x x
I pu = . (1.5.10)
Inserting the expressions for the sound pressure (eq. (1.2.13)) and the particle velocity (eqs.
(1.2.14)) in a plane propagating wave into eq. (1.5.10) shows that
c
p
c
p
I
x
µ µ
2
rms
2
2
ˆ
= = (1.5.11)
in this particular sound field. Moreover, inserting expressions for the sound pressure and the
particle velocity in a simple spherical wave, eqs. (1.2.26) and (1.2.27), into eq. (1.5.10) gives
the same relation for the radial sound intensity:
{ }
( ) ( )
2 2
j j 2 *
* rms
2
ˆ
1 e e 1
ˆ ˆ Re Re 1 .
2 j 2 2
t kr t kr
r r
A p p A A
I pu
r cr kr cr c c
e e
µ µ µ µ
÷ ÷ ÷
¦ ¹
  ¦ ¦
= = ÷ = = =
´ `

¦ ¦ \ .
¹ )
(1.5.12)
It is apparent that there is a simple relation between the sound intensity and the mean square
sound pressure in these two extremely important cases.
35
However, it should be emphasised
that in the general case eq. (1.5.11) is not valid, and one will have to measure both the sound
pressure and the particle velocity simultaneously and average the instantaneous product over
time in order to measure the sound intensity. Equipment for such measurements has been
commercially available since the early 1980s [3].
Example 1.5.1
It follows from eq. (1.5.11) that the sound intensity in a plane propagating wave with an rms sound
pressure of 1 Pa is
2 3 1 2
(1 Pa) (1.2 kgm 343ms ) 2.4 mW m
÷ ÷
· .
Example 1.5.2
The sound intensity in the interference field generated by a plane sound wave reflected from a rigid
surface at normal incidence can be determined by inserting eqs. (1.2.17) and (1.2.18) into eq. (1.5.10):
2
*
i
i
i
2j
j2 1
Re 2 cos sin Re sin 2 0.
2
x
p
p
I p kx kx kx
c c µ µ
¦ ¹
¦ ¹
¦ ¦
= = =
´ ` ´ `
¹ ) ¦ ¦
¹ )
This result shows that there is no net flow of sound energy towards the rigid surface.
Under conditions where the sound pressure and the particle velocity are constant over
a surface in phase as well as in amplitude we can write
35
Eq. (1.5.11) implies that the sound intensity level is almost identical with the sound pressure level in
air at 20°C and 101.3 kPa:
( ) ( ) ( ) ( )
2 2 2 2
ref rms rms ref ref ref ref
10log 10log ( ) 10log 10log 0.14 dB .
I p p
L I I p c I p p cI p L L µ µ = = = ÷ ÷
35
a
ˆ ˆ Z q p = (1.5.13)
(cf. eq. (1.4.4)), and the sound power passing through the surface can be expressed in terms
of the acoustic impedance:
{ } { }
{ }
2
2
*
a a a
ˆ
1 1
ˆ ˆ ˆ Re Re Re .
2 2 2
q
P pq q Z Z = = = (1.5.14)
This expression demonstrates that the radiated sound power of a vibrating surface is closely
related to the volume velocity and to the real part of the radiation impedance.
Equation (1.5.7) implies that one can determine the sound power radiated by a source
by integrating the normal component of the sound intensity over a surface that encloses the
source. This is the sound intensity method of measuring sound power. Note that special
equipment for such measurements is required.
In an environment without reflecting surfaces the sound field generated by a source of
finite extent is locally plane far from the source, as mentioned in section 1.2.2, and therefore
the local sound intensity is to a good approximation given by eq. (1.5.11). With eq. (1.5.7) we
now conclude that one can estimate the radiated sound power of a source by integrating the
mean square pressure generated by the source over a spherical surface centred at the source:
( )
2
a rms
( ) d .
S
P p c S µ =
}
(1.5.15)
However, whereas eq. (1.5.7) is valid even in the presence of sources outside the measure
ment surface eq. (1.5.11) is not; therefore only the source under test must be present. In prac
tice one measures the sound pressure at a finite number of discrete points. This is the free
field method of measuring sound power. Note that an anechoic room (a room without any re
flecting surfaces) is required.
Yet another method of measuring sound power requires a diffuse sound field in a re
verberation room; see chapter 3.
1.5.2 Sound absorption
Most materials absorb sound. As we have seen in chapter 1.2 we need a precise de
scription of the boundary conditions for solving the wave equation, which leads to a descrip
tion of material properties in terms of the specific acoustic impedance, as mentioned in chap
ter 1.4. However, in many practical applications, for example in architectural acoustics, a
simpler measure of the acoustic properties of materials, the absorption coefficient (or absorp
tion factor), is more useful. By definition the absorption coefficient of a given material is the
absorbed fraction of the incident sound power. From this definition it follows that the absorp
tion coefficient takes values between naught and unity. A value of unity implies that all the
incident sound power is absorbed.
Figure 1.5.1 A standing wave tube for measuring the normal incidence absorption coefficient. (From ref. [15].)
36
In general the absorption coefficient of a given material depends on the structure of
the sound field (plane wave incidence of a given angle of incidence, for example, or random
or diffuse incidence in a room). Here we will study only the absorption for plane waves of
normal incidence.
Consider the sound field in a tube driven by a loudspeaker at one end and terminated
by the material under test at the other end, as sketched in figure 1.5.1. This is a one
dimensional field, which means that it has the general form given by eqs. (1.2.15) and
(1.2.16). The amplitudes p
i
and p
r
depend on the boundary conditions, that is, the vibrational
velocity of the loudspeaker and the properties of the material at the end of the tube. The
sound intensity is obtained by inserting eqs. (1.2.15) and (1.2.16) into eq. (1.5.10),
( )
( )
( )( )
2 2 * j * j
i r
i r j j
i r
i r i r
max min
e e
Re e e
2 2
,
2 2
kx kx
kx kx
x
p p
p p
I p p
c c
p p p p
p p
c c
µ µ
µ µ
÷
÷
¦ ¹
÷
÷
¦ ¦
= + =
´ `
¦ ¦
¹ )
+ ÷
= =
(1.5.16)
where the last equation sign follows from eq. (1.2.20). (Note that p
max
and p
min
are ampli
tudes.) The incident sound intensity is the value associated with the incident wave, that is,
.
2
2
i
inc
c
p
I
µ
= (1.5.17)
The absorption coefficient is the ratio of I
x
to I
inc
,
( )
,
1
4
1
1
1 1
2
2
2
2
i
2
r
2
i
inc
s
s
s
s
R
p
p p
I
I
x
+
=

.

\

+
÷
÷ = ÷ =
÷
= = o (1.5.18)
where we have introduced the reflection factor and the standing wave ratio (cf. eqs. (1.2.19)
and (1.2.22)). Note that the absorption coefficient is independent of the phase angle of R,
which shows that there is more information in the complex reflection factor than in the ab
sorption coefficient. Equation (1.5.18) demonstrates that one can determine the normal inci
dence absorption coefficient of a material by exposing it to normal sound incidence in a tube
and measuring the standing wave ratio of the resulting interference field.
Figure 1.5.2 Standing wave pattern for various absorption coefficients: 0.9 (–––); 0.6 (– –); 0.3 (···).
Example 1.5.3
If the material under test is completely reflecting then R = 1, corresponding to an absorption coeffi
cient of zero. In this case the standing wave ratio is infinitely large. If the material is completely absorbing, R =
0, corresponding to an absorption coefficient of unity. In the latter case there is no reflected wave, so the sound
pressure amplitude is constant in the tube, corresponding to a standing wave ratio of one.
37
1.6 RADIATION OF SOUND
Sound can be generated by many different mechanisms. In this note we will study
only the simplest one, which is also the most important: that of a solid vibrating surface. As
we shall see, the most efficient mechanism for radiation of sound involves a net volume dis
placement.
1.6.1 Point sources
The simplest source to describe mathematically is a sphere that expands and contracts
harmonically with spherical symmetry. In free space such a source generates the simple
spherical sound field we studied in section 1.2.2. Say the source has a radius of a. From eq.
(1.2.27) we know that the particle velocity on the surface of the source is
.
j
1
1
e
) ( ˆ
) j(


.

\

+ =
÷
ka a c
A
a u
ka t
r
e
µ
(1.6.1)
The boundary condition on the surface implies that the vibrational velocity
t
U
e j
e must equal
the normal component of the particle velocity; therefore
( )
2 j j
j e j e
,
1 j 4π 1 j
ka ka
cka U Q
A
ka ka
µ µe
= =
+ +
(1.6.2)
where we have introduced the volume velocity of the pulsating sphere,
2
4π , Q a U = (1.6.3)
by multiplying with the surface area of the sphere. Inserting into eq. (1.2.26) gives an expres
sion for the sound pressure generated by the source,
( )
j( ( ))
j e
ˆ .
4π 1 j
t k r a
Q
p
r ka
e
µe
÷ ÷
=
+
(1.6.4)
We can now calculate the radiation impedance of the pulsating sphere. This is the ratio of the
sound pressure on the surface of the sphere to the volume velocity (cf. eq. (1.4.3)):
( )
2
a,r j
ˆ ( ) j j
,
e 4π 1 j 4π 4π
t
p a ck
Z
Q a ka a
e
µe µ eµ
= = +
+
(1.6.5)
where the approximation to the right is based on the assumption that ka << 1. Note that the
imaginary part of the radiation impedance is much larger than the real part at low frequencies,
indicating that most of the force it takes to expand and contract the sphere goes to moving the
mass of the air in a region near the sphere (cf. example 1.4.1). This air moves back and forth
almost as if it were incompressible.
In the limit of a vanishingly small sphere the source becomes a monopole, also known
as a point source or a simple source. With ka << 1, the expression for the sound pressure
generated by a point source with the volume velocity
j
e
t
Q
e
becomes
j( )
j e
ˆ .
4π
t kr
Q
p
r
e
µe
÷
= (1.6.6)
38
A vanishingly small sphere with a finite volume velocity
36
may seem to be a rather academic
source. However, the monopole is a central concept in theoretical acoustics. At low frequen
cies it is a good approximation to any source that produces a net displacement of volume, that
is, any source that is small compared with the wavelength and changes its volume as a func
tion of time, irrespective of its shape and the way it vibrates. An enclosed loudspeaker is to a
good approximation a monopole at low frequencies. A source that injects fluid, the outlet of
an engine exhaust system, for example, is also in effect a monopole.
The sound intensity generated by the monopole can be determined from eq. (1.5.10):
{ }
( )
2
*
*
2 2
1 1 j j 1
ˆ ˆ Re Re 1 .
2 2 4π 4π j 32π
r r
Q
Q Q
I pu
r r c kr r c
µe
µe µe
µ µ
¦ ¹
  ÷
= = ÷ =
´ ` 
\ .
¹ )
(1.6.7)
By multiplying with the surface of the area of a sphere with the radius r we get the sound
power radiated by the monopole,
( )
2
2
2
2
a 2 2
4 .
32π 8π
Q ck Q
P r
r c
µe µ
t
µ
= = (1.6.8)
We could also obtain this result from eqs. (1.5.14) and (1.6.5), of course. Note that the sound
power is proportional to the square of the frequency, indicating that a small pulsating sphere
is not a very efficient radiator of sound at low frequencies.
Reciprocity
The reciprocity principle states that if a monopole source at a given point generates a certain sound
pressure at a another point then the monopole would generate the same sound pressure if we interchange listener
and source position, irrespective of the presence of reflecting or absorbing surfaces. This is a strong statement
with many practical implications.
It is easy to take account of a large reflecting plane surface, say, at z = 0, if one makes
use of the concept of image sources. If the surface is rigid the boundary condition implies
that u
z
= 0 at z = 0, and simple symmetry considerations show that this is satisfied if we re
place the rigid plane with an image source; see figure 1.6.1. The resulting sound pressure is
simply the sum of the sound pressures generated by the source and the image source,
1 2
1 1 2
j( ) j( )
j( ) j ( ) 1
1 2 1 2
j e j e j
ˆ e 1 e .
4π 4π 4π
t kR t kR
t kR k R R
R Q Q Q
p
R R R R
e e
e
µe µe µe
÷ ÷
÷ ÷
 
= + = +

\ .
(1.6.9)
The parenthesis shows the effect of the reflecting plane, that is, it represents the sound pres
sure normalised by the free field value. The normalised equation can be used for studying
outdoor sound propagation over a hard surface, and it is common practice to present the
‘ground effect’, that is, the effect of reflections from the ground on outdoor sound propaga
tion, in this form.
At very low frequencies k(R
1
 R
2
) << 1, and the rigid surface can be seen to have the
effect of increasing the sound pressure by a factor of 1+ R
1
/R
2
. Destructive interference oc
curs when the second term in the parenthesis is real and negative, and the first interference
36
The volume velocity of the monopole is sometimes referred to as the source strength. However,
some authors use other definitions of the source strength. The term ‘volume velocity’ is unambiguous.
39
dip occurs when k(R
1
– R
2
) = π, corresponding to (R
1
– R
2
) being half a wavelength. Figure
1.6.2 shows the sound pressure relative to free field for sound propagation over a rigid plane
surface.
Figure 1.6.1 The sound pressure generated by a monopole above a rigid plane is the sum of two terms: direct
sound and the contribution from the image source.(From ref. [16].)
Figure 1.6.2 The sound pressure in onethird octave bands generated by a monopole above a rigid plane and
shown relative to free field for five different sourcereceiver distances.(From ref. [16].)
If the distance between the source and the observation point is much longer than the
distance between the source and the reflecting plane (see figure 1.6.3) we can make use of the
farfield approximation and let
1 2
r r r in the denominator of eq, (1.6.6). However, the two
contributions will arrive with a different phase no matter how far from the source we are. If
the observation point is sufficiently far we can approximate the two distances by
1
cos r r h u ÷ and
2
cos r r h u + in the complex exponentials. The resulting sound pressure
now becomes
( )
1 2
j( ) j( )
1 2
j( ( cos )) j( ( cos )) j( )
j e j e
ˆ
4π 4π
j j
e e cos( cos ) e .
4π 2π
t kr t kr
t k r h t k r h t kr
Q Q
p
r r
Q Q
kh
r r
e e
e u e u e
µe µe
µe µe
u
÷ ÷
÷ ÷ ÷ + ÷
= +
+ =
(1.6.10)
40
Inspection of eq. (1.6.10) leads to the conclusion that the sound pressure in the far field de
pends on kh and on θ unless kh << 1, in which case the sound pressure is simply doubled.
Figure 1.6.3 Far field sound pressure generated by a monopole near a rigid plane surface.
The sound power of the monopole is affected by the presence of the reflecting surface
unless it is far away, kh >> 1. We can calculate the sound power by integrating the sound in
tensity over a hemisphere, cf. eq. (1.5.7). (Since the normal component of the particle velocity
is zero at all points on the plane between the source and the image source, the normal compo
nent of the intensity is also zero, so this surface does not contribute to the integral.) Moreover,
the considerations that lead to eq. (1.5.15) are also valid for combinations of sources. It fol
lows that
2 2
2
π 2 2π π 2
2 2
a
0 0 0
2 2
2 2
2
0
ˆ
sin d d cos ( cos ) sin d
2 4π
sin(2 )
cos d 1 .
4π 8π 2
kh
p ck Q
P r kh
c
ck Q ck Q
kh
x x
kh kh
µ
u ¢ u u u u
µ
µ µ
= =
 
= = +

\ .
} } }
}
(1.6.11)
Figure 1.6.4 shows the factor in parentheses. It is apparent that the sound power is doubled if
the source is very close to the surface, and that the rigid surface has an insignificant influence
on the sound power output of the source when h exceeds a quarter of a wavelength, corre
sponding to kh = π/2.
Figure 1.6.4 The influence of a rigid surface on the sound power of a monopole.
Example 1.6.1
It can be deduced from eq. (1.6.11) that two identical monopoles in close proximity (two enclosed loud
speakers driven with the same signal, for example) at very low frequencies will radiate twice as much sound
power as they do when they are far from each other. The physical explanation is that the radiation load on each
source is doubled; the sound pressure on each source is not only generated by the source itself but also by the
neighbouring source. Alternatively one might regard the two loudspeakers as one compound source with twice
the volume velocity of each loudspeaker. Because of the quadratic relation between volume velocity and power
(cf. eq. (1.6.8)) this source will radiate four times more sound power than one single loudspeaker in isolation.
41
Two monopoles of the same volume velocity but vibrating in antiphase constitute a
point dipole if the distance between them is much less than the wavelength; see figure 1.6.5. It
is clear that the combined source has no net volume velocity. A point dipole is a good ap
proximation to a small vibrating body that does not change its volume as a function of time.
Such a source exerts a force on the fluid. The oscillating sphere shown in figure 1.6.6, for ex
ample, is in effect a dipole, and so is an unenclosed loudspeaker unit. Other examples include
vibrating beams and wires.
Figure 1.6.5 A point dipole.
The sound pressure generated by the two monopoles is
1 2
j( ) j( )
1 2
j e j e
ˆ .
4π 4π
t kr t kr
Q Q
p
r r
e e
µe µe
÷ ÷
= ÷ (1.6.12)
The near field of this combination of sources is fairly complicated. However, the far field is
relatively simple. We can calculate the sound pressure in the far field in the same way we
used in deriving eq. (1.6.10),
( )
j( ( cos )) j( ( cos )) j( )
2
j( )
j
ˆ e e sin( cos ) e
4π 2π
cos e .
2π
t k r h t k r h t kr
t kr
Q Q
p kh
r r
chk Q
r
e u e u e
e
µe µe
u
µ
u
÷ + ÷ ÷ ÷
÷
÷ =
(1.6.13)
Note that the sound pressure is proportional to hQ, varies as cosθ and is identically zero in
the plane between the two monopoles.
37
The sound power of the dipole is calculated by integrating the mean square sound
pressure over a spherical surface centred midway between the two monopoles:
2 2
2 4
π 2π π
2 2
a
0 0 0
2 2
2 4 2 4
1
2
1
ˆ
sin d d cos sin d
2 4π
d .
4π 6π
p ch k Q
P r
c
ch k Q ch k Q
x x
µ
u ¢ u u u u
µ
µ µ
÷
= =
= =
} } }
}
(1.6.14)
Note that the sound power of the dipole is proportional to the fourth power of the frequency,
indicating very poor sound radiation at low frequencies. The physical explanation of the poor
radiation efficiency of the dipole is of course that the two monopoles almost cancel each
other.
37
The quantity 2hQ is referred to by some authors as the dipole strength. However, other authors use
other definitions.
42
Figure 1.6.6 Fluid particles in the sound field generated by an oscillating sphere. (From ref. [1].)
1.6.2 Sound radiation from a circular piston in an infinite baffle
Apart from the pulsating sphere, a vibrating circular piston in an infinite, rigid baffle is
one of the simplest cases of a spatially extended sound source that can be dealt with analyti
cally. It is often used in connection with loudspeaker modelling.
The basic approach to extended sound sources is to consider them as composed of
many simple sources, just as a dipole is made up of two monopoles. Thus, the piston is the
sum of many monopoles that all radiate in phase. Because of the infinite baffle each mono
pole gives rise to an image source which coincides with the monopole, cf. eqs. (1.6.9) and
(1.6.10)); in other words, the baffle has the effect of doubling the volume velocity of each
monopole. Let the piston vibrate with the velocity
j
e
t
U
e
. It follows that the volume velocity
of each elementary monopole is UdS. By linear superposition we conclude that the sound
pressure radiated by the piston can be evaluated at any position in front of the baffle simply
by integrating over the surface of the piston,
j( )
e
ˆ j d
2π
t kh
S
p U S
h
e
eµ
÷
=
}
, (1.6.15)
where h is the distance between the observation point and the running position on the piston,
and S is the surface of the piston of radius a (see figure 1.6.7).This is a special case of what is
known as Rayleigh’s integral, which can be used for computing the sound radiation into half
space of any plane infinite surface with a given vibrational velocity [17]. Note the factor of
two in the denominator instead of four for the monopole, which is due to the contribution of
the image sources.
Figure 1.6.7 Definition of the variables. (From ref. [18].)
43
The far field sound pressure, that is, the sound pressure at long distances from the cen
tre of the piston compared with the radius and the wavelength, can be calculated by expanding
h in the complex exponential,
2 2
2 sin cos 1 2 sin cos
sin cos ,
y
h r y ry r
r
r y
u ¢ u ¢
u ¢
= + ÷ ÷
÷
(1.6.16)
while retaining only the first term of eq. (1.6.16) in the denominator (cf. eq. (1.6.10)). Thus
the expression for the sound pressure becomes
2π
j( ) j sin cos
0 0
j
ˆ ( , ) e e d d
2π
a
t kr ky
U
p r y y
r
e u ¢
eµ
u ¢
÷
} }
. (1.6.17)
The calculation makes use of the Bessel functions J
0
(z) and J
1
(z), defined by
2π
j cos
0
0
1
J ( ) e d
2π
z
z

 =
}
(1.6.18)
and
1 0
0
1
J ( ) J ( )d
z
z
z
   =
}
(1.6.19)
(see figure 1.6.8), and leads to the following expression for the far field sound pressure,
j( ) j( ) 1 1
J ( sin ) 2J ( sin ) j j
ˆ ( , ) e e
sin 2π sin
t kr t kr
a ka ka U Q
p r
r k r ka
e e
u u eµ eµ
u
u u
÷ ÷
(
= =
(
¸ ¸
, (1.6.20)
where we have introduced the volume velocity of the piston, Q = π a
2
U. The factor in brack
ets is called the directivity of the piston, which is a frequency dependent function that de
scribes the directional characteristics of the source in the far field,
1
2J ( sin )
( , )
sin
ka
D f
ka
u
u
u
(
=
(
¸ ¸
. (1.6.21)
This function has its maximum value, unity, when θ = 0, indicating maximum radiation in the
axial direction all frequencies. Figure 1.6.9 shows the directivity for different values of the
normalised frequency ka. Note that the piston is an omnidirectional source (a monopole
placed on a rigid surface) at low frequencies, just as one would expect. At high frequencies
the radiation of the piston is concentrated in a beam near the axial direction.
Figure 1.6.8 Bessel functions.
44
E
Figure 1.6.9 Directivity of the piston as a function of the normalised frequency ka. (From ref. [18].)
The sound pressure on the axis of the piston can be evaluated fairly easily. Since sin θ
= 0 on the axis, the expression for the distance h reduces to
2 2
, h r y = + (1.6.22)
from which,
2 2
d
d d
y y y
h y
h
r y
= =
+
. (1.6.23)
Thus the sound pressure on the axis is given by
2 2
2 2 2π
j j j j j
0
j
ˆ e e d d e e e .
2π
r a
t  kh t  kr  k r a
r
U
p h cU
e e
eµ
¢ µ
+
+
(
= = ÷
(
¸ ¸
} }
(1.6.24)
If we introduce the quantity
( ) , 2
2 2
r a r ÷ + = A (1.6.25)
the sound pressure can be written
  ( )
). sin( e j 2 ˆ
j
A =
A + ÷
k cU p
r k t e
µ (1.6.26)
It can be seen that the sound pressure is zero when k) is a multiple of π, that is, when ) is a
multiple of half a wavelength, corresponding to the positions
1
2 2
a n
r a
n a
ì
ì
(
= ÷
(
¸ ¸
(1.6.27)
on the axis, where n is a positive integer. In a similar way, the sound pressure assumes a
maximum value for
2 ) 1 2 ( 2
2 2
ì + = ÷ + = A m r a r (1.6.28)
(where m is a positive integer), that is, for
45
1 2 1
.
2 1 4
a m
r a
m a
ì
ì
+ (
= ÷
(
+
¸ ¸
(1.6.29)
Figure 1.6.10 shows the normalised sound pressure on the axis of the piston as a function of
the distance, which for a given frequency is defined by the corresponding kafactor.
Figure 1.6.10 Sound pressure on the axis of a baffled piston for ka/2t = 5.5. (From ref. [19].)
It may seem surprising that the sound pressure is zero at some positions right in front of the
vibrating piston. The explanation is destructive interference, caused by the fact that the dis
tance from such a position to the various parts of the piston varies in such a manner that the
contributions cancel out.
Example 1.6.2
In the far field, when r >> a and r >> a
2
/λ, one obtains
2 2
2
1
1 ,
2 4 2
a a
r r
r r
(
 
A + ÷ =
( 
( \ .
¸ ¸
and the sound pressure reduces to
2
j( ) j( )
j
ˆ j e e .
2 2π
t kr t kr
ka ckQ
p cU
r r
e e
µ
µ
÷ ÷
 
= =

\ .
This expression agrees with eq. (1.6.20) for θ = 0 (D(f) = 1), as of course it should. This asymptotic expression is
plotted as a dashed line in figure 1.6.10.
Example 1.6.3
The distances at which the minima occur, normalised by the radius of the piston, are given in terms of
normalised frequencies by
π
.
4π
ka n
r
a
n ka
(
= ÷
(
¸ ¸
Minima of order n only occur for ka ≥ 2πn > 6. Thus for a loudspeaker with a radius of 50 mm, minima only
occur at frequencies higher than 6900 Hz, that is, far above the frequencies at which the piston approximation is
valid. It follows that the minima are never observed in front of loudspeakers in real life.
In the near field there is no possible approximation except on the axis. However, by
developing the spherical monopole field in cylindrical coordinates, the force exerted on the
piston can be calculated analytically. The calculations are rather complicated (see ref. [19] or
46
[20] for a complete treatment), and lead to an expression in terms of special functions such as
Bessel and Struve functions. The result is,
2 j 1 1
J (2 ) H (2 )
ˆ
ˆd π e 1 j ,
t
S
ka ka
F p S c a U
ka ka
e
µ
(
= = ÷ +
(
¸ ¸
}
(1.6.30)
where H
1
is the first Struve function.
The radiation impedance is the impedance seen by the piston, that is, the ratio of the
average sound pressure to the volume velocity,
.
e
ˆ
e
ˆ
j j r a, t t
SQ
F
Q
p
Z
e e
=
> <
= (1.6.31)
Combining eqs.(1.6.30) and (1.6.31) gives
1 1
a,r 2
J (2 ) H (2 )
1 j .
π
ka ka c
Z
a ka ka
µ (
= ÷ +
(
¸ ¸
(1.6.32)
Figure 1.6.11 shows the normalised, dimensionless radiation impedance (the bracket in eq.
(1.6.32)),
2
a,r
1 1
π
j .
Z a
R X
c µ
= + (1.6.33)
At low frequencies and at high frequencies the radiation impedance takes simple expressions:
2
a,r a,r a,r 2
1 8
1 j j ,
2π 3 π
ka Z r m ck
a
e µ eµ << = + = + (1.6.34a)
a,r 2 2 2 3 2
2 4 π
1 j 1 j .
π π π 2
c c
ka Z
a k a a ka
µ µ
eµ
 
>> = + = +

\ .
(1.6.34b)
Figure 1.6.11 Radiation impedance of a piston as a function of the normalised frequency. (From ref. [19].)
47
The first expression is fundamental for designing loudspeakers. Note that the real part of the
radiation impedance equals that of a small pulsating sphere, eq. (1.6.5), multiplied by a factor
of two because of the rigid plane. The quantity m
a,r
can be interpreted as the acoustic mass of
the air driven along by the piston. Interference effects in the near field make it different from
the imaginary part of impedance of the pulsating sphere. However, as in the case of the pul
sating sphere, eq. (1.6.5), the imaginary part of the acoustic radiation impedance diverges
when the radius a goes to zero.
Example 1.6.4
The mechanical radiation impedance is given by eqs. (1.4.4) and (1.6.33) as Z
m,r
= ρcπa
2
(R
1
+jX
1
). Its
low frequency approximation is therefore:
4 2 3
m,r
π 8
j .
2 3
a ck a
Z
µ
eµ = +
The imaginary part of this impedance is the impedance of the mass of a layer of air in front of the piston. This
layer of air is moving back and forth as if it were incompressible.
The radiated sound power is defined in chapter 1.5 as the integral of the normal com
ponent of the sound intensity over a surface than encloses the source. This method can also be
used for computing the sound power of a piston in an infinite baffle. However, by far the sim
plest approach is to use eq. (1.5.14), which expresses the sound power in terms of the mean
square volume velocity and the real part of the acoustic radiation impedance:
{ }
2 2 2
1 1 1 1
a a,r 1 2 2 2 2 2
J (2 )
Re 1 .
π π
ka c c
P Q Z Q R Q
a a ka
µ µ (
= = = ÷
(
¸ ¸
(1.6.35)
At low frequencies this becomes, with eq. (1.6.34a),
2
2
a
,
4π
ck Q
P
µ
= (1.6.36)
which is just what we would expect since the piston acts as a monopole on a rigid plane in
this frequency range (cf. eq. (1.6.11)).
Example 1.6.5
Instead of using the volume velocity and the acoustic impedance we could equally well compute the
sound power from the mean square velocity and the real part of the mechanical radiation impedance, since, with
eq. (1.4.4),
{ } { }
2 2
1 1
a a,r m,r 2 2
Re Re . P Q Z U Z = =
Example 1.6.6
Equation (1.6.36) shows that the sound power of the piston is proportional to ωQ
2
at low frequencies,
that is, the sound power is independent of the frequency if the volume acceleration is independent of the fre
quency. This implies that the displacement of the piston should be inversely proportional to the square of the
frequency if we want the sound power to be independent of the frequency. In other words, it implies very large
displacements at low frequencies. Since mechanical systems such as loudspeakers only allow a limited excur
sion, the low frequency sound power output of a loudspeaker is always limited: the only way to increase the
sound power is to increase the size of the membrane. This explains why very large loudspeakers are found in
subwoofers.
48
The directivity factor of a source is defined as the sound intensity on the axis in the far
field normalised by the sound intensity of an omnidirectional source with the same sound
power. From eq. (1.6.20) the sound intensity on the axis is
2
2
2 2
1


.

\

=
r
Q
ck I
r
t
µ (1.6.37)
(see also example 1.6.2). Normalising with P
a
/4πr
2
(eq. (1.6.35)) gives the directivity factor
Q(f),
( ) ( )
.
) 2 ( J
1
) (
1
2
1
2
ka
ka
ka
R
ka
f Q
÷
= = (1.6.38)
The directivity factor of the piston is plotted in figure 1.6.12 as a function of the normalised
frequency ka. Note that the directivity factor approaches two at low frequencies rather than
one, reflecting the fact that all the sound power is radiated in only half a sphere.
In practice, one often uses the directivity index, defined by
). ( log 10 ) ( f Q f DI = (1.6.40)
Figure 1.6.12 Directivity factor of a piston in a baffle. (From ref. [18].)
ka
0
5
10
15
Q(f)
4 3 2 1 0
49
REFERENCES
1 L. Cremer and M. Hubert: Vorlesungen über Technische Akustik (3
rd
edition). Springer
Verlag, Berlin, 1985.
2 T.D. Rossing, F.R. Moore and P.A. Wheeler: The Science of Sound (3
nd
edition). Addi
son Wesley, San Francisco, CA, 2002.
3 M.J. Crocker and F. Jacobsen: Sound intensity. Chapter 156 in Encyclopedia of Acous
tics, ed. M.J. Crocker. John Wiley & Sons, New York, 1997.
4 F. Jacobsen: A note on instantaneous and timeaveraged active and reactive sound in
tensity. Journal of Sound and Vibration 147, 1991, 489496.
5 F. Jacobsen: An elementary introduction to applied signal analysis. Acoustic Technol
ogy, Department of Electrical Engineering, Technical University of Denmark, Note no
7001, 2010.
6 R.B. Randall: Frequency analysis (3
rd
edition). Brüel & Kjær, Nærum, 1987.
7 D.W. Martin and W.D. Ward: Subjective evaluation of musical scale temperament in
pianos. Journal of the Acoustical Society of America 33, 1961, 582585.
8 R.W. Krug: Sound level meters. Chapter 155 in Encyclopedia of Acoustics, ed. M.J.
Crocker, John Wiley & Sons, New York, 1997.
9 J. Pope: Analyzers. Chapter 107 in Handbook of Acoustics, ed. M.J. Crocker. John
Wiley & Sons, New York, 1998.
10 P.V. Brüel, J. Pope and H.K. Zaveri: Introduction to acoustical measurement and in
strumentation. Chapter 154 in Encyclopedia of Acoustics, ed. M.J. Crocker. John Wiley
& Sons, New York, 1997.
11 Anon.: Microphone Handbook. Brüel & Kjær, Nærum, 1996.
12 F. Jacobsen: Propagation of sound waves in ducts. Acoustic Technology, Department of
Electrical Engineering, Technical University of Denmark, Note no 31260, 2010.
13 K. Rasmussen: Analogier mellem mekaniske, akustiske og elektriske systemer (4
th
editi
on). Polyteknisk Forlag, Lyngby, 1994.
14 W. Marshall Leach, Jr.: Introduction to Electroacoustics and Audio Amplifier Design
(2
nd
edition). Kendall/Hunt Publishing Company, Dubuque, IA, 1999.
15 Z. Maekawa and P. Lord: Environmental and Architectural Acoustics. E & FN Spon,
London, 1994.
16 E.M. Salomons: Computational Atmospheric Acoustics. Kluwer Academic Publishers,
Dordrecht, 2001.
17 W.S. Rayleigh: On the passage of waves through apertures in plane screens, and allied
theorems. Philosophical Magazine 43, 1897, 259272.
18 K. Rasmussen: Lydfelter. Department of Acoustic Technology, Technical University of
Denmark, Note no 2107, 1996.
19 A.D. Pierce: Acoustics. An Introduction to Its Physical Principles and Applications. The
Acoustical Society of America, New York, 1989.
20 P.M. Morse and K.U. Ingard: Theoretical Acoustics. McGrawHill, New York, 1968.
21 L.E. Kinsler, A.R. Frey, A.B. Coppens and J.V. Sanders: Fundamentals of Acoustics
(4
th
edition). John Wiley & Sons, New York, 2000.
50
BIBLIOGRAPHY
Recommended reading includes the following. The book by Everest and Pohlmann
manages to deal with many acoustic phenomena practically without mathematics. The books
by Fahy, Beranek, and Kinsler et al. are also introductory. More advanced treatments can be
found in the Nelson’s chapter, in the books by Morse, Morse & Ingard, and Filippi et al., and
in Pierce’s chapters and book.
1. F. Alton Everest and K.C. Pohlmann: Master Handbook of Acoustics (5
th
edition).
McGrawHill, New York, 2009.
2. L.L. Beranek: Acoustics (2
nd
edition). The American Institute of Physics for the
Acoustical Society of America, Cambridge, MA, 1986.
3. F. Fahy: Foundations of Engineering Acoustics. Academic Press, San Diego, 2000.
4. L.E. Kinsler, A.R. Frey, A.B. Coppens and J.V. Sanders: Fundamentals of Acoustics
(4
th
edition). John Wiley & Sons, New York, 2000.
5. P.A. Nelson: An introduction to acoustics. Chapter 1 in Fundamentals of Noise and
Vibration, ed. F.J. Fahy and J. Walker. E & FN Spon, London, 1998.
6. P.M. Morse: Vibration and Sound (2
nd
edition). The American Institute of Physics,
New York, 1983.
7. P.M. Morse and K.U. Ingard: Theoretical Acoustics. McGrawHill, New York, 1968.
8. A.D. Pierce: Acoustics. An Introduction to Its Physical Principles and Applications.
The Acoustical Society of America, New York, 1989.
9. A.D. Pierce: Mathematical theory of wave propagation. Chapter 2 in Encyclopedia of
Acoustics, ed. M.J. Crocker, John Wiley & Sons, New York, 1997.
10. A.D. Pierce: Basic linear acoustics. Chapter 3 in Springer Handbook of Acoustics,
ed. T.D. Rossing, Springer, New York, 2007.
11. P. Filippi, D. Habault, J.P. Lefebre and A. Bergassoli: Acoustics: Basic Physics,
Theory and Methods. Academic Press, London, 1999.
51
1.9 APPENDIX: COMPLEX NOTATION
In a harmonic sound field the sound pressure at any point is a function of the type
cos(Tt + n). It is common practice to use complex notation in such cases. This is a symbolic
method that makes use of the fact that complex exponentials give a more condensed notation
that trigonometric functions because of the complicated multiplication theorems of sines and
cosines.
We recall that a complex number A can be written either in terms of its real and
imaginary part or in terms of its magnitude (also called absolute value or modulus) and phase
angle,
, e j
j
i r
A
A A A A
¢
= + = (1.9.1)
where
1 j ÷ = (1.9.2)
is the imaginary unit, and
{ } { } , sin Im , cos Re
i r A A
A A A A A A ¢ ¢ = = = = (1.9.3, 1.9.4)
2
i
2
r
A A A + = (1.9.5)
(see figure 1.9.1). The complex conjugate of A is
; e j
j
i r
*
A
A A A A
¢ ÷
= ÷ = (1.9.6)
therefore the magnitude can also be written
.
*
A A A · = (1.9.7)
Multiplication and division of two complex numbers are most conveniently carried out if they
are given in terms of magnitudes and phase angles,
. e , e
) j( ) j(
B A B A
B
A
B A B A AB
¢ ¢ ¢ ¢ ÷ +
= = (1.9.8, 1.9.9)
Figure 1.9.1. Complex representation of a harmonic signal.
52
Complex representation of harmonic signals makes use of the fact that
x x
x
sin j cos e
j
+ = (1.9.10)
(Euler’s equation) or, conversely,
( ) ( )
j j j j
1 1
cos e e , sin j e e
2 2
x x x x
x x
÷ ÷
= + = ÷ ÷ . (1.9.11a, 1.9.11b)
In a harmonic sound field the sound pressure at a given position can be written
, e ˆ
j t
A p
e
= (1.9.12)
where A is the complex amplitude of the sound pressure. The real, physical sound pressure is
of course a real function of the time,
{ } { } ), cos( e Re ˆ Re
) j(
A
t
t A A p p
A
¢ e
¢ e
+ = = =
+
(1.9.13)
which is seen to be an expression of the form cos(ωt + φ). The magnitude of the complex
quantity A is called the amplitude of the pressure, and φ
A
is its phase. It can be concluded
that complex notation implies the mathematical trick of adding another solution, an expres
sion of the form sin(ωt + φ), multiplied by a constant, the imaginary unit j. This trick relies on
linear superposition.
Figure 1.9.2. Two simple harmonic signals with identical frequencies. (From ref. [21].)
The mathematical convenience of the complex representation of harmonic signals can
be illustrated by an example. A sum of two harmonic signals of the same frequency, A
1
e
jωt
and A
2
e
jωt
, is yet another harmonic signal with an amplitude of A
1
+ A
2
 (see figure 1.9.2).
Evidently, this can also be derived without complex notation,
( ) ( )
( ) ( )
1 1 2 2
1 1 2 2 1 1 2 2
½
2 2
1 1 2 2 1 1 2 2
cos( ) cos( )
cos cos cos sin sin sin
cos cos sin sin cos( ),
p A t A t
A A t A A t
A A A A t
e ¢ e ¢
¢ ¢ e ¢ ¢ e
¢ ¢ ¢ ¢ e ¢
= + + +
= + ÷ +
(
= + + + +
(
¸ ¸
(1.9.14)
where
53
,
cos cos
sin sin
arctan
2 2 1 1
2 2 1 1
¢ ¢
¢ ¢
¢
A A
A A
+
+
= (1.9.15)
but the expedience and convenience of the complex method seems indisputable.
Since
, e j e
d
d
j j t t
t
e e
e = (1.9.16)
it follows that differentiation with respect to time corresponds to multiplication by a factor of
jω. Conversely, integration with respect to time corresponds to division with jω. If, for exam
ple, the vibrational velocity of a surface is, in complex representation,
j( ) j
ˆ e e ,
B
t t
v B B
e ¢ e +
= = (1.9.17)
which means that the real, physical velocity is
{ } ), cos( ˆ Re
B
t B v v ¢ e + = = (1.9.18)
then the acceleration is written
, ˆ j ˆ v a e = (1.9.19)
which means that the physical acceleration is
{ } { }
j
ˆ Re Re j e sin( )
t
B
a a B B t
e
e e e ¢ = = = ÷ + , (1.9.20)
and this is seen to agree with the fact that
). sin( ) cos(
d
d
B B
t t
t
¢ e e ¢ e + ÷ = + (1.9.21)
In a similar manner we find the displacement,
,
j
ˆ
ˆ
e
ç
v
= (1.9.22)
which means that
{ } ), sin(
1
e
j
1
Re
ˆ
Re
j
B
t
t B B ¢ e
e e
ç ç
e
+ =
)
`
¹
¹
´
¦
= = (1.9.23)
in agreement with the fact that
54
). cos( ) sin(
1
d
d
B B
t t
t
¢ e ¢ e
e
+ =

.

\

+ (1.9.24)
Acoustic secondorder quantities involve time averages of squared harmonic signals
and, more generally, products of harmonic signals. Such quantities are dealt with in a special
way, as follows. Expressed in terms of the complex pressure amplitude pˆ , the mean square
pressure becomes
, 2 ˆ
2
2
rms
2
p p p = = (1.9.25)
in agreement with the fact that the average value of a squared cosine is ½. Note that it is the
squared magnitude of pˆ that enters into the expression, not the square of pˆ , which in general
would be a complex number proportional to e
2jωt
.
The time average of a product is given by the following expression
{ } { }
1 1
ˆˆ ˆ ˆ Re Re
2 2
xy xy x y
 
= = . (1.9.26)
This can be seen as follows,
{ } { }
j( )
j( )
1 1 1
ˆˆ ˆ ˆ ˆ ˆ Re Re e e cos( ),
2 2 2
y
x
t
t
x y
xy x y x y
e 
e 
 
÷ +
+ 
= = ÷ (1.9.27)
which is seen to in agree with
). cos( ˆ ˆ
2
1
) cos( ˆ ) cos( ˆ
y x y x
y x t y t x xy ¢ ¢ ¢ e ¢ e ÷ = + + = (1.9.28)
Note that either xˆ or yˆ must be conjugated.½
Ear, Hearing and Speech
2 Ear, Hearing and Speech
Torben Poulsen
2.1 Introduction
The aim of the present chapter is to give the student a basic understanding of the function of
the ear, the perception of sound and the consequences for speech understanding. The content
covers the basic psychoacoustic aspects of a situation where two persons speak to each other.
The major topics are: the ear and its functional principles, basic psychoacoustics (hearing
threshold, loudness, masking) and speech intelligibility.
2.2 The Ear
Figure 2.2.1 Drawing of the ear. A is the outer ear. B is the middle ear. C is the inner ear. From [1]
The ear may be divided into four main parts: The outer ear, the middle ear, the inner ear and
the nerve connection to the brain. The first three parts (the peripheral parts) are shown in
Figure 2.2.1. Part A being the outer ear, B is the middle ear and C is the inner ear. The sound
will reach the outer ear, progress through the outer ear canal, reach the tympanic membrane
(the ear drum), transmit the movements to the bones in the middle ear, and further transmit
the movements to the fluid in the inner ear. The fluid movements will be transformed to
nerve impulses from the hair cells in the inner ear and the impulses are transmitted to the
brain through the auditory nerve.
55
Ear, hearing and speech
2.2.1 The outer ear
The outer ear consists of the pinna (or the auricle) and the ear canal. The Pinna plays an
important role for our localisation of sounds sources. The special shape of pinna produces
reflections and diffraction so that the signal that reaches the ear will be dependent on the
direction to the sound. The pinna has common features from person to person but there are
big individual differences in the details. Localisation of sound sources is difficult if a hearing
protector or a crash helmet covers the pinna. The outer part of the ear canal is relatively soft
whereas the inner part is stiff and bony. At the end of the ear canal the tympanic membrane is
situated. The length of the ear canal is approximately 25 mm and the diameter is
approximately 7 mm. The area is approximately 1 cm
2
. These numbers are approximate and
vary from person to person.
The ear canal may be looked upon as a tube that is closed in one end and open in the other.
This will give resonances for frequencies where the length of the ear canal corresponds to 1/4
of the wavelength of the sound. With a length of 25 mm and a speed of sound of 340 m/s the
resonance frequency will be
kHz
m
s m
f
res
4 , 3
025 , 0 * 4
/ 340
= =
This calculation is correct if the ear canal was a cylindrical tube. Most ear canals will have
one or two bends. This implies that it is usually not possible from the outside to see the
tympanic membrane at the end of the ear canal. It’s necessary to make the canal straighter,
which may be done by pulling pinna upward and backwards.
The tympanic membrane is found at the end of the canal. The membrane is not perpendicular
to the axis of the ear canal but tilted approx. 30 degrees. The tympanic membrane is shaped
like a cone with the top of the cone pointing inwards into the middle ear. The thickness is
approx. 0.1 mm.
2.2.2 The middle ear
The middle ear consists of three small bones: hammer, anvil and stirrup. The Latin names are
also often used: Malleus, Incus and Stapes. These bones are the smallest bones in the human
body. A drawing is shown in Figure 2.2.2. The function of the middle ear is to transmit the
vibrations of the tympanic membrane to the fluid in the inner ear. From Figure 2.2.2 it is seen
that the hammer (Maleus, M) is fixed to the tympanic membrane (1) from the edge and into
the centre of the membrane (the top of the cone). The anvil (Incus, I, 2) connects the hammer
and the stirrup (Stapes, S) and the footplate of the stirrup makes the connection into the inner
ear. This connection is sometimes called the oval window. The footplate rotates around the
point marked (3). The middle ear is filled with air and is connected to the nose cavity (and
thus the atmospheric pressure) through The Eustachian tube (ET, 4). The fluid in the inner
ear is incompressible and an inwards movement of the stirrup will be equalised by a
corresponding outward movement by the round window (5).
56
Ear, Hearing and Speech
Figure 2.2.2 Drawing of the middle ear. See text for details. From [2]
Usually the Eustachian tube is closed but opens up when you swallow or yawn. When the
tube is open, the pressure at the two sides of the tympanic membrane is equalised. If the
Eustachian tube becomes blocked (which is typically the case when you catch a cold) the
equalisation will not take place and after some time the oxygen in the middle ear will be
assimilated by the tissue and an underpressure will build up in the middle ear. This causes
the tympanic membrane to be pressed inwards and thus the sensitivity of the hearing is
reduced.
The chain of middle ear bones forms a lever function that  together with the area ratio
between the tympanic membrane and the footplate of stapes  makes an impedance match
between the air in the outer ear and the liquid in the inner ear. The lever ratio is approx. 1.3
and the area ratio is approx. 14. The total ratio is thus 18, which corresponds to approx.
25 dB.
Two small muscles, tensor tympani (6) and stapedius (7), see Figure 2.2.2, are attached to the
bones and will be activated by the socalled middle ear reflex. The reflex is elicited when the
ear is exposed to sounds above approx. 70 dB SPL whereby the transmission through the
middle ear is reduced. The reduction is about 20 dB at 125 Hz, 10 dB at 1000 Hz and less
than 5 dB at frequencies above 2000 Hz. The middle ear reflex can to some extent protect the
inner ear from excessive exposure. Because the reflex is activated by a signal from the brain
there will be a delay of about 25 to 150 ms before the effect is active. The reflex has
therefore no protective effect on impulsive sounds.
2.2.3 The inner ear
The inner ear consists of a snailshell shaped structure in the temporal bone called Cochlea.
The cochlea is filled with lymph and is closely connected to the balance organ that contains
the three semicircular canals. There are 2.75 turns in the snail shell and the total length from
the base to the top is 32 mm. A cross section of one of the turns is shown in Figure 2.2.3.
57
Ear, hearing and speech
This figure shows that the cochlea is divided into three channels (latin: Scala) called scala
vestibuli (1), scala media (2), and scala tympani (3).
Figure 2.2.3 Cross section of a cochlea turn. See text for details. From [1]
There are two connections (windows) from cochlea to the middle ear cavity. The oval
window is the footplate of the stirrup and is connected to Scala Vestibuli (1). The round
window is connected to Scala Tympani (3). The round window prevents an overpressure to
build up when the oval window moves inwards. Scala Vestibuli and Scala Tympani are
connected at the top of the cochlea with a hole called Helicotrema.
The Basilar membrane (6 in Figure 2.2.3) divides scala tympani from scala media. The width
of the basilar membrane (BM) changes from about 0.1 mm at the base of the cochlea to about
0.5 mm at the top of the cochlea (at helicotrema). The change of the BMwidth is thus the
opposite of the width of the snail shell. The function of the BM is very important for the
understanding of the function of the ear.
A structure  called the organ of Corti  is positioned on top of the Basilar Membrane in Scala
Media. The organ of Corti consists of one row of inner hair cells (7 in Figure 2.2.3) and three
rows of outer hair cells (8 in Figure 2.2.3). The designations ‘inner’ and ‘outer’ refer to the
centre axis of the snail shell which is to the left in Figure 2.2.3. The hair cells are special
nerve cells where small hairs protrude from the top of the cells. There are approx. 3000 inner
hair cells and about 12000 outer hair cells. A soft membrane (5 in Figure 2.2.3) covers the
top of the hair cells. The organ of Corti transforms the movements of the Basilar membrane
to nerve impulses that are then transmitted to the hearing centre in the brain.
The inner hair cells are the main sensory cells. Most of the nerve fibres are connected to the
inner hair cells. When sound is applied to the ear, the basilar membrane and the organ of
Corti will vibrate and the hairs on the top of the hair cells will bend back and forth. This will
trigger the (inner) hair cells to produce nerve impulses.
The outer hair cells contain muscle tissue and these cells will amplify the vibration of the
basilar membrane when the ear is exposed to weak sounds so that the vibrations are big
enough for the inner hair cells to react. The amplification function of the outer hair cells is
nonlinear which means that they have an important effect at low sound levels whereas they
58
Ear, Hearing and Speech
are of almost no importance at high sound levels. The amplifier function  sometimes called
the cochlear amplifier  is destroyed if the ear is exposed to loud sounds such as gunshots or
heavy industrial noise. This is called a noise induced hearing loss. The amplifier function
also deteriorates with age. This is called an age related hearing loss.
2.2.4 The frequency analyzer at the Basilar membrane
The basilar membrane acts like a frequency analyser. When the ear is exposed to a pure tone
the movement of the basilar membrane will show a certain pattern and the pattern is
connected to a certain position on the basilar membrane. If the frequency is changed, the
pattern will not change but the position of the pattern will move along the basilar membrane.
This is illustrated in Figure 2.2.4 for the frequencies 400 Hz, 1600 Hz and 6400 Hz. The
400 Hz component produce BMmovement close to the top of the cochlea. 6400 Hz produces
a similar pattern but close to the base of the cochlea. Note that a single frequency produces
movements of the basilar membrane over a broad area. This means that even for a single
frequency many hair cells are active at the same time. Note also that the deflection of the BM
is asymmetrical. The envelope of the deflection (shown dotted in Figure 2.2.4) has a steep
slope towards the low frequency side and a much less steep slope towards the high frequency
side. The same different slopes are also found in masking thresholds and it can be shown that
masking is closely related to the basilar membrane movements.
Figure 2.2.4 Movement of the basilar membrane (b) when the ear is exposed to a combination of 400
Hz, 1600 Hz and 6400 Hz (a). O.W.: Owal window (base of cochlea). Hel: Helicotrema (top of
cochlea). From [3]
The nonlinear behaviour of the outer hair cells and their influence on the BM movement is
illustrated in Figure 2.2.5. This figure shows the BMamplitude at a certain position of the
basilar membrane as a function of the stimulus frequency. (Note that this is different from
Figure 2.2.4 where the amplitude is shown as a function of basilar membrane position for
different frequencies). There are at least three nonlinear phenomena illustrated in the figure.
1) At low exposure levels (20 dB) the amplitude is very selective and a ‘high’ amplitude is
achieved only in a very narrow frequency range. For high exposure levels (80 dB) the
59
Ear, hearing and speech
‘high’ amplitude is achieved at a much wider frequency range. Thus, the filter
bandwidth of the auditory analyser changes with the level of the incoming sound.
2) The frequency where the maximum amplitude is found change with level. At high
levels it is almost one octave below the maxamplitude frequency at low levels.
3) The maximum amplitude grows nonlinearly with level. At low levels (20 dB) the
maximum BMamplitude is about 60 dB (with some arbitrary reference). At an input
level of 80 dB the maximum BM amplitude is about 85 dB. In other words the change
in the outside level from 20 dB to 80 dB, i.e., 60 dB, is reduced (compressed) to a
change in the maximum BMamplitude of only 25 dB.
These nonlinear phenomena are caused by the function of the outer hair cells. The increase
of amplitude at low levels is sometimes called ‘the cochlear amplifier’. In a typical cochlear
hearing loss, the outer hair cells are not functioning correctly or may be destroyed. In other
words: The cochlear amplifier does not work. This will be seen as an elevated hearing
threshold and this is called a hearing loss.
10
20 30 5 3 2
Frequency, kHz
0
20
40
60
80
B
a
s
i
l
a
r
m
e
m
b
r
a
n
e
m
o
v
e
m
e
n
t
,
d
B
80 dB SPL
60 dB
40 dB
20 dB SPL
Figure 2.2.5 Movement of the Basilar membrane at a fixed point for stimulus levels from 20 dB SPL
to 80 dB SPL. Redrawn from [4]
60
Ear, Hearing and Speech
2.3 Human hearing
The human hearing can handle a wide range of frequencies and sound pressure levels. The
weakest audible sound level is called the hearing threshold and the sound level of the loudest
sound is called the threshold of discomfort or the threshold of pain.
2.3.1 The hearing threshold
The hearing threshold is frequency dependent, see Figure 2.3.1. At 1000 Hz the threshold is
about 2 dB SPL whereas it is about 25 dB SPL at 100 Hz and about 15 dB at 10000 Hz.
10 100 1000 10000
20 50 200 500 2000 5000 20000
Frequency, Hz
0
20
40
60
80
100
S
o
u
n
d
p
r
e
s
s
u
r
e
l
e
v
e
l
,
d
B
Figure 2.3.1 The binaural hearing threshold in a free field. From [5]
The threshold curve in Figure 2.3.1 is measured under the following conditions:
• Free field (no reflections from walls, floor, ceiling)
• Frontally incoming sound (called frontal incidence)
• signals are single pure tones
• binaural listening (i.e. listening with both ears)
• no background noise
• test subjects between 18 and 25 years of age
• the threshold is determined by means of either the ascending or the bracketing method
The curve is the median value (not the mean) over the subject’s data. The sound pressure
level, which is shown in the figure, is the level in the room at the position of the test subject’s
head but measured without the presence of the test subject. This curve is also called the
absolute threshold (in a free field) and data for the curve may be found in ISO 3897 [6] and
in ISO 226 [5].
61
Ear, hearing and speech
In ISO 3897 also threshold data for narrow band noise in a diffuse sound field are found.
The threshold curve is similar to the curve in Figure 2.3.1 and deviates from the pure tone
curve only by a few dB (2 to +6) in the frequency range 500 Hz to 16 kHz.
2.3.2 Audiogram
For practical use it is not convenient to measure the hearing threshold in a free or a diffuse
sound field in the way described in the previous section. For practical and clinical purposes,
usually only the deviation from normal hearing is of interest. Such deviations are determined
by means of a calibrated audiometer and the result of the measurement is called an
audiogram.
125 250 500 1k 2k 4k 8k
Frequency, Hz
80
70
60
50
40
30
20
10
0
10
H
e
a
r
i
n
g
t
h
r
e
s
h
o
l
d
l
e
v
e
l
,
d
B
H
L
80
70
60
50
40
30
20
10
0
10
Figure 2.3.2 Audiogram for a typical age related hearing loss.
Figure 2.3.2 shows an audiogram for a person in the frequency range 125 Hz til 8000 Hz.
The zero line indicates the average threshold for young persons and a normal audiogram will
give data points within 10 to 15 dB from the zero line. An elevated hearing threshold (i.e. a
hearing loss) is indicated downwards in an audiogram and the values are given in dB HL.
The term ‘HL’ (hearing level) is used to emphasise that it is the deviation from the average
normal hearing threshold.
The measurements are performed with headphones for each ear separately. The results from
the left ear are shown with '×' and the results from the right ear are shown with '○'.
Sound pressure level, dB SPL, and hearing level, dB HL, is not the same. An example: From
Figure 2.3.1 it can be seen that the hearing threshold at 125 Hz is 22 dB SPL (measured in
the way described previously). If a person has a hearing loss of 5 dB HL at this frequency the
62
Ear, Hearing and Speech
63
threshold would be 27 dB SPL. In an audiogram the 5 dB hearing loss will be shown as a
point 5 dB below the zero line (e.g. right ear, Figure 2.3.2). Another example: At 4000 Hz the
free field threshold is 6 dB (see Figure 2.3.1). A hearing loss of 50 dB HL (e.g. left ear,
Figure 2.3.2) will give a threshold of 44 dB SPL.
In order for the audiometry to give correct results, the audiometer must be calibrated
according to the ISO 389 series of standards. These standards specify the SPL values that
shall be measured in a specific coupler (an artificial ear) when the audiometer is set to 0 dB
HL. The values in the standards are headphone specific, which means that the audiometer
ust be recalibrated if the headphone is exchanged with another headphone. m
Table 2.3.1 shows reference values for two headphones commonly used in audiometry.
F, Hz
125
250
500
1k
2k
3k
4k
6k
8k
10k
12,5k
14k
16k
TDH 39
HDA 200
45,0
30,5
25,5
18,0
11,5
11,0
7,0
5,5
9,0
4,5
10,0
2,5
9,5
9,5
15,5
17,0
13,0
17,5

22,0

28,0

36,0

56,0
Table 2.3.1. Calibration values in dB SPL for a Telephonics TDH 39 earphone and a Sennheiser HDA
200 earphone. The TDH 39 earphone can not be used above 8 kHz. The TDH 39 data are from ISO
3891 [7]. The HDA 200 data are from ISO 3895 [8] and ISO 3898 [9].
2.3.3 Loudness Level
The definition of loudness levels is as follows: For a given sound, A, the loudness level is
defined as the sound pressure level (SPL) of a 1000Hz tone which is perceived equally loud
as sound A. The unit for loudness level is Phon (or Phone). In order to measure loudness
level a 1 kHz tone is needed and this tone should then be adjusted up and down in level until
it is perceived just as loud as the other sound. When this situation is achieved, the sound
pressure level of the 1 kHz tone is per definition equal to the loudness level in phone. For a
1000Hz tone the value in dB SPL and in Phone will be the same.
The loudness level for pure tones has been measured for a great number of persons with
normal hearing under the same conditions as for the absolute threshold (Figure 2.3.1). The
result is shown in Figure 2.3.3.
Ear, hearing and speech
10 100 1000 10000
20 50 200 500 2000 5000 20000
Frequency, Hz
0
20
40
60
80
100
120
140
S
o
u
n
d
p
r
e
s
s
u
r
e
l
e
v
e
l
,
d
B
Figure 2.3.3 Equal loudness level contours. Redrawn from [5]
Some examples, see Figure 2.3.3: A 4000Hz tone at 26 dB SPL will be perceived with the
same loudness as a 1000Hz tone at 30 dB SPL and thus the loudness level of the 4000 Hz
tone is 30 Phone. A 125Hz tone at 90 dB SPL will have a loudness level of 80 Phone.
The curves in Figure 2.3.3 are  in principle  valid only for the special measurement situation
where the tones are presented one at a time. They should not be used directly to predict the
perception of more complicated signals such as music and speech because the curves do not
take masking and temporal matters into account. Reflections in a room are not taken into
account either.
Translations of Loudness Level:
Danish: Hørestyrkeniveau (enhed: Phon)
German: Lautstärkepegel (Einheit: Phon)
French: Niveau de Sonie.
2.4 Masking
The term ‘Masking’ is used about the phenomenon that the presence of a given sound (sound
A) can make another sound (sound B) inaudible, in other words A masks B or B is masked
by A. Masking is a very common phenomenon which is experienced almost every day, e.g.
when you need to turn down the radio in order to be able to use the telephone.
The situation described above is also called simultaneous masking because both the masking
signal and the masked signal are present at the same time. This is not the case in backward
and forward masking. Backward and forward refer to time. E.g. forward masking means
masking after a signal has stopped (i.e. forward in time). Simultaneous masking is best
64
Ear, Hearing and Speech
described in the frequency domain and is closely related to the movements of the Basilar
membrane in the inner ear.
The masking phenomenon is usually investigated by determining the hearing threshold for a
pure tone when various masking signals are present. The threshold determined in this
situation is called the masked threshold contrary to the absolute threshold.
2.4.1 Complete masking
If the ear is exposed to white noise, the hearing threshold (i.e. masked threshold) will be as
shown in Figure 2.4.1 where also the absolute threshold is shown. The masked threshold is
shown for different levels of the white noise.
Figure 2.4.1 Masking from white noise. The curves show the masked threshold for different spectrum
levels of white noise. From [3]
The masked thresholds are almost independent of frequency up to about 500 Hz. Above
500 Hz the threshold increases by about 10 dB per decade (= 3 dB/octave). A 10dB change
in the level of the noise will also change the masked threshold by 10 dB.
If a narrow band signal is used instead of the white noise, the masked threshold will be as
shown in Figure 2.4.2. Here the masked threshold is shown for a narrow band signal centred
at 250 Hz, 1 kHz and 4 kHz respectively. Generally the masking curves have steep slopes
(about 100 dB/octave) towards the low frequency side and less steep slopes (about
60 dB/octave) towards the high frequency side.
65
Ear, hearing and speech
Figure 2.4.2 Masking from narrow band noise. The curves show the masked threshold when the ear is
exposed to narrow band noise (critical band noise) at 250 Hz, 1 kHz and 4 kHz respectively. From
[3]
The masking curves for narrow band noise are very level dependent. This is illustrated in
Figure 2.4.3. The slope at the low frequency side is almost independent of level but the slope
at the high frequency side depends strongly on the level of the narrow band noise. The dotted
lines near the top of the curves indicate experimental difficulties due to interference between
the noise itself and the pure tone used to determine the masked threshold.
Figure 2.4.3 The influence of level on the masked threshold. The slope towards higher frequencies
decreases with increasing level, i.e. masking increases nonlinearly with level. From [3]
The masked threshold for narrow band noise is mainly caused by the basilar membrane
motion. The different slopes towards the low and the high frequency side are also seen here
and also the nonlinear level dependency is seen. Compare with Figure 2.2.4.
66
Ear, Hearing and Speech
67
2.4.2 Partial masking
The term ‘Complete masking’ is used when the presence of a given sound (sound A) can
make another sound (sound B) inaudible. Partial masking is a situation where sound A
influences the perception of sound B even though sound B is still audible. The influence is
mainly seen in the loudness of sound B.
An example: When you listen to a standard carradio while you are driving at, e.g. 100 km/h,
you will adjust the level of the radio to a comfortable level. There will be some background
noise from the engine, the tires, and the wind around the car (at least in ordinary cars). Then,
when you come to a crossing or a traffic light and have to stop you will hear that the radio
volume is much too high. This is an example of partial masking where the background noise
masks part of the radio signal and when the background noise disappears the masking
disappears too and the radio signal becomes louder than before. (Some modern car radios are
equipped with a speed dependent automatic level control. The example above is therefore not
fully convincing in this situation.)
2.4.3 Forward masking
It has been shown that a strong sound signal can mask another (weak) signal which is
presented after the strong signal. This kind of masking goes forward in time and is therefore
called forward masking. The effect lasts for about 200 ms after the end of the strong signal.
Forward masking is also called postmasking.
2.4.4 Backward masking
It has been shown that a strong sound signal can mask another (weak) signal which appears
before the strong signal. This kind of masking goes back in time and is therefore called
backward masking. The effect is restricted to about 20 ms before the start of the strong
signal.
Backward masking is also called premasking.
Ear, hearing and speech
2.5 Loudness
The term ‘loudness’ denotes the subjective perception of strength or powerfulness of the
sound signal. The unit for loudness is Son or Sone. Note that ‘loudness’ and ‘loudness level’
are two different concepts. Translation of terms:
Loudness Loudness Level
Danish Hørestyrke Hørestyrkeniveau
German Lautheit Lautstärkepegel
French Sonie Niveau de Sonie
2.5.1 The loudness curve
The Sone scale was established in order to avoid the confusion between dB SPL values and
the perception of loudness: A 1 kHz tone at 80 dB SPL is not perceived double as loud as the
same tone at 40 dB SPL. Figure 2.5.1 shows the relation between the Sone and the Phone
scales. (Hint: for a 1 kHz tone, phone and dB SPL is the same number). Arbitrarily it has
been decided that one sone should correspond to 40 phones. The curve is based on a great
number of loudness comparisons. The curve is called a loudness curve.
Figure 2.5.1 The loudness curve for a normal hearing test subject (solid line) and for a person with a
cochlear hearing loss (dashed)
68
Ear, Hearing and Speech
The straight part of the solid line in Figure 2.5.1 corresponds to Stevens’ power law:
10 / ) 40 (
2
−
=
L
N
where N is the loudness (in sone) and L is the loudness level ( in phones). The curve shows
that a doubling of the loudness corresponds to a 10phone increase in loudness level (or a 10
dB increase in SPL if we are dealing with a 1 kHz tone). For many daily life sounds a rule of
thumb says that a 10dB increase is needed in order to perceive a doubling of the loudness.
The loudness curve becomes steeper near the hearing threshold. This is also the case for a
person with a cochlear hearing loss (e.g., the very common hearing impairment caused by
age). An example of such a hearing loss is shown by the dashed curve in Figure 2.5.1 where
the threshold (1 kHz) is a little less than 40 dB SPL. The steeper slope means that  near the
threshold  the loudness increases rapidly for small changes in the sound level. This effect is
called loudness recruitment. Recent research have shown that – for this kind of hearing loss –
the loudness at threshold has a value significantly different from nil as indicated in the figure
[10]. In other words, listeners with cochlear hearing loss have softness imperception, rather
than loudness recruitment. Note that at higher sound levels the loudness perception is the
same for both normal and impaired listeners.
2.5.2 Temporal integration
The perception of loudness needs some time to build up. This means that short duration
sounds (less than one second) are perceived as less loud than the same sound with longer
duration. The growth of loudness as a function of duration is called temporal integration. The
growth resembles the exponential growth of a time constant. It has been shown that the time
constant is about 100 ms.
Short sounds  like a pistol shot, fireworks, handclap, etc.  are perceived as weak sounds
although their peak sound pressure levels may be well above 150 dB SPL. This is one of the
reasons why impulsive sounds generally are more dangerous than other sounds.
2.5.3 Measurement of loudness
Many years ago it was thought that a sound level meter with filters corresponding to the ears’
sensitivity (described by the equal loudness level contours (Figure 2.3.3)) could be used to
easure loudness. This is not the case. m
Figure 2.5.2 show the characteristics for the commonly used A and C filters, but due to
masking and other phenomena these filters will not give a result that corresponds to loudness.
For the determination of loudness, special calculation software is needed. For stationary
sounds two procedures can be found in [11]. For nonstationary sound, loudness calculations
are found in professional Sound Quality calculation software. For research purposes loudness
models (software) can be found on the Internet (e.g. at
http://hearing.psychol.cam.ac.uk/Demos/demos.html )
69
Ear, hearing and speech
10 100 1000 10000
20 50 200 500 2000 5000 20000 5
Frequency, Hz
80
70
60
50
40
30
20
10
0
10
20
W
e
i
g
h
t
i
n
g
c
h
a
r
a
c
t
e
r
i
s
t
i
c
,
d
B
Weighting filters
Afilter
Cfilter
Dfilter
Figure 2.5.2 Filter characteristics for the A, C and D filter. The data for the A and the C filter are
from [12]. The data for the D filter is from [13].
The main effect of the Afilter is that it attenuates the low frequency part of the signal. The
attenuation is e.g. 20 dB at 100 Hz and 30 dB at 50 Hz. Wind noise and other low frequency
components are attenuated by the Afilter and is therefore very practical for many noise
measurement situations.
The Cfilter is ‘flat’ in the major part of the audible frequency range. It may me used as an
approximation to a measurement with linear characteristic.
The Dfilter is mainly used in connection with evaluation of aircraft noise. The frequency
range around 3 kHz is known to be annoying and therefore this frequency range is given a
higher weight.
70
Ear, Hearing and Speech
2.6 The auditory filters
The movements of the basilar membrane in the inner ear constitute a frequency analyser
where the peak of the envelope moves along the basilar membrane as a function of
frequency. See Figure 2.2.4. The width of the envelope peak may be seen as an indication of
the selectivity of the analyser filter and it has been common practice to describe the
frequency selectivity of the ear as a set of filters, a filter bank, which cover the audible
frequency range. It should be noted though that the concept of a filter bank is a very coarse
description and should be seen as a typical engineering approximation to the real situation.
Frequency selectivity is important for the perception of the different frequencies in complex
sound signals such as speech and music. We rely e.g. on our frequency selectivity when we
distinguish different vowels from each other.
The concept of frequency discrimination is different from frequency selectivity. Frequency
discrimination is the ability to hear the difference between two tones that are close in
frequency (one frequency at a time).
2.6.1 Critical bands
The bandwidth of the filters in the filter bank can be determined by means of various
psychoacoustic experiments. Many of these are masking experiments and led to the
formulation of the critical band model. It is outside the scope of the present text to go into
the background and the details of this model.
The results of the investigations are shown in Figure 2.6.1. It is seen that the bandwidth
(Critical Bands) is almost constant at 100 Hz up to a centre frequency of about 500 Hz and
above this frequency the bandwidth increases. The increase in bandwidth above 500 Hz is
similar to the increase in bandwidth for onethirdoctave filters.
The critical bandwidth may be calculated from the empirical formula:
69 , 0 2
) 4 , 1 1 ( 75 25 f CB + + =
where CB is the bandwidth in Hz of the critical band and f is the frequency in kHz (not in
Hz).
71
Ear, hearing and speech
10 100 1000 10000
20 50 200 500 2000 5000 20000
Frequency, Hz
10
100
1000
10000
B
a
n
d
w
i
d
t
h
,
H
z
Bandwidth
Crit. Band
ERB
1/3 octave
Figure 2.6.1 Bandwidth of critical bands and Equivalent Rectangular bandwidth, ERB. The
bandwidth of 1/3octave filters (straight line) is shown for comparison. The curves are computed from
the formulas given in the text.
If the audible frequency range is ‘filled up’ with consecutive critical bands from the lowest
frequency to the highest frequency, it is seen that 24 critical bands will cover the whole
frequency range. Each of the ‘filters’ has been given a number called Bark. Bark number one
is the band from zero to 100 Hz; Bark number two is the band from 100 Hz to 200 Hz, etc.
Band no. 8 has a centre frequency of 1000 Hz and goes from 920 Hz to 1080 Hz. The band
around 4000 Hz is no. 17 and has a bandwidth of 700 Hz.
The critical bands are not fixed filters similar to the filters in a physical filter bank as the
numbers given above may indicate. The critical bands are a result of the incoming sound
signal and as such much more ‘flexible’ than physical filters would be.
2.6.2 Equivalent Rectangular Bands
The auditory filters have also been determined by means of notched noise measurements
where the threshold of a pure tone is determined in the notch of a broadband noise as a
function of the width of the notch. This leads to the concept of equivalent rectangular
bandwidth, i.e. the bandwidth of a rectangular filter that transmits the same amount of energy
as the auditory filter. The bandwidth of such rectangular filters is shown in Figure 2.6.1 as a
function of centre frequency.
The rectangular bandwidth may be calculated from the empirical formula:
) 1 37 , 4 ( 7 , 24 + = f ERB
where ERB is the bandwidth in Hz and f is the centre frequency in kHz.
72
Ear, Hearing and Speech
2.7 Speech
A speech signal is produced in the following way. Air is pressed from the lungs up through
the vocal tract, through the mouth cavities and/or the nose cavities and the sound is radiated
from the mouth and the nose. The vocal folds will vibrate when voiced sounds are produced.
2.7.1 Speech production
A schematic illustration of the production of voiced sounds is given in Figure 2.7.1 where the
vocal folds vibrate. The source spectrum is a line spectrum where the distance between the
lines corresponds to the fundamental frequency. The fundamental frequency is around
125 Hz for men, around 250 Hz for woman and around 300 for children, but there are big
individual variations. There are thus more lines in a male spectrum compared to a female.
The source spectrum decreases with the square of the frequency (1/f
2
). The source spectrum
is transformed by the ‘tube’ consisting of trachea, throat (pharynx) and the mouth. This
structure is simulated in Figure 2.7.1 by a cylindrical tube of length 17 cm.
Figure 2.7.1 The principle of vowel generation. From [14]
The tube has pronounced resonances (where the length of the tube corresponds to the odd
multiples of 1/4 wavelength) indicated by the peaks at 500, 1500 and 2500 Hz. The final
spectrum radiated from the mouth is then the product of the two spectra. The final spectrum
is a line spectrum with characteristic peaks caused by the transfer function. The peaks are
called formants and the formants are positioned differently for each vowel. Table 2.7.1 shows
the formants frequencies (in round numbers) for the three most different vowels. The sounds
are /i/: as in eve, /a/ as in father, /u/ as in moon. There are individual differences from person
to person.
73
Ear, hearing and speech
74
/i/
/a/
/u/
1. formant
2. formant
3. formant
225
2200
3000
700
1200
2500
250
700
2200
Table 2.7.1 Formant frequencies in Hz of the vowels /i/, /a/ and /u/.
The unvoiced sounds are produced in many different ways, e.g. by pressing air out through
the teeth /s/, out between the lips and the teeth /f/, by sudden opening of the lips /p/, sudden
opening between tongue and teeth /t/ and between tongue and palate /k/. These sounds are
called unvoiced because the vocal folds do not vibrate but stays open in order for the air to
pass.
2.7.2 Speech spectrum, speech level
A general longterm speech spectrum is shown in Figure 2.7.2 that is based on the average of
18 speech samples from 12 languages.
The spectrum is a onethird octave spectrum which means that the curves are tilted
3 dB/octave compared to the result of a FFTcalculation. (The result of a FFT is a density
spectrum).
It is worth to note that the speech spectrum is almost independent of the language. This is not
surprising when the speech production mechanism is taken into account. The spectrum in
Figure 2.7.2 is based on English (several dialects), Swedish, Danish, German, French
(Canadian), Japanese, Cantonese, Mandarin, Russian, Welsh, Singhalese and Vietnamese. A
total of 392 talkers participated in the investigation.
The spectrum for women falls off below 200 Hz because their fundamental frequency
typically is around 250 Hz. The maximum is found around 500 Hz for both gender and above
500 Hz the two curves are almost identical. The slope above 500 Hz is approximately minus
10 dB per decade (or 3 dB/octave).
Ear, Hearing and Speech
100 1000 10000
200 500 2000 5000 20000 50
Frequency, Hz
30
40
50
60
70
1
/
3
o
c
t
a
v
e
l
e
v
e
l
Speech spectrum
Male
Female
3 dB /oct
Figure 2.7.2 The longterm speech spectrum for male and female speech shown as a 1/3octave
spectrum. For comparison a line with slope –3 dB per octave (= –10 dB per decade) is shown.
Redrawn from [15]
The average level of male speech is about 65 dB SPL, measured at 1 m in front of the mouth.
For women the level is typically 3 dB lower, i.e. 63 dB. (Compare the number of lines in the
spectrum). During normal speech the level will vary ±15 dB around the mean value.
2.7.3 Speech intelligibility
The speech intelligibility of a transmission system is usually measured by means of a list of
words (or sentences) where the percentage of correctly understood words gives the
intelligibility score. The transmission system could be almost anything, e.g. a telephone line
or a room. The intelligibility depends on the word material (sentences, single words,
numbers, etc.), the speaker, the listener, the scoring method and the quality of the
transmission system.
Often the intelligibility score is given as a function of the signaltonoise ratio. An example
of this is shown in Figure 2.7.3 for the wordmaterial on the Dantale CD. This CD contains
eight tracks of 25 words each. The words are common Danish singlesyllable words that are
distributed phonetically balanced over the eight lists so that the lists can be regarded as
equivalent. The words are recorded on the left channel of the CD and on the right channel a
noise signal is recorded with (almost) the same spectrum as the words. The noise signal is
amplitude modulated in order to make it resemble normal speech. The Dantale CD is
described in [16]
The result in Figure 2.7.3 is obtained with the words and the noise on the Dantale CD with
untrained Danish normal hearing listeners. It is seen that even at a signaltonoise ratio of
75
Ear, hearing and speech
0 dB almost all words are understood. It is also seen that an increase of just 10 dB in SNR
can change the situation from impossible to reasonable, e.g. from 15 dB (10%) to  5 dB
(70%). It is a general finding that such a relatively small improvement of the signaltonoise
ratio can improve the intelligibility situation dramatically. In other words, if the background
noise in a room is a problem for the understanding of speech in the room, then just a small
reduction of the background noise will be beneficial.
25 20 15 10 5
SNR, dB
0
0
10
20
30
40
50
60
70
80
90
100
W
o
r
d
s
c
o
r
e
,
%
Figure 2.7.3 Word score for the speech material DANTALE as a function of speechtonoise ratio
(SNR). Redrawn from [17]
It is time consuming and complicated to measure speech intelligibility with test subjects.
Therefore measurement and calculation methods have been developed for the estimation of
the expected speech intelligibility in a room or on a transmission line.
Articulation Index, AI [18]: Determination of the signaltonoise ratio in frequency bands
(usually one octave or onethird octave). The SNR values are weighted according to the
importance of the frequency band. The weighted values are added and the result normalised
to give an index between zero and one. The index can then be translated to an expected
intelligibility score for different speech materials.
Speech Intelligibility Index, SII [19]: This method is based on the AI principle, but the
weighting functions are changed and a number of ‘corrections’ to the AImethod are
implemented. One of these is the correction for the change in speech spectrum according to
the vocal effort (shouting, raised voice, low voice).
Speech Transmission Index, STI [20]: In this method the modulation transfer function, MTF,
from the source (the speaker) to the receiver (the listener) is determined. The MTF is
determined for octave bands of noise (125 Hz to 8 kHz) and for a number of modulation
76
Ear, Hearing and Speech
77
frequencies (0,63 Hz to 12,5 Hz). The reduction in modulation is transformed to an
equivalent signaltonoise ratio and as in the AI method these values are added and
normalised in order to yield an index between zero and one. The index can then be translated
to an expected intelligibility score for different speech materials.
Rapid Speech Transmission Index, RASTI [21]: This is an abbreviated version of STI. Only
the frequency bands 500 Hz and 2 kHz and only nine different modulation frequencies are
used. The result is an index which is used in the same way as in STI.
Ear, hearing and speech
78
2.8 References
1. Hougaard, S., et al., Sound and Hearing. 2 ed. 1995: Widex.
2. Engström, H. and Engström, B., A short survey of some common or important ear
diseases. 1979: Widex.
3. Zwicker, E. and Fastl, H., Psychoacoustics. Facts and models. 2 ed. 1999: Springer.
4. Kemp, D.T., Developments in cochlear mechanics and techniques for noninvasive
evaluation. Adv Audiol, 1988. 5: p. 2745.
5. ISO226, Acoustics  Normal equalloudnesslevel contours, in FDIS, May 2002,
N327. 2002, International Standardization Organization: Geneva.
6. ISO3897, Acoustics  Reference zero for the calibration of audiometric equipment
Part 7: Reference threshold of hearing under freefield and diffusefield listening
conditions. 1996, International Organization for Standardization: Geneva,
Switzerland.
7. ISO3891, Acoustics  Reference zero for the calibration of audiometric equipment 
Part 1: Reference equivalent threshold sound pressure levels for pure tones and
supraaural earphones. 1991, International Organization for Standardisation: Geneva,
Switzerland.
8. ISO3895, Acoustics  Reference zero for the calibration of audiometric equipment 
Part 5: Reference equivalent threshold sound pressure levels for pure tones in the
frequency range 8 kHz to 16 kHz“. 1998, International Organization for
Standardization: Geneva, Switzerland.
9. ISO3898, Acoustics  Reference zero for the calibration of audiometric equipment 
Part 8: Reference equivalent threshold sound pressure levels for pure tones and
circumaural earphones (ISO/DIS). 2001, International Organization for
Standardization: Geneva, Switzerland.
10. Florentine, M. and Buus, S. Evidence for normal loudness growth near threshold in
cochlear hearing loss. in 19 Danavox Symposium. 2001. Kolding, Denmark. p. xxyy.
11. ISO532, Acoustics  Method for calculating loudness level. 1975, International
Organisation for Standardisation: Geneva, Switzerland.
12. IEC651, Sound level meters. 1979, International Electrotecnical Commission:
Geneva, Switzerland.
13. IEC537, Frequency weighting for the measurement of aircraft noise (Dweighting).
1976, International Electrotechnical Commission: Geneva, Switzerland.
Ear, Hearing and Speech
79
14. Borden, G. and Harris, K., Speech science primer. 1980: Williams & Wilkins.
15. Byrne, D., Ludvigsen, C., and al., e., Longterm average speech spectra ... J. Acoust.
Soc. Am., 1994. 96(no. 4): p. 2110?2120?
16. Elberling, C., Ludvigsen, C., and Lyregaard, P.E., DANTALE, a new Danish Speech
material. Scandinavian Audiology, 1989. 18: p. 169175.
17. Keidser, G., Normative data in quiet and in noise for DANTALE  a Danish speech
material. Scandinavian Audiology, 1993. 22: p. 231236.
18. ANSIS3.5, American National Standard methods for the calculation of the
Articulation Index. 1969, American National Standards Institute, Inc.: New York.
19. ANSIS3.5, American National Standard methods for the calculation of the Speech
Intelligibility Index. 1997, American National Standards Institute, Inc.: New York.
20. Steeneken, H. and Houtgast, T., A physical method for measuring speech
transmission quality. J. Acoust. Soc. Am., 1980. 67: p. 318326.
21. IEC26816, Sound system equipment  Part 16: The objective rating of speech
intelligibility in auditoria by the RASTI method. 1988, International Electrotechnical
Commission.
Further reading:
Plack, C. J. (2005). The sense of hearing. Lawrence Earlbaum Associates. ISBN: 08058
48843
Moore, B. C. J. (2003). An introduction to the psychology of hearing. 5th Edition. Academic
press ISBN: 0125056281
Yost, W. A. (2000). Fundamentals of hearing. An introduction. 4th Edition. Academic press.
ISBN: 0127756957
About standardized audiological, clinical tests see
http://www.phon.ucl.ac.uk/home/andyf/natasha/
3. An introduction to room acoustics
Jens Holger Rindel
3.1 SOUND WAVES IN ROOMS
3.1.1 Standing waves in a rectangular room
A rectangular room has the dimensions l
x
, l
y
, and l
z
. The wave equation can then be written
0
2
2
2
2
2
2
2
= +
∂
∂
+
∂
∂
+
∂
∂
p k
z
p
y
p
x
p
(3.1.1)
where p is the sound pressure and k = ω /c is the angular wave number, ω is the angular
frequency and c is the speed of sound in air. The equation can be solved by separation of the
variables and it is assumed that the solution can be written in the form:
t
z Z y Y x X p
ω j
e ) ( ) ( ) ( ⋅ ⋅ ⋅ =
Insertion in (3.1.1) and division by p gives
0
1 1 1
2
2
2
2
2
2
2
= +
∂
∂
+
∂
∂
+
∂
∂
k
z
Z
Z y
Y
Y x
X
X
This can be separated, and for the xdirection it yields
0
1
2
2
2
= +
∂
∂
x
k
x
X
X
Similar equations hold for the y and zdirections. The angular wave number k has been divided
into three
(3.1.2)
2 2 2 2
z y x
k k k k + + =
The general solution to the above onedimensional equation is
) cos( ) (
x x x
x k C x X ϕ + =
in which the constants C
x
and ϕ
x
are determined from the boundary conditions.
The room surfaces are now assumed to be rigid, i.e. the normal component of the particle velocity
is zero at the boundaries
0
j
1
=
∂
∂
− =
x
p
u
x
ωρ
for x = 0 and x = l
x
This means that ϕ
x
= 0 and
x
x
x
n
l
k ⋅ =
π
where n
x
= 0, 1, 2, 3, … (3.1.3)
Two similar boundary conditions hold for the y and zdirections. With these conditions the
solution to (3.1.1) is
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⋅
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⋅
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⋅ =
z
z
y
y
x
x
l
z
n
l
y
n
l
x
n p p π π π cos cos cos
0
(3.1.4)
The time factor e
jωt
is understood. The amplitude of the sound pressure does not move with time,
so the waves that are solutions to (3.1.4) are called standing waves. They are also called the
modes of the room, and each of them is related to a certain natural frequency (or eigenfrequency)
given by
2 2 2
2 2 2
z y x
n
n
k k k
c k c
f + + = = =
π π π
ω
81
2
2
2
2
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
=
z
z
y
y
x
x
n
l
n
l
n
l
n c
f (3.1.5)
The modes can be divided into three groups:
Axial modes are onedimensional, only one of n
x
, n
y
, n
z
is > 0.
Tangential modes are twodimensional, two of n
x
, n
y
, n
z
are > 0.
Oblique modes are threedimensional, all three of n
x
, n
y
, n
z
are > 0.
Some examples are shown in Fig. 3.1.1. It is observed that the set of numbers (n
x
, n
y
, n
z
) indicate
the number of nodes (places with p = 0) along each coordinate axis.
Figure 3.1.1. Examples of room modes. (2,0,0) is onedimensional and (2,1,0) is two
dimensional. The lines are isosound pressure amplitude curves.
n
x
n
y
n
z
f
n
(Hz)
0 1 0 25
1 0 0 30
0 0 1 36
1 1 0 39
0 1 1 43
1 0 1 47
0 2 0 49
1 1 1 53
1 2 0 58
2 0 0 60
0 2 1 61
2 1 0 65
1 2 1 68
2 0 1 70
0 0 2 72
Table 3.1.1. Calculated natural frequencies at low frequencies using (3.1.5) in a rectangular
room with dimensions 5.7 m, 7.0 m, 4.8 m.
82
3.1.2 Transfer function in a room
The transfer function is the frequency response from a source position to a receiver position in a
room. A measured transfer function is shown in Fig. 3.1.2. It fluctuates very much with frequency
and the maxima can be identified as the natural frequencies of the room. The example in Fig.
3.1.2 has the same room dimensions as was used for the calculations in Table 3.1.1.
Figure 3.1.2. Transfer function in a rectangular room. At low frequencies it is possible to identify
the modes by their modal numbers.
3.1.3 Density of natural frequencies
A closer inspection of equation (3.1.5) shows that the natural frequencies of a rectangular room
may be interpreted in a geometrical way. A threedimensional frequency space is shown in Fig.
3.1.3. The natural frequencies of the onedimensional modes are marked on each of the axes,
representing the axial modes of the length, the width and the height, respectively. The interesting
observation is now that the points in the grid represent the oblique modes, and the distance to
each point from the origin is the natural frequency of that mode. So, the number of oblique modes
below a certain frequency f is equal to the number of grid points inside the sphere with radius f.
The volume is 1/8 of the sphere with radius f, i.e. (4 π f
3
/ 3) / 8 = π f
3
/ 6. Each mode occupies a
volume c
3
/ (8 l
x
l
y
l
z
) = c
3
/ (8 V). So, the number of oblique modes below f is approximately:
3
3
3
3
3
4 8
6 c
f V
c
V f
N
obl
π π
= =
The tangential modes are found in the plane between two of the axes. If these and the axial modes
are also taken into account, the number of modes with natural frequencies below the frequency f
is:
c
f L
c
f S
c
f V
N
8 4 3
4
2 3
+
⎟
⎠
⎞
⎜
⎝
⎛
+
⎟
⎠
⎞
⎜
⎝
⎛
=
π π
(3.1.6)
V is the volume of the room, S = 2( l
x
l
y
+ l
x
l
z
+ l
y
l
z
) is the total area of the surfaces, and L = 4 (l
x
+ l
y
+ l
z
) is the total length of all edges. It should be noted that the modal points of the tangential
and axial modes in Fig. 3.1.3 are located on the coordinate planes and axes, respectively.
Therefore we count the tangential points only as halves and those on the axes only as quarters.
At high frequencies the oblique modes dominate, and the first term in (3.1.6) is a good
approximation for any room, not only for rectangular rooms.
83
Figure 3.1.3. Frequencygrid, in which each grid point represents a room mode.
The modal density is the average number of modes per hertz.
c
L
f
c
S
f
c
V
f
N
8 2
4
d
d
2
2
3
+ + =
π
π (3.1.7)
In Fig. 3.1.4 this is compared to the actual modal density in a room. For high frequencies it is
sufficient to use the first term (oblique modes) for the modal density:
2
3
4
d
d
f
c
V
f
N
π ≅ (3.1.8)
84
Figure 3.1.4. Modal density as a function of frequency. Actual number of modes per 10 Hz in a
rectangular room and estimated by (3.1.7).
3.2 STATISTICAL ROOM ACOUSTICS
3.2.1 The diffuse sound field
In this chapter the acoustical behaviour of a room is treated from a statistical point of view, based
on energy balance considerations. It is assumed that the modal density is high enough, so the
influence of single modes in the room can be neglected. It is also assumed that the reflection
density is high enough, so the phase relations between individual reflections can be neglected.
This means that the reflections in the room are assumed to be uncorrelated and their contribution
can be added on an energy basis.
The diffuse sound field is defined as a sound field in which:
The energy density is the same everywhere
All directions of sound propagation occur with the same probability
It is obvious that the direct sound field near a sound source is not included in the diffuse sound
field. Neither are the special interference phenomena that are known to give increased energy
density near the room boundaries and corners. The diffuse sound field is an ideal sound field that
does not exist in any room. However, in many cases the diffuse sound field can be a good and
very practical approximation to the real sound field.
85
3.2.2 Incident sound power on a surface
In a plane propagating sound wave the relation between rms sound pressure p
1
and sound
intensity I
1
is
c I p ρ ⋅ =
1
2
1
In a diffuse sound field the rms sound pressure p
diff
is the result of sound waves propagating in all
directions, and all having the sound intensity I
1
. By integration over a sphere with the solid angle
ψ = 4π the rms sound pressure in the diffuse sound field is
(3.2.1) c I c I p
diff
ρ π ψ ρ
π ψ
⋅ ⋅ = ⋅ =
∫
=
1
4 d
4
1
2
In the case of a plane wave with the angle of incidence θ relative to the normal of the surface, the
incident sound power per unit area on the surface is
θ
ρ π
θ
θ
cos
4
cos
2
1
c
p
I I
diff
= = (3.2.2)
where p
diff
is the rms sound pressure in the diffuse sound field. This is just the sound intensity in
the plane propagating wave multiplied by the cosine, which is the projection of a unit area as seen
from the angle of incidence, see Fig. 3.2.1.
p
diff
p
1
I
inc
I
θ
θ
b
a
Figure 3.2.1. a: Plane wave at oblique incidence on a surface. b: Diffuse incidence on a surface.
The total incident sound power per unit area is found by integration over all angles of incidence
covering a half sphere in front of the surface, see Fig. 3.2.2. The integration covers the solid angle
ψ = 2π.
2
1
2
1
) d(sin sin 2
4
1
d d sin cos
4
1
d
2
1
0
2
2
0
2 /
0
2
2
⋅ ⋅ = ⋅ ⋅ =
= =
∫
∫ ∫ ∫
=
c
p
c
p
c
p
I I
diff diff
diff
inc
ρ
θ θ
ρ
π
π
ϕ θ θ θ
ρ π
ψ
π π
π ψ
θ
c
p
I
diff
inc
ρ 4
2
= (3.2.3)
It is noted that this is four times less than in the case of a plane wave of normal incidence.
86
Figure 3.2.2. Definition of angles of incidence in a diffuse sound field.
3.2.3 Equivalent absorption area
The absorption coefficient α is defined as the ratio of the nonreflected sound energy to the
incident sound energy on a surface. It can take values between 0 and 1, and α = 1 means that all
incident sound energy is absorbed in the surface. An example of a surface with absorption
coefficient, α = 1 is an open window.
The product of area and absorption coefficient of a surface material is the equivalent absorption
area of that surface, i.e. the area of open windows giving the same amount of sound absorption as
the actual surface. The equivalent absorption area of a room is
m
i
i i
S S S S A α α α α = + + = =
∑
.....
2 2 1 1
(3.2.4)
where S is the total surface area of the room and α
m
is the mean absorption coefficient. The unit
of A is m
2
. In general, the equivalent absorption area may also include sound absorption due to
the air and due to persons or other objects in the room.
3.2.4 Energy balance in a room
The total acoustic energy in a room is the sum of potential energy and kinetic energy, or twice the
potential energy, since the time average of the two parts must be equal. The total energy E is the
energy density multiplied by the room volume V:
V
c
p
V w V w w E
pot kin pot
2
2
2 ) (
ρ
= = + = (3.2.5)
Here and in the following, p denotes the rms sound pressure in the diffuse sound field (called p
diff
in section 3.2.2). The energy absorbed in the room is the incident sound power per unit area
(3.2.3) multiplied by the total surface area and the mean absorption coefficient, i.e. the equivalent
absorption area (3.2.4),
A
c
p
A I S I P
inc m inc abs a
ρ
α
4
2
,
= = = (3.2.6)
If P
a
is the sound power of a source in the room, the energy balance equation of the room is
87
t
E
P P
abs a a
d
d
,
= − (3.2.7a)
) (
d
d
4
2
2
2
p
t c
V
A
c
p
P
a
ρ ρ
= − (3.2.7b)
With a constant sound source a steady state situation is reached after some time, and the right side
of the equation is zero. So, the absorbed power equals the power emitted from the source, and the
steady state sound pressure in the room is
c
A
P
p
a
s
ρ
4
2
= (3.2.8)
This equation shows that the sound power of a source can be determined by measuring the sound
pressure generated by the source in a room, provided that the equivalent absorption area of the
room is known. It also shows how the absorption area in a room has a direct influence on the
sound pressure in the room. For some cases it is more convenient to express eq. (3.2.8) in terms
of the sound pressure level L
p
and the sound power level L
W
,
⎟
⎠
⎞
⎜
⎝
⎛
+ ≅
A
A
L L
W p
0
4
log 10 (dB) (3.2.9)
where A
0
= 1 m
2
is a reference area. The approximation comes from neglecting the term with the
constants and reference values
dB 0 dB 14 . 0
) 10 20 ( 1
10 343 204 . 1
log 10 log 10
2 6
12
2
0
≅ =
⋅ ⋅
⋅ ⋅
=
−
−
ref
ref
p A
P c ρ
3.2.5 Reverberation time. Sabine´s formula
If the sound source is turned off after the sound pressure has reached the stationary value, the first
term in the energy balance equation (3.2.7b) is zero, and the rms sound pressure is now a function
of time:
( ) 0 ) (
d
d
) (
4
2
2
2
= + t p
t c
V
t p
c
A
ρ ρ
(3.2.10)
The solution to this equation can be written
t
V
A c
s
p t p
4 2 2
e ) (
−
= (3.2.11)
where p
s
2
is the mean square sound pressure in the steady state and t = 0 is the time when the
source is turned off. It is seen that the mean square sound pressure, and hence the sound energy,
follows an exponential decay function. On a logarithmic scale the decay is linear, and this is
called the decay curve, see Fig. 3.2.3.
If instead the source is turned on at the time t = 0, the sound buildup in the room follows a
similar exponential curve, also shown in Fig. 3.2.3.
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
− =
− t
V
A c
s
p t p
4 2 2
e 1 ) ( (3.2.12)
88
Figure 3.2.3. Buildup and decay of sound in a room. Here the source is turned on a t = 0 and
turned off at t = 1 s. Top: linear scale (sound pressure squared). Bottom: logarithmic scale (dB).
The reverberation time T
60
is defined as the time it takes for the sound energy in the room to
decay to one millionth of the initial value, i.e. a 60 dB decay of the sound pressure level. Hence,
for t = T
60
,
60
4 2 6 2 2
e 10 ) (
T
V
A c
s s
p p t p
−
−
= =
So, the reverberation time is
A c
V
A c
V
T
3 . 55 4
) 10 ln( 6
60
= ⋅ ⋅ = (3.2.13)
This is Sabine’s formula named after Wallace C. Sabine, who introduced the reverberation time
concept around 1896. He was the first to demonstrate that T
60
is inversely proportional to the
equivalent absorption area A.
Note: Sabine’s formula is often written as T
60
= 0.16 V/A. However, this implies that V must be in
m
3
and A in m
2
.
3.2.6 Stationary sound field in a room. Reverberation distance
A reverberation room is a special room with long reverberation time and a good diffusion. In
such a room the diffuse sound field is a good approximation, and the results for stationary
conditions (3.2.8) and for sound decay (3.2.13) can be applied to measure the sound power of a
sound source:
60
2
3 . 55
4 T c
V
c
p
P
s
a
⋅ =
ρ
(3.2.14)
The reverberation time and the average sound pressure level in the reverberation room are
measured, and the sound power level is calculated from
89
dB
t
T
V
V
L
T c P
V p
L L
p
ref
ref
p W
14 log 10 log 10
4
3 . 55
log 10
0
60
0
60
2
2
− − + =
⋅ ⋅
⋅ ⋅
+ =
ρ
(3.2.15)
where V
0
= 1 m
3
and t
0
= 1 s.
In most ordinary rooms the diffuse sound field is not a good approximation. Each of the
following conditions may indicate that the sound field is not diffuse
An uneven distribution of sound absorption on the surfaces, e.g. only one surface is highly
absorbing
A lack of diffusing or sound scattering elements in the room
The ratio of longest to shortest room dimension is higher than three
The volume is very large, say more than 5000 m
3
A rather simple modification to the stationary sound field is to separate the direct sound. The
sound power radiated by an omnidirectional source is the sound intensity at the distance r in a
spherical sound field multiplied by the surface area of a sphere with radius r
(3.2.16)
2
4 r I P
r a
π ⋅ =
Thus, the sound pressure squared of direct sound in the distance r from the source is
c
r
P
p
a
dir
ρ
π
2
2
4
= (3.2.17)
The stationary sound is described by (3.2.8)
c
A
P
p
a
s
ρ
4
2
=
The reverberation distance r
rev
is defined as the distance where p
dir
2
= p
s
2
when an omni
directional point source is placed in a room. It is a descriptor of the amount of absorption in a
room, since the reverberation distance depends only on the equivalent absorption area
A
A
r
rev
14 . 0
16
= =
π
(3.2.18)
At a distance closer to the source than the reverberation distance, the direct sound field
dominates, and this is called the direct field. At longer distances the reverberant sound field
dominates, and in this socalled far field the stationary, diffuse sound field may be a usable
approximation.
An expression for the combined direct and diffuse sound field can derived by simple addition of
the squared sound pressures of the two sound fields. However, since the direct sound is treated
separately, it should be extracted from the energy balance equation, which was used to describe
the diffuse sound field. To do this, the sound power of the source should be reduced by a factor of
(1  α
m
), which is the fraction of the sound power emitted to the room after the first reflection. So,
the squared sound pressure in the total sound field is
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
− + = − + =
m
rev
s m s dir total
r
r
p p p p α α 1 ) 1 (
2
2
2 2 2 2
(3.2.19)
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
− + ⋅ = ) 1 (
4
4
1
2
2
m a total
A r
c P p α
π
ρ (3.2.20)
90
Normal sound sources like a speaking person, a loudspeaker or a musical instrument radiate
sound with different intensity in different directions. The directivity factor Q is the ratio of the
intensity in a certain direction to the average intensity,
a
P
r
I Q
2
4π
⋅ = (3.2.21)
So, the squared sound pressure of the direct sound is
c
r
P Q
p
a
dir
ρ
π
2
2
4
⋅
= (3.2.22)
This leads to a general formula for the sound pressure level as a function of the distance from a
sound source in room.
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
− + + + ≅
m
rev
W p
r
r
Q
A
A
L L α 1 log 10
4
log 10
2
2
0
(dB) (3.2.23)
where A
0
= 1 m
2
. In a reverberant room with little sound absorption (say, α
m
< 0.1) the sound
pressure level in the far field will be approximately as predicted by the diffuse field theory, i.e.
the last term will be close to zero. In the case of a highly directive sound source like a trumpet (Q
>> 1) the direct field can be extended to distances much longer than the reverberation distance. In
the latter situation the last term in (3.2.23) raises the sound pressure level above the diffuse field
value.
Figure 3.2.4. Relative sound pressure level as a function of distance in a room with
approximately diffuse sound field. The source has a directivity factor of one. The parameter on
the curves is A / (1  α
m
) in m
2
.
In large rooms with medium or high sound absorption (say, α
m
> 0.2) the sound pressure level
will continue to decrease as a function of the distance, because the diffuse field theory is not valid
in such a room. Instead, the slope of the spatial decay curve may be taken as a measure of the
degree of acoustic attenuation in a room. So, in large industrial halls the attenuation in dB per
doubling of the distance may be a better descriptor than the reverberation time.
91
3.3 GEOMETRICAL ROOM ACOUSTICS
3.3.1 Sound rays and a general reverberation formula
In geometrical acoustics rays are used to describe the sound propagation. The concept of rays
implies that the wavelength and the phase of the sound are neglected, and only the direction of
sound energy propagation is treated in geometrical acoustics.
The sound decay shall now be studied by following a plane wave travelling as a ray from wall to
wall, see Fig. 3.3.1. The energy of the wave is gradually decreased due to absorption at the
surfaces, all of which are assumed to have the mean absorption coefficient α
m
.
The ray representing a plane wave may start in any direction and it is assumed that the decay of
energy in the ray is representative for the decay of energy in the room. The room may have any
shape.
Figure 3.3.1. A plane wave travelling as a ray from wall to wall in a room.
By each reflection the energy is reduced by a factor (1  α
m
). The initial sound pressure is p
0
and
after n reflections the squared sound pressure is
) 1 ln( 2
0
2
0
2
e ) 1 ( ) (
m
n n
m
p p t p
α
α
− ⋅
⋅ = − ⋅ = (3.3.1)
The distance of the ray from one reflection to the next is l
i
and the total distance traveled by the
ray up to the time t is
(3.3.2)
m
i
i
l n t c l ⋅ = ⋅ =
∑
where l
m
is the mean free path. So, the squared sound pressure is
t
l
c
m
m
p t p
⋅ − ⋅
⋅ =
) 1 ln(
2
0
2
e ) (
α
(3.3.3)
When the squared sound pressure has dropped to 10
 6
of the initial value, the time t is by
definition the reverberation time T
60
:
60
) 1 ln(
6
) 1 ln( ) 10 ln( 6 e 10
60
T
l
c
m
m
T
l
c
m
m
⋅ − ⋅ = ⋅ − ⇒ =
⋅ − ⋅
−
α
α
This leads to an interesting pair of general reverberation formulas:
m
m
m
m
c
l
c
l
T
α α ⋅
⋅
≈
− ⋅ −
⋅
=
8 . 13
) 1 ln(
8 . 13
60
(3.3.4)
The last approximation is valid if α
m
< 0.3, i.e. only in rather reverberant rooms. The
approximation comes from:
L + + + =
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
−
= − −
3 2 1
1
ln ) 1 ln(
3 2
α α
α
α
α
m
m
92
With the assumption that all directions of sound propagation appear with the same probability, it
can be show (Kosten, 1960) that the mean free path in a threedimensional room is
S
V
l
m
4
= (3dimensional) (3.3.5)
where V is the volume and S is the total surface area.
Similarly, the mean free path in a twodimensional room can be derived. This could be the
narrow air space in a double wall, or structureborne sound in a plate. The height or thickness
must be small compared to the wavelength. In this case the mean free path is
U
S
l
x
m
π
= (2dimensional) (3.3.6)
where S
x
is the area and U is the perimeter. The onedimensional case is just the sound travelling
back and forth between two parallel surfaces with the distance l = l
m
.
Insertion of (3.3.5) in the last part of (3.3.4) gives the Sabine formula (3.2.13), whereas insertion
in the first part of (3.3.4) leads to the socalled Eyring’s formula for reverberation time in a room:
) 1 ln(
3 . 55
60
m
S c
V
T
α − ⋅ ⋅ −
⋅
= (3.3.7)
In a reverberant room (α
m
< 0.3) it gives the same result as Sabine’s formula, but in highly
absorbing rooms Eyring’s formula is theoretically more correct. In practice the absorption
coefficients are not the same for all surfaces and the mean absorption coefficient is calculated as
in (3.2.4):
∑
⋅ =
i
i i m
S
S
α α
1
(3.3.8)
In the extreme case of an anechoic room (α
m
= 1) Eyring’s formula gives correctly a
reverberation time of zero, whereas Sabine’s formula is obviously wrong, giving the value T
60
=
55.3 V/c S. However, in normal rooms with a mixture of different absorption coefficients it is
recommended to use Sabine’s formula.
3.3.2 Sound absorption in the air
A sound wave travelling through the air is attenuated by a factor m, which depends on the
temperature and the relative humidity of the air, see Fig. 3.3.2. The unit of the air attenuation
factor is m
 1
. If this attenuation is included in (3.3.3) the squared sound pressure in the decay is
( )
m m
m
m
m
l m
l
t c
t c m
t
l
c
p p t p
⋅ − − ⋅
−
⋅ − ⋅
⋅ = ⋅ ⋅ =
) 1 ln(
2
0
) 1 ln(
2
0
2
e e e ) (
α α
(3.3.9)
The general reverberation formula then becomes
) (
8 . 13
) ) 1 ln( (
8 . 13
60
m m
m
m m
m
l m c
l
l m c
l
T
⋅ +
⋅
≈
⋅ + − −
⋅
=
α α
(3.3.10)
In the threedimensional case with (3.3.5) we then have
) 4 (
3 . 55
) 4 ) 1 ln( (
3 . 55
60
mV S c
V
mV S c
V
T
m m
+ ⋅
⋅
≈
+ − ⋅ −
⋅
=
α α
(3.3.11)
These two expressions are the Eyring and the Sabine formula, respectively, with the air
absorption included. By comparison with (3.2.13) it is seen that the equivalent absorption area
including air absorption is
(3.3.12) mV S A
i
i i
4 + =
∑
α
Some typical values of m are found later in Table 3.4.3.
93
Figure 3.3.2. The air attenuation factor m as a function of the relative humidity. The air
temperature is 20 °C. (Ref.: Harris1966).
3.3.3 Sound reflections and image sources
The direction of a sound reflection from a large plane surface follows the same geometrical law,
as known from optics, i.e. the angle of reflection is equal to the angle of incidence. This means
that the reflected sound can be interpreted as sound coming from an image source behind the
reflecting surface, see Fig. 3.3.3. This principle can be extended to higher order reflections.
Figure 3.3.3. Reflection in one surface (a) and in two surfaces (b). A is the source and R is the
receiver. First order image sources are indicated by A’ and second order image sources by A’’.
94
Echo is a wellknown acoustic phenomenon. It is defined as a single sound reflection that is
clearly audible as separate from the direct sound. The human ear is able to hear a reflection as an
echo if the time delay is approximately 50 ms. The socalled echoellipse is shown in Fig. 3.3.4.
Any point E on the ellipse represents a potential reflection with a delay of 50 ms, i.e. the distance
LE + EP = 17 m. Reflections from room surfaces outside the ellipse (as R
2
on the figure) are
delayed more than 50 ms and may cause an echo at the receiver point.
Figure 3.3.4. The echoellipse in the longitudinal section of an auditorium. L is the source and P
the receiver. (Ref.: Petersen 1984).
3.3.4 Reflection density in a room
The image source principle can easily be applied to higher order reflections in a rectangular
room. An infinite number of image rooms make a grid, and each cell in the grid is an image room
containing an image source. The principle is shown for the twodimensional case in Fig. 3.3.5.
Figure 3.3.5. Rectangular room with a sound source and image sources, here shown in two
dimensions. Image sources located inside the circle with radius ct will contribute reflections up
to time t.
95
If an impulse sound is emitted the number of reflections that will arrive within the time t can be
calculated as the volume of a sphere with radius ct divided by the room volume V:
V
ct
t N
3
3
4
) (
) (
π
= (3.3.13)
The reflection density is then the number of reflections within a small time interval dt, and by
differentiation:
2
3
4
d
d
t
V
c
t
N
π = (3.3.14)
The reflection density increases with the time squared, so the higher order reflections are
normally so dense in arrival time that it is impossible to distinguish separate reflections. If
(3.3.14) is compared to (3.1.8), it is striking to observe the analogy between reflection density in
the time domain and modal density in the frequency domain.
3.4 ROOM ACOUSTICAL DESIGN
3.4.1 Choice of room dimensions
The room dimensions determine the natural frequencies of a room. A good acoustical design of a
room implies that the transfer function should be as smooth as possible. With reference to Fig.
3.1.2 is clear that the room dimensions of a rectangular room should not be identical, because in a
cubic room many modes will have the same natural frequency, and thus there will be bigger gaps
in the transfer function. This would be very unfortunate, especially at low frequencies in small
rooms for speech, music or acoustic measurements. The dimensions of such rooms should be
designed after calculations of the normal modes below 100 Hz, see also Table 3.1.1.
3.4.2 Reflection control
In room with an audience it is very important to design the room surfaces with respect to the early
reflections. First of all in order to avoid problems with echo and focusing, but also to ensure a
good distribution of reflections to the audience area, see Fig. 3.4.1. In rooms for speech the
ceiling reflections are most important, whereas rooms for music should not give too much
reflection directly from the ceiling. In such room the ceiling should rather give diffuse reflections,
but the side walls are important because lateral reflections contribute to the acoustic of a concert
hall, see Fig. 3.4.2.
Figure 3.4.1. Ceiling reflections in auditoriums. a) concave ceiling causing focusing and uneven
sound distribution. b) plane reflectors causing an even sound distribution. (Ref.: Petersen 1984).
96
Figure 3.4.2. Wall reflections in auditoriums. a) rectangular room, b) fan shape room, c) inverse
fan shape room.
3.4.3 Calculation of reverberation time
Sabine’s formula (3.2.13) is the most well known and simple method for calculation of
reverberation time in a room
A
V
A c
V
T
16 . 0 3 . 55
60
≅ = (3.4.1)
with volume V in m
3
and A in m
2
. The equivalent absorption area is calculated as in (3.3.12), but
in addition to absorption from surfaces and air, the absorption from persons or other items in the
room should be included, if relevant
mV A n S A
j
j j
i
i i
4 + + =
∑ ∑
α (3.4.2)
Here n
j
is the number of items, each contributing with an absorption area A
j
. Examples of
absorption coefficients of common materials and absorption areas for persons are given in Table
3.4.1 and 3.4.2, respectively. The air attenuation can be taken from Table 3.4.3.
Frequency (Hz)
Material 125 250 500 1000 2000 4000
Brick, bare concrete
Parquet floor on studs
Needlepunch carpet
Window glass
Curtain draped to half
its area, 100 mm air
space
0.01
0.16
0.03
0.35
0.10
0.02
0.14
0.04
0.25
0.25
0.02
0.11
0.06
0.18
0.55
0.02
0.08
0.10
0.12
0.65
0.03
0.08
0.20
0.07
0.70
0.04
0.07
0.35
0.04
0.70
Table 3.4.1. Typical values of the absorption coefficient α for some common materials.
Frequency (Hz)
Persons 125 250 500 1000 2000 4000
Standing, normal
clothing
Standing, with
overcoat
Sitting musician with
instrument
0.12
0.17
0.60
0.24
0.41
0.95
0.59
0.91
1.06
0.98
1.30
1.08
1.13
1.43
1.08
1.12
1.47
1.08
Table 3.4.2. Typical values of absorption area A in m
2
for persons.
97
Frequency Relative
humidity (%) 1 kHz 2 kHz 4 kHz 8 kHz
40
50
60
70
80
0.0011
0.0010
0.0009
0.0009
0.0008
0.0026
0.0024
0.0023
0.0021
0.0020
0.0072
0.0061
0.0056
0.0053
0.0051
0.0237
0.0192
0.0162
0.0143
0.0133
Table 3.4.3. Examples of air attenuation factor m (m
1
) at a temperature of 20°C.
3.4.4 Reverberation time in nondiffuse rooms
In a room with the sound absorption unequally distributed on the surfaces the assumption of a
diffuse sound field is not fulfilled, and thus Sabine’s formula will not be reliable. The measured
reverberation time may be either shorter or longer than predicted by Sabine’s formula.
A shorter reverberation time will appear in a room in which the first reflections are directed
towards the most absorbing surface. In an auditorium this is typically the floor with the audience,
see Fig. 3.4.1 b.
In a rectangular room without sound scattering surfaces or elements, there is a possibility of
prolonged decay in certain directions. In order to give an idea of the problem it is possible to
calculate the different reverberation times associated to onedimensional decays in each of the
three main directions using the general reverberation formula (3.3.4).
m
m
m
m
l
c
l
T
α α
⋅ ≈
⋅
⋅
≈ 04 . 0
8 . 13
60
(l
m
in m) (3.4.3)
Figure 3.4.3. A rectangular room with indicated absorption coefficients.
As an example the room in Fig. 3.4.3 is considered. The ceiling has a high absorption coefficient
(α = 0.8), but all other surfaces are acoustically hard (α = 0.1).
Volume V = 5 ⋅ 10 ⋅ 20 = 1000 m
3
Surface area S = 700 m
2
Equivalent absorption area A = 200 ⋅ 0.8 + 500 ⋅ 0.1 = 210 m
2
Mean absorption coefficient α
m
= A / S = 210 / 700 = 0.30
Mean absorption coefficient (height) α
m
= (0.8 + 0.1) / 2 = 0.45
Mean free path (3dim.) l
m
= 4 V / S = 4 ⋅ 1000 / 700 = 5.7 m
Mean free path (2dim.) l
m
= π S
x
/ U = π ⋅ 200 / 60 = 10.5 m
98
The results are shown in Table 3.4.4. A twodimensional reverberation in the horizontal plane
between the walls has also been calculated (4.2 s). The onedimensional decays are the extreme
cases with the longest reverberation time being 20 times the shortest one, 8.0 s and 0.4 s,
respectively!
Direction
l
m
(m) α
m
T
60
(s)
3dim. (Sabine)
3dim. (Eyring)
2dim. (horizontal)
1dim. (length)
1dim. (width)
1dim. (height)
5.7
5.7
10.5
20
10
5
0.30
0.30
0.10
0.10
0.10
0.45
0.8
0.6
4.2
8.0
4.0
0.4
Table 3.4.4. Calculation of the onedimensional reverberation times of the rectangular room in
Fig. 3.4.3.
The real decay that is measured in the room will be a mixture of these different decays, and the
reverberation time will be considerably longer than predicted from Sabine’s formula. Eyring’s
formula is even worse. The measured decay curve will be bent, and thus the measuring result
depends on which part of the decay curve is considered for the evaluation of reverberation time.
In a room with long reverberation time due to nondiffuse conditions and at least one sound
absorbing surface, introducing some sound scattering elements in the room can have a significant
effect. It could be furniture or machines on the floor or some diffusers on the walls. This will
make the sound field more diffuse, and the reverberation time will be reduced, i.e. it will come
closer to the Sabine value. In other words: The sound absorption available in the room becomes
more efficient when scattering elements are introduced to the room.
Note. In the onedimensional case it is strictly not correct to use the arithmetic average of the
absorption coefficients, if one of them is high. By inspection of (3.3.1) it is seen that the mean
absorption coefficient should be calculated from
( )( )
2 1
1 1 ) 1 ( α α α − − = −
m
(3.4.4)
So, if one of the surfaces is reflective and the other is totally absorbing, α
m
= 1 and hence the
reverberation time is zero.
3.4.5 Optimum reverberation time and acoustic regulation of rooms
The optimum reverberation time depends of the activities in the room. It is important to choose
the room volume and the surface materials with such sound absorbing properties that the
reverberation time can get the right value for the purpose. In workshops with noise sources it is
important to have a reverberation time as short as possible. In schools the classrooms should have
a reverberation time between 0.6 and 0.9 s and independent of the frequency between 100 and
4000 Hz in order to obtain good acoustical conditions for speech. In concert halls the
reverberation time should be around 1.5 to 2.2 s at mid frequencies (500 – 1000 Hz) with the
longer values in the bigger halls. For music the reverberation time may be up to 50% longer at
low frequencies (125 Hz) and somewhat shorter at high frequencies. The latter is unavoidable in a
big hall due to the air attenuation.
99
Use of room Optimum reverberation time, s
(500 – 1000 Hz)
Cinema
Rock concert
Lecture
Theatre
Opera
Symphony concert
Choir concert
Organ music
0,4 – 1,0
0,8 – 1,1
0,8 – 1,2
1,0 – 1,2
1,3 – 1,7
1,5 – 2,2
1,7 – 2,5
2,0 – 3,0
Table 3.4.5. Optimum reverberation time at mid frequencies for various purposes in rooms with
an audience.
3.4.6 Measurement of reverberation time
The reverberation time in a room can be measured with a noise signal or with an impulse. The
traditional method uses white noise emitted by a loudspeaker and a microphone to measure the
sound pressure level as a function of time after the source is switched off. This gives a decay
curve and a typical example is shown in Fig. 3.4.4.
Figure 3.4.4. Typical decay curve measured with noise interrupted at the time t = 0.
From the microphone the signal is led to a frequency filter, which is either an octave filter of a
onethird octave filter. If the sound in the room is sufficiently diffuse and a sufficient large
number of modes are excited the decay curve is close to a straight line between the excitation
level and the background level. The dynamic range is seldom more than around 50 dB and the
whole range of the measured decay curve is not used. The lower part of the decay curve is
influenced by the background noise and the upper part may be influenced by the direct sound,
which gives a steeper start of the curve. So, the part of the decay curve used for evaluation begins
5 dB below the average stationary level and ends normally 35 dB below the same level. The
evaluation range is thus 30 dB and the slope is determined by fitting a straight line or
100
automatically by calculating the slope of a linear regression line. From the slope of the decay
curve in dB per second is calculated the reverberation time, which is the time for a 60 dB drop
following the straight line. The result is sometimes denoted T
30
in order to make it clear that the
actually used evaluation range is 30 dB.
If the background noise is too high and a sufficient dynamic range is not available the
reverberation time can instead be measured as T
20
. In this case the slope of the decay curve is
evaluated between –5 dB and –25 dB below the excitation level.
The reverberation time is measured in a number of source and receiver positions, and in each
position the decay is determined as an average of a number of excitations. White noise is a
random noise signal and thus the measured decay curves are always a little different.
Sometimes the decay curves are not nice and straight and it is difficult to measure a certain
reverberation time. One reason can be that it is a measurement at low frequencies in a small room
and maybe only two or three modes are excited within the frequency band of the measurement. In
this case there may be interference between the modes causing very irregular decay curves.
Another difficult situation is coupled rooms, i.e. a room divided into sections with different
reverberation times. A typical example is a theatre with a reverberant stage house and a rather
dead auditorium. In this case the decay curve will be bent, i.e. the upper part shows a short
reverberation time and the lower part shows a longer reverberation time. It might be possible to
determine both of these reverberation times, however, the shorter one representing the initial
decay is the most important one, because the subjective evaluation of the reverberation is related
to the initial decay.
3.5 REFERENCES
C.M. Harris (1966). Absorption of sound in air versus humidity and temperature. JASA 40, pp
148159.
C.W. Kosten (1960). The mean free path in room acoustics. Acoustica 10, pp 245250.
J. Petersen 1984). Rumakustik (in Danish). SBIanvisning 137. Danish Building Research
Institute.
W.C. Sabine (1922). Collected papers on acoustics. Dover Publications, Inc. 1964, New York.
101
4 Sound absorbers and their application in room design
Anders Chr. Gade
4.1 Introduction
The reverberation time T
60
as defined in Section 3.2.5 is the most important descriptor of the
acoustics of a room. Therefore, calculating predictions of T
60
(e.g. according to Equation 3.4.1) is a
very basic part of room acoustical design which in turn calls for the availability of reliable data on
the frequency dependant sound absorption characteristics of materials used for room surface
cladding and for furnishing of rooms (such as furniture, people and machinery).
In Table 3.4.1 absorption coefficients per octave band were listed for some materials generally
found in rooms. The values indicate that some of these, e.g. windows and wooden floors on studs,
primarily absorb low frequency sounds. On the other hand, curtains and persons (see Table 3.4.2)
mainly absorb middle and high frequencies. In order to obtain a well balanced T
60
versus frequency
for a given type of room it is therefore important to mix properly different types of materials when
designing the room.
In this chapter we will give a basic introduction to the physical mechanisms involved in sound
absorption and present some types of sound absorption materials well suited for  or specifically
designed for  sound absorption and reverberation control. The absorption properties will be
described in terms of the sound absorption coefficient as defined in Section 1.5.2.
For certain types of rooms, such as schools and work rooms, general demands on reverberation
control exist. Therefore the last section in this chapter is devoted to examples on how sound
absorbing materials can be applied in the design of such rooms.
4.2 The room method for measurement of sound absorption.
In Section 1.5.2, a method for measuring the absorption coefficient, the tube method, was presented
which reveals the absorption coefficient for a single angle of incidence (usually normal incidence as
illustrated to the left in Figure 4.2.1). However, the absorption will normally depend on the
direction of sound incidence
1
. Materials applied in rooms with a (more or less) diffuse sound field
will be exposed to sound arriving from many different directions as illustrated in Fig. 4.2.1(c).
Therefore we will start this chapter by presenting a method for measurement of sound absorption,
which provides the relevant diffuse field absorption coefficient: the reverberation room method.
Figure 4.2.1 Different conditions for sound incidence on a surface. From [1]
1
The absorption for oblique incidence as illustrated in case (b) in Figure 4.2.1 – or as a function of angle of incidence 
can be measured using various techniques using separation in time or subtraction of incident and reflected sound pulses.
However, these techniques are not always very reliable.
103
The measurement takes place in a reverberation room, with highly irregular or non parallel surfaces
and/or suspended, sound diffusing elements. Hereby it can be assumed that the sound field will
fulfil the requirements for application of the Sabine reverberation equation. Assume the room has a
volume V, total surface area S and that α
empty
is the absorption coefficient of the room surfaces
(which ideally should all be made from the same, acoustically hard material). In this case equations
3.4.1 and 3.4.2 (disregarding air absorption) yields:
60,
0.16
empty
Room empty
V
T
S α
= (4.1)
If now we place a test sample of a material with area S
sample
(usually 10 m
2
) in the room, the
equation changes into:
( )
60,
0.16
sample
sample sample Room sample empty
V
T
S S S α α
=
+ −
(4.2)
in which we have considered that an area, S
sample
, of the room surface has now been covered by the
sample. Combining equations 4.1 and 4.2 by eliminating S yields for the unknown absorption
coefficient, α
sample
, of the test sample:
60, 60,
0,16 1 1
sample empty
sample sample empty
V
S T T
α α
⎡ ⎤
= −
⎢ ⎥
⎢ ⎥
⎣ ⎦
+ (4.3)
The measurement is normally carried out in 1/1 or 1/3 octave bands from 100 to 5000 Hz.
If absorption measurements using the room method is carried out on small sized samples, these
sometimes appear to have a absorption coefficient larger than 1.0, as seen in Figure 4.2.2. Of course
this is not logical, if the absorption power should be related
solely to the physical area of the sample. The phenomenon
is probably due to diffraction of sound around the edges of
the sample, which dominates the behaviour in cases where
the linear dimension of the sample approaches the wave
length of the sound, i.e. the effect is more pronounced at low
frequencies.
Although a complication in documentation of absorption
properties, this phenomenon can be applied successfully in
Fig. 4.2.2 Absorption coefficients practice by providing increased absorption effect, if the
of different materials versus area available absorption material can be provided in smaller
(measured in square feet). From [1]. pieces and spread out over the room surfaces.
4.3 Different types of sound absorbers
In this section the three most common types of sound
absorbing constructions will be described, each with its
own characteristic frequency dependency of the
absorption coefficient as sketched in Fig. 4.3.1.
Fig. 4.3.1 Typical behaviour of absorption versus
frequency for Porous, resonating and membrane
absorbers respectively.
104
4.3.1 Porous absorbers
Porous absorbers are present in rooms in the form of textiles like curtains, carpets and furniture
upholstery, porous mortar in (unpainted !) brick walls and not least as a wide variety of dedicated
sound absorbing products for suspended ceilings.
Figure 4.3.2 Left: Standing wave pattern formed by an incident and a reflected sound wave in front
of a porous material of a certain thickness flush mounted on a heavy and hard surface. Right:
Absorption versus frequency of a thin, porous sheet placed in front of a hard surface. From [1].
Porous materials are characterized by having an open structure of e.g. of fibres glued or woven
together which is accessible by the air. Thus, air can be pressed through the material more or less
easily depending on the flow resistance (determined e.g. by how densely a fabric is woven – try for
yourself by blowing through clothing or curtains !). The absorption properties are caused by viscous
friction between the moving air molecules in the sound waves and the often huge internal surface
area of the structure whereby the (kinetic) sound energy is converted into heat.
Fig. 4.3.3 Absorption coefficients for mineral
wool (glasswool) with thickness as parameter
(a) and with wall distance as parameter (b).
f the
r
, for a
tion
If a porous sheet of a certain thickness is placed
flush on a rigid surface and hit by an normal
incidence sound wave a standing wave pattern
will be created with pressure amplitude as
indicated to the left in Fig. 4.3.2. As seen from
Figure 1.2.10 (c), with a rigid termination, the
particle velocity and so the kinetic energy o
sound field will be high where the pressure
amplitude (the potential energy) is low. In othe
words, for the absorber to be efficient (with
normal incidence of the sound wave), the
thickness of the porous layer need to be at least
λ/4, so that friction takes place where the
energy is primarily kinetic. In other words
given thickness of the material, there is a lower
limiting frequency below which the absorp
drops off because the material can no longer
“reach” the region of high kinetic energy. On
the other hand, as the absorber is not absorbing
the potential energy anyway, one can save
material and just place a thin sheet (but still
105
with a suitable flow resistance) at a certain dista from the rigid wall (like a curtain in front of
window). In the case of normal incidence, a
nce a
pplying a thin sheet will cause the absorption to drop
mineral wool mats of
ifferent thickness (upper graph) and different distances to the rigid wall (lower graph). It is seen
m melted
lass (Glasswool) or stone (Rockwool) much like “Candy Floss”. Mineral wool is used as porous
,
ete
l
rane absorbers
membrane absorber is characterized by consisting of a non porous sheet or panel placed at a
backing whereby an air filled cavity is formed. This system can
on
again at a higher frequency where the distance between sheet and hard wall equals λ/2; but with
diffuse field incidence this dip will not be very pronounced. Diffuse field incidence also causes the
absorbers to be effective (α > 0,8) if just the thickness/distance is > λ/8.
Fig. 4.3.3 shows how the absorption coefficient varies with frequency for
d
that more low frequencies are absorbed as the thickness or the wall distance increases.
Mineral wool consists of thin fibres pressed and glued together. The fibres are made fro
g
sound absorbers, very often in the form of tiles which can be mounted in a suspended ceiling
system. Such ceilings will often be placed below ventilation ducts and other technical installations
whereby a large distance (typically between 20 cm and one metre) is ensured to the hard concr
deck behind. Hereby the ceiling can absorb efficiently over a wide frequency range – as well as hide
the installations. Mineral wool ceiling tiles are normally given a carefully controlled layer of specia
paint from the factory to make them look like normal (white) plaster ceilings as much as possible.
However, if one tried to repaint them, the porous properties and so the absorption normally
disappears.
4.3.2 Memb
A
certain distance from a hard
resonate at frequency determined by the mass per unit area of the plate, m, and the spring functi
of the enclosed air, which is determined by the depth, L, of the cavity:
2
0
1
2
c
f
mL
ρ
π
= (4.4)
owever, Equation 4.4 only apply if the plate is complete
frequency is also determined by the plate stiffness and mode of plate vibration, of which a few are
ional
H ly limp. Normally, the resonance
illustrated in Fig. 4.3.4, with p and q being integers determining the shape of the two dimens
oscillation pattern of the plate.
Fig. 4.3.4 Different modes of vibration in a stiff plate.
follows: In this case the resonance frequency can be described as
( )
2
2 2
2 4
1
r
c p q
f
ρ π
3
2
2 12 1
Eh
mL m a b π ν
⎡ ⎤
⎛ ⎞ ⎛ ⎞
= + +
⎢ ⎥
− ⎝ ⎠ ⎝ ⎠
⎢ ⎥
⎣ ⎦
(4.5)
which a and b are the dimensions of the plate (or the distance between
plate), h is the thickness while E and ν are the Young’s modulus and the Poisson ratio respectively.
⎜ ⎟ ⎜ ⎟
in studs supporting the
106
From this formula it is seen that a resonance frequency is determined completely by the stiffnes
the depth of the cavity is infinitely deep – as is the case e.g. with a single pane window.
s if
Fig. 4.3.5 absorption versus frequency of
membrane absorber for two different plate
ood placed
ency
ms in the for
he effect is a controlled low frequency T
60
value as opposed to rooms
thicknesses and with and without mineral
wool in the cavity. From [1]
Membrane absorbers are often found in roo
oard or wood panel walls. T
Fig. 4.3.5 show absorption versus frequency
for two different thickness of plyw
45 mm from a hard backing – with and
without mineral wool in the cavity. As
expected it is seen that the thicker and heavier
plate result in the lowest resonance frequ
as expected from equations 4.4 and 4.5.
Besides, it is observed that the mineral wool
inlay, which increases the internal damping of
the construction causes a significant
improvement in the absorption around the
resonance frequency and also causes the
resonance frequency to become lower.
m of wooden floors on joists or as gypsum
b
made entirely from heavy concrete or masonry which causes the sound to be “dark” and blurred at
low frequencies.
Fig. 4.3.6 Example of membrane absorbers attached to the concrete side wall in the multi purpose
all (Kolding Teater). Besides controlling low frequency reverberation, the panels also provide
ome diffusion of the sound.
h
s
107
4.3.3 Resonator absorbers
In stead of having a plate forming the mass of the resonating system, the mass can be oscillating air
an opening between a closed cavity and the open atmosphere. Also in this case, the enclosed air
and resonating panel (right). From [1
in the cavity provides the spring function. An example of such a single resonator, called a Helmholz
n be experienced by
lowing across the opening of a bottle) is given by:
in
Fig. 4.3.7 Single resonator (left) ].
resonator, is illustrated in Figure 4.3.7. The resonance frequency (which ca
b
( )
0
2
c S
f
V l π δ
=
+
(
S V
4.6)
with being the area of the opening, being the enclosed volume, l the length of the neck and δ a
orrection to the neck length which is due to the fact that th
with very high velocity  is not confined to the physical length of the neck; but some of the air
owever, if a perforated panel is placed in front of a cavity as seen to the right in Fig. 4.3.7, then
c e oscillating air mass  often moving
outside both ends of the neck will be moving as well.
Resonators like the build in “bottle” in the left side of Fig. 4.3.7 are not very practical, as the
frequency range of the absorption is normally very limited around the sharp resonance frequency.
H
this construction can be regarded as a large number of single resonators put together, and the
physical proportions in this case often causes a much more useful frequency range of absorption.
For the resonance frequency of the panel we have:
( )
0
2
c P
f
L l π δ
=
+
(4.
which is almost identical with Equation 4.6 except for the opening area being rep
7)
laced by the
egree of perforation, P, of the panel and the volume V bein
If the holes are circular with diameter d, we have for the end correction: δ ≈ 0.8 d. Resonating
y
ised by placing a thin layer of mineral wool or glass felt
alled vlies) in the cavity. Like in the case of the membrane absorber, it is important to adjust the
d g changed into the depth of the cavity L.
panels will often have a higher resonance frequency and absorb efficiently in a wider frequenc
range than the membrane absorbers.
Regarding damping, the viscous damping can be significant if the hole/slit dimensions are small;
but often the absorption can be optim
(c
damping to achieve optimal absorption.
108
Perforated panels are found in the form of perforated gypsum board or steel plates (used e.g. for
suspended ceilings)
2
, or as panels made of wooden boards with slits between the individual boards
s illustrated to the left in Figure 4.3.8. Other possibilities are walls made from perforated tiles,
e
n
ontrolled gaps in front of a former window niche filled with mineral wool. The panel controls low
equency reverberation in a former power plant building made from heavy masonry converted into
multi
room acoustic design
he main purpose of introducing absorption for reverberation control in rooms is to reduce noise
Danish Building Law
ygningsreglementet af 1995, BR95) [2] contains demands on maximum T
60
values in school class
re
a
which make use of the cavity already present in a double masonry wall as shown to the right in th
same Figure.
Fig. 4.3.8 Resonating panel constructions i practice. Left: Wooden boards separated by
c
fr
a concert hall (Værket, Randers). Right: Perforated bricks on the rear wall in a sports and
purpose hall. By making this wall absorbing, echoes back to the stage placed more than 50 m away
are avoided (Frihedshallen, Sønderborg).
4.4 Application of sound absorbers in
T
levels (see Fig. 3.2.4) and in some cases to increase intelligibility. The
(B
rooms, day care institutions and apartment buildings, whereas the Danish Working Environment
Agency have issued rules for industrial buildings and offices [3]. These current Danish rules a
2
It should be added that in many cases with perforated gypsum or steel plates used as suspended ceilings, the
combinations of perforation and cavity depth causes the absorber to act more like a porous absorber but with reduced
performance at high frequencies due to the panel shielding off the porous layer to some degree.
109
briefly listed in Figure 4.4.1. Recommendable values for other types of rooms – including auditoria
and concert halls were listed in Table 3.4.5. Special standards exist for design of cinemas and stud
control rooms and listening rooms. In Denmark, no rules exist for other public spaces like traffic
terminals, sports arenas and restaurants  although the acoustic conditions in these places are often
horrible. However, acoustic concerns a generally included in modern design of these spaces as well.
io
ximum values of reverberation time in buildings.
corridors in office buildings just reflect common
s
ecified in terms of a required minimum absorption area. The reason for this is that
ften calculation as well as measurement of T
60
is often questionable in these rooms.
ause here the
ften delicate absorption materials are not subject to mechanical damage.
eiling of mineral wool tiles with integrated light fixtures. Right: Vertical Mineral wool baffles.
Fig. 4.4.1 Listing of Danish rules regarding ma
(The values listed for single person offices and
design practice.)
As indicated in Figure 4.4.1 the rules for large industrial halls as well as open plan areas in office
and schools are sp
o
In most cases the ceiling is the most obvious surface to treat with absorption, as it constitutes a large
area which is normally available apart from a few light or ventilation fixtures and bec
o
Fig. 4.4.2 Examples of acoustic treatment mounted in ceiling in industrial halls. Left: suspended
c
110
In Figure 4.4.2 are shown two examples of acoustic treatment of ceilings. To the left a normal
suspended ceiling of mineral wool tiles with integrating lighting and ventilation. This type of
ceiling is often found in offices, schools, shops etc.. The vertical mineral wool baffles shown to the
ot
lways sufficient to place the absorption in the ceiling surface alone; but also available wall areas
right can be a solution when the ceiling is already heavily occupied by technical installations.
In rooms where practically all the absorption is placed in the ceiling, the reverberation time
basically becomes a function of the room height as shown in Figure 4.4.3. In high rooms, it is n
a
must be used as illustrated by the mineral wool tiles to the right in Figure 4.4.3.
ig. 4.4.3 Simplified calculation of T
60
in room with all ab
urface(left) shows the need for additional absorption on walls in tall rooms (right).
coustic
nsure proper
telligibility of speech (often emitted through loudspeakers). In Figure 4.4.4 is illustrated how a
Fig. 4.4.4 Schematic illustration of the influence of reverberation on the intelligibility of speech.
F sorption placed on the ceiling
s
In many public places like traffic terminals, department stores, sports halls etc., the room a
absorption treatment is not only done with the purpose of reducing noise but also to e
in
long room decay can cover (mask) the weak phonems illustrated schematically as vertical bars. In
speech the consonant sounds are often the weaker elements; but they contain most of the
information. Therefore, a long reverberation can seriously deteriorate intelligibility.
111
In room acoustic design not
only consists of reverberation con surfaces. In these rooms
also the design of the room the
surfaces. And
in order to support intelligi s) after the direct
sound.
Even in norma pplied by
leaving a central part of the ce rface areas can be found to
provide the required reverberation
ODEON programm e part of the
terms of the
Fig. 4.4.5 Illustrations from the room acoustic simulation programme ODEON of a class room
design with a partly absorbing (dark) and reflective (lighter grey) ceiling.
References
[1] Z. Maekawa and P. Lord: Environmental and Architectural Acoustics. E & FN Spon,
London, 1994.
[2] Bygningsreglementet; Bygge og Boligstyrelsen 1995. Publications from the Danish
National Agency for Enterprise and Construction (in Danish) can be found at:
http://www.ebst.dk/pub_lydforhold/0/8/0
[3] AT anvisning nr.1.1.0.1, november 1995; Akustik i arbejdsrum
(Acoustics in work places) http://www.at.dk/sw5110.asp
s dedicated for speech like auditoria, class rooms and theatres, the room
trol by absorption treating of the room
geometry is important to ensure proper propagation of sound from
source to the listeners through reflection of the sound waves off non absorbing room
bility, these reflections must arrive not long (up to 40 m
l sized class rooms this concern about supporting reflections may be a
iling reflective (given that enough other su
control). Thus, Fig. 4.4.5 illustrates such a case in which the
e was used to balance the application of absorbing and reflectiv
ceiling for a school project and to predict reverberation time and the intelligibility in
Speech Transmission Index mentioned in Section 2.7.
112
5. An introduction to sound insulation
Jens Holger Rindel
5.1 THE SOUND TRANSMISSION LOSS
5.1.1 Definition
A sound wave incident on a wall or any other surface separating two adjacent rooms partly
reflects back to the source room, partly dissipates as heat within the material of the wall, partly
propagates to other connecting structures, and partly transmits into the receiving room.
The power incident on the wall is P
1
and the power transmitted into the receiving room is P
2
.
The sound transmission coefficient τ is defined as the ratio of transmitted to incident sound
power
1
2
P
P
= τ (5.1.1)
However, the sound transmission coefficients are typically very small numbers, and it is more
convenient to use the sound transmission loss R with the unit deciBel (dB). It is defined as
(dB) log 10
1
log 10 log 10
2
1
τ
τ
− = = =
P
P
R (5.1.2)
Another name for the same term is the sound reduction index.
5.1.2 Sound insulation between two rooms
Figure 5.1.1. Airborne sound transmission from source room (1) to receiving room (2)
The most common case is the sound insulation between two rooms. With the assumption of
diffuse sound fields in both rooms it is possible to derive a simple relation between the
transmission loss and the sound pressure levels in the two rooms. The rooms are called the
source room and the receiving room, respectively. In the first room is a sound source that
generates the average sound pressure p
1
. The sound power incident on the wall is, see eq.
(3.2.6)
113
c
S p
S I P
inc
ρ 4
2
1
1
= = (5.1.3)
The area of the wall is S. In the receiving room the average sound pressure p
2
is generated from
the sound power P
2
radiated into the room, see eq. (3.2.8)
c
A
P
p ρ
2
2 2
2
4
= (5.1.4)
Here A
2
denotes the absorption area in the receiving room. Insertion in the definition (5.1.2)
gives
(dB) log 10 log 10
2
2 1
2
2
2
2
1
A
S
L L
A p
S p
R + − = = (5.1.5)
Here L
1
and L
2
are the sound pressure levels in the source and receiving room, respectively.
This important result is the basis for transmission loss measurements.
5.1.3 Measurement of sound insulation
Sound insulation is measured in onethird octave bands covering the frequency range from 100
Hz to 3150 Hz. In recent years the international standards for measurement of sound insulation
have been revised and it is recommended to extend the frequency range down to 50 Hz and up
to 5000 Hz. One reason for this is that the low frequencies 50 – 100 Hz are very important for
the subjective evaluation of the sound insulation properties of lightweight constructions. In
recent years lightweight constructions have been more commonly used in new building
technology, whereas heavy constructions have traditionally been used for sound insulation.
The sound pressure levels are measured as the average of a number of microphone positions or
as the average from microphones slowly moving on a circular path. The results are averaged
over two different source positions. More details are given in ISO 140 Part 3 and 4.
In addition to the two sound pressure levels it is also necessary to measure the reverberation
time in the receiving room in order to calculate the absorption area. Sabine’s equation is used
for this, see eq. (3.2.13)
2
2
2
3 . 55
T c
V
A = (5.1.6)
Only under special laboratory conditions it is possible to measure the transmission loss of a
wall without influence from other transmission paths. In a normal building the sound will not
only be transmitted through the separating construction, but the flanking constructions will also
influence the result, see later in section 5.5.4.
For measurements of sound insulation in buildings the apparent sound transmission loss is
(dB) log 10
2
2 1
A
S
L L R + − = ′ (5.1.7)
The apostrophe after the symbol indicates that flanking transmission can be assumed to
influence the result.
5.1.4 Multielement partitions and apertures
A partition is often divided into elements with different sound insulation properties, e.g. a wall
with a door. Each element is described by the area S
i
and the transmission coefficient τ
i
. If the
sound intensity incident on the surfaces of the source room is denoted I
inc
the total incident
sound power on the partition is
114
∑
=
= =
n
i
inc inc i
I S I S P
1
1
The total area is called S. The total sound power transmitted through the partition is
∑
=
=
n
i
inc i i
I S P
1
2
τ
Thus, the transmission coefficient of the partition is
∑ = =
=
n
i
i i res
S
S P
P
1 1
2
1
τ τ (5.1.8)
The same result can also be written in terms of the transmission losses R
i
of each element
⎟
⎠
⎞
⎜
⎝
⎛
− = − =
∑
=
−
n
i
R
i res res
i
S
S
R
1
1 , 0
10
1
log 10 log 10 τ (5.1.9)
In the simple case on only two elements the graph in Fig. 5.1.2 may be used.
Figure 5.1.2. Graph for estimating the transmission loss of a multielement partition
An aperture in a wall is a special example of an element with different transmission properties.
As an approximation it can be assumed that the transmission coefficient of the aperture is 1. If
also the area of the aperture S
ap
is very small compared to the total area, this leads to the
following result for the resulting transmission loss of the wall with aperture:
( )
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+ − ≅
⎟
⎠
⎞
⎜
⎝
⎛
+ − =
− −
S
S
S S
S
R
ap
R
ap
R
res
1 1
1 , 0 1 , 0
1
10 log 10 10
1
log 10 (5.1.10)
Fig. 5.1.3 can illustrate the result. It is seen that the relative area of the aperture defines an
upper limit of the sound insulation that can be achieved.
115
Figure 5.1.3. Graph for estimating the transmission loss of a construction with an aperture
116
5.2 SINGLE LEAF CONSTRUCTIONS
5.2.1 Sound transmission through a solid material
The solid material is supposed to have the shape of a large plate with thickness h. The material
is characterised by the density ρ
m
and the speed of longitudinal waves c
L
. The surface of the
material defines two transition planes where the sound waves change from one medium to
another. It is assumed that the medium on either side is air with the density ρ and the speed of
sound c (also longitudinal waves). The symbols and notation are explained in Fig. 5.2.1.
h
p
i
p
1
p
2
p
t
p
r
p
4
p
3
Figure 5.2.1. Thick wall with incident, reflected and transmitted sound waves
The sound pressure is equal on either side of the two transition planes:
(5.2.1)
3 2
4 1
p p p
p p p p
t
r i
+ =
+ = +
Also the particle velocity is equal on either side of the two transition planes:
3 2
4 1
u u u
u u u u
t
r i
− =
− = −
(5.2.2)
The characteristic impedance in the surrounding medium (air) is denoted Z
0
and that in solid
material is denoted Z
m
. Thus the ratio of sound pressure to particle velocity in each of the plane
propagating waves is:
L m m
t
t
r
r
i
i
c Z
u
p
u
p
u
p
u
p
c Z
u
p
u
p
u
p
ρ
ρ
= = = = =
= = = =
4
4
3
3
2
2
1
1
0
(5.2.3)
Using (5.2.3) in (5.2.2) leads to:
( )
( )
3 2
0
4 1
0
p p
Z
Z
p
p p
Z
Z
p p
m
t
m
r i
− =
− = −
(5.2.4)
Assuming propagation from one side of the material to the other without losses means that there
is only a phase difference between the pressure at the two intersections:
117
(5.2.5)
h k
h k
m
m
p p
p p
j
3 4
j
1 2
e
e
−
−
=
=
Here k
m
= ω /c
L
is the angular wave number for longitudinal sound propagation in the solid
material.
From the above equations (5.2.1), (5.2.4) and (5.2.5) can be derived the ratio between the sound
pressures p
i
and p
t
and thus the transmission loss can be expressed by:
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+ + = = ) ( sin
4
1
) ( cos log 10 log 10
2
2
0
0 2
2
0
h k
Z
Z
Z
Z
h k
p
p
R
m
m
m
m
t
i
(5.2.6)
R
0
, dB
Fig. 5.2.2. Transmission loss at normal incidence of sound on a 600 mm thick concrete wall.
At high frequencies some dips can be observed in the transmission loss curve. They occur at
frequencies where the thickness is equal to half a wavelength in the solid material, or a multiple
of half wavelengths. However, the dips are very narrow and they are mainly of theoretical
interest.
Two special cases can be studied. First the case of a thin wall: Z
m
>> Z
0
and k
m
h << 1
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+ ≅
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+ ≅
2
2
2
0
0
2
1 log 10 ) ( sin
2
1 log 10
c
h
h k
Z
Z
R
m
m
m
ρ
ρ ω
(5.2.7)
The other special case is a very thick wall: Z
m
>> Z
0
and k
m
h >> 1
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
≅
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
≅
c
c
Z
Z
R
L m m
ρ
ρ
2
log 20
2
log 10
2
0
0
(5.2.8)
118
The crossover frequency from (5.2.7) to (5.2.8) is the frequency f
h
at which k
m
h = 1:
h
c
f
L
h
π 2
= (5.2.9)
This is the frequency at which the thickness is approximately one sixth of the longitudinal
wavelength λ
L
in the material:
π
λ
π 2 2
L L
f
c
h = =
The result for the thin wall is the socalled mass law, which will de derived in a different way in
the next section. The result for a very thick wall (5.2.8) means that there is an upper limit on the
sound insulation that can be achieved by a singleleaf construction, and this limit depends on
the density of the material. For wood it is 68 dB, for concrete 80 dB and for steel 94 dB. (These
numbers should be reduced by 5 dB in the case of random incidence instead of normal
incidence, see section 5.2.3).
5.2.2 The mass law
p
i
p
r
p
t
v
n
v
t
= v
n
/ cos θ
θ
Figure 5.2.3. Thin wall with sound pressures and particle velocities
A thin wall with the mass per unit area m is considered, see Fig. 5.2.3. The application of
Newton’s second law (force = mass ⋅ acceleration) gives:
n
n
t r i
v m
t
v
m p p p p ω j
d
d
= = − + = Δ (5.2.10)
where v
n
is the velocity of the wall vibrations (in the direction normal to the wall). The
separation impedance Z
w
is introduced:
m
v
p
Z
n
w
ω j =
Δ
= (5.2.11)
The separation impedance will be more complicated if the bending stiffness of the wall is also
taken into account, see below.
The particle velocities in the sound waves are called u with the same indices as the
corresponding sound pressures. Due to the continuity requirement the normal component of the
velocity on both sides of the wall is:
θ θ cos ) ( cos
r i t n
u u u v − = = (5.2.12)
which leads to
119
θ cos
0
n
r i t
v
Z p p p = − = (5.2.13)
The sound transmission loss R
θ
at a certain angle of incidence θ is:
(dB)
2
cos
1 log 10 log 10
2
0
2
Z
Z
p
p
R
w
t
i
θ
θ
+ = = (5.2.14)
In the special case of normal sound incidence (θ = 0) the insertion of (5.2.11) gives the
important mass law of sound insulation:
(dB) log 20
2
j 1 log 10
2
0
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
≅ + =
c
m f
c
m
R
ρ
π
ρ
ω
(5.2.15)
Since m = ρ
m
h this result is the same as derived above in (5.2.7).
5.2.3 Sound insulation at random incidence
The transmission coefficient at the angle of incidence θ is from (5.2.14)
θ
ρ
ω
θ τ
2
2
cos
2
1
1
) (
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+
=
c
m
(5.2.16)
Random incidence means that the sound field on the source side of the partition is
approximately a diffuse sound field. In a diffuse sound field the incident sound power P
1
on a
surface is found by integration over the solid angle ψ = 2 π assuming the same sound intensity
I
1
in all directions. The principle is the same as used in section 3.2.2. Since, in each direction
the transmitted sound power is equal to the incident sound power multiplied by the transmission
coefficient, the ratio between transmitted and incident power is:
∫
∫
∫
∫
= = =
=
=
2 /
0
2 /
0
2
1
2
1
1
2
d sin cos
d sin cos ) (
d
d ) (
π
π
π ψ
π ψ
θ θ θ
θ θ θ θ τ
ψ
ψ θ τ
τ
S I
S I
P
P
∫ ∫
= =
1
0
2
2 /
0
) d(cos ) ( d sin cos ) ( 2 θ θ τ θ θ θ θ τ τ
π
( )
( ) ( )
2
2
1
0
2 2
2
2 1 ln
2
cos 2 1
) d(cos
c m
m
c
c m
ρ ω
ω
ρ
θ ρ ω
θ
τ +
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
=
+
=
∫
( ) (dB) 23 , 0 log 10 log 10
0 0
R R R − = − = τ (5.2.17)
This is the theoretical result for random incidence, and for typical values (R
0
between 30 and 60
dB) it means that R is 8 to 11 dB lower than R
0
. However, in real life this is not true and it can
be shown that the result is related to partitions of infinite size. Taking the finite size into
account the result is approximately:
(5.2.18) dB 5
0
− ≅ R R
This is in good agreement with measuring results on real walls.
120
5.2.4 The critical frequency
The bending stiffness per unit length of a plate with thickness h is:
) 1 ( 12
2
3
ν −
=
h E
B (5.2.19)
where E is Young’s modulus of the material and ν is Poisson’s ratio. (ν ≅ 0.3 for most rigid
materials).
The speed of propagation of bending waves in a plate with bending stiffness per unit width B
and mass per unit area m is (see section 6.3.3):
c
b
f
f
c
m
B
c = =
4
ω (5.2.20)
Here f
c
is introduced as the critical frequency. It is defined as the frequency at which the speed
of bending waves equals the speed of sound in air, c
b
= c.
The critical frequency is:
B
m c
f
c
π 2
2
= (5.2.21)
A sound wave with the angle of incidence θ propagates across the wall with the phase speed
c / sin θ , i.e. the phase speed is in general higher than c, see Fig. 5.2.4. If the bending wave
speed happens to be equal to the phase speed of the incident sound wave, this is called
coincidence:
θ sin / c c
b
=
Figure 5.2.4. Thin wall with bending wave and indication of speed of propagation along the
wall
The coincidence leads to a significant dip in the sound transmission loss. The coincidence dip
will be at a frequency higher than or equal to the critical frequency:
121
θ
2
sin
c co
f f = (5.2.22)
The separation impedance (5.2.11) is replaced by:
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
− = θ ω
4
2
sin 1 j
c
w
f
f
m Z (5.2.23)
Insertion in the general equation (5.2.14) leads to the sound transmission loss at a certain angle
of incidence:
( ) (dB) sin 1 log 20 cos log 20
4 2
0
θ θ
θ c
f f R R − + + = (5.2.24)
5.2.5 A general model of sound insulation of single constructions
The general model of sound insulation is based on mass law as given in (5.2.15). However, the
following results are valid for sound insulation between rooms with approximately diffuse
sound fields. In the frequency range below the critical frequency, f < f
c
:
( ) dB 5 1 log 20
2
0
− − + ≅
c
f f R R (5.2.25)
In the frequency range above the critical frequency, f ≥ f
c
:
(dB)
2
log 10
0
c
f
f
R R
π
η
+ ≅ (5.2.26)
where η is the loss factor (see section 6.2.2.3).
The upper limit for sound insulation of a singleleaf construction is, according to (5.2.8):
dB 5
2
log 20 −
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
≤
c
c
R
L m
ρ
ρ
(5.2.27)
A sketch of the transmission loss as a function of frequency is shown in Fig. 5.2.5
R, dB
f
c
Frequency (log)
Figure 5.2.5. Sound insulation of a singleleaf construction, f
c
is the critical frequency and the
upper limit is the dotted line.
122
5.3 DOUBLE LEAF CONSTRUCTIONS
5.3.1 Sound transmission through a double construction
m
1
d m
2
p
i
p
1
p
2
p
t
p
r
p
4
p
3
v
1
v
2
Fig. 5.3.1. A double construction with indication of sound pressures and particle velocities
A double construction with two plates in the distance d is considered, see Fig. 5.3.1. The
separation impedance of the two plates is denoted Z
1
and Z
2
, respectively. As for the single
construction in (5.2.10) the movement of each wall is:
(5.3.1)
2 2 3 2
1 1 4 1
) (
v Z p p p
v Z p p p p
t
r i
= − +
= − − +
The velocity of each wall equals the particle velocity on either side:
t t
r i r i
p
Z
u v
p p
Z
u u v
p p
Z
u u v
p p
Z
u u v
0
2
3 2
0
3 2 2
4 1
0
4 1 1
0
1
1
) (
1
) (
1
) (
1
= =
− = − =
− = − =
− = − =
(5.3.2)
Assuming propagation from one side of the cavity to the other without losses means that there
is only a phase difference between the pressure at the two intersections:
(5.3.3)
kd
d k
p p
p p
j
3 4
j
1 2
e
e
−
−
=
=
From the above equations (5.3.1), (5.3.2) and (5.3.3) can be derived the ratio between the sound
pressures p
i
and p
t
and thus the transmission loss can be expressed by:
123
2
2
0
2 1
0
2 1
0
2 1
2
0
) sin(
2 2
1 j ) cos(
2
1 log 10
log 10
kd
Z
Z Z
Z
Z Z
kd
Z
Z Z
p
p
R
t
i
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+
+
+ +
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛ +
+ =
=
(5.3.4)
If only the mass of each wall is taken into account the separation impedances are:
2 2
1 1
j
j
m Z
m Z
ω
ω
=
=
(5.3.5)
Neglecting the smaller parts and inserting Z
0
= ρ c together with (5.3.5) yields:
⎥
⎥
⎦
⎤
⎢
⎢
⎣
⎡
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
−
+
+
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛ +
≅
2
2
2 1
2
2 1
2
2 1
0
) sin(
) ( 2
) cos(
2
) (
) sin(
2
) (
log 10 kd
c
m m
kd
c
m m
kd
c
m m
R
ρ
ω
ρ
ω
ρ
ω
(5.3.6)
This result will be discussed and simplified below.
5.3.2 The massairmass resonance frequency
The transmission loss is minimum when the last term is zero, i.e.
ω
ρc
m m
m m
kd
2 1
2 1
) tg(
+
= (5.3.7)
For a cavity that is narrow compared to the wave length (kd << 1) we get:
ω
ρ ω c
m m
m m
c
d
kd kd
2 1
2 1
) tg(
+
= = ≈
The solution is the massairmass resonance frequency f
0
= ω
0
/ 2 π
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+ =
2 1
0
1 1
2 m m d
c
f
ρ
π
(5.3.8)
If the depth d of the cavity is comparable to the wavelength there are many solutions to (5.3.7)
and they are approximately kd = n π. The dips in the sound insulation occur at frequencies at
which the cavity depth equals one or more half wavelenghts: d = n λ /2.
However, more important than these dips is the shift from low to highfrequency behaviour of
the air cavity. The crossover frequency has no particular physical meaning, but it is the
frequency f
d
at which kd = 1:
d
c
f
d
π 2
= (5.3.9)
This is quite similar to the result (5.2.9) found for the sound transmission through a solid
material. Only, in this case the transmission is through air. The springlike behaviour of the air
cavity changes from that of a simple spring below the crossover frequency to that of a
transmission channel at higher frequencies.
5.3.3 A general model of sound insulation of double constructions
The result (5.3.6) can be simplified in different way depending on the frequency range. In the
frequency range below the resonance frequency, f < f
0
:
) 2 1 (
2 1
0
2
) (
log 20
+
=
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛ +
≈ R
c
m m
R
ρ
ω
(5.3.10)
124
This means that the construction behaves as a single construction with the mass per unit area
(m
1
+ m
2
). In the frequency range above the resonance frequency, f
0
< f < f
d
:
) 2 log( 20
2
log 20
2 1
3 2
2 1
3
0
kd R R
c
d m m
R + + ≈
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
≈
ρ
ω
(5.3.11)
In this a much better sound insulation can be obtained, and it depends on the product of the
three parameters m
1
, m
2
and d. At frequencies above f
d
where the cavity is wide compared to the
wavelength, sin (kd) is replaced by its maximum value 1, and for f ≥ f
d
:
dB 6
) ( 2
log 20
2 1
2
2 1
2
0
+ + ≈
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
≈ R R
c
m m
R
ρ
ω
(5.3.12)
In this highfrequency range, d is no longer an important parameter.
A sketch of the transmission loss as a function of frequency is shown in Fig. 5.3.2.
f
0
f
d
Frequency (log)
R, dB
Figure 5.3.2. Sound insulation of a doubleleaf construction, f
0
is the resonance frequency and
f
d
is the crossover frequency of the cavity.
R, dB
f
0
f
d
f
c1
f
c2
Frequency (log)
Figure 5.3.3. Sound insulation of an asymmetric doubleleaf construction with two thin plates
having different critical frequencies, f
c1
and f
c2
, respectively.
125
5.4 FLANKING TRANSMISSION
Fig. 5.4.1. Direct transmission and three flanking transmission paths via the floor.
The transmission of sound from a source room to a receiver room can be via flanking
constructions like the floor, the ceiling or the façade. When all relevant transmission paths are
considered the sound insulation is described by the apparent sound transmission loss:
(dB) log 10 log 10
2
2 1
3 2
1
A
S
L L
P P
P
R + − =
+
= ′ (5.4.1)
where P
2
is the sound power transmitted through the partition wall to the receiver room and P
3
is the sound power radiated to the receiver room from the flanking surfaces and other flanking
paths:
(5.4.2)
∑
=
i
i F
P P
, 3
Each single flanking transmission path i can be characterised by the flanking transmission loss,
R
F,i
:
(dB) log 10
,
1
,
i F
i F
P
P
R = (5.4.3)
It is convenient to keep the incident sound power P1 on the partition wall as a reference for all
the flanking transmission losses. In this way it is very simple to add all the contributions
together, and the apparent transmission loss is calculated from:
(5.4.4) (dB) 10 10 log 10
,
1 , 0 1 , 0
⎟
⎠
⎞
⎜
⎝
⎛
+ − = ′
∑
− −
i
R R i F
R
In the typical case of horizontal transmission through a wall the will be 12 flanking paths,
namely three possible paths for each of the four surrounding flanking constructions, see Fig.
5.4.1.
126
5.5 ENCLOSURES
A noise source is supposed to radiate the sound power P
a
. The noise source is totally covered
by an enclosure with surface area S, absorption coefficient α on the inside, and the enclosure is
made from a plate with transmission loss R or transmission coefficient τ. The average sound
pressure in the enclosure p
encl
can be estimated, if a diffuse sound field is assumed:
c
S
P
p
a
encl
ρ
α
4
2
= (5.5.1)
The sound power incident on the inner surface of the enclosure is (still with the assumption of a
diffuse sound field):
c
S p
P
encl
inc
ρ 4
2
= (5.5.2)
The sound power transmitted through the enclosure is then:
a inc out
P P P
α
τ
τ = = (5.5.3)
The insertion loss of the enclosure is the difference in radiated sound power level without and
with the enclosure:
(dB) log 10 log 10 log 10 α
τ
α
+ = = = Δ R
P
P
L
out
a
(5.5.4)
This result cannot be considered to be very accurate. Especially the assumption of a diffuse
sound field inside the enclosure is doubtful. However, the result is not bad as a rough estimate
for the design of an enclosure. It is clearly seen from (5.5.4) that both transmission loss and
absorption coefficient are important for an efficient reduction of noise by an enclosure.
5.6 IMPACT SOUND INSULATION
The noise generated from footsteps on floors is characterised by the impact noise level. It is
measured according to ISO 140 Part 6 and 7 by a standardised tapping machine. The main data
for the tapping machine are:
• The noise is generated by steel hammers with a fall height of 40 mm
• Each steel hammer has a mass of 500 g
• The number of taps per second is 10.
In the source room the tapping machine is placed on the floor in a number of positions. In the
room below  or any other room in the building – the calibrated sound pressure level L
2
is
measured. The reverberation time in the receiving room must also be measured in order to
calculate the absorption area A
2
. The impact sound pressure level is the sound pressure level in
dB re 20 μPa that would be measured if the absorption area is A
0
= 10 m
2
:
2
0
0
2
2
m 10 (dB) log 10 = + = A
A
A
L L
n
(5.6.1)
The frequency range is the same as for airborne sound insulation, i.e. the 16 onethird octave
bands from 100 Hz to 3150 Hz. However, it is recommended to extend the frequency range
down to 50 Hz, especially in the case of lightweight floor constructions.
127
Fig. 5.6.1. Principle of measuring the impact sound pressure level from a floor to a receiving
room (2)
5.7 SINGLENUMBER RATING OF SOUND INSULATION
5.7.1 The weighted sound reduction index
The singlenumber rating of sound insulation is practical for several purposes:
• to characterise the measuring result of a building construction,
• for quick comparison of the sound insulation obtained with different constructions, and
• to specify requirements for sound insulation.
The weighted sound reduction index R
w
is based on a standardised reference curve that is
defined in onethird octaves in the frequency range 100 Hz – 3150 Hz. The reference curve is
made from three straight lines with a slope of 9 dB per octave from 100 to 400 Hz, 3 dB per
octave from 400 to 1250 Hz, and 0 dB per octave from 1250 to 3150 Hz.
128
The measured transmission loss is compared to the reference curve, and the sum of
unfavourable deviations is calculated. An unfavourable deviation is the deviation between the
reference curve and the measured curve if the measured sound insulation is lower than the value
of the reference curve.
The reference curve is shifted up or down in steps of 1 dB, and the correct position of the
reference curve is found when the sum of unfavourable deviations is as large as possible, but do
not exceed 32 dB. The value of the reference curve at 500 Hz is taken as the singlenumber
value of the measuring result. The method is also shown in Fig. 5.7.1.
T
r
a
n
s
m
i
s
s
o
n
L
o
s
s
,
R
,
d
B
Frequency, Hz
Fig. 5.7.1. Determination of the weighted sound reduction index. M is the measured curve, V
1
is
the reference curve in position 52 dB, and V
2
is the shifted reference curve. The result is R
w
=
60 dB.
5.7.2 The weighted impact sound pressure level
The weighted impact sound pressure level L
n,w
is very similar to the weighted sound reduction
index. It is based on a standardised reference curve that is defined in onethird octaves in the
frequency range 100 Hz – 3150 Hz. The reference curve is made from three straight lines with a
slope of 0 dB per octave from 100 to 315 Hz, 3 dB per octave from 315 to 1000 Hz, and 9 dB
per octave from 1000 to 3150 Hz.
The measured impact sound pressure level is compared to the reference curve, and the sum of
unfavourable deviations is calculated. An unfavourable deviation is the deviation between the
129
reference curve and the measured curve if the measured impact sound pressure level is higher
than the value of the reference curve.
The reference curve is shifted up or down in steps of 1 dB, and the correct position of the
reference curve is found when the sum of unfavourable deviations is as large as possible, but do
not exceed 32 dB. The value of the reference curve at 500 Hz is taken as the singlenumber
value of the measuring result. The method is also shown in Fig. 5.7.2.
I
m
p
a
c
t
S
o
u
n
d
P
r
e
s
s
u
r
e
L
e
v
e
l
,
L
n
,
w
,
d
B
Frequency, Hz
Fig. 5.7.2. Determination of the weighted impact sound pressure level. M is the measured
curve, V
1
is the reference curve in position 60 dB, and V
2
is the shifted reference curve. The
result is L
n,w
= 47 dB.
130
5.8 REQUIREMENTS FOR SOUND INSULATION
The Danish requirements for new buildings are laid down in “Bygningsreglement 1995” (BR
95) and in “Bygningsreglement for småhuse 1998” (BRS 98).
For dwellings in multistorey houses and for hotels the main requirements are:
• The airborne sound insulation shall be R´
w
≥ 52 dB in horizontal directions and R´
w
≥ 53 dB
in vertical directions.
• The impact sound pressure level shall be L´
n,w
≤ 58 dB.
• Between rooms for common service or commercial use and dwellings the airborne sound
insulation shall be R´
w
≥ 60 dB and the impact sound pressure level shall be L´
n,w
≤ 48 dB.
For rowhouses or semidetached houses the main requirements are:
• The airborne sound insulation shall be R´
w
≥ 55 dB.
• The impact sound pressure level shall be L´
n,w
≤ 53 dB.
In schools the main requirements are:
• Between classrooms the airborne sound insulation shall be R´
w
≥ 48 dB in horizontal
directions and R´
w
≥ 51 dB in vertical directions.
• The impact sound pressure level in classrooms shall be L´
n,w
≤ 63 dB.
• From rooms for music or workshops to classrooms the airborne sound insulation shall be
R´
w
≥ 60 dB and the impact sound pressure level shall be L´
n,w
≤ 53 dB.
The sound insulation of facades is not specified directly, but in buildings where then outdoor
traffic noise exceeds L
Aeq, 24
≥ 55 dB, the indoor noise in living rooms shall not exceed L
Aeq, 24
≤
30 dB.
5.9 REFERENCES
ISO 1403 (1995): Acoustics. Measurement of sound insulation in buildings and of building
elements. Part 3: Laboratory measurements of airborne sound insulation of building elements.
ISO 1404 (1998): Acoustics. Measurement of sound insulation in buildings and of building
elements. Part 4: Field measurements of airborne sound insulation between rooms.
ISO 1406 (1998): Acoustics. Measurement of sound insulation in buildings and of building
elements. Part 6: Laboratory measurements of impact sound insulation of floors.
ISO 1407 (1998): Acoustics. Measurement of sound insulation in buildings and of building
elements. Part 7: Field measurements of impact sound insulation of floors.
131
ISO 7171 (1996): Acoustics. Rating of sound insulation in buildings and of building elements.
Part 1: Airborne sound insulation.
ISO 7172 (1996): Acoustics. Rating of sound insulation in buildings and of building elements.
Part 2: Impact sound insulation.
BR95. (1995). Bygningsreglement (Building regulations, in Danish). Byggeog Boligstyrelsen,
Copenhagen.
BRS 98. (1998). Bygningsreglement for småhuse (Building regulations for small houses, in
Danish). Byggeog Boligstyrelsen, Copenhagen.
132
133
6 MECHANICAL VIBRATION AND STRUCTUREBORNE SOUND
Mogens Ohlrich
6.1 INTRODUCTION
Audio frequency vibration of mechanical systems and waves in solid structures form an
integral part of engineering acoustics in describing the dynamic phenomena in solids and
fluids, and their interaction. This subject, referred to as phenomena of structureborne sound
or vibroacoustics, is important because sound or noise is very often generated directly by
mechanical vibration of solid bodies or by waves transmitted in solid structures, and
eventually radiated into the fluid as audible sound. Examples are musical sound from a string
instrument or noise from a pump in a central heating system.
Vibration of simple resonant systems (resonators) is characterised by mass and stiffness
properties and by some form of damping mechanism, which dissipate vibrational energy. The
simplest description of dynamic behaviour applies to resonators that can be modelled as a
(minimal) combination of discrete or ‘lumped’ elements. If the response of the resonator
primarily occurs in only one direction, ie in a single motion coordinate, then the system is said
to have a single degree of freedom (sdof). Figure 6.1.1 shows examples of sdofresonators.
The mathematical description of the vibration of such systems is governed by an ordinary
secondorder differential equation. This is usually derived from a force balance of the mass
element. Solution of the equation shows that such systems have a single preferred ‘natural’
frequency of vibration, which can exist in the absence of external excitation.
Figure 6.1.1 Examples of single degree of freedom resonators. After ref. [1].
Vibration of more complex systems requires more than one motion coordinate for a
complete description. For example, in the case of a loudspeaker three degrees of freedom are
required for describing the designed translational motion of the ‘piston cone’ and its
unintentional rocking motions, which can occur in two planes. In general such motions will be
governed by three coupled, secondorder differential equations. However, by using a special
set of coordinates these equations can be uncoupled and solved independently, as is the case
134
for the sdofresonator.
Vibration of different phase, ie, structural wave motion, can occur when the wavelength
of vibration in a solid structure is less than one of its typical dimensions. If this is the case it is
natural to threat the system as a continuous one. The response of such a system is governed by
a partial differential equation, because the response depends upon both time and a spatial
position coordinate that specifies the location at which the response is to be determined.
6.1.1 SOURCES OF VIBRATION
There are many types of excitation mechanisms that generate vibration and waves in solid
structures. Such sources are associated with nature or they involve the employment of
machines in the broadest sense, that is, devices that do work, ranging from a miniature loud
speaker in a hearing aid to a combustion engine of a truck, say. The sources can be classified
by their temporal variations for which there are two types, transient and continuous that
includes time variation of either deterministic (periodic) or random nature.
Examples of sources of vibration are shown in Figure 6.1.2. Transient sources
representing local impact are very common both as a single impact and in repetition, in which
case the excitation timehistory becomes periodic. The hammer impact symbolizes a variety
of excitation mechanisms such as musical percussion (drums, xylophones), impulsive sources
of vibration and noise in buildings (footfalls, door slamming), impacts in production
machinery (punch presses, forge hammers) and periodic impacts in combustion engines
(valves, piston slab). Figure 6.1.2b illustrates force excitation caused by an unbalanced
rotating mass; such excitation is often of a harmonic (pure tone) nature. Other sources of
vibration and noise are random variation of surface roughness, eg in wheel/surface contacts,
or distributed excitation of a structure, eg caused by a sound field.
Figure 6.1.2 Examples of sources that generate vibration and structureborne sound.
6.1.2 MEASUREMENT QUANTITIES
Investigations of vibration in solid structures are usually carried out by measuring a local
quantity at a specific position on the structure. Distribution of vibration over a larger area can
be determined by measurements in a number of discrete positions. The local measurement
quantity is either a motion (displacement, velocity or acceleration) or a force. Both types of
variables are vectors, and thus assigned to a certain orientation or direction.
Vibratory motion is usually measured unidirectional with a small transducer of the
accelerometertype that is fastened to the structure’s surface. The accelerometer is based upon
the piezoelectric principle with an output signal proportional to the acceleration a = a(t) of
the vibrating surface. Accelerometers are available with different sensitivities. The velocity v
or displacement ξ of the vibratory motion is obtained by integration of the acceleration signal.
135
A localised (point) force F = F(t) is mostly measured with a piezoelectric force
transducer, which produces an output proportional to the force. The measurement is carried
out by inserting the transducer between a source (eg a vibration exciter) and the measurement
object. This arrangement is mostly used for measuring the dynamic properties of structures,
for example, the impedances or the mobilities.
6.1.3 LINEAR MECHANICAL SYSTEMS
The dynamic properties of a physical system depend upon its mass and stiffness distribution
and damping losses. These properties are attempted described by mathematical models in the
form of one or more differential equations of motion. The system is said to be linear if the
dependent response variables are of first order. When this is the case, one can use the very
important superposition principle. This means that the response contributions from
independent excitations can be superimposed or summed as vectors.
Herein we assume that systems considered are linear, which is often the case when
vibration or waves have small amplitudes. System dynamics can therefore be described by
linear differential equations. These can be based either on a discrete model or on a continuous
model. In the discrete model the properties of system components are described by discrete
(‘lumped’) quantities, represented by ideal masses, massless springs and dampers, see Figure
6.1.3. The physical properties of the continuous model are functions of the spatial coordinates.
Dynamic properties of the system are therefore described by partial differential equations.
Figure 6.1.3 Lumped model of a physical system, where the physical properties are represented by
ideal discrete elements of point masses, massless springs and dampers.
The choice between the two models depends upon a number of factors such as frequency
range of interest, structural shape and forms of excitation. However, the actual decision of the
type of model is usually not strictly scientific, but is often based on intuition and practical
experience. In this note we shall focus mainly on the analysis of discrete models, whereas
only a brief summary will be given of wave motion in continuous structures (structureborne
sound).
Figure 6.1.4a shows the basic lumped elements; the quantity s represents the spring
constant (stiffness), m is the mass and r is the damping constant of a viscous damper; for
translatory motion these quantities have units of [N/m], [kg] and [kg/s], respectively. The
viscous damper represents a velocity proportional resistance that results in energy losses.
Symbolically, the viscous damping is thought caused by motion of a piston in a fluidfilled
cylinder.
The properties of the elements are independent of time t, and there is a linear relation
between forces F
i
= F
i
(t) and changes in, respectively, displacement ξ = ξ(t) , velocity v = v(t)
136
and acceleration a = a(t) over the terminals of the elements. Thus, for the ideal spring there is
proportionality between force and deformation according to Hooke’s law. The viscous
damping force is proportional to the velocity of the ‘deformation’ in the massless damper.
Figure 6.1.4 (a) Forceresponserelations for ideal lumped elements.(b) Excitation (action) and
reaction by compression of spring.
Note that both motion and force variables are vector quantities, as shown by the example
in Figure 6.1.4b . Both quantities are defined as positive in the direction of the vector; the
motion variables are thus defined as positive in the xdirection. In Figure 6.1.4a, the positive
force F required for accelerating the mass m is therefore F = ma , which is Newton’s second
law of motion in its simplest form.
6.2 SIMPLE MECHANICAL RESONATORS
Figure 6.2.1a shows a model of a single degree of freedom system that is connected to a rigid
foundation. The system consists of a mass m , a spring of spring constant s , and a velocity
proportional viscous damper of damping constant r.
Figure 6.2.1 (a) Viscously damped simple resonator driven by an external force F ; (b) diagram which
shows the forces acting on the mass m .
137
6.2.1 EQUATION OF MOTION FOR SIMPLE RESONATOR
The system is assumed excited by a timevarying external force F = F(t) and it is understood
that the system can vibrate only translatory, to and fro, in the direction of the force, that is, in
the horizontal plane in this example. The motion of the mass from its equilibrium position is
denoted by the displacement ξ = ξ(t) , and this is taken positive towards the righthand side.
The vibration response caused by the external force is uniquely defined by the
instantaneous value ξ . This displacement of the mass results in a compression of the spring
that produces a restoring, elastic spring force
. ξ s F
s
− = (6.2.1)
Thus, the reaction on the mass that is caused by the spring force, acts in the opposite direction
of the displacement imposed by the external force. If viscous damping is assumed as
illustrated by the parallelcoupled dashpot in Figure 6.2.1 then this element will exert a
corresponding restoring damping force
,
d
d
t
r F
r
ξ
− = (6.2.2)
that is, a force which is also directed opposite to that of the motion of the mass and in
proportion to its vibration velocity v = dξ /dt .
The vector sum of forces that act on the mass, that is, F + F
s
+ F
r
= F − sξ − rv , thus
serves to accelerate the mass. So, according to Newton’s second law of motion, this sum must
be equal to the product of mass m and acceleration a = d
2
ξ/dt
2
, ie
∑
= = .
d
d
2
2
t
m ma F
i
ξ
(6.2.3)
The equation of motion for the system therefore becomes
.
d
d
d
d
2
2
F s
t
r
t
m = + + ξ
ξ ξ
(6.2.4a)
This equation is often written in a reduced form as
,
d
d
d
d
2
0
2
2
m
F
t m
r
t
= + + ξ ω
ξ ξ
(6.2.4b)
where ω
0
is the natural angular frequency in [rad/s] of the corresponding undamped system
(r = 0), defined as
.
0
m
s
= ω (6.2.5)
In the literature the fraction r/m in eq. (6.2.4b) is often replaced either by 2δ or by 2ζω
0
where δ is the damping coefficient and ζ is the nondimensional viscous damping ratio.
Their definitions are respectively
.
2
2
and
2
0
ms
r
m
r
m
r
= = =
ω
ζ δ (6.2.6a,b)
Moreover, from Figure 6.2.1b it is seen that the total force F
f
acting on the rigid
foundation is equal to the sum of the spring force and the damping force, that is,
.
d
d
ξ
ξ
s
t
r F
f
+ = (6.2.7)
138
6.2.2 FORCED HARMONIC RESPONSE OF SIMPLE RESONATOR
Let us assume that the excitation force F in eq. (6.2.4) varies harmonically with time as
F = F
1
cosωt with angular frequency ω. After a certain builtup of vibration the mass will
then also execute stationary, harmonic vibration with the same angular frequency ω. Herein
we shall only deal with the stationary vibration of the system, since it is assumed that the
initial builtup of vibration caused by ‘starting’ the force has completely decayed because of
damping effects, see Figure 6.2.2.
Figure 6.2.2 Time history of vibration builtup in the case of harmonic force excitation of a simple,
damped resonator when ω < ω
0
. The vibration builtup response is succeeded by a stationary
vibration at the angular frequency ω of the excitation.
6.2.2.1 Undamped system
Initially, we shall disregard the damping of the considered system by setting r = 0 . Thus for
harmonic excitation the equation of motion (6.2.4) reduces to
. cos
d
d
1 2
0
2
2
t
m
F
t
ω ξ ω
ξ
= + (6.2.8)
The complete solution for ξ = ξ(t) of such a differential equation has the wellknown form
equation the to solutions cos
1
s homogeneou t + = ω ξ ξ (6.2.9)
where the first term represents the stationary harmonic vibration and the second term
represents the abovementioned phenomenon of vibration builtup or decay.
The displacement amplitude ξ
1
of the stationary vibration is obtained directly from eq.
(6.2.8) by substituting the assumed solution ξ = ξ
1
cosωt :
( ) , cos cos
1
1
2
0
2
t
m
F
t ω ω ξ ω ω = + −
which gives
2
0
2 2
0
2
1
2 2
0
1
1
1
1
1
1
1
ω ω
ξ
ω ω ω ω
ξ
−
=
−
=
−
=
stat
s
F
m
F
(6.2.10 a,b,c)
where eq. (6.2.10b) follows from eq. (6.2.5). Furthermore, the quantity ξ
stat
represents the so
called static displacement, which is the compression or extension of the spring caused by the
force F = F
1
cosωt when ω = 0:
.
1
s
F
stat
= ξ (6.2.11)
The stationary part of the solution (6.2.9), which describes the forced harmonic motion of the
139
resonator, is thus given by
. cos
1
1
cos
2
0
2
1
t t
stat
ω
ω ω
ξ ω ξ ξ
−
= = (6.2.12)
The fraction 1/(1 − ω
2
/ω
0
2
) represents the variation of the vibration amplitude with respect to
the excitation frequency ω and it is sometimes referred to as the response amplification factor;
this quantity also reveals the phase relation between the displacement response and excitation
force. Figure 6.2.3 shows the variation of this quantity ξ
1
/ξ
stat
with angular frequency; in
Figure 6.2.3b the same quantity is shown as absolute value (modulus) and phase.
From the figure it can be seen that the vibration amplitude grows towards infinity when
the excitation frequency ω approaches the undamped natural frequency ω
0
of the system; this
excitation condition is called resonant excitation, and the frequency at which ω = ω
0
is the
resonance frequency. At ω = ω
0
, the response ξ
1
is also seen to undergo a change in sign,
which corresponds to a phase change of π radians. Physically, this simply means that the
quantities ξ
1
and F
1
 are inphase at low frequencies, that is, for ω < ω
0
where the system
behaves springlike, whereas they are in antiphase for ω > ω
0
where the response is lagging
the harmonic force excitation by 180 degrees because of the system mass (inertia).
Figure 6.2.3 (a) Relative displacement response ξ
1
/ξ
stat
for an undamped simple resonator; (b) the
same response function plotted as modulus and phase.
For this undamped case the force F
f
that is transmitted to the foundation is caused by
the spring force and is given by F
f
= sξ , which follows from eq. (6.2.7) for r= 0. The disturb
ance force on the foundation thus follows directly by substituting the solution eq. (6.2.12)
. cos
1
1
cos
2
0
2
1
t F t F F
f f
ω
ω ω
ω
−
= = (6.2.13)
This force ratio F
f
/F
1
 has the same frequency variation as the motion ratio ξ
1
/ξ
stat
shown in
Figure 6.2.3. For excitation frequencies below the natural frequency of the system, that is for
ω < ω
0
, the mass has a negligible influence. This means that the excitation force is in
equilibrium with the spring force, which is transmitted unchanged to the foundation. Thus, if
the force on the foundation is to be reduced by vibration isolation it is required that natural
frequency of the system is designed in such a way that ω
0
<< ω/√2 is fulfilled. For a set
140
excitation frequency and system mass this is accomplished by selecting a ‘soft’ spring
element with an appropriately small spring constant s .
6.2.2.2 Viscously damped system
The influence of damping is now being considered. When damping losses are assumed to be
of the viscous type as in Figure 6.2.1 then eq. (6.2.4) applies.
By using complex notation the harmonic excitation force F(t) = F
1
cosωt can be
expressed as F(t) = Re{F
1
e
iωt
}, where F
1
is the complex amplitude of the force. The solution
of the equation of motion is assumed to be of the same form ξ(t) = Re{ξ
1
e
iωt
}, where
ξ
1
= ξ
1
e
iφ
is the complex amplitude of the harmonic displacement with φ being the phase
angle between the displacement response and the driving force. Physical quantities are of
course always real, and it is therefore necessary to take the real part of the mathematical
solution when we want the time variation of the physical motion. This yields
{ } . ) cos( Re ) (
1
i
1
ϕ ω ξ ξ ξ
ω
+ = = t e t
t
(6.2.14)
By performing in eq. (6.2.4a) substitutions of F(t) ≡ F
1
e
iωt
and ξ(t) ≡ ξ
1
e
iωt
result in the
solution for the stationary, harmonic vibration
1
:
( )
t t
e F e s r m
ω ω
ξ ω ω
i
1
i
1
2
i = + + − (6.2.15)
⇔ .
i ) (
i ) (
2 2
0
1
2
1
1
r m
F
r m s
F
ω ω ω ω ω
ξ
+ −
=
+ −
= (6.2.16a, b)
Hereby, the problem is basically solved. (If the time variation of the response is sought then
this is obtained by substitution in eq. (6.2.14).) Furthermore, since the squared modulus is
given by ξ
1
ξ
1
*
= ξ
1

2
, we get
.
) (
2 2 2 2 2
0
2
2
1
2
1
r m
F
ω ω ω
ξ
+ −
= (6.2.16c)
Thus, ξ
1
 is obtained by simply taking the squareroot of the expression (6.2.16c).
The force transmitted to the foundation follows similarly from eq. (6.2.7)
, ) i (
i
1
i t t
f
e s r e F
ω ω
ξ ω + = (6.2.17)
which by substituting eq. (6.2.16b) gives
.
) (
) (
and ,
i ) (
) i (
2 2 2 2 2
0
2
2 2 2
2
1
2
2 2
0
1
r m
r s F
F
r m
r s F
F
f f
ω ω ω
ω
ω ω ω
ω
+ −
+
=
+ −
+
= (6.2.18a,b)
1
) Here the symbol Re{··} is left out. This does not result in any trouble as long as one is strictly dealing
with field quantities (displacement, velocity, force etc). However, when dealing with energy or power quantities,
one must only include the real part of the field quantity. The time variation e
iωt
is also often left out in the
analyses, but it is of course to be recalled and taken into account when necessary.
141
Solution in sum form. The solution (6.2.16) for the complex displacement can also be written
in terms of its real and imaginary parts
. i
1 im re
ξ ξ ξ + = (6.2.19a)
In the following we shall assume that the arbitrary phase of F
1
is set equal to zero by a
suitable choice of timereference (t = 0); this means that the force amplitude is assumed to be
real, ie F
1
= F
1
. Thus, by transforming the denominator in eq. (6.2.16b) to a real quantity this
yields
.
) (
) (
i
) (
) (
2 2 2 2 2
0
2
1
2 2 2 2 2
0
2
1
2 2
0
1
r m
F r
r m
F m
ω ω ω
ω
ω ω ω
ω ω
ξ
+ −
−
+
+ −
−
= (6.2.19b)
The frequency variations of this solution are sketched in Figure 6.2.4a. Shown is the real and
imaginary parts of the displacement response of the viscously damped resonator when this is
driven by a harmonic force of constant amplitude F
1
. The damping is seen to limit the
displacement response in the frequency range around ω~ω
0
where the response ξ
1
is
controlled largely by its imaginary part ξ
im
.
Figure 6.2.4 Frequency variation of displacement ξ
1
for a viscously damped simple resonator driven
by a harmonic force of constant amplitude. (a) Real and imaginary parts; (b) Modulus and phase.
Solution in product form. The solution for the complex displacement response eq. (6.2.16) or
(6.2.19) is often written in the alternative ‘product form’
ϕ
ξ ξ
i
1 1
e = (6.2.20a)
where the modulus ξ
1
 and phase angle φ as usual are determined from eq. (6.2.19):
. tan and
2 2
2
1 re im im re
ξ ξ ϕ ξ ξ ξ = + =
The squared modulus of the displacement is already given by eq. (6.2.16c), whereas the phase
angle is found directly from eq. (6.2.19b), ie
.
) (
tan
2 2
0
ω ω
ω
φ
−
−
=
m
r
(6.2.20b)
142
Note that the phase angle becomes φ = −π/2 at resonant excitation. As previously, the actual
physical time variation of the vibration response follows from eq. (6.2.14)
{ } { } ) cos( Re Re ) (
1
i i
1
i
1
ϕ ω ξ ξ ξ ξ
ω ϕ ω
+ = = = t e e e t
t t
⇔ . ) cos(
) (
) (
2 2 2 2 2
0
2
1
ϕ ω
ω ω ω
ξ +
+ −
= t
r m
F
t (6.2.20c)
Figure 6.2.4b shows how the modulus and phase of the displacement varies with frequency
for harmonic force excitation. This type of graph is the most commonly used form of
presentation for frequency response functions.
The vibration velocity v(t) of the resonator is often of interest and this follows simply by
taking the time derivative of the displacement response, eq. (6.2.16) or (6.2.20):
, ) sin(
d
) ( d
) (
1
ϕ ω ξ ω
ξ
+ − = = t
t
t
t v
or (6.2.21)
( ) { } . i e wher , Re
d
d
Re ) (
1 1
i
1
i
1
ωξ ξ
ω ω
= =
⎭
⎬
⎫
⎩
⎨
⎧
= v e v e
t
t v
t t
So, with respect to the complex amplitudes a differentiation is simply archived by a
multiplication with iω ; evidently integration is performed by a division by iω. Moreover, the
acceleration a(t) of the motion is obtained similarly by the time derivative of velocity or by
the second derivative of displacement.
Nondimensional form. It is often convenient to introduce nondimensional parameters that
enable solutions for a class of systems to be presented in a general form. For simple
resonators the frequency ratio Ω is readily used as frequency parameter
. /
0
ω ω = Ω (6.2.22)
By substituting this as well as the dimensionless viscous damping ratio ζ into eqs. (6.2.16c)
and (6.2.20b) we obtain the general expressions for the displacement ratio ξ
1
/ξ
stat
and for the
phase angle φ :
,
2
tan and
4 ) 1 (
1
2 2 2
2
1
2 2 2
stat
Ω  1
Ω
Ω Ω
ζ
ϕ
ζ ξ
ξ
−
=
+ −
= (6.2.23a,b)
here, it is recalled that the static displacement is ξ
stat
= F
1
/s . Amplitude and phase
characteristics for the displacement ratio (6.2.23), are shown logarithmically in Figure 6.2.5a
for different values of damping ratio ζ . It is clearly seen that the damping has a dominant
influence on the response in the frequency range Ω ~ 1, which is close to the natural
frequency of the system.
Similar expressions for the force ratio F
f
/ F
1
 are obtained by substituting the non
dimensional parameters in eq. (6.2.18). Amplitude and phase characteristics for this ratio
between transmitted force and driving force are shown in Figure 6.2.5b.
143
In forced harmonic vibration the displacement response of the system reaches its
maximum value ξ
max
 at, say, Ω
r
= ω
r
/ω
0
where ω
r
is the resonance frequency. The actual
value of Ω
r
is determined by differentiating eq. (6.2.23a) with respect to Ω and by setting
the obtained expression equal to zero. This gives the value
2
2 1 ζ − = ≡
r
Ω Ω (6.2.24a)
⇔ ; 1 2 n whe , 1
2 2
<< − ≅ ζ ζ
r
Ω (6.2.24b)
in the last approximate expression use have been made of the truncated series:
(1−x)
½
≅ 1 − x/2 provided that x << 1. The maximum displacement thus occurs at an angular
frequency, which is slightly lower than the angular natural frequency of the undamped
system. By substituting eq. (6.2.24a) in (6.2.23a) we get
.
) 1 ( 4
1
2 2 2
2
max
ζ ζ ξ
ξ
−
=
stat
(6.2.25)
Figure 6.2.5 Amplitude and phase characteristics for: (a) Displacement ratio ξ
1
/ξ
stat
, and (b) Force
ratio F
f
/F
1
 . From ref. [2].
However, when the damping is small (ζ << 0.05) the resonance frequency will nearly
coincide with the natural frequency ω
0
of the undamped system, that is, ω
r ≅
ω
0
; the
maximum displacement thus becomes
144
.
2
1
0
1
max
r
F
stat
ω ζ
ξ ξ = ≅ (6.2.26)
The displacement at resonance is thus equal to ξ
stat
divided by 2ζ .
Similarly, the vibration velocity of the system can be shown to take its maximum value
v
max
 at ω = ω
0
, that is, at Ω =1 . Since v = ω ξ this yields
.
2
1
1
0 max
r
F
stat
= ≅
ζ
ξ ω ξ (6.2.27)
Relations for maximum acceleration can be derived in the same manner.
Finally, the modulus and phase of the frequency response functions for displacement and
velocity, respectively, are sketched in loglog format in Figure 6.2.6.
Figure 6.2.6 Logarithmic plots of the frequency response functions of a simple resonator represented
as displacement and velocity. A unit force excitation is assumed.
Characteristic properties. As apparent from previous discussions the dynamic properties of
the resonator are predominantly springlike at low frequencies (Ω << 1) and predominantly
masslike at high frequencies (Ω >> 1) ; the asymptotes shown in Figure 6.2.6 actually
represent the dynamic properties of the individual elements s, m and r under the action of the
force F
1
. The dynamic properties of the resonator (ie, the combined system) are therefore
characterised as being:
. e wher , 1 for controlled Mass
, where , 1 at controlled Damping
, where , 1 for controlled Stiffness
2
1
1
1
1
1
1
ω
ξ
ω
ξ
ξ
m
F
Ω
r
F
Ω
s
F
Ω
≈ >> •
≈ ≅ •
≈ << •
(6.2.28)
These asymptotic values for the displacement response ξ
1
 follow directly from eq. (6.2.16c).
Similar relations can be determined for velocity and acceleration.
145
6.2.2.3 Structurally damped systems
So far we have only considered damping of the viscous type. A second type is structural
damping, which is proportional to changes in elastic deformation, like the displacement of a
spring. Such structural damping is therefore appropriately modelled by assigning the inherent
losses to the spring element. For harmonic motion this can be represented by a complex
stiffness s = s(1 + iη) where η is the damping loss factor and s is the real part of the
complex spring constant. The loss factor thus defines the phase lag (hysteresis) between
harmonic driving force and spring displacement. By using the loss factor the equation of
motion for a single massspring resonator becomes
, ) i 1 (
d
d
i
1 2
2
t
e F s
t
m
ω
ξ η
ξ
= + + (6.2.29)
which, similar to eq. (6.2.15), has the solution ξ(t) = Re{ξ
1
e
iωt
}, where ξ
1
= ξ
1
e
iφ
is the
complex amplitude:
.
i ) (
i ) (
2 2
0
1
2
1
1
η ω ω η ω
ξ
s m
F
s m s
F
+ −
=
+ −
= (6.2.30)
This ‘complex stiffness’ approach is very convenient, because the equation of motion
can be formulated initially without regard to damping and finally the spring constant is
replaced by its complex value s = s(1 + iη) .
Now, comparing eq. (6.2.30) with (6.2.15) shows that sη corresponds to ωr . The
equivalent damping ‘constant’ r
eq
for a structurally damped spring thus becomes frequency
dependent, and so does the equivalent damping ratio ζ
eq
, ie
. ) 2 /( ) ( and / ) (
0
ω ηω ω ζ ζ ω η ω = = = =
eq eq eq eq
s r r (6.2.31a,b)
Alternatively, the loss factor of a parallel combination of an ideal spring and a viscous damper
of constant r may be expressed as η = rω/s . Note also that the equivalent damping ratio eq.
(6.2.31b) becomes ζ
eq
= η/2 at resonance. This relation may be used as an approximation
for other frequencies that are close to resonance.
6.2.3 FREQUENCY RESPONSE FUNCTIONS
The frequency response of a system is defined as the ratio of complex amplitudes of two
quantities representing the response to a certain excitation. This broad characterisation by the
term ‘frequency response’ is often imprecise because the response quantity can be either
displacement or one of its time derivatives: velocity and acceleration. It is therefore
customary to assign specific names and symbols to the various types of frequency response
functions.
6.2.3.1 Receptance
So far we have been dealing with ratios of response over force. When the system response is
characterised by its displacement the complex frequency response is called the receptance
H(ω) . So, this is defined as
, ) ( ) (
i
1
i
1 ) ( i
t
t
e F
e
e H H
ω
ω
ω ϕ
ξ
ω ω = = (6.2.32)
146
where the notation with angular frequency dependence, H(ω), implies that the quantity is a
continuous function of ω ; its amplitude spectrum H(ω) and phase spectrum φ(ω) can be
determined from
{ }
{ }
.
) ( Re
) ( Im
) ( tan and ) ( ) ( ) (
2
ω
ω
ω ϕ ω ω ω
H
H
H H H = =
∗
(6.2.33)
The definition eq. (6.2.32) states that ξ
1
e
iωt
= H(ω)F
1
e
iωt
, which means that the time variation
of the displacement for harmonic excitation is
{ } , ) ) ( cos( ) ( ) ( Re ) (
1
i
1
ω ϕ ω ω ω ξ
ω
+ = = t F H e F H t
t
(6.2.34)
where the force amplitude is assumed to be real.
Receptances of the discrete elements: spring s, damper r and mass m , follow
respectively from the fundamental relations between harmonic force and the associated
motion for such elements
.
1
) ( and
i
1
) ( ,
1
) (
2
m
H
r
H
s
H
m r s
ω
ω
ω
ω ω
−
= = = (6.2.35a,b,c)
Since the ideal spring and damper are massless it is assumed in the definition of their
receptances that one of their terminals is blocked and that a harmonic force drives the other,
free end.
It is sometimes useful to use the reciprocal of the receptance function; this is called
dynamic stiffness [3].
6.2.3.2 Mobility and Impedance
The velocity response is often of interest in vibroacoustics, for instance, because the radiated
sound power from a vibrating structure is proportional to its surface velocity. The complex
ratio between response velocity and driving force is called the mobility Y(ω) (or sometimes
admittance) and is defined as
, ) ( ) (
i
1
i
1 ) ( i
t
t
e F
e v
e Y Y
ω
ω
ω θ
ω ω = = (6.2.36)
There is, of course, a very simple relation between mobility and receptance since the complex
velocity amplitude is v
1
= iωξ
1
, ie
. ) ( i ) ( ω ω ω H Y = (6.2.37)
The mobilities of the ideal components are therefore easily determined either from the
fundamental relations or directly from eq. (6.2.35). Thus
.
i
1
) ( and
1
) ( ,
i
) (
m
Y
r
Y
s
Y
m r s
ω
ω ω
ω
ω = = = (6.2.38)
The reciprocal of a mobility function is named the impedance Z(ω)
.
) (
1
) (
ω
ω
Y
Z = (6.2.39)
147
These different frequency response functions are summarized in Table 6.2.1 together
with corresponding functions that involve acceleration response. The latter is called
accelerance and its reciprocal, the apparent mass. The accelerance is sometimes used because
acceleration is the response quantity that is usually measured directly.
Table 6.2.1 Definition of frequency response functions R/F and F/R , where F is the force and R is
the response that represents either displacement, velocity or acceleration.

Response Name of frequency response function
quantity
R R/F F/R

Displacement ξ Receptance H(ω) Dynamic stiffness S(ω)

Velocity v Mobility Y(ω) Impedance Z(ω)

Acceleration a Acceleration A(ω) Apparent mass M(ω)

6.2.4 FORCED VIBRATION CAUSED BY MOTION EXCITATION
Vibratory disturbances like motion excitation is very common and occurs, for example, in
transportation of any kind, in machinery and in certain cases also in buildings. In all these
examples and in vibration isolation of delicate equipment from disturbing environments, the
‘foundation’ has a given motion ξ
f
= ξ
f
(t) as shown in Figure 6.2.7. Thus we want to find the
imposed/generated motion ξ = ξ(t) of the mass.
Figure 6.2.7 Motion excitation of a damped simple resonator.
There are two motion coordinates, but despite of this the system has only one degree of
freedom, because the motion of the system is uniquely described by a socalled generalized
coordinate q = q(t) ; in this case by the motion differences
.
. .
.
and
ξ ξ ξ ξ − = − =
f f
q q (6.2.40)
The quantities q and
.
q describe, respectively, the compression (or elongation) of the spring
and the velocity difference over the damper. Since the total force on the mass in Figure 6.2.7
readily can be written down, is it not necessary to use q explicitly. From eq. (6.2.3) follows
directly
148
.
..
)
. .
( ) (
ξ ξ ξ ξ ξ m r s F
f f i
= − + − =
∑
(6.2.41)
This gives the equation of motion
.
.
. ..
f f
s r s r m ξ ξ ξ ξ ξ + = + + (6.2.42)
It is seen that there is a clear analogy between this expression and eq. (6.2.4a), if the right
handside of eq. (6.2.42) simply is interpreted as a special ‘forcing function’.
In the case of steadystate harmonic motion excitation ξ
f
e
iωt
, the solution to eq. (6.2.42)
can be assumed to be ξ ≡ ξ
1
e
iωt
; by substituting these quantities we obtain the solution for the
complex amplitude of the displacement ξ = Re {ξ
1
e
iωt
}
.
i ) (
i
2 2
0
1
r m
r s
f
ω ω ω
ω
ξ ξ
+ −
+
= (6.2.43)
This expression has the same form as eq. (6.2.18a). In motion excitation the ratio between
displacements is thus identical to the ratio between forces in the case of force excitation
(Figure 6.2.1). The frequency variation of ξ
1
/ξ
f
is therefore exactly identical to that of F
f
/F
1
shown in Figure 6.2.5b.
This finishes the analysis of simple sdof mechanical resonators. A treatment of free
vibration of such systems and an analysis of more complicated multidegree of freedom
systems is outside the scope of this introductory note on discrete systems. We will therefore
proceed with a brief introduction of continuous structures.
6.3 VIBRATION AND WAVES IN CONTINUOUS SYSTEMS
Distributed solid structures become ‘dynamically elastic’ and exhibit wavetype vibratory
behaviour as the frequency is increased to an extent, where the wavelength become
comparable to, or less than, the physical dimensions of the structure. Although discrete
models can be used for analysing wave motion at the lower frequencies, it becomes expedient
to use wavetype analysis in problems where the wavelength is short. Thus, a brief
introduction will be given to vibration and wave motion in continuous systems. Only systems
of one and two dimensions will be considered here, because most engineering structures have
at least one dimension, which is small in comparison with the relevant structural wavelength
of vibration. In the audible frequency range this is the case for basic engineering components,
such as strings, rods, beams, membranes, plates, shells, pipes etc.
Equations of motion that describe different wave types and vibroacoustic phenomena
have been formulated for many types of continuous structures [4,5,6]. Usually each wave type
is treated separately, although wave conversion between different types generally occurs at
most structural discontinuities, such as edges, corners and cross sectional changes.
The most important wave types in structures are considered to be (a) longitudinal waves,
(b) shear or torsional waves and (c) bending waves, which are also called flexural waves, see
Figure 6.3.1. In the following an introduction of these waves in plane structures will be given.
149
Figure 6.3.1 Different wave types: (a) Longitudinal wave (the lateral deformations are exaggerated),
(b) Torsional wave and (c) Bending wave. After ref. [7].
6.3.1 LONGITUDINAL WAVES
Longitudinal waves in onedimensional structures like rods and beams are compressiontype
waves that are similar to plane sound waves in a fluid. The local structural deformation in
connection with longitudinal wave motion is primarily in the direction of wave propagation,
although there is also a small lateral deformation normal to the structural surface. However,
this deformation is generally so small that it can be neglected as a radiator of sound to the
surrounding fluid. It should also be mentioned that the impedance of longitudinal waves in
solids generally is very high.
The equation of motion for longitudinal waves in an undamped beam can be written in a
compact form; the longitudinal displacement in the wave motion will be denoted by
u = u(x, t), where x represents its spatial dependence. If we assume purely harmonic
excitation and harmonic wave motion u = u(x)e
iωt
this reads
{ } , ) ( ) ( ) (
2
x F x u m x u L ′ = ′ − ω (6.3.1)
where L{····} is a differential operator that describes the force gradient in the beam, m' is its
mass per unit length and F '(x) is an external force excitation per unit length. For
longitudinal waves the operator is given by −ES d
2
/dx
2
, where E in [N/m
2
] is Young’s
modulus of elasticity of the beam material and S is the cross sectional area of the beam.
Two field variables are required for describing the longitudinal wave motion; these are
the already mentioned displacement u = u(x)e
iωt
– or its timederivative, the velocity
v = iωu(x)e
iωt
= v(x)e
iωt
– and the internal force F = F(x)e
iωt
associated with the wave
motion. This is given by
.
x
u
ES F
∂
∂
− = (6.3.2)
Moreover, the wave speed c
l2
of a freely propagation longitudinal wave in the beam is
150
ρ
E
c
l
2
= , (6.3.3)
where ρ is the material mass density; index 2 on c
l2
indicates that the structure has two
surfaces that are small compared with the wavelength of the motion. The corresponding wave
speed in a flat, homogenous plate is slightly higher (by about 5%):
,
) 1 (
2
1
ν ρ −
=
E
c
l
(6.3.4)
where ν is Poisson’s ratio, which is a material constant that expresses the ratio between
deformations in the lateral and lengthwise directions of the structure. For common solid
material ν ≈ 0.3 , and for rubberlike materials ν ≈ 0.5 .
A listing of material properties and wave speeds are given in Table 6.3.1. Note that the
wave speed in metals is about 3000 to 5000 m/s, that is, a magnitude higher than for sound in
air. Furthermore, the mass density for metals is seen to be up to 7000 times higher than for
air. This means that the characteristic impedance (ρc
l
) for compression waves in solid
structures is much higher than for air; for example, the characteristic impedance for steel is
10
5
times higher than in air, but only 27 times higher than the impedance in water.
Table 6.3.1 Material properties and wave speeds (phase speeds) for solid structures. After ref. [8].
6.3.2 SHEAR WAVES
In this wave type only shear deformations occur, but no volume changes. Moreover, the
direction of the ‘particle’ motion is perpendicular to the direction of propagation. Shear waves
are of importance in plates that are builtup of several layers of material with different
properties, eg sandwich honeycomb panels.
151
The equation of motion for shear waves is governed by a second order partial differential
equation [5] of a general form similar to that of longitudinal waves; the details shall not be
given here, though. The wave speed c
s
for shear waves in a plate is given by
,
) 1 ( 2
ν ρ ρ +
= =
E G
c
s
(6.3.5)
where G is the shear modulus of the material. From the righthandside of this equation it is
clear that there is a unique relation between Young’s modulus E and the shear modulus G, ie
.
) 1 ( 2
ν +
=
E
G (6.3.6)
Shear waves in rods are called torsional waves. This type of wave motion that involves
twisting of the cross section of the rod was shown in Figure 6.3.1b. If the rod has a circular
cross section then the wave speed is as given by eq. (6.3.5); otherwise the wave speed will be
lower.
The two wave types discussed so far have high characteristic impedances. These waves
may therefore be important for the wave transmission over large distances (eg in buildings
and ships) and in wave conversion to bending waves, which is the dominant wave type when
it comes to sound radiation to the surrounding fluid media, being air or water.
6.3.3 BENDING WAVES
Bending waves in beams and plates are characterised by the motion being perpendicular to
both the direction of propagation, and the surface of the structure, see Figure 6.3.1c. Bending
waves do therefore play a dominant role in sound radiation from structures. The reasons for
this are that the wave motion has a good ‘match’ to the adjacent air, and that bending waves
are easily generated, because of their low characteristic impedance.
The equation of motion for bending waves in an undamped beam can be written in the
previous compact form, but with the transverse displacement of the bending wave motion
being denoted by w=w(x, t). If we again assume purely harmonic excitation and harmonic
wave motion w=w(x)e
iωt
, we get
{ } , ) ( ) ( ) (
2
x F x w m x w L ′ = ′ − ω (6.3.7)
where the differential operator L{···} that describes the shear force gradient in the beam now
takes the form B d
4
/dx
4
. Here, B is the bending stiffness of the beam, m' is its mass per unit
length and F'(x) is an external force excitation per unit length. The operator is of fourth order,
and four field variables are thus required for describing the bending wave motion. There are
two motion variables, the transverse displacement w = w(x)e
iωt
and the angular displacement
β = β(x)e
iωt
, which is the first spatial derivative of w , ie dw /dx . Two force variables are
associated with the wave motion, the internal shear force F
y
= F
y
(x)e
iωt
and the internal
bending moment M
z
= M
z
(x)e
iωt
; these are given by
. and
2
2
3
3
x
w
B M
x
w
B F
z y
∂
∂
− =
∂
∂
= (6.3.8)
Moreover, the wave speed c
b
of a freely propagation bending wave in the beam is
152
,
4
1
2
1
⎟
⎠
⎞
⎜
⎝
⎛
′
=
m
B
c
b
ω (6.3.9)
which is seen to depend upon frequency; this special phenomenon is called dispersion. Such
dependence results in complicated sound radiation properties for plates and builtup
structures. The wave speed or phase speed is furthermore noticed to depend upon the bending
stiffness and the mass per unit length.
The phase speed of bending waves in a thin homogeneous beam with a rectangular
crosssection and of thickness h in the direction of the motion, is given by
, 8 . 1
2
f h c c
l b
≅ (6.3.10)
where f is the frequency (in Hz) and c
l2
is given by eq. (6.3.3).
Moreover, the phase speed in a thin homogeneous plate of thickness h is given by
, 8 . 1
1
f h c c
l b
≅ (6.3.11)
where c
l1
is given by eq. (6.3.4).
6.3.4 INPUT MOBILITY OF INFINITE SYSTEMS
Finally, in this brief introduction it is appropriate to list some input mobilities for point force
excitation. Or more specifically, input mobilities relating translational velocity v e
iωt
to
translational force F e
iωt
, both at the same point and in the same coordinate (direction). The
corresponding point impedances are the reciprocal of the given point mobilities.
6.3.4.1 Beam or rod
Longitudinal vibration. In the case of a semiinfinite (s∞) beam driven axially at the end, the
input mobility is
.
1
2 l
s
c m
Y
′
=
∞
(6.3.12)
where m' is mass per unit length and c
l2
is given by eq. (6.3.3).
Bending vibration. The input mobility of a semiinfinite beam driven at the end is
.
i 1
b
s
c m
Y
′
−
=
∞
(6.3.13)
where m' is mass per unit length and c
b
is given by eq. (6.3.9), or by eq. (6.3.10), provided
that the beam is of rectangular crosssection and is vibrating in the direction in which the
beam thickness h is measured.
The input mobility of an infinite beam driven in the ‘middle’ is given by
.
4
i 1
b
c m
Y
′
−
=
∞
(6.3.14)
Note that this is four times lower than the input mobility of the semiinfinite beam, eq.
(6.3.13).
6.3.4.2 Plate
Bending vibration. The input mobility of a semiinfinite plate driven normal to its surface and
at the end (edge) is
153
,
5 . 3
1
m B
Y
s
′ ′ ′
=
∞
(6.3.15)
where m'' is the mass per unit area, and for a homogeneous plate of thickness h the bending
stiffness B' is
.
) 1 ( 12
2
3
v
h E
B
−
= ′ (6.3.16)
It is noted that this input mobility, eq. (6.3.15), is purely real, provided that the plate is
undamped as is assumed here.
The input mobility of an infinite plate driven in the ‘middle’ is also real and is given by:
.
8
1
m B
Y
′ ′ ′
=
∞
(6.3.17)
Other point mobilities relating angular velocity to moment excitation, as well as cross
mobilities, are given in ref. [5].
6.4 VIBRATION ISOLATION AND POWER TRANSMISSION
Vibration isolation is one of the most effective ways of reducing the transmission of audio
frequency vibration from a disturbing source (machine, apparatus, etc) to a connected
receiver structure. This is generally accomplished by ‘disconnecting’ the transmission paths
between the two systems. In practice vibration isolation is done by inserting resilient
mechanical connections or rubber elements that are much more compliant (ie, dynamically
soft), than both the source structure and the receiving structure. Such vibration isolators have
springlike properties and are often made of vulcanised rubber elements, metal springs or
combinations thereof. The isolation principle is depicted in Figure 4.1a, and Figure 4.1b
shows an example of measured mobilities of a rubber isolator, engine source and elastic
receiver.
Figure 6.4.1 (a) Vibration isolated diesel engine on elastic ship foundation; (b) Mobilities of isolator,
engine and ship foundation. From ref. [9].
The principle of vibration isolation has already been described in Chapter 6.2. Thus, in
the case of a harmonically driven simple source of mass m resting on a spring s attached to
154
an idealised rigid foundation, it was found that vibration isolation is achieved when the
angular natural frequency ω
0
of the system is somewhat lower than the frequency component
ω of the excitation force.
6.4.1 ESTIMATION OF SPRING STIFFNESS AND NATURAL FREQUENCY
It is often easy
2
to determine the important quantities (m, s and m s /
0
= ω ) for uncritical
arrangements of simple machinery sources that are mounted on vibration isolators (springs).
Usually the mass m of the machine is known. For a vertically loaded spring the static force
F
0
= mg from the mass results in a static deflection (compression) of the spring of magnitude
ξ
0
= F
0
/s . These two relations enable the determination of the static stiffness of the spring, ie
,
0
ξ
g m
s = (6.4.1)
where g (= 9.81m/s
2
) is the gravitational acceleration. The designed natural frequency of the
system can therefore be determined by a very simple formula:
. [Hz]
5 . 0
2
, [rad/s]
0
0
0
0
0
ξ π
ω
ξ
ω ≅ = = = ⇔ f
g
m
s
(6.4.2)
If the spring element is slender and rodlike with a cross sectional area S , length
(height) d and made from a material of Young’s modulus E , then the static spring constant
s can be calculated from
s = ES/d . (6.4.3)
Note that the dynamic stiffness of rubberlike material generally differs from this value of
static spring constant or stiffness s. This will be treated in more details in Section 6.4.4.
Figure 6.4.2 Static deflection of spring, which in the unloaded condition has the length d .
It was mentioned previously that the vibration isolation can be improved by reducing ω
0
,
that is to say, by increasing the static deflection ξ
0
. This can be accomplished by reducing s,
but this results in a more laterally unstable arrangement. As a compromise for a number of
practical source cases it is therefore often ‘common’ to choose values in the approximate
range of 0.004 m < ξ
0
< 0.01 m, which corresponds to 8 > f
0
> 5 Hz.
2
) It should be recalled, however, that the simple oscillator model is a coarse simplification of the
reality, where an extended rigid body on springs will have six degrees of freedom and thus six natural
frequencies, eg see ref. [2].
155
6.4.2 TRANSMISSION OF POWER IN RIGIDLY COUPLED SYSTEMS
In contrast to the idealised model of a simple source on a rigid foundation we shall now
examine the more realistic case of source and foundation or receiving structure of finite
mobilities or impedances. It is reasonable to expect that the dynamic properties of the source
and receiver will effect the vibration isolation that is achievable in practice.
For reference we shall initially address the situation where the vibration source is rigidly
connected to the receiving structure, and it is assumed that source and receiver are connected
via a single motion coordinate (or terminal). First consider the source in a free uncoupled state
in which the vibration activity of the source can be characterised by its free terminal velocity
v
free
and its ability to transmit power by its terminal mobility Y
S
, see Figure 6.4.3a . These
source quantities are suitably combined into a single descriptor [10] called the terminal source
strength J
term
 :
,
2
S
free
term
Y
v
J = (6.4.4)
where
2
free
v is the timeaverage meansquare value of the free velocity v
free
= v
free
(t). This
source strength J
term
, with units of power [W], is useful when comparing different vibratory
sources.
Figure 6.4.3 Systems with a single coupling coordinate: (a) Free vibration source, (b) Source coupled
rigidly to receiving structure, (c) Reaction forces on systems.
In the analysis that follows we assume harmonic vibration v
free
≡ v
free
e
iωt
. The source is
now being connected to a receiving structure, which is characterised by the input mobility Y
R
.
This loading of the source causes the free velocity to change to v
R
, because of the force
reaction (−F) on the source, ie
. F Y v v
S free R
− = (6.4.5)
Since per definition v
R
= Y
R
F, we find directly for the rigid coupled system:
( ) ( ) . and
1 1
free R S R R free R S
v Y Y Y v v Y Y F
− −
+ = + = (6.4.6a,b)
156
The force and velocity at the coupling point have hereby been determined for this case of
rigid coupling.
The power that is transmitted to the receiving structure is given by the wellknown
relations:
{ } { } { }. Re Re Re
2
2
1
2
2
1
2
1
R R R R
Z v Y F Fv P = = =
∗
(6.4.7)
By substituting the expressions from eq. (6.4.6) herein yields
{ } { }
.
Re
Re
2
2
2
1
2
2
2
1
R
R
R
R S
R
free
Y
Y
v
Y Y
Y
v P =
+
= (6.4.8)
For further evaluation of the transmitted power this can be written in a convenient
alternative form. Introducing the terminal source strength J
term
 , eq. (6.4.4), and a power
coupling factor C
P
yields
,
P term
C J P = (6.4.9)
where
,
cos 2
cos
cos
2
θ
ϕ
ϕ
+ +
=
+
=
S R R S
R
R
R S
R S
P
Y Y Y Y
Y Y
Y Y
C (6.4.10)
and φ
R
is the phase angle of the receiver mobility and θ = φ
R
− φ
S
is the phase difference
between receiver and source mobilities. This takes values in the interval: 0 ≤ θ ≤ π . The
power coupling factor is noted to be symmetric with respect to the logarithm of the mobility
ratio Y
R
/Y
S
 . For further details see ref. [10, 11].
6.4.3 VIBRATION ISOLATED SOURCE
The effect of a vibration isolator is now considered. The source is connected to the receiver
via a vibration isolator as schematically shown in Figure 6.4.4a. For simplicity it is assumed
that the isolator can be modelled as an ideal spring with a spring constant s . Thus, because
the spring is assumed massless, this implies that the force on the lefthandside of the spring
Figure 6.4.4 (a) Block diagram of vibration isolation of a source with a single coupling coordinate.
(b) Diagram that shows the forces on the system elements.
157
is identical to the force on the receiver, F
1
= F
R
. The velocities are different, of course, and
similar to before given by
. and
1 1 1
F Y v F Y v v
R R S free
= ′ − = (6.4.11a,b)
The force and the velocities are related according to Hooke’s law as
,
i
i
1
1
⎟
⎠
⎞
⎜
⎝
⎛
′
− =
ω ω
R
v v
s F (6.4.12)
which, together with eq. (6.4.11) give
( ) ( ) . i and i
1 1
1 free R S R R free R S
v Y Y s Y v v Y Y s F
− −
+ + = ′ + + = ω ω (6.4.13a,b)
By substituting F
1
into the general relation, eq. (6.4.7b), gives the transmitted power to the
receiver
{ }
.
i
Re
2
2
2
1
R S
R
free
Y Y s
Y
v P
+ +
= ′
ω
(6.4.14)
So, this gives
,
P term
C J P ′ = ′ (6.4.15)
where
. cos
i
2 R
R S
R S
P
Y Y s
Y Y
C ϕ
ω + +
= ′ (6.4.16)
These results for the vibrationisolated source have to be compared with those for the
rigid coupled case in order to realistically evaluate the influence of the vibration isolator. This
influence is most suitably described by the effectiveness E
iso
= E
iso
(ω) of the vibration
isolator, also called its insertion loss. This is defined as the ratio between the squared
magnitudes of the receiver velocities before and after the installation of the vibration isolator 
or for that matter  as the ratio of the corresponding injected powers. Eqs. (6.4.6b) and
(6.4.13b) thus give
.
/ i
1
2
2
2
R S
R
R
iso
Y Y
s
v
v
E
+
+ =
′
=
ω
(6.4.17)
From this equation it is evident that a high effectiveness (ie, large number) requires that
the isolator mobility iω/s ≡ Y
I
is much higher (ie, much more mobile or compliant) than the
sum of the source and receiver mobilities, that is,
Y
I
= iω/s >> Y
S
+ Y
R
. (6.4.18)
Such a large value of inequality is not easily accomplished over the broad audible frequency
range, because lightly damped resonance in elastic source and receiving structures will occur
and limit the effectiveness of the isolator. Furthermore, at high frequencies the mass of the
isolator can no longer be ignored and resonance occur in the isolator itself, which also limit
the effectiveness. In the case of a symmetric vibration isolator, such modal behaviour can be
accounted for in a prediction by replacing iω/s in eq. (6.4.17) with the actual mobility of the
isolator Y
I
, see also ref. [12].
158
At first, the definition of the isolator effectiveness in eq. (6.4.17) does not seem to apply
to the ideal case of a rigid (immoveable) foundation that was assumed in Chapter 6.2.
However, this is not so, because E
iso
might as well be defined as the ratios of forces acting on
the receiver, whether this is moving or not. This follows from the fact that velocities and
forces are related via the receiver mobility. So, for the general elastic receiver the
effectiveness also reads E
iso
= F
R

2
/ F
R
'

2
, where the dash refers to the case with the source
resiliently connected to the receiver. Hence, by substituting the derived expressions for the
corresponding forces, eq. (6.4.6a) and (6.4.13a), respectively, we obtain exactly eq. (6.4.17).
Example 6.4.1 The isolation effectiveness E
iso
is to be determined for a harmonically driven massspring
resonator, which is connected to a rigid foundation, similar to the systems in Figure 6.2.1 or 6.4.2. The
undamped natural frequency of the resonator is m s /
0
= ω , where m is its mass and s is the spring stiffness. It
is here assumed that the system is structurally damped and that this is accounted for by taken the spring stiffness
to be complex s = s(1+iη).
The source, being the mass m , has the mobility Y
S
= (iωm)
–1
and the mobility of the receiver in the form
of a rigid foundation is Y
R
= 0 . Substituting these into eq. (6.4.15) gives
; i 1
) i 1 (
1
2
2
0
2
0
2
2
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
+
⎟
⎟
⎠
⎞
⎜
⎜
⎝
⎛
− ≅
+
− =
ω
ω
η
ω
ω
η
ω
s
m
E
iso
(6.4.19)
in the last approximation it is assumed that η << 1, so that 1 + η
2
≈ 1. By comparison it is seen that eq.(6.4.19)
is equal to the reciprocal of the results for F
f

2
/ F
1

2
in Figure 6.2.5b. This can also be deduced from eq.
(6.2.18), if the damping constant r is replaced by the equivalent constant r
eq
for a structurally damped spring
r
eq
= sη/ω .
Figure 6.4.5 shows an example of measured and predicted values of the isolation
effectiveness for a complicated vibration source (the diesel engine in Figure 6.4.1a), which is
resiliently mounted on an elastic foundation. The source is mounted on ten multidirectional
isolators; note that these isolators have a much higher mobility than the isolator example
shown in Figure 6.4.1b. The effectiveness is seen to be rather good, about 25 dB on average.
Also shown are two course estimations based upon, respectively, a simple massspringmass
model (LFprediction of resemblance to eq. (6.4.19)), and a simple monocoupled model,
where measured isolator mobility and average point mobilities of source and receiver have
Figure 6.4.5 Effectiveness of vibration isolation 10 log E
iso
of a multicoupled machinery source on
an elastic receiving structure.
159
been used in eq. (6.4.17). Despite of the coarse simplifications in these models, a reasonable
agreement with measurement is found in the frequency range up to 800 Hz.
Another example of predicted isolation effectiveness is shown in Figure 6.4.6. Here, a
105 m tall building structure is mounted on large, flat rubber pads that allow thermal
expansion or contraction of the huge building. Calculations were carried out in order to
estimate their isolation effectiveness against structureborne sound transmission from
disturbing underground rail traffic. It is apparent from Figure 6.4.6 that these thermal
expansion devises are not very useful as vibration isolators; their static deformation is simply
too small – in other words – the stiffness of the isolators is too high. At the fundamental
natural frequency of the system vibration amplification is observed and in the frequency range
above 90 Hz the effectiveness is seen to become very small at certain frequencies. These
correspond to the natural frequencies of the foundation columns (≈ ‘source’), on which the
rubber pads and building structure rest.
Figure 6.4.6 Isolation effectiveness of rubber expansion devises that support a tall building.
6.4.4 DESIGN CONSIDERATIONS FOR RESILIENT ELEMENTS
It was mentioned in Section 6.4.1 that the dynamic stiffness of rubberlike material generally
differs from the static spring stiffness s determined by static measurement. When such
isolators are used it is therefore necessary to insert the dynamic stiffness value s
dyn
instead of
s in the equation for the natural frequency, eg eq. (6.4.2a).
6.4.4.1 Rubberlike materials
The dynamic stiffness of rubber isolators depends upon a number parameters. An important
parameter is the rubber hardness, which is usually characterised in °Shore A of hardness. The
typical hardnessrange of commercial rubber isolators is from about 40°Shore A (for soft
isolators) to 80°Shore A , which is rather hard. Table 6.4.1 presents a coarse guide that shows
approximate, empirical values for the relation between rubber hardness, static Young’s
modulus E and dynamic Young’ modulus E
dyn
, or more specifically their ratio E
dyn
/E .
Thus, for a slender rubber isolator (of static stiffness given by eq. (6.4.3)), the
appropriate dynamic stiffness s
dyn
becomes
s
dyn
= E
dyn
S/d . (6.4.20)
160
However, this is generally not the final estimate, because the stiffness of rubber isolators also
depends upon another important parameter, which is basically the compactness of the isolator.
Generally, the stiffness of a short rubber block is found to be much higher than the stiffness of
a long slender sample. (Note, that this effect of course is accounted for when estimations are
based on a static loaddeflection test, ie on eq. (6.4.1).)
Table 6.4.1 Approximate values for the relation between rubber hardness, static Young’s modulus E
and dynamic Young’s modulus E
dyn
. The results apply to natural rubber.

Rubber hardness Static Young’s modulus E Ratio: E
dyn
/E
°Shore A 10
6
N/m
2


40 1.5 1.2

50 2.5 1.4

60 4.0 1.8

70 6.0 2.2

The stiffness expressed by eq. (6.4.18) therefore has to be corrected for the ‘bulkiness’ of
the rubber isolator. This can be characterised by an area ratio (or shape factor) R
S
=S
const
/S
free
,
in which the area S
constr
represents the total constrained or loaded area of the isolator, and
S
free
is the total free surface area of the isolator. Figure 6.4.7 shows the stiffness correction
factor C
s
to be used for a given area ratio R
S
. Thus, s
dyn
is to be multiplied with C
s
to give
the actual, corrected dynamic stiffness.
Figure 6.4.7 Stiffness correction C
s
to be used as a function of the area ratio R
S
of the vibration
isolator. After ref. [13].
6.4.4.2 Metal and other elastic solids
As oppose to the rubberlike materials, the static and dynamic elastic properties for most
engineering materials are found to be practical identical. For a given elastic material this
means that its Young’s modulus E ≅ E
dyn
and its shear modulus G ≅ G
dyn
. Furthermore,
161
since ν ≈ 0.3 for most solid materials, we have E ≈ 3G .
Resilient elements of metal may take many different forms. Usually they are extended,
continuous components with distributed mass and stiffness, and basically they are designed to
achieve a specified small stiffness at low frequencies. However, at mid and high frequencies
such a resilient element can support different wave types, and resonances will occur in the
resilient element because it is of finite size. This will diminish the isolator effectiveness,
unless damping and/or rubber elements are incorporated into the final design of the resilient
element.
The most common resilient element of metal is probably the helical spring, which is
often made of harden steel. The static and low frequency stiffness in the axial direction of the
spring is
,
8
3
4
D n
d G
s = (6.4.21)
where G is the shear modulus of the material, D is the average diameter of the spring, d is
the diameter of the coil and n is the number of coils or windings.
Other types of resilient elements are leaf springs, which may be thin metal beams or
plates. One example is a socalled cantilever beam, which is rigidly builtin at the receiver
end and is completely free at the other end, where it supports the source to be isolated. For a
beam with constant thickness h and constant rectangular crosssection S the spring stiffness
is
,
4
3
2
L
h S E
s = (6.4.22)
in which E is Young’s modulus and L is the length of the beam. However, usually the
source will be bolted to the beam and this will hinder angular motion at its ‘free’ end. Thereby
the spring stiffness of the resilient element will increase by a factor of four, to become
s = E S h
2
/L
3
. This clearly illustrates the importance of the boundary conditions at mounting
positions.
162
6. 5 REFERENCES
1. Resonators by M. Heckl, Chapter 20 in ‘Modern methods in analytical acoustics: Lecture
Notes’ (ed. D. G. Crighton, A. P. Dowling, J. E. Ffowcs Williams, M. Heckl and F. G.
Leppington), SpringerVerlag, London Ltd. 1994.
2. C. M. Harris and C. E. Crede: Shock and vibration handbook, 2
nd
ed. McGrawHill, New
York 1976.
3. D. J. Mead: Passive vibration control. John Wiley & Sons, Chichester 1999.
4. E. Skudrzyk: Simple and complex vibratory systems. Penn State University Press 1968.
5. L. Cremer , M. Heckl and E. E. Ungar: Structureborne sound, 2
nd
ed., Springer Verlag,
Berlin 1988.
6. L. Cremer und M. Heckl: Körperschall, 2
nd
ed. Springer Verlag, Berlin 1996.
7. Vibration of one and twodimensional continuous systems by M. Heckl, Chapter 64 in
‘Encyclopedia of acoustics’ (ed. M.J. Crocker), John Wiley & Sons, Ney York 1997.
8. F. J. Fahy: Sound and structural vibration. Academic Press, London 1985.
9. M. Ohlrich and F. Jacobsen: Isolation of structural vibration from machinery.
Proceedings of Nordic Acoustical Meeting, NAM82, Stockholm, 1982, pp. 309312.
10. M. Ohlrich: Vibrational source strength as a prerequisite for response prediction by SEA.
NOVEM 2000, Proceedings of Intern. Conf. on Noise & Vibration Predesign and
Characterisation using Energy Methods, Lyon, 2000, on CDROM, pp.12.
11. J. M. Mondot and B. Petersson: Characterisation of structureborne sound sources: The
source descriptor and the coupling function. Journal of Sound and Vibration 114, 1987,
pp. 507518.
12. E. E. Ungar and C. W. Dietrich: Highfrequency vibration isolation. Journal of Sound
and Vibration 4(2), 1966, pp. 224241.
13. VDI 2062 Blatt 2 Vibration isolation: Resilient elements (In German), 1976.
163
LIST OF SYMBOLS
a radius of sphere [m]; acceleration [m/s
2
]
A equivalent absorption area [m
2
]; accelerance [m/Ns
2
]
A
0
reference area [m
2
]
B bending stiffness per unit length [Nm]; bending stiffness [Nm
2
]
B´ bending stiffness per unit width [Nm]
c speed of sound [m/s]
c
b
speed of bending waves [m/s]
c
L
speed of longitudinal waves [m/s]
C
P
power coupling factor [dimensionless]
d length [m]
D directivity [dimensionless]
DI directivity index [dB]
E total acoustic energy [J]; Young’s modulus of elasticity [N/ m
2
]
E
iso
vibration isolation effectiveness; insertion loss [dimensionless]
f frequency [Hz]
f
0
resonance frequency [Hz]
f
c
critical frequency [Hz]
F force [N]
G shear modulus [N/m
2
]
h distance [m]; plate thickness [m]
H receptance [m/N]
H
1
Struve function
I sound intensity [W/m
2
]
I
ref
reference sound intensity [W/m
2
]
I
x
component of sound intensity [W/m
2
]
J
m
Bessel fuction
J
term

terminal source strength [W]
k wavenumber [m
1
]
K stiffness constant [N/m]
K
s
adiabatic bulk modulus [N/m
2
]
l length [m]
l
m
mean free path [m]
L loudness level [phone]; total length of edges [m]; length [m]
L
A
Aweighted sound pressure level [dB re p
ref
]
L
Aeq
equivalent Aweighted sound pressure level [dB re p
ref
]
L
AE
sound exposure level [dB re p
ref
]
L
C
Cweighted sound pressure level [dB re p
ref
]
L
eq
equivalent sound pressure level [dB re p
ref
]
L
I
sound intensity level [dB re I
ref
]
L
n
impact sound pressure level [dB re p
ref
]
L
p
sound pressure level [dB re p
ref
]
L
W
sound power level [dB re P
ref
]
L
Z
sound pressure level measured without frequency weighting [dB re p
ref
]
m air attenuation factor [m
1
]; mass [kg]; mass per unit area [kg/m
2
]
m´ mass per unit length [kg/m]
164
m´´ mass per unit area [kg/m
2
]
M mass [kg]
n natural number [dimensionless]
N loudness [sone]; number of modes [dimensionless]
p sound pressure [Pa]
p
A
(t) instantaneous Aweighted sound pressure [Pa]
p
ref
reference sound pressure [Pa]
p
rms
rms value of sound pressure [Pa]
p
0
static pressure [Pa]
P power [W]
P
a
sound power [W]
P
ref
reference sound power [W]
q volume velocity associated with a fictive surface [m
3
/s]; generalised coordinate [m]
Q volume velocity of source [m
3
/s]; directivity factor [dimensionless]
r radial distance in spherical coordinate system [m]; damping constant of viscous
damper [kg/s]
r
rev
reverberation distance in a room [m]
R gas constant [m
2
s
2
K
1
]; reflection factor [dimensionless]; transmission loss [dB]
R
0
transmission loss at normal incidence [dB]
s standing wave ratio [dimensionless]; spring constant [N/m]
S surface area [m
2
]; cross sectional area [m
2
]
t time [s]
T absolute temperature [K]; averaging time [s]
T
60
reverberation time [s]
u longitudinal displacement [m]
u particle velocity [m/s]
u
x
component of the particle velocity [m/s]
U velocity [m/s]
v velocity [m/s]
V volume [m
3
]
w transverse displacement [m]
w
kin
kinetic energy density [J/m
3
]
w
pot
potential energy density [J/m
3
]
x, y, z Cartesian coordinates [m]
Z
a
acoustic impedance [kg m
4
s
1
]
Z
a, r
acoustic radiation impedance [kg m
4
s
1
]
Z
m
mechanical impedance [kg/s]
Z
m, r
mechanical radiation impedance [kg/s]
Z
w
separation impedance [kg m
2
s
1
]
Y mobility (mechanical admittance) [s/kg]
α absorption coefficient [dimensionless]
α
m
mean absorption coefficient [dimensionless]
β angular displacement [radian]
γ ratio of specific heats [dimensionless]
δ damping coefficient [s
1
]; end correction [m]
ΔL insertion loss [dB]
165
ΔV volume displacement [m
3
]
ζ viscous damping ratio [dimensionless]
η loss factor [dimensionless]
θ polar angle in spherical coordinate system [dimensionless]
λ wavelength [m]
ν Poisson’s ratio [dimensionless]
ξ displacement [m]
ρ density [kgm
3
]
τ time constant [s]; transmission coefficient [dimensionless]
φ phase angle [radian]; azimuth angle in spherical coordinate system [radian]
ω angular frequency [radian/s]
Ω frequency ratio [dimensionless]
^ indicates complex representation of a harmonic variable
166
167
INDEX
Absorption area, 87, 97, 110
Absorption coefficient, 29, 36, 87, 103
Absorption, 29, 35
Accelerance, 147
Acceleration, 28, 53, 134, 137, 142, 147
Accelerometer, 134
Acoustic filters, 29
Acoustic impedance, 29
Acoustic properties of materials, 29, 35
Acoustic twoport, 29
Adiabatic bulk modulus, 3
Adiabatic process, 2, 3
Admittance, 27
Afilter
see Aweighting
AI principle, 76
Air attenuation factor, 93, 98
Amplitude, 5, 6, 15, 52
Analogous electrical circuit, 29
Angular displacement, 151
Angular frequency, 5
Antinode
see Node
Antiphase, 9, 41
see also Quadrature
Aperture, 115
Apparent mass, 147
Apparent sound transmission loss, 114, 127
Audible frequency range, 6
Audiogram, 62
Averaging time
see Integration time
Aweighting, 23, 69, 70
Axial modes, 82
Background noise, correction for, 18, 20
Backward masking, 67
Baffle, effect of, 39, 42
Bandpass filters, 16
Bark, 72
Basilar membrane, 58, 59, 66, 71
Beam, 148
Bel, 18
Bending moment, 151
Bending stiffness, 121
of beam, 151
Bending waves on structures, 1, 2
see also Wave types
Bessel function, 43
Boundary conditions, 4
Cancellation of sound, 9
Cartesian coordinate system, 3
Cavity, sound field in, 3, 30, 106
Cfilter
see Cweighting
Characteristic impedance
see Impedance
Cochlea, 57
Coincidence, 121
Combinations of monopoles, 40, 41
Complex amplitude, 6, 52
Complex exponential representation, 52
Complex stiffness, 145
Compliance
see Stiffness
Condenser microphone, 21
Conservation of mass, 2
Conservation of sound energy, 33
Consonant (intelligibility) 111
Constant percentage filters, 15
Constructive interference, 9, 20
Continuous structure, 148
Converging waves, 14
Crest factor, 26
Critical band, 66, 71, 72
Critical frequency, 121
Crossover frequency, 119, 124
Cweighting, 23, 69, 70
Damping coefficient, 137
Damping constant, 135
Damping loss factor, 145
Damping force, 137
Danish Building Law, 109
Danish Working Environment Agency, 109
Dantale, 75
dB HL, 62, 63
168
Decade, 16
Decay curve, 100
Decibels, 18
Density of the medium, 2, 3
Destructive interference, 9, 20
Detection of a pure tone in noise, 17
Deterministic signal, 17
Dfilter
see Dweighting
Diatonic scale, 16
Differentiation with respect to time,
6, 53, 142, 149
Diffraction, 2
Diffuse sound field, 85, 98, 103
Dipole, 41
Dipole strength, 41
Direct field, 90
Directivity, 43
Directivity factor, 48, 91
Directivity index, 48
Dispersion, 2, 152
Displacement, 53, 134, 138, 147
Displacement ratio, 142
Displacement response, 139, 141, 143
Diverging waves, 14
Double construction, 123
Dweighting, 23, 70
Dwellings (reverberation control) 110
Dynamic stiffness, 146, 159
Echo, 95
Echoellipse, 95
Electret microphone, 21
Enclosure
see Cavity, sound field in
Energy balance equation, 87
Energy density in sound field
kinetic, 32
potential, 32
Energy of a signal, 22
Engine exhaust system, 38
Equally tempered scale, 16
Equation of motion for
simple resonator, 137, 138
continuous structures, 148
Equilibrium position, 137
Equivalent integration time, 24
Equivalent rectangular bandwidth, 72
Equivalent sound pressure level, 24
Equivalent viscous damping ratio, 145
ERB, 72
Euler’s equation of motion, 4, 6
Excursion of a loudspeaker membrane, 47
Exponential averaging
see Time averaging
Eyring’s formula, 93, 99
Far field, 90
Far field approximation, 15, 35, 43
FFT analysers, 15, 17
Field variables, 149
Filter, 15
Filter bank analysers, 15
Flanking transmission, 114, 127
Flanking transmission loss, 127
Flexural waves
see Wave types
Fluctuating noise, 25
see also Intermittent noise
Focusing, 96
Force, 27, 135, 138, 140
Force transducer, 135
Formant, 74
Forward masking, 64, 67
Free field, 61, 63
Freefield correction, 22
Freefield method
see Sound power determination
Freefield microphones, 23
Free terminal velocity, 155
Frequency, 5
Frequency analysis, 15
Frequency discrimination, 71
Frequency response of microphone, 22
Frequency selectivity, 71
Frequency weighting filters, 23
Fundamental frequency, 10
Gas constant, 3
Gauss’s theorem, 33
Generalised coordinate, 147
Ground effect, 38
Harmonic sound field, 5, 52
169
Harmonics, 10
Hearing level, 62
Hearing threshold, 55, 60, 61, 62, 65, 69
Helicotrema, 58, 59
Helmholtz equation, 7
Helmholtz resonator, 31
Hooke’s law, 28, 136
Image sources, 38, 42, 94
Impact sound pressure level, 128
Impedance
acoustic, 28
characteristic, 7, 31
mechanical, 27, 146
radiation, 29, 35, 37, 46
specific acoustic, 29
Incident sound intensity, 36
Incident sound power, 86
Incoherent signals
see Uncorrelated signals
Industry (reverberation control) 110
Independent sources, 17, 20
Inhomogeneous medium, 2
Inner ear, 55, 56, 57, 65, 71
Input impedance, 29
Input point mobility
see Mobility
Insertion loss, 128, 157
Instantaneous energy density, 32
Instantaneous sound intensity, 32
Integration time, 25
see also Time averaging
Intelligibility, 76, 109
Intensity
see Sound intensity
Interface between two fluids, 11
Interference effects, 2, 9, 20, 38
Intermittent noise, 26
Inverse distance law, 14, 20
Isolation effectiveness, 157, 161
Junctions between coupled pipes, 29
Kinetic energy
see Energy density
Levels, 18
Linear averaging
see Time averaging
Linear frequency weighting, 23
Linearised wave equation, 2
Linearity, 4
Liquids, sound in, 4
Locally plane waves, 4, 34
Logarithmic frequency scale, 16
Longitudinal waves, 1
see also Wave types
Loss factor, 122
Loudness, 55, 63, 64, 67, 68, 69
Loudness level, 63, 64, 69
Loudspeakers, 47
Lumped elements, 135, 136
Lumped parameter models, 29
Masking, 55, 59, 64, 65, 66, 67, 69
Mass, 135
Mass density, 150
Mass law, 120
Material properties, 150
see also Acoustic properties of
materials
Mean absorption coefficient, 87, 99
Mean free path, 92
Mean square value, 15, 17
Mechanical admittance, 28
see also Mobility
Mechanical oscillator, 28
Mechanical resonators, 136
see also Mechanical oscillator
Mechanical systems, 135
Membrane absorber, 106
Middle ear, 55, 56, 57, 58
Mobility, 146, 152, 155
see also Mechanical admittance
Mobility, input for semiinfinite or infinite
beam or rod, 152
plate, 152
Modal density, 84, 96
Modes, 81
Monopole, 37
Motion excitation, 147
MTF, 76
Musical tones, 10, 16
170
Natural angular frequency, 137
Natural frequency, 81
see also Resonance frequency
Nearfield characteristics, 15
Newton’s second law of motion, 4, 28, 136,
137
Node, 9, 82
Noise
see Random noise
Noise event, 27
Nominal centre frequencies, 16
Normal ambient conditions, 3
Number of modes, 83
Oblique modes, 82
Octave bands, 16, 103
ODEON programme, 112
Office spaces (reverberation control) 110
Omnidirectionality, 23, 43
see also Directivity, Monopole
Onedimensional wave equation, 5
Onethird octave bands, 16
Orders of magnitude of perturbations, 2
Oscillating sphere
see Dipole
Outdoor sound propagation
see Ground effect
Overtones
see Harmonics
Parseval’s formula, 17
Partial masking, 67
Partials
see Harmonics
Particle displacement, 2
Particle velocity, 2, 71
Partitioning into frequency bands, 17
Pascal, 3
Peak level, 26
Phase, 5, 6, 52
Phase speed
see Wave speed
Phon, 63, 64
Phone scale, 68
Phonems, 111
Pink noise, 17
see also White noise
Piston in a baffle, radiation from, 42
Pitch, 10, 16
Plane waves, 4
Plate, 148
Point dipole, 41
Point source
see Monopole
Poisson’s ratio, 106, 150, 161
Porous absorber, 105
Potential energy
see Energy density
Power coupling factor, 156
Power transmission, 155
Pressure microphone, 23
Pressure node
see Node
Psychoacoustics, 55, 71
Pulsating sphere, 37
Puretone source
see Sinusoidal source
Quadrature, 15
see also Antiphase
Radiation impedance
see Impedance
Radiation of sound, 37
Random errors
see Statistical uncertainty
Random incidence microphone, 23
Random noise, 17
Rapid Speech Transmission Index, 77
RASTI
see Rapid Speech Transmission
Index
Ratio of specific heats, 3
Rayleigh’s integral, 42
Reactive sound field, 15
Receiving structure, 153, 156
Receptance, 145, 146
Reciprocity principle, 38
Reduction index, 113
Reference sound intensity, 21
Reference sound power, 21
Reference sound pressure, 18
Reference velocity, 20
Reflection, 2, 8, 94
171
Reflection density, 96
Reflection factor, 10, 36
Refraction, 2
Resilient element, 159, 161
Resonance, 9, 28, 30
Resonance frequency, 9, 28, 30, 106, 124,
139
see also Natural frequency
Resonant excitation, 138
Resonator absorber, 108
Reverberation distance, 90
Reverberation room, 89, 103
Reverberation time, 89, 97, 98, 103, 109
Rigid surface, reflection from, 8, 38
Rms value
see Root mean square value
Rms sound pressure, 15
Rod, 148
Root mean square value, 15
Rubber hardness, 159, 160
Rubber isolator, 159
Sabine’s formula, 89, 97, 98, 99
Scattering, 2, 99
Schools (reverberation control) 110
SEL
see Sound exposure level
Semitone, 16
Sensitivity of auditory system, 23
Separation impedance, 119, 122
Shadow
see Diffraction
Shear modulus, 151
Shear force, 151
Shear waves
see Wave types
Sign convention, 6, 27
SII, 76
Silencers, 29
Simple source
see Monopole
Simultaneous masking, 64
Single degree of freedom system, 136, 147
Sinusoidal source, 5, 6
Solution in
product form, 141
sum form, 141
nondimensional form, 142
Son, 68
Sone scale, 68
Sound absorption, 103
Sound exposure level, 26
Sound intensity, 32
Sound intensity in a plane wave, 34
Sound intensity level, 20
Sound level meter, 21
Sound power, 31, 33
Sound power determination, 35
Sound power level, 21
Sound pressure level, 18
Sound pressure, 1
Sound reflection
see Reflection
Source spectrum, 73
Source strength, 38, 41, 155
Source structure, 153
Sources of vibration, 134, 153
Specific acoustic impedance
see Impedance
Spectral density, 17
Speech intelligibility, 55, 75, 76
Speech intelligibility index, 76, 79, 112
Speech level, 74, 75
Speech spectrum, 74, 75, 76
Speech Transmission Index, 76, 112
Speed of sound, 3, 4
Spherical coordinate system, 13
Spherical sound waves, 13
see also Monopole
Spherical symmetry, 12
see also Monopole
SPL
see Sound pressure level
Spring constant, 135, 136, 154
see also Stiffness
Standing wave pattern, 9
Standing wave ratio, 10
see also Standing wave tube
Standing wave tube, 36
Standing waves, 81
Stapes, 56
Static pressure, 1, 2, 3
Static stiffness, 154
Stationary signals, 17
172
Statistical models of sound fields, 31
Statistical uncertainty in measurements, 25
STI
see Speech Transmission Index
Stiffness, 28
see also Spring constant
Stiffness correction, 160
Stochastic signals
see Random noise
Structureborne sound, 133
Structural damping, 145
Struve function, 46
Subwoofer
see Loudspeakers
Sum of harmonic signals, 20, 52
Suspended ceiling, 109
Tangential modes, 82
Tapping machine, 128
Temperature fluctuations in sound field, 2
Temperature, influence on the speed of
sound, 3
Temporal integration, 69
Thick wall, 118
Time average of a product, 54
Time averaging
exponential, 24
linear, 25
Time constant, 24
Time derivative
see Differentiation with respect to
time
Time integration, 53
see also Time averaging
Time weighting
see Time averaging
Timeaveraged energy density, 33
Timeaveraged sound intensity, 33
Transfer function, 83
Transmission between fluids, 11
Transmission coefficient, 113
Transmission loss, 113, 114
Transmitted force, 142
Transversal waves, 1
Transverse displacement in beams, 151
Twoport, 29
Typical values of sound power levels, 32
Typical values of sound pressure levels, 19
Uncorrelated signals, 17, 18
Undamped simple resonator, 139
Undamped system, 137, 138, 143
Unvoiced, 74
Velocity, 27, 53, 134, 142, 144, 147
Vibrating sphere
see Pulsating sphere, Dipole
Vibration isolation, 139, 153, 156
Vibration isolator, 153
Vibroacoustics, 133, 148
Viscous damper, 135, 136
Viscous damping ratio, 137, 142
Viscous friction, 105
Viscously damped system, 140
Voiced, 73
Volume acceleration, 47
Volume displacement, 3
Volume velocity, 28, 37
Water, 4, 11, 19
Wavelength, 5
Wavenumber, 5
Wave speed for
longitudinal waves, 149
shear waves, 150
bending waves, 152
Wave types, structural
longitudinal waves, 148
shear or torsional waves, 148, 150
bending or flexural waves, 148, 151
Weighted impact sound pressure level, 130
Weighted sound reduction index, 129
White noise, 17, 21, 100
Young’s modulus of elasticity, 106,149,160
ii
CONTENTS Page 1 An elementary introduction to acoustics ....................................................................1 Finn Jacobsen Introduction ....................................................................................................................1 Fundamental acoustic concepts ......................................................................................1 1.2.1 Plane sound waves .............................................................................................4 1.2.2 Spherical sound waves .....................................................................................13 1.3 Acoustic measurements ................................................................................................15 1.3.1 Frequency analysis ...........................................................................................15 1.3.2 Levels and decibels .........................................................................................18 1.3.3 Noise measurement techniques and instrumentation .......................................21 1.4 1.5 The concept of impedance............................................................................................27 Sound energy, sound intensity, sound power and sound absorption ............................31 1.5.1 The energy in a sound field ..............................................................................32 1.5.2 Sound absorption ..............................................................................................35 1.6 Radiation of sound .......................................................................................................37 1.6.1 Point sources ....................................................................................................37 1.6.2 Sound radiation from a circular piston in an infinite baffle .............................42 1.7 1.8 1.9 2 References ....................................................................................................................49 Bibliography .................................................................................................................50 Appendix: Complex notation .......................................................................................51 Ear, Hearing and Speech ...........................................................................................55 Torben Poulsen Introduction ..................................................................................................................55 The ear ..........................................................................................................................55 2.2.1 2.2.2 The outer ear.....................................................................................................56 The middle ear ..................................................................................................56
1.1 1.2
2.1 2.2
iii
.....................7 Speech ........................................63 Masking .....................85 iv ...................................68 2...............................................................................67 2........1 2..............................................81 3..................................3 Measurement of loudness ...................................72 2............................................2 Temporal integration ..69 2.........................................................59 2.............4...........................................6 The auditory filters ....................................................................7....................... speech level ....................1 Standing waves in a rectangular room ...........................................................................................57 2........................................81 3..............................................................5.............................................81 Jens Holger Rindel Sound waves in rooms.............................................1 The loudness curve .......................2 Transfer function in a room ..........................................................................................................64 2..............1 2.......................................................................2 2......................................2.................................4................................................4.......................................4 The hearing threshold ....5.........3 2............................................3 Density of natural frequencies............83 3............................2....................................................67 2.......................2 Statistical room acoustics ...............4....78 An introduction to room acoustics ...........................6......3........2 Partial masking .......3.............1 2..................................................................................................................................1......................................3..............................................................2 Critical bands..........................7....6..........4 The frequency analyzer at the Basilar membrane ..................67 2.............................75 2..............1 Complete masking .......73 Speech spectrum...............................................................69 2....................................2 2...5 Loudness......................2..74 Speech intelligibility ....................................3 Human hearing .....................................................................8 3 References ....73 2........................................................62 Loudness level ............................4 Backward masking ............................................................................................68 2...........................85 3...................3 The inner ear.....................................................65 2........61 2...................................................1 3...1.....................................................2................................................................................71 2...3 Forward masking ..............71 Equivalent rectangular bands .....................................................83 3........................5................................3 Speech production ...........61 Audiogram ...........................1 The diffuse sound field ............1...........................................................................7............
................4..........4 Reflection density in a room ...............................1 v ............ Reverberation distance ...............................................99 Measurement of reverberation time ...................88 Stationary sound field in a room..87 Reverberation time......................................................................................105 Membrane absorbers ....113 5...............................95 3...3 3.............................................................................1 4...........................................3 Sound reflections and image sources ........1 4................................................1 Definition .............113 Anders Christian Gade Introduction .3.................................................................93 3....6 3...................113 5.92 3...........89 Geometrical room acoustics ............................................................................. Sabine’s formula ......3.......101 Sound absorbers and their application in room design ......104 4..............2 3......................4....4............2.....................................................................................................................................................................94 3..............3....3 3.......................................................................4.....96 Calculation of reverberation time ...109 References ..................2 4.......2......4.......................................................................................................2 Sound absorption in the air.....................................................................4 Room acoustical design ...................................113 Jens Holger Rindel The sound transmission loss ....................4 3.........................................5 3....2 4.....3....................................2 3....97 Reverberation time in nondiffuse rooms.........................92 3.......................................6 Choice of room dimensions.......2................103 The room method for measurement of sound absorption ......112 An introduction to sound insulation .................................................................................................4 3....................108 4...............................87 Energy balance in a room ..86 Equivalent absorption area .....................3..5 4 References .........1....4 4.....98 Optimum reverberation time and acoustic regulation of rooms ....3 4.......................................................3 Incident sound power on a surface ......106 Resonator absorbers ...3 Porous absorbers .....................................3...........96 Reflection control .....5 5 Application of sound absorbers in room acoustic design .........................................................................................................2............................................................................................3.............103 Different types of sound absorbers.................3.............1 3................4...................................5 3............96 3.........................................................................................2......100 3..............................................................................................................................1 Sound rays and a general reverberation formula ................................
..8 5..............4 5................................................1.........3 5.............................................................................2..............................................................................2.....137 6...2 5..................122 Double leaf constructions .3......................................................1.............2 Simple mechanical resonators .......145 vi ...........128 5....5.....................123 5............................................3........................................................136 6.................2..3.....................................4 Multielement partitions and apertures ....................7 Flanking transmission ...117 5.........................................2.............................124 5......................................................126 Enclosures ...........9 6 Requirements for sound insulation .......................2 Measurement quantities............129 5..........................................................119 Sound insulation at random incidence ..........128 5..........120 The critical frequency............131 References ............1 5....................................................................................131 Mechanical vibration and structureborne sound ....................................2 The weighted impact sound pressure level .............................................................................................................................................3 Linear mechanical systems..134 6........1...............................................124 5.......1 Sources of vibration ..3 A general model of sound insulation of double constructions .......................1 6............................3 Sound transmission through a solid material .................................................114 5...................................2............6 5...................................................................................7..................................3 Frequency response functions ...................1............................2 Forced harmonic response of simple resonator ......................1..123 5.................2 Single leaf constructions ......................................................................................................127 Singlenumber rating of sound insulation ..............1 The weighted sound reduction index .....................2...................................2.....2 The massairmass resonance frequency ..2....114 5...............................127 Impact sound insulation ...............................134 6.....................2 Sound insulation between two rooms.................. 135 6.....................133 Mogens Ohlrich Introduction ..1........................7..............1 Sound transmission through a double construction..............5 5........................................................................................133 6....................117 The mass law ................................................138 6.......................5 5......................................................................3 Measurement of sound insulation ........113 5..............121 A general model of sound insulation of single constructions ..........................4 5.....................1 Equation of motion for simple resonator.................
.3 Bending waves ...................................................................................................................162 List of symbols ...147 Vibration and waves in continuous systems ..............................................3.149 6.....150 6..4 6...............................................159 6..........................................................5 References ............................4 Vibration isolation and power transmission .................................................156 6.................................4.........................................................2 Transmission of power in rigidly coupled systems .................163 Index… ........167 vii ...............................3 Vibration isolated source .................................................................................4 Input mobilities of infinite systems ..................................................................2..4 Design considerations for resilient elements ..................148 6.................................................6.................................4..3..............................................................................151 6.......3....................153 6..........4.................152 6...............................................1 Longitudinal waves ....155 6................................1 Estimation of spring stiffness and natural frequency .3.................................................3 Forced vibration caused by motion excitation ...................................154 6................................4...............2 Shear waves .............
viii .
The sound pressure. Figure 1. the difference between the instantaneous value of the total pressure and the static pressure. Thus the scope of acoustics ranges from fundamental physical acoustics to.1 AN ELEMENTARY INTRODUCTION TO ACOUSTICS Finn Jacobsen 1. that is.) 1 . Note that sound waves are longitudinal waves. The purpose of this chapter is to give an introduction to fundamental acoustic concepts. gases and liquids. wave motion in gases. 1.2 FUNDAMENTAL ACOUSTIC CONCEPTS One of the characteristics of fluids. Fluids are unable to transmit shearing forces. psychoacoustics and music.1 Fluid particles and compression and rarefaction in the propagating spherical sound field generated by a pulsating sphere. that is. is the lack of constraints to deformation.2. that is.1. which are transversal waves in which the particles move back and forth in a direction perpendicular to the direction of propagation. and the effects of such wave motion. and noise control. say. On the other hand a fluid reacts against a change in its volume with a change of the pressure. sound recording and reproduction. say. see figure 1. density and temperature. [1]. design of theatres and concert halls. Sound waves are compressional oscillatory disturbances that propagate in a fluid. liquids and solids. the density or temperature fluctuations. and includes technical fields such as transducer technology.2. It is also much easier to measure the sound pressure than. and to acoustic measurements. to the physical principles of acoustic wave motion. (From ref. The waves involve molecules of the fluid moving back and forth in the direction of propagation (with no net flow).1 INTRODUCTION Acoustics is the science of sound. bioacoustics. accompanied by changes in the pressure. is the quantity we hear. and therefore they react against a change of shape only because of inertia. unlike bending waves on a beam or waves on a stretched string.
This is a secondorder partial differential equation that. but other waves of interest in acoustics. the acoustic pressure fluctuations are extremely small compared with commonly occurring static pressure variations. ii) the local longitudinal force caused by a difference in the local pressure is balanced by the inertia of the medium. which at 1000 Hz corresponds to a particle displacement of less than 8 μm .In most cases the oscillatory changes undergone by the fluid are extremely small. which means that the speed of such waves depends on the frequency content of the waveform. Moreover. for example. Waves propagating in different directions interfere. and iii) sound is very nearly an adiabatic phenomenon. 3 2 . about 104. they will be scattered by small obstacles. The concept of fluid particles refers to a macroscopic average. which means that they change direction as they propagate. the fractional changes of the density are about 1. a change in altitude of one metre gives rise to a fractional change in the static pressure that is about 400000 times larger. A mathematical description of the wave motion in a fluid can be obtained by combining equations that express the facts that i) mass is conserved. because of diffraction there will only partly be shadow behind a screen. By comparison. At this level the fractional pressure variations (the sound pressure relative to the static pressure) are about 2 10 4 . The observation that most acoustic phenomena involve perturbations that are several orders of magnitude smaller than the equilibrium values of the medium makes it possible to simplify the mathematical description by neglecting higherorder terms. are dispersive.2. The speed with which sound waves propagate in fluids is independent of the frequency.2 Various wave phenomena.4 10 4 . expressed in terms of the sound 1 2 See section 1. see figure 1. and the particle velocity2 is about 50 mm/s. which is a very high sound pressure level. that is. close to the threshold of pain. At these conditions the fractional pressure variations amount to about 2. bending waves on plates and beams.3.2.2 for a definition of the sound pressure level. waves will be reflected by a rigid surface and more or less absorbed by a soft one. One can get an idea about the orders of magnitude of these changes by considering the variations in air corresponding to a sound pressure level1 of 120 dB.02 °C.2. In fact at 1000 Hz the particle displacement at the threshold of hearing is less than the diameter of a hydrogen atom!3 Sound waves exhibit a number of phenomena that are characteristics of waves. In short. the oscillatory changes of the temperature are less than 0. The result is the linearised wave equation. Figure 1. not to individual molecules. there is no flow of heat. and if the medium is inhomogeneous for instance because of temperature gradients the waves will be refracted. therefore the particle velocity can be much less than the velocity of the molecules. inside an aircraft at cruising height the static pressure is typically only 80% of the static pressure at sea level.5 1010 .
The physical explanation for the ‘additional’ pressure is that the pressure increase/decrease caused by the reduced/expanded volume of the cavity is accompanied by an increase/decrease of the temperature.3) At 293. where ( is the ratio of the specific heat at constant pressure to that at constant volume ( 1.2b) which shows that the equilibrium density of a gas can be written as p0 RT . not on the static pressure. the divergence of the gradient. c p0 RT . the fractional pressure variations in a small cavity driven by a vibrating piston. Note that the speed of sound of a gas depends only on the temperature. The fractional variations in the density are of course identical with the fractional change of the volume (except for the sign). equal the fractional density variations multiplied by the ratio of specific heats . and ρ is the equilibrium density of the medium.2.2. the absolute temperature T.3 kPa for air under normal ambient conditions).1) in a Cartesian coordinate system. As we shall see later the quantity c KS (1. Ks = γp0. p V . therefore.2.pressure p. The quantity Ks is the adiabatic bulk modulus. The adiabatic bulk modulus can also be expressed in terms of the gas constant R ( 287 J·kg1K1 for air). 4 The lefthand side of eq.2. Under normal ambient conditions (20°C. and the equilibrium density of the medium.2. The wave equation states that this high local pressure tends to decrease. Figure 1.204 kgm3.2.3 kPa) the density of air is 1.1) is the Laplacian of the sound pressure. say. (1. that is.4 we shall derive a relation between the volume velocity (= the volume displacement V per unit of time) and the resulting sound pressure.3 A small cavity driven by a vibrating piston.15 K = 20°C the speed of sound in air is 343 m/s. p0 V In section 1. For gases.2a) is the speed of sound. 3 . 101. (1. A negative value of this quantity at a certain point implies that the gradient converges towards the point. indicating a high local value.401 for air) and p0 is the static pressure ( 101. Adiabatic compression Because the process is adiabatic.4 The physical unit of the sound pressure is pascal (1 Pa = 1 Nm2). (1. takes the form 2 p 2 p 2 p 1 2 p x 2 y 2 z 2 c 2 t 2 (1. which increases/reduces the pressure even further. a pistonphone for calibrating microphones.
Fortunately. The density of liquids is also much higher. Both the density and the speed of sound depend on the static pressure and the temperature. 4 .1 Plane sound waves The plane wave is a central concept in acoustics.3). For example. It implies that a sinusoidal source will generate a sound field in which the pressure at all positions varies sinusoidally.5 This is an extremely important property. The boundary conditions are often expressed in terms of the particle velocity. 1. say at levels in excess of 140 dB. surfaces that reflect or absorb sound. (1. This information is known as the boundary conditions. which leads to the conclusion that we must supplement eq. the speed of sound in water is about 1500 ms1. Therefore we need an additional equation that relates the particle velocity to the sound pressure. they simply pass through each other (see figure 1.2. we can safely assume linearity under practically all circumstances encountered in daily life. Such waves can propagate in a duct. In a limited area at a distance far from a source of sound in free space the curvature of the spherical wavefronts is negligible and the waves can be regarded as locally plane.1) with some additional information about the sources that generate the sound field.2. Sound in liquids The speed of sound is much higher in liquids than in gases. unlike the sound pressure. the normal component of the particle velocity u is zero on a rigid surface. (1. For example.2.2.2.4 The sound pressure in a plane wave of arbitrary waveform at two different instants of time.The linearity of eq. Figure 1. Plane waves are waves in which any acoustic variable at a given time is a constant on any plane perpendicular to the direction of propagation. and there are no simple general relations corresponding to eqs.2. etc. z )).5).4) which is simply Newton’s second law of motion for a fluid.2. y .6 The diversity of possible sound fields is of course enormous. objects that scatter sound. the density of water is about 1000 kgm3.2b) and (1. 5 6 This follows from the fact that 2 ( p1 p2 ) t 2 2 p1 t 2 2 p2 t 2 . Note that the particle velocity is a vector.2. u p 0. This complicates the analysis enormously. It also implies linear superposition: sound waves do not interact. This relation is known as Euler’s equation of motion.1) is due to the absence of higherorder terms in p in combination with the fact that 2 x 2 and 2 t 2 are linear operators. which is a scalar. the linear approximation is no longer adequate. The operator is the gradient (the spatial derivative ( x . t (1. At very high sound pressure levels. (1.
2.The plane wave is a solution to the onedimensional wave equation. (1. x 2 c 2 t 2 (1. for example a loudspeaker driven with a pure tone. It is easy to show that the expression p f1 (ct x) f 2 (ct x).2. A harmonic plane wave propagating in the xdirection can be written p p1 sin (ct x) p1 sin(t kx ). is a solution to eq. Note that the wavelength is inversely proportional to the frequency.2. Figure 1. and at any fixed time the sound pressure varies sinusoidally with x with the spatial period c 2πc 2π . c.1). At 1000 Hz 5 .7) where 2πf is the angular (or radian) frequency and k c is the (angular) wavenumber.6) where f1 and f2 are arbitrary functions. (1.2. Harmonic waves are generated by sinusoidal sources. At any position in this sound field the sound pressure varies sinusoidally with the angular frequency ω.2. The quantity p1 is known as the amplitude of the wave.2. which is defined as the distance travelled by the wave in one cycle.2.2. and φ is a phase angle (the arbitrary value of the phase angle of the wave at the origin of the coordinate system at t = 0).5 Two plane waves travelling in opposite directions are passing through each other. Since the argument of f1 is constant if x increases as ct it follows that the first term of this expression represents a wave that propagates undistorted and unattenuated in the positive xdirection with constant speed. 2 p 1 2 p . The special case of a harmonic plane progressive wave is of great importance. and it can be shown this is the general solution.2.5).4 and 1. eq. See figures 1. f k (1.5.8) The quantity λ is the wavelength. whereas the second term represents a similar wave travelling in the opposite direction.5) cf. (1. c (1.
the operator t can be replaced by jω (because the derivative of ejωt with respect to time is jωejωt). (1.2. The ejωt convention is common in electrical engineering. As already mentioned. all that remains to be determined is the amplitude and phase at all positions. in audio and in related areas of acoustics. which leads to the conclusion that acousticians are faced with wavelengths (in air) in the range from 17 m at the lowest audible frequency to 17 mm at the highest audible frequency. physical.2.11) If the source emitted any other signal than a sinusoidal the waveform would in the general case change with the position in the sound field.2. It follows that Euler’s equation of motion can now be written ju p 0. It simplifies the analysis enormously if the wavelength is very long or very short compared with typical dimensions. linearity implies that a sinusoidal source with the frequency ω will generate a sound field that varies harmonically with this frequency at all positions.9) (where φ is the phase of the complex amplitude A). Thus at any given position the sound pressure can be written as a complex function of the form8 p A e jt A e j e jt A e j(t ) ˆ (1. Mathematicians and physicists also tend to prefer the symbol ‘i’ rather than ‘j’ for the imaginary unit. and the real. ˆ p Re p Re A e j(t ) A cos(t ).2. The former function takes account of the amplitude and phase. The sign of the argument of the exponential is just a convention. it can be realised that the wide frequency range is one of the challenges in acoustics. where the sound pressure is written as a complex function of the position multiplied with a complex exponential. physicists and acousticians concerned with outdoor sound propagation.10) Since the entire sound field varies as ejωt. With the alternative sign convention t should obviously be replaced by jω.6 The sound pressure in a plane harmonic wave at two different instants of time. Figure 1. Sound fields are often studied frequency by frequency. ˆ ˆ and the wave equation can be simplified to 7 (1. In rough numbers the audible frequency range goes from 20 Hz to 20 kHz. 6 .the wavelength in air is about 34 cm. Since the efficiency of a radiator of sound or the effect of an obstacle on the sound field depends very much on its size expressed in terms of the acoustic wavelength. because the various frequency components would change amplitude and phase relative to each other.9 and the operator 2 t 2 can be replaced by ω2. and the latter describes the time dependence. 8 9 Throughout this note complex variables representing harmonic signals are indicated by carets.7 Since the frequency is given. This explains the usefulness of harmonic analysis. timevarying sound pressure is the real part of the complex pressure. The alternative convention ejωt is favoured by mathematicians. This leads to the introduction of the complex exponential representation.
2 p 2 p 2 p ˆ ˆ ˆ 2 2 k 2 p 0, ˆ 2 x y z
(1.2.12)
which is known as the Helmholtz equation. See the Appendix (section 1.9) for further details about complex representation of harmonic signals. We note that the use of complex notation is mathematically very convenient, which will become apparent later. Written with complex notation the equation for a plane wave that propagates in the xdirection becomes
p pi e j(t kx ) . ˆ
(1.2.13)
Equation (1.2.11) shows that the particle velocity is proportional to the gradient of the pressure. It follows that the particle velocity in the plane propagating wave given by eq. (1.2.13) is
ux ˆ
1 p k p p ˆ ˆ . pi e j(t kx ) i e j(t kx ) j x c c
(1.2.14)
Thus the sound pressure and the particle velocity are in phase in a plane propagating wave (see also figure 1.2.10), and the ratio of the sound pressure to the particle velocity is ρc, the characteristic impedance of the medium. As the name implies, this quantity describes an important acoustic property of the fluid, as will become apparent later. The characteristic impedance of air at 20°C and 101.3 kPa is about 413 kg·m2s1.
Figure 1.2.7 A semiinfinite tube driven by a piston. Example 1.2.1 An semiinfinite tube is driven by a piston with the vibrational velocity Ue jt as shown in figure 1.2.7. Because the tube is infinite there is no reflected wave, so the sound field can be written
ˆ p( x) pi e j(t kx ) , ˆ u x ( x) pi j(t kx ) e . c
The boundary condition at the piston implies that the particle velocity equals the velocity of the piston: p ˆ u x (0) i e jt Ue jt . c It follows that the sound pressure generated by the piston is
ˆ p( x) U ce j(t kx ) .
The general solution to the onedimensional Helmholtz equation is
ˆ p pi e j(t kx ) pr e j(t kx ) ,
7
(1.2.15)
which can be identified as the sum of a wave that travels in the positive xdirection and a wave that travels in the opposite direction (cf. eq. (1.2.6)). The corresponding expression for the particle velocity becomes, from eq. (1.2.11),
ux ˆ 1 p k k ˆ pi e j(t kx ) pr e j(t kx ) j x p p i e j(t kx ) r e j(t kx ) . c c
(1.2.16)
It can be seen that whereas p u x c in a plane wave that propagates in the positive xˆ ˆ direction, the sign is the opposite, that is, p u x c , in a plane wave that propagates in the ˆ ˆ negative xdirection. The reason for the change in the sign is that the particle velocity is a vector, unlike the sound pressure, so u x is a vector component. It is also worth noting that the ˆ general relation between the sound pressure and the particle velocity in this interference field is far more complicated than in a plane propagating wave.
Figure 1.2.8 Instantaneous sound pressure in a wave that is reflected from a rigid surface at different instants of time. (Adapted from ref. [2].)
A plane wave that impinges on a plane rigid surface perpendicular to the direction of propagation will be reflected. This phenomenon is illustrated in figure 1.2.8, which shows how an incident transient disturbance is reflected. Note that the normal component of the gradient of the pressure is identically zero on the surface for all values of t. This is a consequence of the fact that the boundary condition at the surface implies that the particle velocity must equal zero here, cf. eq. (1.2.4). However, it is easier to analyse the phenomenon assuming harmonic waves. In this case the sound field is given by the general expressions (1.2.15) and (1.2.16), and our task is to determine the relation between pi and pr from the boundary condition at the surface, say at x = 0. As mentioned, the rigid surface implies that the particle velocity must be zero here, which with eq. (1.2.16) leads to the conclusion that pi = pr , so the reflected wave has the same amplitude as the incident wave. Equation (1.2.15) now becomes p pi e j(t kx ) e j(t kx ) pi e j kx e j kx e jt 2 pi cos kx e jt , ˆ and eq. (1.2.16) becomes 8 (1.2.17)
ux j ˆ
2 pi sin kx e jt . c
(1.2.18)
Note that the amplitude of the sound pressure is doubled on the surface (cf. figure 1.2.8). Note also the nodal10 planes where the sound pressure is zero at x =  λ/4, x =  3λ/4, etc., and the planes where the particle velocity is zero at x =  λ/2, x =  λ, etc. The interference of the two plane waves travelling in opposite directions has produced a standing wave pattern, shown in figure 1.2.9. The physical explanation of the fact that the sound pressure is identically zero at a distance of a quarter of a wavelength from the reflecting plane is that the incident wave must travel a distance of half a wavelength before it returns to the same point; accordingly the incident and reflected waves are in antiphase (that is, 180° out of phase), and since they have the same amplitude they cancel each other. This phenomenon is called destructive interference. At a distance of half a wavelength from the reflecting plane the incident wave must travel one wavelength before it returns to the same point. Accordingly, the sound pressure is doubled here (constructive interference). The corresponding pattern for the particle velocity is different because the particle velocity is a vector. Another interesting observation from eqs. (1.2.17) and (1.2.18) is that the resulting sound pressure and particle velocity signals at any position are 90° out of phase (otherwise expressed, if the sound pressure as a function of time is a cosine then the particle velocity is a sine). As we shall see later this indicates that there is no net flow of sound energy towards the rigid surface. See also figure 1.2.10.
Figure 1.2.9 Standing wave pattern caused by reflection from a rigid surface at x = 0; amplitudes of the sound pressure and the particle velocity. Example 1.2.2 The standing wave phenomenon can be observed in a tube terminated by a rigid cap. When the length of the tube, l, equals an oddnumbered multiple of a quarter of a wavelength the sound pressure is zero at the input, which means that it would take very little force to drive a piston here. This is an example of an acoustic resonance. In this case it occurs at the frequency
f0 c , 4l
A node on, say, a vibrating string is a point that does not move, and an antinode is a point with maximum displacement. By analogy, points in a standing wave at which the sound pressure is identically zero are called pressure nodes. In this case the pressure nodes coincide with velocity antinodes.
10
9
We tend to determine the pitch on the basis of the spacing between the harmonic components. indicating partial reflection with a phase shift at the reflection plane.10 shows the instantaneous sound pressure and particle velocity at two different instants of time in a tube that is terminated by a material that does not reflect sound at all (case (a)). closed organ pipes are tubes closed at one end and driven at the other. open end.21) it follows that s (1.20b) and when they are in antiphase the sound pressure amplitude assumes the minimum value R s 1 . (1. However. usually consisting of the fundamental and a number of its harmonics. The relative position of a tone on a musical scale is called the pitch [2]. (1. as we have seen. and by a rigid material that gives perfect reflection (case (c)). The ratio of pr to pi is the (complex) reflection factor R. If we introduce the reflection factor in eq. pitch is a subjective phenomenon and not completely equivalent to frequency. Note that the resonances are harmonically related.20a) (1. and thus we can detect the pitch of a musical tone even if the fundamental is missing.19) from which it can be seen that the amplitude of the sound pressure varies with the position in the sound field. See also chapter 1.2.and at oddnumbered multiples of this frequency.2.2. and the fundamental is the first harmonic.21) (1. The pitch of a musical tone essentially corresponds to its fundamental frequency. However. The amplitude of this quantity describes how well the reflecting surface reflects sound. The ratio of pmax to pmin is called the standing wave ratio. also called partials. s 1 (1. 11 10 . The n’th harmonic (or partial) is also called the (n1)’th overtone. 5f0. For example.5. This means that if some mechanism excites the tube the result will be a musical sound with the fundamental frequency f and overtones corresponding to oddnumbered harmonics. pmin 1 R From eq. 3f0. (1. which is also the distance between two adjacent harmonic components. 7f0. A musical (or complex) tone is not a pure (sinusoidal) tone but a periodic signal. by a soft material that partly absorbs the incident sound wave (case (b)). pmin pi 1 R .2. pmax pi 1 R . See also example 1.4. etc.4.22) which leads to the conclusion that it is possible to determine the acoustic properties of a material by exposing it to normal sound incidence and measuring the standing wave ratio in the resulting interference field.15) it becomes ˆ p pi e j(t kx ) R e j(t kx ) . in the general case of a more or less absorbing surface R will be complex and its magnitude less than unity (R ≤ 1). In the case of a rigid plane R = 1. These pure tones occur at multiples of the fundamental frequency. which implies perfect reflection with no phase shift. and such pipes have only oddnumbered harmonics.2.2.2.11 Brass and woodwind instruments are based on standing waves in tubes. pmax 1 R . When the two terms in the parenthesis are in phase the sound pressure amplitude assumes its maximum value.2. Figure 1.
2. (b) case with partial reflection from a soft surface. (From ref.11.) Sound transmission between fluids When a sound wave in one fluid is incident on the boundary of another fluid. Anticipating a reflected wave we can write ˆ p1 pi e j(t kx ) pr e j(t kx ) for fluid 1. (a) Case with no reflection (R = 0).2. For simplicity let us assume that a plane wave in fluid 1 strikes the surface of fluid 2 at normal incidence as shown in figure 1.10 Spatial distributions of instantaneous sound pressure and particle velocity at two different instants of time. it will be partly reflected and partly transmitted. a sound wave in air is incident on the surface of water. say.Figure 1. and ˆ p2 pt e j(t kx ) 11 . (c) case with perfect reflection from a rigid surface (R = 1). [3].
It follows that pi pr pt and p pi pr t . and the particle velocity must be the same in fluid 1 and in fluid 2 (otherwise the fluids would not remain in contact). 12 .2.2. and not reflected at all if ρ2c2 = ρ1c1. pi 2 c2 1c1 which shows that the wave is almost fully reflected in phase ( R 1 ) if ρ2c2 >> ρ1c1.) Figure 1. amplitudes of the sound pressure and the particle velocity. 1c1 2 c2 Combining these equations gives c 1c1 pr R 2 2 . There are two boundary conditions at the interface: the sound pressure must be the same in fluid 1 and in fluid 2 (otherwise there would be a net force). almost fully reflected in antiphase ( R 1 ) if ρ2c2 << ρ1c1. ρ1 and ρ2.12 Reflection of a pressure wave at the interface between a medium of high characteristic impedance and a medium of low characteristic impedance. (Adapted from ref.2. irrespective of the individual properties of c1. [2]. c2. Figure 1.11 Reflection and transmission of a plane wave incident on the interface between two fluids. Figure 1.for fluid 2.13 Standing wave pattern in a medium of high characteristic impedance caused by caused by reflection from a medium of low characteristic impedance.
9 and 1.2.2. becomes p x p . 2 p 1 p 1 p 1 p x 2 1 p 1 p x 2 2 p x 2 p x .2. (1.13. Since the sound pressure depends only on r we have p p r .23a). with r x2 y2 z 2 . eq. θ. However.5).2. 2 r 2 c t 2 (1.2. and so is a sound wave that strikes the airwater interface from the water.2. The resulting equation is more complicated than eq. x 2 r r x r r r r r r r r r r r 2 r 2 r 3 r Combining eq.2 Spherical sound waves The wave equation can be expressed in other coordinate systems than the Cartesian. (It is easy to get from eq.23b) it becomes apparent that this equation is identical in form with the onedimensional wave equation. r 2 r r c 2 t 2 (1. r r r r 2 c 2 t 2 x 2 y 2 z 2 r r r2 r 2 r3 13 . (1.2.2. Compare figures 1.23a): 2 p 2 p 2 p 3 p x 2 y 2 z 2 2 p x 2 y 2 z 2 p 2 p 2 p 1 2 p . (1. x r x which. x r r Similar considerations leads to the following expression for the secondorder derivative. (1. if the source under study is spherically symmetric there can be no angular dependence.1) with this expression and the corresponding relations for y and z finally yields eq. as shown in figure 1. it is more difficult the other way.8 and 1. φ).2.Because of the significant difference between the characteristic impedances of air and water (the ratio is about 1: 3600) a sound wave in air that strikes a surface of water at normal incidence is almost completely reflected.12 2 p 2 p 1 2 p . although p has been replaced by rp.2. (1.2.23b) to eq.2. and the equation becomes quite simple.12.12.24a) 12 This can be seen as follows. (1. If sound is generated by a source in an environment without reflections (which is usually referred to as a free field) it will generally be more useful to express the wave equation in a spherical coordinate system (r.23) can be written rp f1 (ct r ) f 2 (ct r ). and figures 1. (1.2.) It follows that the general solution to eq.23a) If we rewrite in the form 2 (rp ) 1 2 (rp) . 1.2.2.1).2. but in the latter case the phase of the reflected wave is reversed. (1.
.14 (a) Measurement far from a spherical source in free space. instantaneous particle velocity multiplied by Dc. the first term of eq. the sound pressure is seen to decrease in inverse proportion to the distance. [4].6. Accordingly we will ignore the second term when we study sound radiation in chapter 1. . The first term is wave that travels outwards.2.5 and 1.25) e j(t kr ) . r (1.2.6. See chapters 1.26) The inverse distance law is also known as the inverse square law because the sound intensity is inversely proportional to the square of the distance to the source. (b) measurement close to a spherical source. Note that the shape of the wave is preserved. r (1. In principle such a wave could be generated by a reflecting spherical surface centred at the source. a spherical wave travelling inwards.2. This is the inverse distance law.24b) where f1 and f2 are arbitrary functions. ––. but that is a rare phenomenon indeed. 14 .13 The second term represents a converging wave.2.2. Instantaneous sound pressure. (1.) A harmonic spherical wave is a solution to the Helmholtz equation 2 ( rp ) ˆ k 2 rp 0. that is.. However. (From ref. Figure 1. ˆ 2 r Expressed in the complex notation the diverging wave can be written pA ˆ 13 (1.that is p 1 f1 (ct r ) f 2 (ct r ) .6)). away from the source (cf.
a frequency weighting filter16 is usually applied to the signal before the rms value is determined.3. that is. refs.cos2ωt) and thus has a time average of ½ leads to the conclusion that the rms value of a sinusoidal signal with the amplitude A is A / 2 .27) Because of the spherical symmetry there are no components in the other directions.The particle velocity component in the radial direction can be calculated from eq. This topic is outside the scope of this note. 15 .1 The fact that sin2ωt = ½ (1 .1) However.3 ACOUSTIC MEASUREMENTS The most important measure of sound is the rms sound pressure. 0 T T 2 ½ (1.15 defined as prms 1 T p (t ) lim p 2 (t )dt .11). The dimensionless quantity kr is known as the Helmholtz number.2. so that ‘far from’ means that r >> λ (or kr >> 1). Root mean square value.2. usually abbreviated rms. and therefore the rms sound pressure is determined in frequency bands. 90° out of phase. 1. On the other hand. just as ‘near’ means that r << λ (or kr << 1). but see. Alternatively. 1. Quite often such a single value does not give sufficient information about the nature of the sound.3. which is the time average of the squared signal. See figure 1. dimensions are measured in terms of the wavelength. as we shall see. j r c r jkr (1.1 Frequency analysis Single frequency sound is useful for analysing acoustic phenomena. This is the square root of the mean square value. Note that far14 from the source the sound pressure and the particle velocity are in phase and their ratio equals the characteristic impedance of the medium. Example 1.g. Frequency (or spectral) analysis of a signal involves decomposing the signal into its spectral components.3. If the sound is more or less steady. but most sounds encountered in practice have ‘broadband’ characteristics. ur ˆ 1 p A e j(t kr ) 1 p 1 ˆ ˆ 1 c 1 jkr . which means that they cover a wide frequency range. the signal can be passed through a number of 14 In acoustics. [5. e. just as in a plane wave. These are near field characteristics. The resulting sound pressures are practically always compressed logarithmically and presented in decibels. (1. when kr << 1 the particle velocity is larger than p c and the sound pressure and the ˆ particle velocity are almost in quadrature. This analysis can be carried out by means of digital analysers that employ the discrete Fourier transform (‘FFT analysers’).14. it will practically always be more useful to analyse it in the frequency domain than to look at the sound pressure as a function of time. and such a sound field is also known as a reactive field..2. 6]. 16 15 A filter is a device that modifies a signal by attenuating some of its frequency components.
⅓ 17 18 19 Round numbers are convenient. Thus an octave spans eight notes. 1.6. f l f c 2½ . ten octaves or thirty onethird octaves. Figure 1. (1.g. from C to C'. and therefore frequencies are often represented on a logarithmic scale in acoustics. and frequency analysis is often carried out with constant percentage filters. and a onethird octave is almost identical with one tenth of a decade.2b. 1 fc fl fu . Real filters have.2a.3.25. that is. of course.1 The keyboard of a small piano. Constant relative bandwidth corresponds to uniform resolution on a logarithmic frequency scale. (Adapted from ref.5. 8..3. 2.20 As mentioned earlier. the lower limiting frequency of an octave band is half the upper frequency limit. with n = 1. 1. 4... The most common filters in acoustics are octave band filters and onethird octave band filters. 1. 1. hence the name octave (from Latin octo: eight). The standardised nominal centre frequencies are based on the fact that the series 1. say. onethird octave band filters are called Terzfilters in German. Accordingly. 1 fu 2 6 fc .. In a similar manner a onethird octave19 band is a band for which fu = 2⅓ fl . Table 1. [7].1).3.3b. 5.3a.3c) Since 210 1024 103 it follows that 210 3 10 and 21 3 101 10 .3. a certain passband ripple and a finite stopband attenuation. fc fl fu ..’ The filters can have the same bandwidth or they can have constant relative bandwidth. Since 2 = 24/12 it can be seen that a onethird octave is identical with four semitones or a major third (e. which means that the bandwidth is a certain percentage of the centre frequency. but would completely remove frequency components outside the passband. and fl fc 2 6 . and the centre frequency is the geometric mean. The diatonic scale contains seven notes per octave corresponding to the white keys on a piano keyboard. ten onethird octaves very nearly make a decade. a ‘filter bank.2c) where fc is the centre frequency. a range of three decades. 20 16 .) An octave18 is a frequency ratio of 2:1. 3. 2.3. 10 is in reasonable agreement with 10n/10. Accordingly. 10.3. see figure 1.3. the human ear may respond to frequencies in the range from 20 Hz to 20 kHz.3. A semitone is one twelfth of an octave on the equally tempered scale (a frequency ratio of 21/12:1). a fundamental unit in musical scales. that is. The white keys from C to B correspond to the seven notes of the C major scale. cf. An ideal bandpass filter would allow frequency components in the passband to pass unattenuated. that is. 2.1..3. f u 2½ f c .3.3. from C to E. Musical tones an octave apart sound very similar. 6.contiguous analogue or digital bandpass filters17 with different centre frequencies. (1.15. figure 1.1 gives the nominal centre frequencies of standardised octave and onethird octave band filters. 1. Such a scale is in much better agreement with the subjective pitch of musical sounds than a linear scale.
i (1. so that all cross terms vanish. 20 25 31.3.i is the rms value of the output of the i’th filter. say.4) Note that the mean square values of the two signals are added unless ω1 = ω2. which can be determined with a parallel bank of contiguous filters. constant mean square value per hertz.3. the different frequency components are uncorrelated signals. e.g.5) is known as Parseval’s formula. When white noise is passed through a bandpass filter. It follows that when white noise is analysed with constant percentage filters. When several independent sources of noise are present at the same time the mean square sound pressures generated by the individual sources are additive. ( A sin 1t B sin 2t ) 2 A2 sin 2 1t B 2 sin 2 2t 2 ABsin 1t sin 2t ( A2 B 2 ) / 2. octave bands. the mean square value of any stationary signal equals the sum of mean square values of its frequency components. Example 1. hence the name pink noise. Equation (1. less and less noise passes through the filter. the mean square values of the output signals of two adjacent filters differ by a factor of two. It follows that the mean square value of a given pink noise signal in octave bands is three times larger than the mean square value of the noise in onethird octave bands.3.1 Standardised onethird octave and octave (bold characters) band centre frequencies (in hertz). the mean square of the output signal is proportional to the bandwidth of the filter. unlike periodic signals. therefore FFT analysers are particular suitable for detecting tones in noise.. (1.3. Compared with filter bank analysers FFT analysers have the advantage that the spectral resolution can be varied over a wide range [6]. Random noise Many generators of sound produce noise rather than pure tones. Pink noise is stationary noise with constant mean square value in bands with constant relative width.5) where prms. The term white noise is an analogy to white light. if white noise is analysed with a bank of octave band filters.5 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1000 1250 1600 2000 2500 3150 4000 5000 6300 8000 10000 12500 16000 20000 An important property of the mean square value of a signal is that it can be partitioned into frequency bands. Thus compared with white noise low frequencies are emphasised. This is due to the 17 . which is an analogy to an optical phenomenon. The reason is that products of different frequency components average to zero. This means that if we analyse a signal in. Whereas pure tones and other periodic signals are deterministic. Thus 2 2 prms prms. This can be illustrated by analysing a sum of two pure tones with different frequencies. The validity of this rule is not restricted to pure tones of different frequency. onethird octave bands. the sum of the mean square values of the filtered signals equals the mean square value of the unfiltered signal. noise is a stochastic or random phenomenon. has a finite power spectral density (mean square value per hertz) implies that one can detect a pure tone in noise irrespective of the signaltonoise ratio by analysing with sufficiently fine spectral resolution: As the bandwidth is reduced.i .Table 1.2 The fact that noise. White noise is stationary noise with a flat power spectral density. the mean square of the output is proportional to the centre frequency of the filter.3. For example. that is. Stationary noise is a stochastic signal with statistical properties that do not change with time. and the tone will emerge.
fact that independent sources generate uncorrelated signals, that is, signals whose product average to zero; therefore the cross terms vanish:
2 2 ( p1 (t ) p2 (t )) 2 p12 (t ) p2 (t ) 2 p1 (t ) p2 (t ) p12 (t ) p2 (t ).
(1.3.6)
It follows that
2 2 prms, tot prms,i .
(1.3.7)
i
Note the similarity between eqs. (1.3.5) and (1.3.7). It is of enormous practical importance that the mean square values of uncorrelated signals are additive, because signals generated by different mechanisms are invariably uncorrelated. Almost all signals that occur in real life are mutually uncorrelated.
Example 1.3.3 Equation (1.3.7) leads to the conclusion that the mean square pressure generated by a crowd of noisy people in a room is proportional to the number of people. Thus the rms value of the sound pressure in the room is proportional to the square root of the number of people. Example 1.3.4 Consider the case where the rms sound pressure generated by a source of noise is to be measured in the presence of background noise that cannot be turned off. It follows from eq. (1.3.7) that it is possible to correct the measurement for the influence of the stationary background noise; one simply subtracts the mean square value of the background noise from the total mean square pressure. For this to work in practice the background noise must not be too strong, though, and it is absolutely necessary that it is completely stationary.
1.3.2 Levels and decibels The human auditory system can cope with sound pressure variations over a range of more than a million times. Because of this wide range, the sound pressure and other acoustic quantities are usually measured on a logarithmic scale. An additional reason is that the subjective impression of how loud noise sounds correlates much better with a logarithmic measure of the sound pressure than with the sound pressure itself. The unit is the decibel,21 abbreviated dB, which is a relative measure, requiring a reference quantity. The results are called levels. The sound pressure level (sometimes abbreviated SPL) is defined as
L p 10 log10
2 prms p 20 log10 rms , 2 pref pref
(1.3.8)
where pref is the reference sound pressure, and log10 is the base 10 logarithm, henceforth written log. The reference sound pressure is 20 μPa for sound in air, corresponding roughly to the lowest audible sound at 1 kHz.22 Some typical sound pressure levels are given in figure 1.3.2.
As the name implies, the decibel is one tenth of a bel. However, the bel is rarely used today. The use of decibels rather than bels is probably due to the fact that most sound pressure levels encountered in practice take values between 10 and 120 when measured in decibels, as can be seen in figure 1.3.2. Another reason might be that to be audible, the change of the level of a given (broadband) sound must be of the order of one decibel. For sound in other fluids than atmospheric air (water, for example) the reference sound pressure is 1 :Pa. To avoid possible confusion it may be advisable to state the reference sound pressure explicitly, e.g., ‘the sound pressure level is 77 dB re 20 μPa.’
22 21
18
Figure 1.3.2 Typical sound pressure levels. (Source: Brüel & Kjær.)
19
The fact that the mean square sound pressures of independent sources are additive (cf. eq. (1.3.7)) leads to the conclusion that the levels of such sources are combined as follows: 0.1L L p ,tot 10 log 10 p ,i . i (1.3.9)
Another consequence of eq. (1.3.7) is that one can correct a measurement of the sound pressure level generated by a source for the influence of steady background noise as follows: L p ,source 10 log 10
0.1L p ,tot
10
0.1L p ,background
.
(1.3.10)
This corresponds to subtracting the mean square sound pressure of the background noise from the total mean square sound pressure as described in example 1.3.5. However, since all measurements are subject to random errors, the result of the correction will be reliable only if the background level is at least, say, 3 dB below the total sound pressure level. If the background noise is more than 10 dB below the total level the correction is less than 0.5 dB.
Example 1.3.5 Expressed in terms of sound pressure levels the inverse distance law states that the level decreases by 6 dB when the distance to the source is doubled. Example 1.3.6 When each of two independent sources in the absence of the other generates a sound pressure level of 70 dB at a certain point, the resulting sound pressure level is 73 dB (not 140 dB!), because 10 log 2 3 . If one source creates a sound pressure level of 65 dB and the other a sound pressure level of 59 dB, the total level is 10 log(106.5 105.9 ) 66 dB . Example 1.3.7 Say the task is to determine the sound pressure level generated by a source in background noise with a level of 59 dB. If the total sound pressure level is 66 dB, it follows from eq. (1.3.10) that the source would have produced a sound pressure level of 10 log(106.6 105.9 ) 65 dB in the absence of the background noise. Example 1.3.8 When two sinusoidal sources emit pure tones of the same frequency they create an interference field, and depending on the phase difference the total sound pressure amplitude at a given position will assume a value between the sum of the two amplitudes and the difference:
A B Ae jt Be jt A B A e jA B e jB A B . For example, if two pure tone sources of the same frequency each generates a sound pressure level of 70 dB in the absence of the other source then the total sound pressure level can be anywhere between 76 dB (constructive interference) and  ∞ dB (destructive interference). Note that eqs. (1.3.7) and (1.3.9) do not apply in this case because the signals are not uncorrelated. See also figure 1.9.2 in the Appendix.
Other firstorder acoustic quantities, for example the particle velocity, are also often measured on a logarithmic scale. The reference velocity is 1 nm/s = 109 m/s.23 This reference is also used in measurements of the vibratory velocities of vibrating structures. The acoustic secondorder quantities sound intensity and sound power, defined in chapter 1.5, are also measured on a logarithmic scale. The sound intensity level is
23
The prefix n (for ‘nano’) represents a factor of 109.
20
4). the level increases 1 dB from one band to the next (10 log(21 3 ) 1dB) . highly tensioned foil. Pref (1. and this produces an electrical voltage proportional to the instantaneous sound pressure.3 shows a block diagram of a simple sound level meter.12) where Pa is the sound power and Pref = 1 pW.3 A sound level meter. In the following a very brief description of such an instrument will be given. placed very close to the diaphragm (see figure 1.3 Noise measurement techniques and instrumentation A sound level meter is an instrument designed to measure sound pressure levels. When the diaphragm moves in response to the sound pressure. which is amplified and passes through various filters. in agreement with the fact that if the linear quantities are doubled then quantities of second order are quadrupled. (1. see e. whereas levels of secondorder (quadratic) quantities are defined as ten times the logarithm. 1.3. (From ref.3. Today such instruments can be anything from fairly simple devices with analogue filters and detectors and a moving coil meter to advanced digital analysers. which are more stable and accurate than other types. Example 1. the capacitance changes. the back plate.24 and the sound power level is LW 10 log Pa . The most commonly used microphones for this purpose are condenser microphones.3. and the result is finally converted to decibels and shown on a display. The microphone converts the sound pressure to an electrical signal. refs. [8. Inside the housing of the microphone cartridge is the other part of the capacitor.3. 21 .11) where I is the intensity and Iref = 1 pWm2 = 1012 Wm2.) 24 The prefix p (for ‘pico’) represents a factor of 1012. 9] for further details. Note than levels of linear quantities (pressure. Figure 1.9 It follows from the constant spectral density of white noise that when such a signal is analysed in onethird octave bands. Figure 1.3.g. After this the signal is squared and averaged with a detector. [10]. either by an external voltage on the back plate or (in case of prepolarised electret microphones) by properties of the diaphragm or the back plate.3. particle velocity) are defined as twenty times the logarithm of the ratio of the rms value to a reference value.3. The diaphragm of a condenser microphone is a very thin.LI 10 log I I ref . The capacitor is electrically charged.
) Figure 1. and typical measurement microphones are ‘½inch’ microphones with a diameter 22 . [11].3. However. The freefield correction is the fractional increase of the sound pressure (usually expressed in dB) caused by the presence of the microphone in the sound field.6 Freefield response of a microphone of the ‘freefield’ type at axial incidence. this is in conflict with the requirement of a high sensitivity and a low inherent noise level.) The microphone should be as small as possible so as not to disturb the sound field.5 The ‘freefield correction’ of a typical measurement microphone for sound coming from various directions. (From ref.) Figure 1. (From ref. [11]. (From ref.3.Figure 1.3. [11].4 A condenser microphone.
3. much less sensitive to low frequencies than to medium frequencies. from a few kilohertz and upwards the size of the microphone is no longer negligible compared with the wavelength.3.25 If no weighting filter is applied. ‘Randomincidence’ microphones are designed for measurements in a diffuse sound field where sound is arriving from all directions. but it has long ago been realised that the human auditory system is far more complicated than implied by such simple weighting curves. which means that its response varies with the nature of the sound field. at low levels.) 25 In practice the unit is often written dB (A) and dB (C).and Cweighted sound pressure level are denoted LA and LC respectively.7 Standardised frequency weighting curves. The original intention was to simulate a human ear at various levels. is sometimes used in combination with Aweighting. However. On the other hand the Aweighted sound pressure level is the most widely used singlevalue measure of sound.5. the level is sometimes denoted LZ. The results of measurements of the A. because the Aweighted sound pressure level correlates in general much better with the subjective effect of noise than measurements of the sound pressure level with a flat frequency response. and therefore it is no longer omnidirectional. In particular the human ear is. say below 1 kHz.6). This is the background for the standardised frequency weighting filters shown in figure 1. and ‘pressure’ microphones are intended for measurements in small cavities.3. ‘freefield’ microphones are designed to have a flat response for axial incidence (see figure 1. because a large difference between the Aweighted level and the Cweighted level is a clear indication of a prominent content of low frequency noise. and such microphones should therefore be pointed towards the source. At low frequencies. such a microphone is much smaller than the wavelength and does not disturb the sound field appreciably. Figure 1. In this frequency range the microphone is omnidirectional as of course it should be since the sound pressure is a scalar and has no direction.3. [8]. The sensitivity of the human auditory system varies significantly with the frequency in a way that changes with the level. see figure 1. which is essentially flat in the audible frequency range.of about 13 mm. 23 . and the unit is dB. For example. respectively. Cweighting. (From ref. and Band Dweighting filters are little used today. One can design condenser microphones to have a flat response in as wide a frequency range as possible under specified sound field conditions.7.
1 16.5 0.0 14. The alternative to exponential averaging is linear (or integrating) averaging.0 0.3 0. The lowpass filter corresponds to applying a time weighting function.5 6.1 13.3 6.5 4.3.0 0. The equivalent sound pressure level is defined as 24 .0 0.0 0.2 The response of standard A.0 4.2 1.2 0.2 22.0 1.0 0.0 0.3 0.3 0.2 1.2 26.2 In the measurement instrument the frequency weighting filter is followed by a squaring device.6 4.4 56.6 9.1 0. a lowpass filter that smooths out the instantaneous fluctuations.3.1 1.0 0.3 2.5 0.4 10. Centre frequency (Hz) 8 10 12. and F (for ‘fast’) is exponential averaging with a time constant of 125 ms. and a logarithmic converter.0 0.2 1.0 0.5 19.6 30.1 0.5 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1000 1250 1600 2000 2500 3150 4000 5000 6300 8000 10000 12500 16000 20000 Aweighting (dB) 77. which implies that the squared signal is smoothed with a decaying exponential so that recent data are given more weight than older data: 1 t 2 L p (t ) 10 log p 2 (u ) e (t u ) / du pref . in which all the sound is weighted uniformly during the integration.9 0.4 3.2 0.Table 1.2 8.9 8.2 4. (1.6 6.7 39.4 63.0 1.13) Two values of the time constant τ are standardised: S (for ‘slow’) corresponds to a time constant of 1 s.8 1.7 50.0 0.8 70.5 0.8 0.0 2.5 11.4 34.4 6.3 1.3 Cweighting (dB) 20.3 11.0 3.8 0.6 1. The most common time weighting in sound level meters is exponential.and Cweighting filters in onethird octave bands.5 16 20 25 31.1 2.5 44.0 0.8 3.2 8.
1 t2 2 2 Leq 10 log t1 p (t )dt pref . t t 2 1
(1.3.14)
Measurements of random noise with a finite integration time are subject to random errors that depend on the bandwidth of the signal and on the integration time. It can be shown that the variance of the measurement result is inversely proportional to the product of the bandwidth and the integration time [6].26 As can be seen by comparing with eqs. (1.3.1) and (1.3.8), the equivalent sound pressure level is just the sound pressure level corresponding to the rms sound pressure determined with a specified integration period. The Aweighted equivalent sound pressure level LAeq is the level corresponding to a similar time integral of the Aweighted instantaneous sound pressure. Sometimes the quantity is written LAeq,T where T is the integration time. Whereas exponential averaging corresponds to a running average and thus gives a (smoothed) measure of the sound at any instant of time, the equivalent sound pressure level (with or without Aweighting) can be used for characterising the total effect of fluctuating noise, for example noise from road traffic. Typical values of T are 30 s for measurement of noise from technical installations, 8 h for noise in a working environment and 24 h for traffic noise. Sometimes it is useful to analyse noise signals in onethird octave bands, cf. section 1.3.1. From eq. (1.3.5) it can be seen that the total sound pressure level can be calculated from the levels in the individual onethird octave bands, Li, as follows: LZ 10 log 100.1Li . (1.3.15) i In a similar manner one can calculate the Aweighted sound pressure level from the onethird octave band values and the attenuation data given in table 1.3.2, LA 10 log 100.1( Li K i ) , i (1.3.16)
where Ki is the relative response of the Aweighting filter (in dB) in the i’th band, given in table 1.3.2.
Example 1.3.10 A source gives rise to the following onethird octave band values of the sound pressure level at a certain point,
Centre frequency (Hz) 315 400 500 630 800
26
Sound pressure level (dB) 52 68 76 71 54
In the literature reference is sometimes made to the equivalent integration time of exponential detectors. This is two times the time constant (e.g. 250 ms for ‘F’), because a measurement of random noise with an exponential detector with a time constant of τ has the same statistical uncertainty as a measurement with linear averaging over a period of 2τ [9].
25
and less than 50 dB in all the other bands. It follows that
LZ 10 log 105.2 106.8 107.6 107.1 105.4 77.7 dB,
and
LA 10 log 10(5.2 0.66) 10(6.8 0.48) 10(7.6 0.32) 10(7.1 0.19) 10(5.4 0.08) 74.7 dB.
Noise that changes its level in a regular manner is called intermittent noise. Such noise could for example be generated by machinery that operates in cycles. If the noise occurs at several steady levels, the equivalent sound pressure level can be calculated from the formula
t Leq,T 10log i 100.1Li . i T
(1.3.17)
This corresponds to adding the mean square values with a weighting that reflects the relative duration of each level.
Example 1.3.11 The Aweighted sound pressure level at a given position in an industrial hall changes periodically between 84 dB in intervals of 15 minutes, 95 dB in intervals of 5 minutes and 71 dB in intervals of 20 minutes. From eq. (1.3.17) it follows that the equivalent sound pressure level over a working day is
5 20 15 LAeq 10 log 108.4 109.5 107.1 87.0 dB. 40 40 40
Most sound level meters have also a peak detector for determining the highest absolute value of the instantaneous sound pressure (without filters and without time weighting), ppeak. The peak level is calculated from this value and eq. (1.3.8) in the usual manner, that is, L p 20 log ppeak pref . (1.3.18)
Example 1.3.12 The crest factor of a signal is the ratio of its peak value to the rms value (sometimes expressed in dB). From example 3.1 it follows that the crest factor of a pure tone signal is 2 or 3 dB.
The sound exposure level (sometimes abbreviated SEL) is closely related to LAeq, but instead of dividing the time integral of the squared Aweighted instantaneous sound pressure by the actual integration time one divides by t0 = 1 s. Thus the sound exposure level is a measure of the total energy27 of the noise, normalised to 1 s:
In signal analysis it is customary to use the term ‘energy’ in the sense of the integral of the square of a signal, without regard to its units. This should not be confused with the potential energy density of the sound field introduced in chapter 1.5.
27
26
1 LAE 10 log t 0
t2 t1
2 2 pA (t )dt pref
(1.3.19)
This quantity is used for measuring the total energy of a ‘noise event’ (say, a hammer blow or the take off of an aircraft), independently of its duration. Evidently the measurement interval should encompass the entire event.
Example 1.3.13 It is clear from eqs. (1.3.14) and (1.3.19) that LAeq,T of a noise event of finite duration decreases with the logarithm of T if the T exceeds its duration:
1 LAeq,T 10 log T
T 2 2 pA (t )dt pref LAE 10 log . t0
Example 1.3.14 If n identical noise events each with a sound exposure level of LAE occur within a period of T (e.g., one working day) then the Aweighted equivalent sound level is
LAeq,T LAE 10 log n 10 log T , t0
because the integrals of the squared signals are additive; cf. eq. (1.3.7).28
1.4 THE CONCEPT OF IMPEDANCE
By definition an impedance is the ratio of the complex amplitudes of two signals representing cause and effect, for example the ratio of an AC voltage across a part of an electric circuit to the corresponding current, the ratio of a mechanical force to the resulting vibrational velocity, or the ratio of the sound pressure to the particle velocity. The term has been coined from the verb ‘impede’ (obstruct, hinder), indicating that it is a measure of the opposition to the flow of current etc. The reciprocal of the impedance is the admittance, coined from the verb ‘admit’ and indicating lack of such opposition. Note that these concepts require complex representation of harmonic signals; it makes no sense to divide, say, the instantaneous sound pressure with the instantaneous particle velocity. There is no simple way of describing properties corresponding to a complex value of the impedance without the use of complex notation. The mechanical impedance is perhaps simpler to understand than the other impedance concepts, since it is intuitively clear that it takes a certain vibratory force to generate mechanical vibrations. The mechanical impedance of a structure at a given point is the ratio of the complex amplitude of a harmonic point force acting on the structure to the complex amplitude of the resulting vibratory velocity at the same point,29
Strictly speaking this requires that the instantaneous product of the ‘event’ and any of its time shifted versions time average to zero. In practice this will always be the case. Note that the sign of the imaginary part of the impedance changes if the eiωt convention is used instead of the ejωt convention. Cf. footnote no 9 on p. 6.
29
28
27
therefore the mechanical impedance of the spring is Zm ˆ ˆ F F K .1 It takes a force of F = a·M to set a mass M into the acceleration a (Newton’s second law of motion). In practice there will always be some losses.4. The acoustic impedance is associated with average properties on a surface.Zm ˆ F . of course. Z m j M where K K j M j 2 j M 1 0 .4.4. Example 1. In order to set the mass into vibrations one will have to move the mass and displace the spring from its equilibrium value. therefore the mechanical impedance of the mass is Zm ˆ ˆ F F j M .4. as sketched in figure 1. 0 K M is the angular resonance frequency.4. indicating that even a very small harmonic force at this frequency will generate an infinite velocity.1b) This quantity is also known as the mobility. ˆ F (1. The mechanical admittance is the reciprocal of the mechanical impedance. v ˆ (1.3 A simple mechanical oscillator consists of a mass M suspended from a spring with a stiffness constant of K. The unit is s/kg. It is defined as the complex ratio of the average sound pressure to the volume velocity.2 It takes a force of F = ξK to stretch a spring with the stiffness K a length of ξ (Hooke’s law). Ym v ˆ .4.1. 28 . ˆ ˆ v a j Example 1. Note that the impedance is zero at the resonance. which is the surface integral of the normal component of the particle velocity. It follows that the mechanical impedance of this system is the sum of the impedance of the mass and the impedance of the spring. ˆ j ˆ v j Figure 1.1 A mass hanging from a spring.1a) The unit is kg/s. Example 1. This quantity is mainly used under conditions where the sound pressure is more or less constant on the surface. so the impedance is very small but not zero at the resonance frequency.4.
30 The concept of acoustic impedance is essentially associated with approximate lowfrequency models. In many practical applications the properties of acoustic materials are described in terms of absorption coefficients (or absorption factors). which is used mainly in theoretical work. and since q Sun if the velocity is uniform.31 It is possible to calculate the transmission of sound through complicated systems of pipes using fairly simple considerations based the assumption of continuity of the sound pressure and the volume velocity at each junction [12]. a loudspeaker membrane caused by the motion of the medium. an acoustic twoport is completely described by the relations between the sound pressures and the volume velocities at the two terminals. it is a very good approximation to assume that the sound field in a tube is onedimensional when the wavelength is long compared with the crosssectional dimensions of the tube.33 The load of the medium on a vibrating piston can be described either in terms of the acoustic radiation impedance (the ratio of the sound pressure to the volume velocity) or the mechanical radiation impedance (the ratio of the force to the velocity). In other words. This quantity is used for describing the load on. In the general case we need to describe the properties of acoustic materials with the local ratio of the sound pressure on the surface to the resulting vibrational velocity. the acoustic impedance describes the load on a (real or ˆ fictive) piston caused by the medium.ˆ ˆ q u dS . It is possible to calculate the absorption coefficient of a material from its specific acoustic impedance.3) The unit is kgm4s1. This makes it possible to employ socalled lumped parameter models where the system is described by an analogous electrical circuit composed of simple lumped element.15) and (1. Such transducers are usually much smaller than the wavelength in a significant part of the frequency range.2. S (1. 14]. assuming either normal or diffuse sound incidence (see chapter 1. inductors. 29 . (1.5).32 The acoustic impedance is also useful in studying the properties of acoustic transducers. Under such conditions the sound field can be described by eqs. In case of a cylindrical tube such relations can easily be derived from eqs.16) [12]. losses and springs [13.4.2) where S is the surface area. is called the specific acoustic impedance. the voltage and current at the input terminal and the voltage and current at the output terminal. By analogy. Silencers (or mufflers) are composed of coupled tubes. the impedance is called the radiation impedance.2. resistors and capacitors. for example.16). (1.2. Finally it should be mentioned that the acoustic impedance can be used for describing the acoustic properties of materials exposed to normal sound incidence. Since the total force acting on the surface equals the product of the average sound pressure and the area.2. If the piston is real. 32 33 31 30 Such systems act as acoustic filters. and a tube of a given length behaves as an acoustic twoport. it can be seen ˆ ˆ that there is a simple relation between the two impedance concepts under such conditions: Zm Za S 2.4.15) and (1.4) This equation makes it possible to calculate the force it would take to drive a massless piston with the velocity un . ‘Twoport’ is a term from electric circuit theory denoting a network with two terminals. (1. (1. For example. but not the impedance from the absorption coefficient. representing masses. Thus the acoustic impedance is ˆ ˆ Z a pav q . In most literature this quantity. Such a network is completely described by the relations between four quantities.4.
Slk jV c (1. Note that the impedance goes to infinity when l equals a multiple of half a wavelength.2b). Conversely.4. the impedance is zero when l equals an oddnumbered multiple of a quarter of a wavelength. At low frequencies the acoustic impedance of the rigidly terminated tube analysed in example 1.7) in agreement with the considerations on p.4. p0 . (1. Thus the acoustic impedance of a cavity much smaller than the wavelength is springlike.4 can be simplified.2.2). indicating that the air in the tube acts as a spring.18) (with x = l).4. Figure 1.6) it can be seen that the acoustic impedance of a cavity at low frequencies also can be written Za (1.2.4. Example 1.4.4. example 1.Example 1.17) and (1.3.2. at these frequencies the sound pressure on a vibrating piston at the inlet of the tube would vanish. therefore it behaves as a spring with the acoustic impedance 30 . (1. indicating that it would take an infinitely large force to drive a piston at the inlet of the tube at these frequencies (see figure 1. Za j c S cot kl .4. c 2 p0 . from eq.4. Cf.4.3. Since. jV (1. 3. with a stiffness that is inversely proportional to the volume and independent of the shape of the cavity.5 A Helmholtz resonator is the acoustic analogue to the simple mechanical oscillator described in example 1.2. The factor cotkl approaches 1/kl.2 The acoustic input impedance of a tube terminated rigidly.4.5) where V = Sl is the volume of the tube. see figure 1.2. The dimensions of the cavity are much smaller that the wavelength. and the acoustic impedance becomes c2 Za j .4 The acoustic input impedance of a tube terminated by a rigid cap can be deduced from eqs. where l is the length of the tube and S is its crosssectional area.
eq.5 SOUND ENERGY.4. and sound fields are also energy fields in which potential and kinetic energies are generated. 12. However. (1. It is apparent that the radiated sound power is a negligible part of the energy conversion of almost any source.3 we conclude that the angular resonance frequency is 0 c S .4): Za j leff .2. jV where V is the volume. therefore the air in the neck behaves as a lumped mass with the mechanical impedance Z m j Sleff .Za c2 . has already been introduced. cf. where leff is the effective length and S is the crosssectional area of the neck. the characteristic impedance. as we have seen on p. (1. The usefulness is due to the fact that a statistical approach where the energy of the sound field is considered turns out to give very useful approximations in room acoustics and in noise control. SOUND INTENSITY. and it approximates this value in a free field far from the source (cf. As we have seen in section 1. In fact determining the sound power of sources is a central 31 .2. eq.27)). 1. (1. S By analogy with example 1.4.) The corresponding acoustic impedance follows from eq. energy considerations are nevertheless of great practical importance in acoustics.5. because some of the air just outside the neck is moving along with the air in the neck. the complex ratio of the sound pressure to the particle velocity in a plane propagating wave equals the characteristic impedance of the medium (cf.4. However. transmitted and dissipated.1. The unit is kgm2s1. Vleff Note that the resonance frequency is independent of the density of the medium. eq. sources of sound emit sound power.5).14)).2. (The effective length of the neck is somewhat longer than the physical length. Thus.3 A Helmholtz resonator. Yet another impedance concept. SOUND POWER AND SOUND ABSORPTION The most important quantity for describing a sound field is the sound pressure. (1. but it is perhaps less obvious that a smaller neck area gives a lower frequency.1. It is intuitively clear that a larger volume or a longer neck would correspond to a lower frequency. The air in the neck moves back and forth uniformly as if it were incompressible. Some typical sound power levels are given in table 1. the characteristic impedance describes a property of the medium.4. Figure 1.
5.5. and the derivation is similar.5.5. the phenomenon is analogous to the potential energy stored in a compressed or elongated spring. The instantaneous kinetic energy density at a given position in a sound field (the kinetic energy per unit volume) is wkin t 1 2 u t . expresses the magnitude and direction of the instantaneous flow of sound energy per unit area at the given position. and the derivation is similar. Aircraft turbojet engine Gas turbine (1 MW) Small airplane Tractor (150 hp) Large electric motor (0.1 The energy in a sound field It can be shown that the instantaneous potential energy density at a given position in a sound field (the potential sound energy per unit volume) is given by the expression wpot t p 2 t . 2 c 2 (1.5. 2 (1.1 Typical sound power levels.2) This quantity describes the energy per unit volume at the given position represented by the mass of the particles of the medium moving with the velocity u. (1. The value and relevance of knowing the sound power radiated by a source is due to the fact that this quantity is largely independent of the surroundings of the source in the audible frequency range. I t p t u t .point in noise control engineering. or the work done by the sound wave per unit area of an imaginary surface perpendicular to the vector.5 MW) Vacuum cleaner Office machine Speech Whisper 10 kW 32 W 5W 100 mW 10 mW 100 μW 32 μW 10 μW 10 nW 160 dB 135 dB 127 dB 110 dB 100 dB 80 dB 75 dB 70 dB 40 dB 1. The instantaneous sound intensity at a given position is the product of the instantaneous sound pressure and the instantaneous particle velocity.3) This quantity. Table 1. which is a vector. 32 . This corresponds to the kinetic energy of a moving mass.1) This quantity describes the local energy stored per unit volume of the medium because of the compression or rarefaction.
wpot 2 prms . t where I (t ) is the divergence of the instantaneous sound intensity and w(t) is the sum of the potential and kinetic energy densities. wkin 1 2 ˆ u . one can derive the equation w(t ) I (t ) . the relation between the sound pressure and density changes. IdS 0 S (1. If. V where S is the area of an arbitrary. I I(t ) p(t )u(t ). 4 (1.5. and Euler’s equation of motion). which expresses the simple fact that the rate of change of the total sound energy at a given point in a sound field is equal to the flow of converging sound energy.4.5.8a.5. the surface encloses a source the integral equals the radiated sound power of the source.5. 2 c 2 wkin 1 2 urms . This is the equation of conservation of sound energy. as in chapters 1. (1.5.5) is more important than the instantaneous intensity I(t).8b) According to Gauss’s theorem the volume integral of the divergence of a vector equals the corresponding surface integral of the (outward pointing) normal component of the vector. if the sound energy density at the point increases there must be a net flow of energy towards the point. and the timeaveraged sound intensity (which is usually referred to just as the ‘sound intensity’).4) and (1. 1. and if it decreases there must be net flow of energy diverging away from the point. 2 (1.34 V I (t )dV I (t ) dS S t w(t)dV Et(t) . 34 33 .5. This equation shows that the rate of change of the total sound energy within a closed surface is identical with the surface integral of the normal component of the instantaneous sound intensity. closed surface.4b) are more important than the instantaneous quantities.7) Often we will be concerned with harmonic signals and make use of complex notation.5.5.4a.6) in any sound field unless there is generation or dissipation of sound power within the surface S. (1.Energy conservation By combining the fundamental equations that govern a sound field (the conservation of mass. Expressed in the complex notation eqs.2 and 1. Energy conservation considerations lead to the conclusion that the integral of the normal component of the sound intensity over a closed surface is zero.5) become wpot ˆ p 2 2 4 c . 1. on the other hand. V is the volume inside the surface. In practice the timeaveraged energy densities. irrespective of the presence of other sources of noise outside the surface: IdS P S a (1. The global version of this equation is obtained using Gauss’s theorem. and E(t) is the total instantaneous sound energy within the surface.5.
2. Example 1.5.5.14 dB L p . (1. (1.5.35 However. (1. and one will have to measure both the sound pressure and the particle velocity simultaneously and average the instantaneous product over time in order to measure the sound intensity.12) I r Re pu 1 2 jkr 2 cr 2 2 c c r cr 2 2 It is apparent that there is a simple relation between the sound intensity and the mean square sound pressure in these two extremely important cases. (1.10) shows that 2 p ˆ prms Ix 2 c c 2 (1.11) that the sound intensity in a plane propagating wave with an rms sound pressure of 1 Pa is (1 Pa) 2 (1. 2 (1.14)) in a plane propagating wave into eq.5. (1.2 The sound intensity in the interference field generated by a plane sound wave reflected from a rigid surface at normal incidence can be determined by inserting eqs. (1.11) is not valid.5. inserting expressions for the sound pressure and the particle velocity in a simple spherical wave.4 mW m 2 .) The component of the sound intensity in the xdirection is Ix 1 ˆ ˆx Re pu * .5.26) and (1.10) Inserting the expressions for the sound pressure (eq.11) implies that the sound intensity level is almost identical with the sound pressure level in air at 20°C and 101. (1.2. Equipment for such measurements has been commercially available since the early 1980s [3].9) (Note that the two complex exponentials describing the time dependence of the sound pressure and the particle velocity cancel each other because one of them is conjugated. Example 1. 35 34 .2. into eq. (1.3 kPa: 2 2 2 2 LI 10 log I I ref 10 log prms ( c) I ref 10 log prms pref 10 log cI ref pref L p 0.5.18) into eq. Under conditions where the sound pressure and the particle velocity are constant over a surface in phase as well as in amplitude we can write Eq. see the Appendix.10): Ix 2 j pi 2 j2 pi* 1 Re 2 pi cos kx sin kx Re sin 2kx 0. it should be emphasised that in the general case eq. (1.13)) and the particle velocity (eqs.10) gives the same relation for the radial sound intensity: 2 A e jt kr A* e jt kr ˆ A p prms 1 1 * ˆ ˆr Re .2.5.17) and (1.2. 2 (1.5. (1.27).I 1 ˆˆ Re pu* .11) in this particular sound field. (1.5.5. Moreover.1 It follows from eq.2 kgm 3 343 ms 1 ) 2. eqs. c c 2 This result shows that there is no net flow of sound energy towards the rigid surface.2.5.
(1.15) However.11) is not. (1. This is the freefield method of measuring sound power.2 we need a precise description of the boundary conditions for solving the wave equation. As we have seen in chapter 1.5.11).7) is valid even in the presence of sources outside the measurement surface eq. In practice one measures the sound pressure at a finite number of discrete points. which leads to a description of material properties in terms of the specific acoustic impedance. However.2.5. eq. This is the sound intensity method of measuring sound power.5. [15].5. is more useful. With eq.2 Sound absorption Most materials absorb sound. as mentioned in section 1. in many practical applications.7) we now conclude that one can estimate the radiated sound power of a source by integrating the mean square pressure generated by the source over a spherical surface centred at the source: 2 Pa prms ( c) d S . see chapter 3. (1.14) This expression demonstrates that the radiated sound power of a vibrating surface is closely related to the volume velocity and to the real part of the radiation impedance. and therefore the local sound intensity is to a good approximation given by eq. and the sound power passing through the surface can be expressed in terms of the acoustic impedance: ˆ q 1 1 2 ˆˆ ˆ Pa Re pq* Re q Z a Re Z a . In an environment without reflecting surfaces the sound field generated by a source of finite extent is locally plane far from the source. the absorption coefficient (or absorption factor).5. A value of unity implies that all the incident sound power is absorbed. Note that an anechoic room (a room without any reflecting surfaces) is required. therefore only the source under test must be present.4.7) implies that one can determine the sound power radiated by a source by integrating the normal component of the sound intensity over a surface that encloses the source. 2 2 2 2 (1. for example in architectural acoustics.5. whereas eq.) 35 .5. a simpler measure of the acoustic properties of materials. (1.4)). 1.5. Note that special equipment for such measurements is required. From this definition it follows that the absorption coefficient takes values between naught and unity.1 A standing wave tube for measuring the normal incidence absorption coefficient. (From ref.13) (cf. Yet another method of measuring sound power requires a diffuse sound field in a reverberation room.5. By definition the absorption coefficient of a given material is the absorbed fraction of the incident sound power. Figure 1.2. Equation (1.5. S (1.p qZ a ˆ ˆ (1. as mentioned in chapter 1.4. (1.
2 c The absorption coefficient is the ratio of Ix to Iinc. 1 R 1 2 s 1 1 s 2 (1. (1. R = 0. that is.17) 2 p p I x i 2 r I inc pi 2 2 4s s 1 .10). (1.5.3 (···).2. If the material is completely absorbing. Note that the absorption coefficient is independent of the phase angle of R. p* e j kx pr* e j kx pi 2 pr j kx j kx i I x Re pi e pr e 2 c 2 c 2 pmin .19) and (1.2 Standing wave pattern for various absorption coefficients: 0. that is.) The incident sound intensity is the value associated with the incident wave. which shows that there is more information in the complex reflection factor than in the absorption coefficient.6 (– –).3 If the material under test is completely reflecting then R = 1.5. (1. Figure 1. 0.2.2.22)).5.5.16) into eq. 2 c 2 c where the last equation sign follows from eq.18) where we have introduced the reflection factor and the standing wave ratio (cf.9 (–––). (Note that pmax and pmin are amplitudes.5. 36 .20).2.2.15) and (1. or random or diffuse incidence in a room). Here we will study only the absorption for plane waves of normal incidence.18) demonstrates that one can determine the normal incidence absorption coefficient of a material by exposing it to normal sound incidence in a tube and measuring the standing wave ratio of the resulting interference field. (1. This is a onedimensional field. corresponding to an absorption coefficient of unity.5.2.2.16). Equation (1. Consider the sound field in a tube driven by a loudspeaker at one end and terminated by the material under test at the other end. (1. i i max p pr p pr p (1. In this case the standing wave ratio is infinitely large. The amplitudes pi and pr depend on the boundary conditions. eqs. which means that it has the general form given by eqs. the vibrational velocity of the loudspeaker and the properties of the material at the end of the tube. for example.5. The sound intensity is obtained by inserting eqs. Example 1. corresponding to an absorption coefficient of zero. so the sound pressure amplitude is constant in the tube. as sketched in figure 1. corresponding to a standing wave ratio of one. 2 (1.16) p I inc i .5.15) and (1. 0.In general the absorption coefficient of a given material depends on the structure of the sound field (plane wave incidence of a given angle of incidence. In the latter case there is no reflected wave.1.
4πr 1 jka (1.r ˆ p(a) ck 2 j j .5) where the approximation to the right is based on the assumption that ka << 1.6.6. From eq. In free space such a source generates the simple spherical sound field we studied in section 1. Q 4πa 2U . 1.6) . example 1.3) by multiplying with the surface area of the sphere.1.6. (1. (1. (1. which is also the most important: that of a solid vibrating surface.27) we know that the particle velocity on the surface of the source is ur ( a ) ˆ 1 A e j(t ka ) 1 . As we shall see.4.1 Point sources The simplest source to describe mathematically is a sphere that expands and contracts harmonically with spherical symmetry. 1 jka 4π 1 jka (1. Say the source has a radius of a. In this note we will study only the simplest one.2. This is the ratio of the sound pressure on the surface of the sphere to the volume velocity (cf. Q e jt 4πa 1 jka 4π 4πa (1.4. the expression for the sound pressure generated by a point source with the volume velocity Q e jt becomes ˆ p jQ e j(t kr ) .2.2.26) gives an expression for the sound pressure generated by the source.1). Note that the imaginary part of the radiation impedance is much larger than the real part at low frequencies.3)): Z a.2) where we have introduced the volume velocity of the pulsating sphere. With ka << 1.4) We can now calculate the radiation impedance of the pulsating sphere. also known as a point source or a simple source. (1.6. eq.1) The boundary condition on the surface implies that the vibrational velocity U e jt must equal the normal component of the particle velocity.6. 4πr 37 (1. therefore A j cka 2U e j ka jQ e j ka .6. This air moves back and forth almost as if it were incompressible.2.6 RADIATION OF SOUND Sound can be generated by many different mechanisms. Inserting into eq. ˆ p jQ e j(t k ( r a )) . c a jka (1. indicating that most of the force it takes to expand and contract the sphere goes to moving the mass of the air in a region near the sphere (cf. the most efficient mechanism for radiation of sound involves a net volume displacement.6. In the limit of a vanishingly small sphere the source becomes a monopole.
A vanishingly small sphere with a finite volume velocity36 may seem to be a rather academic source. is also in effect a monopole. (1. This is a strong statement with many practical implications. Note that the sound power is proportional to the square of the frequency.7) By multiplying with the surface of the area of a sphere with the radius r we get the sound power radiated by the monopole. and the rigid surface can be seen to have the effect of increasing the sound pressure by a factor of 1+ R1/R2.5. at z = 0. At very low frequencies k(R1 .1.10): 1 1 jQ jQ* 1 Q * ˆ ˆr Re . If the surface is rigid the boundary condition implies that uz = 0 at z = 0. The sound intensity generated by the monopole can be determined from eq. R2 4πR1 4πR2 4πR1 (1. indicating that a small pulsating sphere is not a very efficient radiator of sound at low frequencies. in this form. the effect of reflections from the ground on outdoor sound propagation. At low frequencies it is a good approximation to any source that produces a net displacement of volume.5). and simple symmetry considerations show that this is satisfied if we replace the rigid plane with an image source. I r Re pu 1 2 2 4πr 4πr c jkr 32π 2 r 2 c 2 (1. It is easy to take account of a large reflecting plane surface.5.8) We could also obtain this result from eqs. that is. (1. of course. that is. irrespective of its shape and the way it vibrates.6. The resulting sound pressure is simply the sum of the sound pressures generated by the source and the image source.6. The term ‘volume velocity’ is unambiguous. Pa Q 2 32π 2 r 2 c 4 r 2 ck 2 Q 8π 2 . and the first interference The volume velocity of the monopole is sometimes referred to as the source strength.6. see figure 1. (1.R2) << 1. However. the monopole is a central concept in theoretical acoustics. if one makes use of the concept of image sources. for example. say.14) and (1. Reciprocity The reciprocity principle states that if a monopole source at a given point generates a certain sound pressure at a another point then the monopole would generate the same sound pressure if we interchange listener and source position. some authors use other definitions of the source strength. that is.6. Destructive interference occurs when the second term in the parenthesis is real and negative. An enclosed loudspeaker is to a good approximation a monopole at low frequencies. 36 38 . the outlet of an engine exhaust system. any source that is small compared with the wavelength and changes its volume as a function of time. irrespective of the presence of reflecting or absorbing surfaces. The normalised equation can be used for studying outdoor sound propagation over a hard surface.6. it represents the sound pressure normalised by the free field value.9) The parenthesis shows the effect of the reflecting plane. However. and it is common practice to present the ‘ground effect’. ˆ p R1 jk (R1 R2 ) j Q e j(t kR1 ) j Q e j(t kR2 ) j Q j(t kR1 ) e e 1 . A source that injects fluid.
(From ref. [16].6.6.(From ref.6. Figure 1. (1. the two contributions will arrive with a different phase no matter how far from the source we are.3) we can make use of the farfield approximation and let r1 r2 r in the denominator of eq.10) 39 . If the observation point is sufficiently far we can approximate the two distances by r1 r h cos and r2 r h cos in the complex exponentials.) If the distance between the source and the observation point is much longer than the distance between the source and the reflecting plane (see figure 1. corresponding to (R1 – R2) being half a wavelength. However.2 shows the sound pressure relative to free field for sound propagation over a rigid plane surface.6.dip occurs when k(R1 – R2) = π.1 The sound pressure generated by a monopole above a rigid plane is the sum of two terms: direct sound and the contribution from the image source. Figure 1.6.2 The sound pressure in onethird octave bands generated by a monopole above a rigid plane and shown relative to free field for five different sourcereceiver distances. The resulting sound pressure now becomes ˆ p j Q e j(t kr1 ) j Q e j(t kr2 ) 4πr1 4πr2 j Q j(t k ( r h cos )) j(t k ( r h cos )) e e jrQ cos(kh cos ) e j(t kr ) .) Figure 1.6). [16].6. 4πr 2π (1.
It follows that Pa π 2 0 2π 0 ˆ p 2 2 2 c r sin d d 2 ck 2 Q 4π 2 2 π 2 0 cos 2 (kh cos ) sin d (1.6.8)) this source will radiate four times more sound power than one single loudspeaker in isolation. Figure 1. The sound power of the monopole is affected by the presence of the reflecting surface unless it is far away.6.5. kh >> 1. and that the rigid surface has an insignificant influence on the sound power output of the source when h exceeds a quarter of a wavelength. corresponding to kh = π/2.6. 2kh Figure 1.11) that two identical monopoles in close proximity (two enclosed loudspeakers driven with the same signal.6.3 Far field sound pressure generated by a monopole near a rigid plane surface. so this surface does not contribute to the integral. Example 1. Alternatively one might regard the two loudspeakers as one compound source with twice the volume velocity of each loudspeaker. in which case the sound pressure is simply doubled. eq.11) ck 2 Q 4πkh kh 0 cos 2 xdx ck 2 Q 8π sin(2kh) 1 . cf. for example) at very low frequencies will radiate twice as much sound power as they do when they are far from each other.7). eq. We can calculate the sound power by integrating the sound intensity over a hemisphere.6.4 The influence of a rigid surface on the sound power of a monopole. (Since the normal component of the particle velocity is zero at all points on the plane between the source and the image source. (1. (1. 40 .15) are also valid for combinations of sources.4 shows the factor in parentheses.) Moreover. (1. (1.1 It can be deduced from eq.Inspection of eq. Because of the quadratic relation between volume velocity and power (cf.5. The physical explanation is that the radiation load on each source is doubled.6.6. It is apparent that the sound power is doubled if the source is very close to the surface. the sound pressure on each source is not only generated by the source itself but also by the neighbouring source. (1. the considerations that lead to eq. the normal component of the intensity is also zero. Figure 1.6.10) leads to the conclusion that the sound pressure in the far field depends on kh and on θ unless kh << 1.
Note that the sound power of the dipole is proportional to the fourth power of the frequency. The oscillating sphere shown in figure 1.13) Note that the sound pressure is proportional to hQ. The sound pressure generated by the two monopoles is ˆ p jQ e j(t kr1 ) jQ e j(t kr2 ) . is in effect a dipole.14) .6. indicating very poor sound radiation at low frequencies. It is clear that the combined source has no net volume velocity.12) The near field of this combination of sources is fairly complicated. ˆ p jQ j(t k ( r h cos )) j(t k ( r h cos )) e e Q sin(kh cos ) e j(t kr ) 4πr 2πr 2 chk Q cos e j(t kr ) . varies as cosθ and is identically zero in the plane between the two monopoles. and so is an unenclosed loudspeaker unit. The physical explanation of the poor radiation efficiency of the dipole is of course that the two monopoles almost cancel each other. 41 .5 A point dipole.10). 2πr (1. We can calculate the sound pressure in the far field in the same way we used in deriving eq. However. other authors use other definitions.5.6. A point dipole is a good approximation to a small vibrating body that does not change its volume as a function of time. However.6.6.6. (1.Two monopoles of the same volume velocity but vibrating in antiphase constitute a point dipole if the distance between them is much less than the wavelength. 37 The quantity 2hQ is referred to by some authors as the dipole strength. 4πr1 4πr2 (1. for example.6.6. see figure 1.6. the far field is relatively simple.37 The sound power of the dipole is calculated by integrating the mean square sound pressure over a spherical surface centred midway between the two monopoles: Pa π 0 2π 0 ˆ p 2 2 c r sin d d 2 2 ch 2 k 4 Q 4π 2 2 π 0 cos 2 sin d ch 2 k 4 Q 4π 1 1 x 2 dx ch 2 k 4 Q 6π (1. Such a source exerts a force on the fluid. Other examples include vibrating beams and wires. Figure 1.
6. a vibrating circular piston in an infinite. By linear superposition we conclude that the sound pressure radiated by the piston can be evaluated at any position in front of the baffle simply by integrating over the surface of the piston. Thus. S 2πh (1. Note the factor of two in the denominator instead of four for the monopole.2 Sound radiation from a circular piston in an infinite baffle Apart from the pulsating sphere.6.) 42 . which is due to the contribution of the image sources. (1.6.6.10)). [1].Figure 1. in other words. (From ref.This is a special case of what is known as Rayleigh’s integral. cf.6. ˆ p j e j(t kh ) UdS .6. The basic approach to extended sound sources is to consider them as composed of many simple sources. (From ref.9) and (1.6.7 Definition of the variables. It is often used in connection with loudspeaker modelling. and S is the surface of the piston of radius a (see figure 1. [18]. just as a dipole is made up of two monopoles. It follows that the volume velocity of each elementary monopole is UdS.15) where h is the distance between the observation point and the running position on the piston. the baffle has the effect of doubling the volume velocity of each monopole. Let the piston vibrate with the velocity Ue jt . Because of the infinite baffle each monopole gives rise to an image source which coincides with the monopole.6 Fluid particles in the sound field generated by an oscillating sphere. eqs. the piston is the sum of many monopoles that all radiate in phase.) 1. Figure 1.7). rigid baffle is one of the simplest cases of a spatially extended sound source that can be dealt with analytically. which can be used for computing the sound radiation into half space of any plane infinite surface with a given vibrational velocity [17].
the sound pressure at long distances from the centre of the piston compared with the radius and the wavelength. 0 0 e 2πr (1.9 shows the directivity for different values of the normalised frequency ka. Q = π a2U. (1.The far field sound pressure. unity.21) This function has its maximum value. Figure 1.6. The factor in brackets is called the directivity of the piston.17) The calculation makes use of the Bessel functions J0(z) and J1(z). e e r k sin 2πr ka sin (1. Thus the expression for the sound pressure becomes ˆ p (r . which is a frequency dependent function that describes the directional characteristics of the source in the far field. can be calculated by expanding h in the complex exponential. ˆ p(r .8).6. At high frequencies the radiation of the piston is concentrated in a beam near the axial direction.20) where we have introduced the volume velocity of the piston. just as one would expect.6.6. defined by J0 ( z) and J1 ( z ) 1 z J 0 ( )d z 0 jU a J1 (ka sin ) j(t kr ) j Q 2 J1 (ka sin ) j(t kr ) . eq.16) while retaining only the first term of eq.19) 1 2π jz cos e d 2π 0 (1. y h r 2 y 2 2ry sin cos r 1 2 sin cos r r y sin cos .18) (see figure 1.6. that is. (1.6. ) jU j(t kr ) 2π a jkysin cos e ydyd .6. and leads to the following expression for the far field sound pressure. when θ = 0.10)). Figure 1.16) in the denominator (cf.6. (1. Note that the piston is an omnidirectional source (a monopole placed on a rigid surface) at low frequencies. ) 1 .6.6. indicating maximum radiation in the axial direction all frequencies. ka sin (1.6.8 Bessel functions. 2 J (ka sin ) D( f . ) (1. 43 .
where n is a positive integer. that is.22) (1. h (1.27) on the axis.26) It can be seen that the sound pressure is zero when k) is a multiple of π. Since sin θ = 0 on the axis. from which.23) Thus the sound pressure on the axis is given by ˆ p jU jt 2π e 0 r 2π r 2 a2 E e.jkr e. dh ydy r 2 y2 y dy .25) the sound pressure can be written p 2 jcU e jt k r sin(k). the expression for the distance h reduces to h r 2 y2 .6.6. the sound pressure assumes a maximum value for 2 r 2 a 2 r (2m 1) 2 (1. ˆ (1. when ) is a multiple of half a wavelength. corresponding to the positions 1 a n r a 2n 2 a (1. for 44 .6.6.6. (From ref. that is.Figure 1.6.) The sound pressure on the axis of the piston can be evaluated fairly easily. In a similar way. (1.28) (where m is a positive integer).jk r 2 a2 . [18]. (1.6.6.jkh dhd cUe jt e.24) If we introduce the quantity r 2 a2 r 2 .9 Directivity of the piston as a function of the normalised frequency ka.
are given in terms of normalised frequencies by r ka πn . one obtains a2 1 r 1 2 2 2r a2 .10 Sound pressure on the axis of a baffled piston for ka/2= 5. This asymptotic expression is plotted as a dashed line in figure 1. It follows that the minima are never observed in front of loudspeakers in real life. However. far above the frequencies at which the piston approximation is valid.6. by developing the spherical monopole field in cylindrical coordinates. 1 a 2m 1 r a .10 shows the normalised sound pressure on the axis of the piston as a function of the distance. [19].) It may seem surprising that the sound pressure is zero at some positions right in front of the vibrating piston. minima only occur at frequencies higher than 6900 Hz. Example 1. Thus for a loudspeaker with a radius of 50 mm. e 2πr 2r This expression agrees with eq.6. r 4r and the sound pressure reduces to ka 2 j(t kr ) j ckQ j(t kr ) ˆ p j cU e .5. The explanation is destructive interference. In the near field there is no possible approximation except on the axis. Figure 1. as of course it should. normalised by the radius of the piston. (1. caused by the fact that the distance from such a position to the various parts of the piston varies in such a manner that the contributions cancel out.3 The distances at which the minima occur. [19] or 45 . which for a given frequency is defined by the corresponding kafactor.20) for θ = 0 (D(f) = 1).6. 4 a 2m 1 (1.6. that is. a 4πn ka Minima of order n only occur for ka ≥ 2πn > 6.6.29) Figure 1. when r >> a and r >> a2/λ.2 In the far field. (From ref. Example 1. The calculations are rather complicated (see ref.6.6. the force exerted on the piston can be calculated analytically.10.
[20] for a complete treatment). the ratio of the average sound pressure to the volume velocity.r 1 8 ck 2 j . 2 3 πk a πa 2ka 2 Figure 1.30) where H1 is the first Struve function. Z a.r πa 2 c R1 jX 1.30) and (1.31) gives Z a. (1.6.33) At low frequencies and at high frequencies the radiation impedance takes simple expressions: ka 1 ka 1 Z a.32)). 2π 3aπ 2 (1. The result is. H (2ka) J (2ka) ˆ ˆ j 1 F pdS cπa 2U e jt 1 1 .6.11 Radiation impedance of a piston as a function of the normalised frequency.6.) 46 .6. jt Qe SQ e jt (1.6.31) Combining eqs.r 1 πa 2 c J1 (2ka) H (2ka) j 1 .r ra. [19].r Z a.r ˆ p F ˆ . dimensionless radiation impedance (the bracket in eq. that is. ka ka (1. (From ref.11 shows the normalised.6.34a) (1.6. The radiation impedance is the impedance seen by the piston. and lead to an expression in terms of special functions such as Bessel and Struve functions. (1.6.6.(1.32) Figure 1. S ka ka (1. Z a.34b) c πa 2 j 2 c 4 π 2 1 j .6.6.r j ma.
multiplied by a factor of two because of the rigid plane. (1.r can be interpreted as the acoustic mass of the air driven along by the piston.r . This method can also be used for computing the sound power of a piston in an infinite baffle. (1. (1.6. Pa 1 2 Q Re Z a.6.4) and (1. However. (1.The first expression is fundamental for designing loudspeakers.6. the imaginary part of the acoustic radiation impedance diverges when the radius a goes to zero.r πa 4 ck 2 8a 3 . (1.r 1 Q 2 2 2 2 c πa 2 R1 1 Q 2 2 1 πa 2 c J1 (2ka ) . the sound power is independent of the frequency if the volume acceleration is independent of the frequency. Example 1.6 Equation (1. as in the case of the pulsating sphere. by far the simplest approach is to use eq. Note that the real part of the radiation impedance equals that of a small pulsating sphere. Example 1.4).4.5 as the integral of the normal component of the sound intensity over a surface than encloses the source.36) which is just what we would expect since the piston acts as a monopole on a rigid plane in this frequency range (cf.5). with eq.4. ka (1.36) shows that the sound power of the piston is proportional to ωQ2 at low frequencies.6. (1. since. which expresses the sound power in terms of the mean square volume velocity and the real part of the acoustic radiation impedance: Pa 1 Q Re Z a.6. (1. with eq.r = ρcπa2(R1+jX1). However. Interference effects in the near field make it different from the imaginary part of impedance of the pulsating sphere. 47 .5 Instead of using the volume velocity and the acoustic impedance we could equally well compute the sound power from the mean square velocity and the real part of the mechanical radiation impedance.14). Pa ck 2 Q 4π 2 . (1.35) At low frequencies this becomes. j 2 3 The imaginary part of this impedance is the impedance of the mass of a layer of air in front of the piston.6.6.5). the low frequency sound power output of a loudspeaker is always limited: the only way to increase the sound power is to increase the size of the membrane.6.5.11)).6. This layer of air is moving back and forth as if it were incompressible.6.6. This implies that the displacement of the piston should be inversely proportional to the square of the frequency if we want the sound power to be independent of the frequency. 2 2 2 Example 1. The radiated sound power is defined in chapter 1. eq.34a). In other words.4 The mechanical radiation impedance is given by eqs. Its low frequency approximation is therefore: Z m. Since mechanical systems such as loudspeakers only allow a limited excursion.r 1 U Re Z m. that is. This explains why very large loudspeakers are found in subwoofers. eq. it implies very large displacements at low frequencies. The quantity ma. eq.33) as Zm.
Note that the directivity factor approaches two at low frequencies rather than one. (1. [18].6.) 48 . (1.The directivity factor of a source is defined as the sound intensity on the axis in the far field normalised by the sound intensity of an omnidirectional source with the same sound power. (1.12 as a function of the normalised frequency ka. (1. reflecting the fact that all the sound power is radiated in only half a sphere. Normalising with Pa/4πr2 (eq.35)) gives the directivity factor Q(f).40) Q(f) 15 10 5 0 ka 0 1 2 3 4 Figure 1.6. (From ref.2).6.20) the sound intensity on the axis is Q 1 I r ck 2 2r 2 2 (1.6.6. one often uses the directivity index.37) (see also example 1.6.38) The directivity factor of the piston is plotted in figure 1. Q( f ) ka 2 R1 ka 2 J (2ka) 1 1 ka . In practice.12 Directivity factor of a piston in a baffle. defined by DI ( f ) 10 log Q( f ).6. From eq.6.
W. 2000. Jr. Journal of Sound and Vibration 147. 1961. 489496. Lord: Environmental and Architectural Acoustics. John Wiley & Sons. A. 1996.B. L. Salomons: Computational Atmospheric Acoustics. Crocker. Krug: Sound level meters. New York. K. Journal of the Acoustical Society of America 33. J. SpringerVerlag. Rasmussen: Lydfelter. Anon. Technical University of Denmark. A. 49 . 1997.J. and allied theorems. San Francisco. M. Ward: Subjective evaluation of musical scale temperament in pianos. Lyngby. Note no 31260. Chapter 107 in Handbook of Acoustics. New York. John Wiley & Sons. Dordrecht. Maekawa and P. J.D. D. Department of Acoustic Technology. ed. McGrawHill.M. Moore and P. 2010.V. Martin and W. Pope: Analyzers.: Introduction to Electroacoustics and Audio Amplifier Design (2nd edition). Coppens and J. akustiske og elektriske systemer (4th edition). New York. Acoustic Technology.E. Philosophical Magazine 43. ed. 2010.J. Pierce: Acoustics.U. Jacobsen: Sound intensity. Nærum. 1997. Crocker. Brüel & Kjær. K. M. Randall: Frequency analysis (3rd edition). W. John Wiley & Sons. New York. Wheeler: The Science of Sound (3nd edition). 1999. Polyteknisk Forlag. 2001. E & FN Spon. Rasmussen: Analogier mellem mekaniske. Department of Electrical Engineering. The Acoustical Society of America. 2002. Chapter 156 in Encyclopedia of Acoustics. 1987.J. An Introduction to Its Physical Principles and Applications. New York. Note no 2107. Technical University of Denmark. Marshall Leach. F. Department of Electrical Engineering. 582585.R. Morse and K. M. Crocker and F. ed. John Wiley & Sons. 1994. F. Kluwer Academic Publishers. Acoustic Technology. 259272.: Microphone Handbook.A. R. Hubert: Vorlesungen über Technische Akustik (3rd edition). Dubuque. E.REFERENCES 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 L. 1897.M. New York. Sanders: Fundamentals of Acoustics (4th edition).R.S. Technical University of Denmark. A. Rossing. Kendall/Hunt Publishing Company. 1991.J. Nærum.W. F. 1997. Chapter 154 in Encyclopedia of Acoustics.K. Kinsler.D. Addison Wesley. 1989. Zaveri: Introduction to acoustical measurement and instrumentation. 1994. Crocker. Chapter 155 in Encyclopedia of Acoustics. 1968. Cremer and M. T. Rayleigh: On the passage of waves through apertures in plane screens. P. CA. Z.B.V. Ingard: Theoretical Acoustics. W. New York. Note no 7001. John Wiley & Sons. P.J.D. IA. F. 1998. Pope and H. Brüel & Kjær. Berlin. London. R. M. Brüel. Frey. M. Jacobsen: Propagation of sound waves in ducts. 1996. ed. 1985. Crocker. Jacobsen: An elementary introduction to applied signal analysis. Jacobsen: A note on instantaneous and timeaveraged active and reactive sound intensity.
L. New York. Bergassoli: Acoustics: Basic Physics. Lefebre and A. 2000. and Kinsler et al. in the books by Morse.U. Pierce: Mathematical theory of wave propagation. A. F.D. A. D.B. L. Ingard: Theoretical Acoustics. Rossing. Beranek. 2009. McGrawHill. Fahy and J. Coppens and J. 6. A. 1986. 1989. 7. and in Pierce’s chapters and book. J. F.R.D.V.D. P. 1998. The American Institute of Physics.M. Cambridge. Morse and K. 2. and Filippi et al. San Diego. 1. A. ed. Nelson: An introduction to acoustics. 8. McGrawHill. 1999.BIBLIOGRAPHY Recommended reading includes the following. Chapter 2 in Encyclopedia of Acoustics. 1997. Pierce: Acoustics.. 11. The books by Fahy. Frey. New York. 50 . Academic Press. The Acoustical Society of America. An Introduction to Its Physical Principles and Applications. Kinsler. are also introductory. Crocker. Chapter 3 in Springer Handbook of Acoustics. Morse: Vibration and Sound (2nd edition). Walker. 2007. Pierce: Basic linear acoustics. Alton Everest and K. London. 5.A. John Wiley & Sons. Beranek: Acoustics (2nd edition). P. 9. Filippi.P. 2000. New York. John Wiley & Sons. 4. F. M. P. P. Academic Press. Sanders: Fundamentals of Acoustics (4th edition). New York.J. E & FN Spon. London.J. A. New York. 1968. L.M. Morse & Ingard. Pohlmann: Master Handbook of Acoustics (5th edition). The American Institute of Physics for the Acoustical Society of America. More advanced treatments can be found in the Nelson’s chapter. 10. New York. Springer. T. Habault. The book by Everest and Pohlmann manages to deal with many acoustic phenomena practically without mathematics. New York.E. MA. Chapter 1 in Fundamentals of Noise and Vibration. Fahy: Foundations of Engineering Acoustics.D.C. ed. Theory and Methods. 1983. ed. 3.
3. (1.9 APPENDIX: COMPLEX NOTATION In a harmonic sound field the sound pressure at any point is a function of the type cos(Tt + n). A B A j( A B ) e . 51 .9.5) A Ar2 Ai2 (see figure 1. The complex conjugate of A is A* Ar jAi A e j A .4) (1.9.1) where j 1 is the imaginary unit. This is a symbolic method that makes use of the fact that complex exponentials give a more condensed notation that trigonometric functions because of the complicated multiplication theorems of sines and cosines.9. (1. 1.9.7) Multiplication and division of two complex numbers are most conveniently carried out if they are given in terms of magnitudes and phase angles. (1. A Ar jAi A e j A .1. B (1.9.1. It is common practice to use complex notation in such cases.6) therefore the magnitude can also be written A A A* . Ai ImA A sin A .1). (1.8. We recall that a complex number A can be written either in terms of its real and imaginary part or in terms of its magnitude (also called absolute value or modulus) and phase angle.9.2) (1. 1.9) Figure 1. AB A B e j( A B ) .9. Complex representation of a harmonic signal.9.9.9. and Ar ReA A cos A .9.
cos x 1 jx e e jx .11a.9. 1.Complex representation of harmonic signals makes use of the fact that e jx cos x jsin x (Euler’s equation) or. this can also be derived without complex notation. 2 2 (1. Two simple harmonic signals with identical frequencies. and φA is its phase.9. is yet another harmonic signal with an amplitude of A1 + A2 (see figure 1. physical sound pressure is of course a real function of the time. [21].) The mathematical convenience of the complex representation of harmonic signals can be illustrated by an example. the imaginary unit j. p Rep Re A e j(t A ) A cos(t A ).2. Evidently.13) which is seen to be an expression of the form cos(ωt + φ).12) where A is the complex amplitude of the sound pressure. It can be concluded that complex notation implies the mathematical trick of adding another solution.9. sin x j 1 e jx e jx .2). (From ref. ˆ (1. The magnitude of the complex quantity A is called the amplitude of the pressure. multiplied by a constant.9. p A1 cos(t 1 ) A2 cos( t 2 ) A1 cos 1 A2 cos 2 cos t A1 sin 1 A2 sin 2 sin t 2 2 A1 cos 1 A2 cos 2 A1 sin 1 A2 sin 2 cos( t ). This trick relies on linear superposition. The real. A1ejωt and A2ejωt. ˆ (1.9. conversely.14) where 52 . Figure 1. an expression of the form sin(ωt + φ).11b) (1.9.9.10) In a harmonic sound field the sound pressure at a given position can be written p Ae jt . A sum of two harmonic signals of the same frequency. ½ (1.9.
9.9. integration with respect to time corresponds to division with jω.19) (1.9.15) but the expedience and convenience of the complex method seems indisputable. arctan A1 sin 1 A2 sin 2 .9.16) it follows that differentiation with respect to time corresponds to multiplication by a factor of jω. physical velocity is v Rev B cos(t B ). ˆ (1. j (1.9. for example. A1 cos 1 A2 cos 2 (1.18) then the acceleration is written a j v . ˆ v Be jt B e j(t B ) . ˆ ˆ which means that the physical acceleration is ˆ a Re a Re j Be jt B sin( t B ) .20) (1. dt In a similar manner we find the displacement.21) ˆ v ˆ .9. Conversely. (1.9. (1. dt (1. If.17) which means that the real. and this is seen to agree with the fact that d cos(t B ) sin(t B ). Since d jt e je jt .9.22) which means that Re ˆ Re 1 1 Be jt B sin(t B ). j (1.9. in complex representation. the vibrational velocity of a surface is.23) in agreement with the fact that 53 .
(1. The time average of a product is given by the following expression xy 1 1 ˆˆ ˆ ˆ Re xy Re x y . 2 2 (1. as follows. Note that it is the squared magnitude of p that enters into the expression. products of harmonic signals.d1 sin(t B ) cos(t B ). Such quantities are dealt with in a special way. ˆ ˆ 2 (1.9.24) Acoustic secondorder quantities involve time averages of squared harmonic signals and. more generally.9. which in general ˆ ˆ would be a complex number proportional to e2jωt. ˆ ˆ Re xy Re x e j(t x ) y e 2 2 2 which is seen to in agree with xy x cos(t x ) y cos(t y ) ˆ ˆ 1 x y cos( x y ).9. the mean square ˆ pressure becomes 2 p 2 prms p ˆ 2 2. not the square of p .28) (1.25) in agreement with the fact that the average value of a squared cosine is ½.½ ˆ ˆ 54 .9. dt (1.9.27) Note that either x or y must be conjugated. 1 1 1 ˆˆ ˆ ˆ j(t y ) x y cos(x y ). Expressed in terms of the complex pressure amplitude p .26) This can be seen as follows.
Hearing and Speech 2 Ear.Ear. transmit the movements to the bones in the middle ear. the inner ear and the nerve connection to the brain.1. masking) and speech intelligibility. Part A being the outer ear. B is the middle ear.2. 55 . loudness. C is the inner ear. The first three parts (the peripheral parts) are shown in Figure 2. reach the tympanic membrane (the ear drum). The fluid movements will be transformed to nerve impulses from the hair cells in the inner ear and the impulses are transmitted to the brain through the auditory nerve. progress through the outer ear canal. and further transmit the movements to the fluid in the inner ear. 2. The major topics are: the ear and its functional principles. B is the middle ear and C is the inner ear. A is the outer ear. The content covers the basic psychoacoustic aspects of a situation where two persons speak to each other.2. From [1] The ear may be divided into four main parts: The outer ear. the perception of sound and the consequences for speech understanding. basic psychoacoustics (hearing threshold. The sound will reach the outer ear. the middle ear.1 Introduction The aim of the present chapter is to give the student a basic understanding of the function of the ear. Hearing and Speech Torben Poulsen 2.1 Drawing of the ear.2 The Ear Figure 2.
The tympanic membrane is found at the end of the canal. From Figure 2.2. 0. This implies that it is usually not possible from the outside to see the tympanic membrane at the end of the ear canal. S) and the footplate of the stirrup makes the connection into the inner ear. The length of the ear canal is approximately 25 mm and the diameter is approximately 7 mm. hearing and speech 2.Ear. A drawing is shown in Figure 2.2 it is seen that the hammer (Maleus. 4). The outer part of the ear canal is relatively soft whereas the inner part is stiff and bony.1 mm. The thickness is approx. The area is approximately 1 cm2. 2) connects the hammer and the stirrup (Stapes. Localisation of sound sources is difficult if a hearing protector or a crash helmet covers the pinna. At the end of the ear canal the tympanic membrane is situated. The Latin names are also often used: Malleus. The footplate rotates around the point marked (3). I. These numbers are approximate and vary from person to person. Most ear canals will have one or two bends. 30 degrees. 2. The ear canal may be looked upon as a tube that is closed in one end and open in the other. Incus and Stapes. M) is fixed to the tympanic membrane (1) from the edge and into the centre of the membrane (the top of the cone).4 kHz This calculation is correct if the ear canal was a cylindrical tube.2. The middle ear is filled with air and is connected to the nose cavity (and thus the atmospheric pressure) through The Eustachian tube (ET.2. 56 . This will give resonances for frequencies where the length of the ear canal corresponds to 1/4 of the wavelength of the sound. The anvil (Incus. The Pinna plays an important role for our localisation of sounds sources.2.2 The middle ear The middle ear consists of three small bones: hammer. The special shape of pinna produces reflections and diffraction so that the signal that reaches the ear will be dependent on the direction to the sound.1 The outer ear The outer ear consists of the pinna (or the auricle) and the ear canal. The fluid in the inner ear is incompressible and an inwards movement of the stirrup will be equalised by a corresponding outward movement by the round window (5). anvil and stirrup. The tympanic membrane is shaped like a cone with the top of the cone pointing inwards into the middle ear.025 m = 3. The pinna has common features from person to person but there are big individual differences in the details. which may be done by pulling pinna upward and backwards. With a length of 25 mm and a speed of sound of 340 m/s the resonance frequency will be f res = 340 m / s 4 * 0. These bones are the smallest bones in the human body. It’s necessary to make the canal straighter. The function of the middle ear is to transmit the vibrations of the tympanic membrane to the fluid in the inner ear. This connection is sometimes called the oval window.2. The membrane is not perpendicular to the axis of the ear canal but tilted approx.
The middle ear reflex can to some extent protect the inner ear from excessive exposure.3 and the area ratio is approx. 70 dB SPL whereby the transmission through the middle ear is reduced. See text for details.3. see Figure 2. Because the reflex is activated by a signal from the brain there will be a delay of about 25 to 150 ms before the effect is active. When the tube is open. 25 dB. 14. tensor tympani (6) and stapedius (7). the pressure at the two sides of the tympanic membrane is equalised. A cross section of one of the turns is shown in Figure 2. are attached to the bones and will be activated by the socalled middle ear reflex. The reduction is about 20 dB at 125 Hz.2.2 Drawing of the middle ear. 57 .2.makes an impedance match between the air in the outer ear and the liquid in the inner ear.2. From [2] Usually the Eustachian tube is closed but opens up when you swallow or yawn. which corresponds to approx. 10 dB at 1000 Hz and less than 5 dB at frequencies above 2000 Hz.2. 2. If the Eustachian tube becomes blocked (which is typically the case when you catch a cold) the equalisation will not take place and after some time the oxygen in the middle ear will be assimilated by the tissue and an underpressure will build up in the middle ear. The reflex has therefore no protective effect on impulsive sounds.together with the area ratio between the tympanic membrane and the footplate of stapes . 1. There are 2. This causes the tympanic membrane to be pressed inwards and thus the sensitivity of the hearing is reduced.75 turns in the snail shell and the total length from the base to the top is 32 mm. The total ratio is thus 18. The reflex is elicited when the ear is exposed to sounds above approx. Two small muscles.2. The cochlea is filled with lymph and is closely connected to the balance organ that contains the three semicircular canals.Ear. The chain of middle ear bones forms a lever function that . The lever ratio is approx.3 The inner ear The inner ear consists of a snailshell shaped structure in the temporal bone called Cochlea. Hearing and Speech Figure 2.
The oval window is the footplate of the stirrup and is connected to Scala Vestibuli (1). When sound is applied to the ear. hearing and speech This figure shows that the cochlea is divided into three channels (latin: Scala) called scala vestibuli (1).1 mm at the base of the cochlea to about 0.3.2.3) divides scala tympani from scala media. 3000 inner hair cells and about 12000 outer hair cells. the basilar membrane and the organ of Corti will vibrate and the hairs on the top of the hair cells will bend back and forth.is positioned on top of the Basilar Membrane in Scala Media. The inner hair cells are the main sensory cells. The designations ‘inner’ and ‘outer’ refer to the centre axis of the snail shell which is to the left in Figure 2. The function of the BM is very important for the understanding of the function of the ear. Scala Vestibuli and Scala Tympani are connected at the top of the cochlea with a hole called Helicotrema.3) covers the top of the hair cells. The width of the basilar membrane (BM) changes from about 0. The Basilar membrane (6 in Figure 2.2. A soft membrane (5 in Figure 2. The organ of Corti transforms the movements of the Basilar membrane to nerve impulses that are then transmitted to the hearing centre in the brain. The organ of Corti consists of one row of inner hair cells (7 in Figure 2.5 mm at the top of the cochlea (at helicotrema). The change of the BMwidth is thus the opposite of the width of the snail shell.3). Figure 2. A structure .2. The outer hair cells contain muscle tissue and these cells will amplify the vibration of the basilar membrane when the ear is exposed to weak sounds so that the vibrations are big enough for the inner hair cells to react. There are approx.3 Cross section of a cochlea turn. The round window prevents an overpressure to build up when the oval window moves inwards.2. The round window is connected to Scala Tympani (3).3) and three rows of outer hair cells (8 in Figure 2. The amplification function of the outer hair cells is nonlinear which means that they have an important effect at low sound levels whereas they 58 .2. From [1] There are two connections (windows) from cochlea to the middle ear cavity.Ear.2. This will trigger the (inner) hair cells to produce nerve impulses.called the organ of Corti . The hair cells are special nerve cells where small hairs protrude from the top of the cells. Most of the nerve fibres are connected to the inner hair cells. and scala tympani (3). See text for details. scala media (2).
Hel: Helicotrema (top of cochlea).is destroyed if the ear is exposed to loud sounds such as gunshots or heavy industrial noise. 1600 Hz and 6400 Hz.2. 1) At low exposure levels (20 dB) the amplitude is very selective and a ‘high’ amplitude is achieved only in a very narrow frequency range. O. Hearing and Speech are of almost no importance at high sound levels. The envelope of the deflection (shown dotted in Figure 2.4 where the amplitude is shown as a function of basilar membrane position for different frequencies).: Owal window (base of cochlea). (Note that this is different from Figure 2. The 400 Hz component produce BMmovement close to the top of the cochlea. If the frequency is changed. The same different slopes are also found in masking thresholds and it can be shown that masking is closely related to the basilar membrane movements. There are at least three nonlinear phenomena illustrated in the figure.sometimes called the cochlear amplifier . Figure 2.W. 2.2. Note also that the deflection of the BM is asymmetrical.4 Movement of the basilar membrane (b) when the ear is exposed to a combination of 400 Hz. This is illustrated in Figure 2. the pattern will not change but the position of the pattern will move along the basilar membrane. When the ear is exposed to a pure tone the movement of the basilar membrane will show a certain pattern and the pattern is connected to a certain position on the basilar membrane.4 for the frequencies 400 Hz. This is called an age related hearing loss.5. 1600 Hz and 6400 Hz (a).Ear.2. The amplifier function . This figure shows the BMamplitude at a certain position of the basilar membrane as a function of the stimulus frequency.2. Note that a single frequency produces movements of the basilar membrane over a broad area.4 The frequency analyzer at the Basilar membrane The basilar membrane acts like a frequency analyser. 6400 Hz produces a similar pattern but close to the base of the cochlea. For high exposure levels (80 dB) the 59 .2. This is called a noise induced hearing loss.2. This means that even for a single frequency many hair cells are active at the same time. From [3] The nonlinear behaviour of the outer hair cells and their influence on the BM movement is illustrated in Figure 2.4) has a steep slope towards the low frequency side and a much less steep slope towards the high frequency side. The amplifier function also deteriorates with age.
kHz Figure 2. In other words the change in the outside level from 20 dB to 80 dB. This will be seen as an elevated hearing threshold and this is called a hearing loss. In other words: The cochlear amplifier does not work. dB 80 dB SPL 60 60 dB 40 40 dB 20 20 dB SPL 0 2 3 5 20 30 10 Frequency. i. In a typical cochlear hearing loss. At high levels it is almost one octave below the maxamplitude frequency at low levels. The increase of amplitude at low levels is sometimes called ‘the cochlear amplifier’.Ear. Thus.5 Movement of the Basilar membrane at a fixed point for stimulus levels from 20 dB SPL to 80 dB SPL. hearing and speech ‘high’ amplitude is achieved at a much wider frequency range. Redrawn from [4] 60 .e.. 80 Basilar membrane movement. the outer hair cells are not functioning correctly or may be destroyed. 2) The frequency where the maximum amplitude is found change with level. At an input level of 80 dB the maximum BM amplitude is about 85 dB. the filter bandwidth of the auditory analyser changes with the level of the incoming sound. These nonlinear phenomena are caused by the function of the outer hair cells. 60 dB. At low levels (20 dB) the maximum BMamplitude is about 60 dB (with some arbitrary reference). is reduced (compressed) to a change in the maximum BMamplitude of only 25 dB.2. 3) The maximum amplitude grows nonlinearly with level.
3.3. Hz 10000 Figure 2. The weakest audible sound level is called the hearing threshold and the sound level of the loudest sound is called the threshold of discomfort or the threshold of pain. At 1000 Hz the threshold is about 2 dB SPL whereas it is about 25 dB SPL at 100 Hz and about 15 dB at 10000 Hz. is the level in the room at the position of the test subject’s head but measured without the presence of the test subject.3. From [5] The threshold curve in Figure 2.1 is measured under the following conditions: • Free field (no reflections from walls. 2.Ear.1 The binaural hearing threshold in a free field. ceiling) • Frontally incoming sound (called frontal incidence) • signals are single pure tones • binaural listening (i.3.1.1 The hearing threshold The hearing threshold is frequency dependent.e. dB 80 60 40 20 0 20 50 200 500 2000 5000 20000 10 100 1000 Frequency. listening with both ears) • no background noise • test subjects between 18 and 25 years of age • the threshold is determined by means of either the ascending or the bracketing method The curve is the median value (not the mean) over the subject’s data. Hearing and Speech 2. 100 Sound pressure level. which is shown in the figure. 61 . This curve is also called the absolute threshold (in a free field) and data for the curve may be found in ISO 3897 [6] and in ISO 226 [5]. floor. see Figure 2.3 Human hearing The human hearing can handle a wide range of frequencies and sound pressure levels. The sound pressure level.
Figure 2. The term ‘HL’ (hearing level) is used to emphasise that it is the deviation from the average normal hearing threshold. An elevated hearing threshold (i. If a person has a hearing loss of 5 dB HL at this frequency the 62 . dB SPL.1 and deviates from the pure tone curve only by a few dB (2 to +6) in the frequency range 500 Hz to 16 kHz.2 shows an audiogram for a person in the frequency range 125 Hz til 8000 Hz.2 Audiogram For practical use it is not convenient to measure the hearing threshold in a free or a diffuse sound field in the way described in the previous section. and hearing level. For practical and clinical purposes.2 Audiogram for a typical age related hearing loss. The results from the left ear are shown with '×' and the results from the right ear are shown with '○'. The threshold curve is similar to the curve in Figure 2.3. dB HL 10 20 30 40 50 60 70 80 250 500 1k 2k 4k 8k 10 0 10 20 30 40 50 60 70 80 Figure 2. hearing and speech In ISO 3897 also threshold data for narrow band noise in a diffuse sound field are found.Ear. Frequency. Hz 125 10 0 Hearing threshold level. a hearing loss) is indicated downwards in an audiogram and the values are given in dB HL.1 it can be seen that the hearing threshold at 125 Hz is 22 dB SPL (measured in the way described previously). The zero line indicates the average threshold for young persons and a normal audiogram will give data points within 10 to 15 dB from the zero line.3. The measurements are performed with headphones for each ear separately. Sound pressure level. dB HL. usually only the deviation from normal hearing is of interest.3. Such deviations are determined by means of a calibrated audiometer and the result of the measurement is called an audiogram. An example: From Figure 2. is not the same.3.e.3. 2.
5 9.5 3k 10. F.5k 28.0 4.3 Loudness Level The definition of loudness levels is as follows: For a given sound.5 6k 15. 63 .0 14k 36.3.5 2k 9.5 11. The TDH 39 data are from ISO 3891 [7].1.3. left ear.0 500 11. Table 2. In order for the audiometry to give correct results. the loudness level is defined as the sound pressure level (SPL) of a 1000Hz tone which is perceived equally loud as sound A.5 4k 9.5 18.2).3.0 30.g.2) will give a threshold of 44 dB SPL. Hz TDH 39 HDA 200 125 45. the sound pressure level of the 1 kHz tone is per definition equal to the loudness level in phone. Figure 2. The HDA 200 data are from ISO 3895 [8] and ISO 3898 [9]. right ear.3.1).1).0 8k 13.0 17. The result is shown in Figure 2.3. These standards specify the SPL values that shall be measured in a specific coupler (an artificial ear) when the audiometer is set to 0 dB HL.0 12. Figure 2.3. A. The loudness level for pure tones has been measured for a great number of persons with normal hearing under the same conditions as for the absolute threshold (Figure 2. The TDH 39 earphone can not be used above 8 kHz. which means that the audiometer must be recalibrated if the headphone is exchanged with another headphone. For a 1000Hz tone the value in dB SPL and in Phone will be the same.3.0 2.3.3.0 Table 2.Ear.0 16k 56.0 1k 7. In an audiogram the 5 dB hearing loss will be shown as a point 5 dB below the zero line (e. The values in the standards are headphone specific.5 10k 22.g. Another example: At 4000 Hz the free field threshold is 6 dB (see Figure 2.0 5.5 250 25. Hearing and Speech threshold would be 27 dB SPL. In order to measure loudness level a 1 kHz tone is needed and this tone should then be adjusted up and down in level until it is perceived just as loud as the other sound. When this situation is achieved. the audiometer must be calibrated according to the ISO 389 series of standards. A hearing loss of 50 dB HL (e. 2.5 17.1 shows reference values for two headphones commonly used in audiometry. Calibration values in dB SPL for a Telephonics TDH 39 earphone and a Sennheiser HDA 200 earphone. The unit for loudness level is Phon (or Phone).
forward masking means masking after a signal has stopped (i. Redrawn from [5] Some examples. 2. forward in time). when you need to turn down the radio in order to be able to use the telephone. The situation described above is also called simultaneous masking because both the masking signal and the masked signal are present at the same time.3 Equal loudness level contours. see Figure 2. This is not the case in backward and forward masking.g. dB 100 80 60 40 20 0 10 20 50 100 1000 Frequency.3. Hz 200 500 2000 5000 10000 20000 Figure 2.e.g.3. They should not be used directly to predict the perception of more complicated signals such as music and speech because the curves do not take masking and temporal matters into account. e.3. in other words A masks B or B is masked by A. Simultaneous masking is best 64 .4 Masking The term ‘Masking’ is used about the phenomenon that the presence of a given sound (sound A) can make another sound (sound B) inaudible. Translations of Loudness Level: Danish: Hørestyrkeniveau (enhed: Phon) German: Lautstärkepegel (Einheit: Phon) French: Niveau de Sonie. Backward and forward refer to time. Reflections in a room are not taken into account either. The curves in Figure 2. A 125Hz tone at 90 dB SPL will have a loudness level of 80 Phone. E. hearing and speech 140 120 Sound pressure level.3 are .in principle .3: A 4000Hz tone at 26 dB SPL will be perceived with the same loudness as a 1000Hz tone at 30 dB SPL and thus the loudness level of the 4000 Hz tone is 30 Phone.Ear.valid only for the special measurement situation where the tones are presented one at a time. Masking is a very common phenomenon which is experienced almost every day.
e. Generally the masking curves have steep slopes (about 100 dB/octave) towards the low frequency side and less steep slopes (about 60 dB/octave) towards the high frequency side. Figure 2. the hearing threshold (i. Hearing and Speech described in the frequency domain and is closely related to the movements of the Basilar membrane in the inner ear. The masked threshold is shown for different levels of the white noise.1 Masking from white noise. Here the masked threshold is shown for a narrow band signal centred at 250 Hz. A 10dB change in the level of the noise will also change the masked threshold by 10 dB.1 where also the absolute threshold is shown. From [3] The masked thresholds are almost independent of frequency up to about 500 Hz.4. 2. Above 500 Hz the threshold increases by about 10 dB per decade (= 3 dB/octave). The threshold determined in this situation is called the masked threshold contrary to the absolute threshold.4. If a narrow band signal is used instead of the white noise. the masked threshold will be as shown in Figure 2. The curves show the masked threshold for different spectrum levels of white noise. 65 .Ear.1 Complete masking If the ear is exposed to white noise. masked threshold) will be as shown in Figure 2.4.4.2. 1 kHz and 4 kHz respectively. The masking phenomenon is usually investigated by determining the hearing threshold for a pure tone when various masking signals are present.
masking increases nonlinearly with level.2.4. Figure 2. The slope at the low frequency side is almost independent of level but the slope at the high frequency side depends strongly on the level of the narrow band noise.4.e.Ear. Compare with Figure 2. 66 . This is illustrated in Figure 2. The different slopes towards the low and the high frequency side are also seen here and also the nonlinear level dependency is seen.2 Masking from narrow band noise. hearing and speech Figure 2. The slope towards higher frequencies decreases with increasing level. From [3] The masked threshold for narrow band noise is mainly caused by the basilar membrane motion.3 The influence of level on the masked threshold. The dotted lines near the top of the curves indicate experimental difficulties due to interference between the noise itself and the pure tone used to determine the masked threshold.4. The curves show the masked threshold when the ear is exposed to narrow band noise (critical band noise) at 250 Hz.4.3. From [3] The masking curves for narrow band noise are very level dependent. i. 1 kHz and 4 kHz respectively.
the tires.Ear. e.4. An example: When you listen to a standard carradio while you are driving at.4. you will adjust the level of the radio to a comfortable level. and the wind around the car (at least in ordinary cars).2 Partial masking The term ‘Complete masking’ is used when the presence of a given sound (sound A) can make another sound (sound B) inaudible. 100 km/h. This is an example of partial masking where the background noise masks part of the radio signal and when the background noise disappears the masking disappears too and the radio signal becomes louder than before. 67 . This kind of masking goes forward in time and is therefore called forward masking.4 Backward masking It has been shown that a strong sound signal can mask another (weak) signal which appears before the strong signal. The effect is restricted to about 20 ms before the start of the strong signal. Then.) 2. The influence is mainly seen in the loudness of sound B. when you come to a crossing or a traffic light and have to stop you will hear that the radiovolume is much too high. The effect lasts for about 200 ms after the end of the strong signal. Hearing and Speech 2.4. Backward masking is also called premasking. The example above is therefore not fully convincing in this situation. Partial masking is a situation where sound A influences the perception of sound B even though sound B is still audible. (Some modern car radios are equipped with a speed dependent automatic level control.3 Forward masking It has been shown that a strong sound signal can mask another (weak) signal which is presented after the strong signal. There will be some background noise from the engine. 2. This kind of masking goes back in time and is therefore called backward masking.g. Forward masking is also called postmasking.
The curve is based on a great number of loudness comparisons. phone and dB SPL is the same number).1 shows the relation between the Sone and the Phone scales.1 The loudness curve for a normal hearing test subject (solid line) and for a person with a cochlear hearing loss (dashed) 68 .5.1 The loudness curve The Sone scale was established in order to avoid the confusion between dB SPL values and the perception of loudness: A 1 kHz tone at 80 dB SPL is not perceived double as loud as the same tone at 40 dB SPL. The curve is called a loudness curve. Figure 2.Ear. The unit for loudness is Son or Sone. (Hint: for a 1 kHz tone. Figure 2. Arbitrarily it has been decided that one sone should correspond to 40 phones.5.5. Translation of terms: Loudness Hørestyrke Lautheit Sonie Loudness Level Hørestyrkeniveau Lautstärkepegel Niveau de Sonie Danish German French 2. hearing and speech 2. Note that ‘loudness’ and ‘loudness level’ are two different concepts.5 Loudness The term ‘loudness’ denotes the subjective perception of strength or powerfulness of the sound signal.
Note that at higher sound levels the loudness perception is the same for both normal and impaired listeners. This effect is called loudness recruitment. special calculation software is needed.ac.g. etc. The steeper slope means that .1 where the threshold (1 kHz) is a little less than 40 dB SPL. Recent research have shown that – for this kind of hearing loss – the loudness at threshold has a value significantly different from nil as indicated in the figure [10]. fireworks.html ) 69 .and C. 2.like a pistol shot..psychol. In other words. It has been shown that the time constant is about 100 ms. loudness calculations are found in professional Sound Quality calculation software.3. Short sounds . listeners with cochlear hearing loss have softness imperception.3 Measurement of loudness Many years ago it was thought that a sound level meter with filters corresponding to the ears’ sensitivity (described by the equal loudness level contours (Figure 2. Figure 2. The growth resembles the exponential growth of a time constant. For many daily life sounds a rule of thumb says that a 10dB increase is needed in order to perceive a doubling of the loudness.filters. This is not the case.5. handclap. This means that short duration sounds (less than one second) are perceived as less loud than the same sound with longer duration. An example of such a hearing loss is shown by the dashed curve in Figure 2. 2. The growth of loudness as a function of duration is called temporal integration. at http://hearing. For nonstationary sound.Ear.near the threshold .are perceived as weak sounds although their peak sound pressure levels may be well above 150 dB SPL.2 Temporal integration The perception of loudness needs some time to build up.5. This is one of the reasons why impulsive sounds generally are more dangerous than other sounds. For the determination of loudness.2 show the characteristics for the commonly used A. For stationary sounds two procedures can be found in [11].cam.1 corresponds to Stevens’ power law: N = 2 ( L − 40 ) / 10 where N is the loudness (in sone) and L is the loudness level ( in phones).5. Hearing and Speech The straight part of the solid line in Figure 2. the very common hearing impairment caused by age). The loudness curve becomes steeper near the hearing threshold.the loudness increases rapidly for small changes in the sound level. The curve shows that a doubling of the loudness corresponds to a 10phone increase in loudness level (or a 10dB increase in SPL if we are dealing with a 1 kHz tone).uk/Demos/demos.g. This is also the case for a person with a cochlear hearing loss (e.5. .5. rather than loudness recruitment.3)) could be used to measure loudness. For research purposes loudness models (software) can be found on the Internet (e. but due to masking and other phenomena these filters will not give a result that corresponds to loudness.
70 . It may me used as an approximation to a measurement with linear characteristic. 20 dB at 100 Hz and 30 dB at 50 Hz. C and D filter.g. The Dfilter is mainly used in connection with evaluation of aircraft noise. The attenuation is e.2 Filter characteristics for the A.5. The data for the D filter is from [13]. The frequency range around 3 kHz is known to be annoying and therefore this frequency range is given a higher weight. The main effect of the Afilter is that it attenuates the low frequency part of the signal. The Cfilter is ‘flat’ in the major part of the audible frequency range. Wind noise and other low frequency components are attenuated by the Afilter and is therefore very practical for many noise measurement situations. hearing and speech 20 10 Weighting characteristic.Ear. Hz 10000 Figure 2. dB 0 10 20 30 40 50 60 70 80 5 20 50 200 500 2000 5000 20000 Weighting filters Afilter Cfilter Dfilter 10 100 1000 Frequency. The data for the A and the C filter are from [12].
a filter bank.g.6.69 where CB is the bandwidth in Hz of the critical band and f is the frequency in kHz (not in Hz).6 The auditory filters The movements of the basilar membrane in the inner ear constitute a frequency analyser where the peak of the envelope moves along the basilar membrane as a function of frequency. on our frequency selectivity when we distinguish different vowels from each other. The critical bandwidth may be calculated from the empirical formula: CB = 25 + 75 (1 + 1.2. Many of these are masking experiments and led to the formulation of the critical band model. It should be noted though that the concept of a filter bank is a very coarse description and should be seen as a typical engineering approximation to the real situation.6. It is seen that the bandwidth (Critical Bands) is almost constant at 100 Hz up to a centre frequency of about 500 Hz and above this frequency the bandwidth increases. The increase in bandwidth above 500 Hz is similar to the increase in bandwidth for onethirdoctave filters. It is outside the scope of the present text to go into the background and the details of this model.1 Critical bands The bandwidth of the filters in the filter bank can be determined by means of various psychoacoustic experiments. See Figure 2.1. 2. The results of the investigations are shown in Figure 2. Frequency discrimination is the ability to hear the difference between two tones that are close in frequency (one frequency at a time). Frequency selectivity is important for the perception of the different frequencies in complex sound signals such as speech and music.4. The width of the envelope peak may be seen as an indication of the selectivity of the analyser filter and it has been common practice to describe the frequency selectivity of the ear as a set of filters. We rely e. which cover the audible frequency range.4 f 2 ) 0.Ear. The concept of frequency discrimination is different from frequency selectivity. Hearing and Speech 2. 71 .
The curves are computed from the formulas given in the text. If the audible frequency range is ‘filled up’ with consecutive critical bands from the lowest frequency to the highest frequency.6. it is seen that 24 critical bands will cover the whole frequency range. i. Hz 1000 100 10 10 20 50 100 1000 Frequency. etc. ERB. The rectangular bandwidth may be calculated from the empirical formula: ERB = 24. 72 . This leads to the concept of equivalent rectangular bandwidth. The critical bands are a result of the incoming sound signal and as such much more ‘flexible’ than physical filters would be. Each of the ‘filters’ has been given a number called Bark. Hz 200 500 2000 5000 10000 20000 Figure 2. Bark number two is the band from 100 Hz to 200 Hz. The band around 4000 Hz is no. The bandwidth of such rectangular filters is shown in Figure 2.Ear. 8 has a centre frequency of 1000 Hz and goes from 920 Hz to 1080 Hz. Band no.6. hearing and speech 10000 Bandwidth Crit. 2.e.1 Bandwidth of critical bands and Equivalent Rectangular bandwidth. the bandwidth of a rectangular filter that transmits the same amount of energy as the auditory filter.6. Bark number one is the band from zero to 100 Hz. 17 and has a bandwidth of 700 Hz.7 ( 4. Band ERB 1/3 octave Bandwidth.1 as a function of centre frequency. The bandwidth of 1/3octave filters (straight line) is shown for comparison. The critical bands are not fixed filters similar to the filters in a physical filter bank as the numbers given above may indicate.2 Equivalent Rectangular Bands The auditory filters have also been determined by means of notched noise measurements where the threshold of a pure tone is determined in the notch of a broadband noise as a function of the width of the notch.37 f + 1) where ERB is the bandwidth in Hz and f is the centre frequency in kHz.
The peaks are called formants and the formants are positioned differently for each vowel.1 by a cylindrical tube of length 17 cm. through the mouth cavities and/or the nose cavities and the sound is radiated from the mouth and the nose.1 shows the formants frequencies (in round numbers) for the three most different vowels. The sounds are /i/: as in eve. 73 .7. This structure is simulated in Figure 2. Air is pressed from the lungs up through the vocal tract.7. throat (pharynx) and the mouth.1 Speech production A schematic illustration of the production of voiced sounds is given in Figure 2. There are thus more lines in a male spectrum compared to a female. There are individual differences from person to person. The source spectrum is a line spectrum where the distance between the lines corresponds to the fundamental frequency. Figure 2. The source spectrum is transformed by the ‘tube’ consisting of trachea. From [14] The tube has pronounced resonances (where the length of the tube corresponds to the odd multiples of 1/4 wavelength) indicated by the peaks at 500. 1500 and 2500 Hz. around 250 Hz for woman and around 300 for children.7 Speech A speech signal is produced in the following way.7.7.Ear.1 where the vocal folds vibrate. /a/ as in father. but there are big individual variations. The source spectrum decreases with the square of the frequency (1/f2). /u/ as in moon. 2. The fundamental frequency is around 125 Hz for men.7. The vocal folds will vibrate when voiced sounds are produced. The final spectrum is a line spectrum with characteristic peaks caused by the transfer function. Table 2.1 The principle of vowel generation. Hearing and Speech 2. The final spectrum radiated from the mouth is then the product of the two spectra.
formant 225 2200 3000 /a/ 700 1200 2500 /u/ 250 700 2200 Table 2.2 Speech spectrum. formant 2. German. /a/ and /u/. Swedish. French (Canadian). Singhalese and Vietnamese.Ear. out between the lips and the teeth /f/.7. The maximum is found around 500 Hz for both gender and above 500 Hz the two curves are almost identical. Japanese. Welsh. The spectrum for women falls off below 200 Hz because their fundamental frequency typically is around 250 Hz. The slope above 500 Hz is approximately minus 10 dB per decade (or 3 dB/octave). sudden opening between tongue and teeth /t/ and between tongue and palate /k/. The spectrum in Figure 2. by pressing air out through the teeth /s/. Russian. speech level A general longterm speech spectrum is shown in Figure 2. 74 . It is worth to note that the speech spectrum is almost independent of the language.7. This is not surprising when the speech production mechanism is taken into account. The unvoiced sounds are produced in many different ways. These sounds are called unvoiced because the vocal folds do not vibrate but stays open in order for the air to pass.1 Formant frequencies in Hz of the vowels /i/.g. hearing and speech /i/ 1. A total of 392 talkers participated in the investigation.2 is based on English (several dialects). 2.7. (The result of a FFT is a density spectrum). Cantonese. e. The spectrum is a onethird octave spectrum which means that the curves are tilted 3 dB/octave compared to the result of a FFTcalculation. Danish. formant 3.7. Mandarin.2 that is based on the average of 18 speech samples from 12 languages. by sudden opening of the lips /p/.
3 Speech intelligibility The speech intelligibility of a transmission system is usually measured by means of a list of words (or sentences) where the percentage of correctly understood words gives the intelligibility score. Often the intelligibility score is given as a function of the signaltonoise ratio. For women the level is typically 3 dB lower. numbers. e. The words are recorded on the left channel of the CD and on the right channel a noise signal is recorded with (almost) the same spectrum as the words. (Compare the number of lines in the spectrum). The noise signal is amplitude modulated in order to make it resemble normal speech. 2. 63 dB. the listener. An example of this is shown in Figure 2. a telephone line or a room.7.2 The longterm speech spectrum for male and female speech shown as a 1/3octave spectrum. Hearing and Speech 70 Speech spectrum Male Female 3 dB /oct 60 1/3 octave level 50 40 30 50 100 200 1000 Frequency. The Dantale CD is described in [16] The result in Figure 2.7. measured at 1 m in front of the mouth. During normal speech the level will vary ±15 dB around the mean value. the scoring method and the quality of the transmission system.g. single words. The words are common Danish singlesyllable words that are distributed phonetically balanced over the eight lists so that the lists can be regarded as equivalent.e.Ear. i. For comparison a line with slope –3 dB per octave (= –10 dB per decade) is shown. This CD contains eight tracks of 25 words each. Hz 500 2000 5000 10000 20000 Figure 2. etc. Redrawn from [15] The average level of male speech is about 65 dB SPL.7. The transmission system could be almost anything. It is seen that even at a signaltonoise ratio of 75 .3 for the wordmaterial on the Dantale CD.3 is obtained with the words and the noise on the Dantale CD with untrained Danish normal hearing listeners.). The intelligibility depends on the word material (sentences. the speaker.7.
It is a general finding that such a relatively small improvement of the signaltonoise ratio can improve the intelligibility situation dramatically. Therefore measurement and calculation methods have been developed for the estimation of the expected speech intelligibility in a room or on a transmission line. The index can then be translated to an expected intelligibility score for different speech materials. but the weighting functions are changed and a number of ‘corrections’ to the AImethod are implemented. if the background noise in a room is a problem for the understanding of speech in the room. Articulation Index. In other words. from the source (the speaker) to the receiver (the listener) is determined. hearing and speech 0 dB almost all words are understood.Ear. Redrawn from [17] It is time consuming and complicated to measure speech intelligibility with test subjects. Speech Intelligibility Index. STI [20]: In this method the modulation transfer function. raised voice. then just a small reduction of the background noise will be beneficial. The SNR values are weighted according to the importance of the frequency band.g. AI [18]: Determination of the signaltonoise ratio in frequency bands (usually one octave or onethird octave). SII [19]: This method is based on the AI principle. One of these is the correction for the change in speech spectrum according to the vocal effort (shouting. It is also seen that an increase of just 10 dB in SNR can change the situation from impossible to reasonable. 100 90 Word score.5 dB (70%). dB Figure 2. Speech Transmission Index. The MTF is determined for octave bands of noise (125 Hz to 8 kHz) and for a number of modulation 76 .7. from 15 dB (10%) to . MTF. % 80 70 60 50 40 30 20 10 0 25 20 15 10 5 0 SNR. low voice). The weighted values are added and the result normalised to give an index between zero and one. e.3 Word score for the speech material DANTALE as a function of speechtonoise ratio (SNR).
The index can then be translated to an expected intelligibility score for different speech materials. Hearing and Speech frequencies (0. Rapid Speech Transmission Index. The result is an index which is used in the same way as in STI.Ear. 77 .63 Hz to 12.5 Hz). Only the frequency bands 500 Hz and 2 kHz and only nine different modulation frequencies are used. RASTI [21]: This is an abbreviated version of STI. The reduction in modulation is transformed to an equivalent signaltonoise ratio and as in the AI method these values are added and normalised in order to yield an index between zero and one.
1975. Sound level meters. 1999: Springer.. E. May 2002.Normal equalloudnesslevel contours. A short survey of some common or important ear diseases. International Organization for Standardization: Geneva. ISO3895.8 References 1. and Fastl. ISO3897. B. Acoustics .T. Switzerland. xxyy. 2.Reference zero for the calibration of audiometric equipment Part 1: Reference equivalent threshold sound pressure levels for pure tones and supraaural earphones. Engström. Denmark. and Engström. et al. 2001. Acoustics . . 3. 1988. Facts and models. 1995: Widex.Reference zero for the calibration of audiometric equipment Part 8: Reference equivalent threshold sound pressure levels for pure tones and circumaural earphones (ISO/DIS). 1979. S. 1998. 1979: Widex. Florentine. 1991. International Organization for Standardisation: Geneva. 12. and Buus. ISO3891. ISO226. M..Method for calculating loudness level. ISO532. 2 ed. 1996. D. Switzerland. International Organization for Standardization: Geneva. 11. S.Ear. International Electrotecnical Commission: Geneva. Evidence for normal loudness growth near threshold in cochlear hearing loss. 5: p. IEC537. International Standardization Organization: Geneva. 10. in FDIS.. H. Switzerland. N327. Switzerland. Switzerland. 13. Acoustics . Acoustics . International Electrotechnical Commission: Geneva. Switzerland. in 19 Danavox Symposium. 9. 6. p. Adv Audiol. International Organisation for Standardisation: Geneva. Kemp. 1976. Acoustics . ISO3898.Reference zero for the calibration of audiometric equipment Part 5: Reference equivalent threshold sound pressure levels for pure tones in the frequency range 8 kHz to 16 kHz“. Psychoacoustics. 2745. Frequency weighting for the measurement of aircraft noise (Dweighting). Sound and Hearing. Acoustics . H. 78 7. 2001. 2002. IEC651. 8. 4. hearing and speech 2. 2 ed. 5. International Organization for Standardization: Geneva.. Zwicker. Hougaard. Developments in cochlear mechanics and techniques for noninvasive evaluation. Switzerland.. Kolding.Reference zero for the calibration of audiometric equipmentPart 7: Reference threshold of hearing under freefield and diffusefield listening conditions.
. 318326. Fundamentals of hearing. Soc.5. Scandinavian Audiology. IEC26816. A..Ear. An introduction. Academic press ISBN: 0125056281 Yost.a Danish speech material. Inc. 96(no.ac. and Harris.. 18. Inc. The sense of hearing. Am. 16. 2110?2120? Elberling. ISBN: 0127756957 About standardized audiological. W. 1988.phon. (2000). J.Part 16: The objective rating of speech intelligibility in auditoria by the RASTI method. 17. 1989. 1997. P. Academic press. American National Standard methods for the calculation of the Speech Intelligibility Index.: New York. 18: p. Ludvigsen. G.. An introduction to the psychology of hearing. 21.: New York. G.. Keidser. DANTALE. Sound system equipment . C. ISBN: 0805848843 Moore. T. Borden. e. Am. J. 1980. and al.. 1993. Lawrence Earlbaum Associates. B. 20. and Houtgast. 4): p. C. Longterm average speech spectra . J. Steeneken. 1969. Hearing and Speech 14. 1980: Williams & Wilkins. 1994.. J. 169175. 67: p... H. 15. Scandinavian Audiology. (2005).E. Acoust. 4th Edition. A physical method for measuring speechtransmission quality. 231236. C. American National Standards Institute. Normative data in quiet and in noise for DANTALE .. clinical tests see http://www. Soc. C. (2003). International Electrotechnical Commission..ucl. C. 22: p.5. 5th Edition..uk/home/andyf/natasha/ 79 . ANSIS3. and Lyregaard. Byrne. Ludvigsen. Speech science primer. Further reading: Plack.. 19. K. a new Danish Speech material. American National Standard methods for the calculation of the Articulation Index. American National Standards Institute.. ANSIS3. D. Acoust.
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and lz. The room surfaces are now assumed to be rigid. ω is the angular frequency and c is the speed of sound in air. i. and each of them is related to a certain natural frequency (or eigenfrequency) given by ω ck c 2 fn = n = = k x2 + k y + k z2 2π 2π 2π 81 . The wave equation can then be written ∂2 p ∂2 p ∂2 p + + + k2 p = 0 (3.1.3) k x = ⋅ nx lx Two similar boundary conditions hold for the y.1.4) are called standing waves.1.1) ∂ x2 ∂ y2 ∂ z2 where p is the sound pressure and k = ω /c is the angular wave number. 3.1. With these conditions the solution to (3.3.1 SOUND WAVES IN ROOMS 3. so the waves that are solutions to (3.and zdirections.1. 1.and zdirections. and for the xdirection it yields 1 ∂2 X + k x2 = 0 X ∂ x2 Similar equations hold for the y. The equation can be solved by separation of the variables and it is assumed that the solution can be written in the form: p = X ( x) ⋅ Y ( y ) ⋅ Z ( z ) ⋅ e jω t Insertion in (3. the normal component of the particle velocity is zero at the boundaries 1 ∂p ux = − = 0 for x = 0 and x = lx jωρ ∂ x This means that ϕx = 0 and π where nx = 0.1 Standing waves in a rectangular room A rectangular room has the dimensions lx.1) is ⎛ ⎛ ⎛ x⎞ y⎞ z⎞ ⎜ (3.2) The general solution to the above onedimensional equation is X ( x) = C x cos(k x x + ϕ x ) in which the constants Cx and ϕx are determined from the boundary conditions. An introduction to room acoustics Jens Holger Rindel 3. … (3. The amplitude of the sound pressure does not move with time. They are also called the modes of the room. ly.1.4) p = p0 ⋅ cos⎜ π n x ⎟ ⋅ cos⎜ π n y ⎟ ⋅ cos⎜ π n z ⎟ ⎜ ⎟ ⎜ ⎟ lx ⎠ ly ⎠ lz ⎟ ⎠ ⎝ ⎝ ⎝ The time factor ejωt is understood. The angular wave number k has been divided into three 2 2 k 2 = k x + k y + k z2 (3. 2.1) and division by p gives 1 ∂ 2 X 1 ∂ 2Y 1 ∂ 2 Z + + + k2 = 0 2 2 2 X ∂x Y ∂y Z ∂z This can be separated.1.e.1.
1.5) in a rectangular room with dimensions 5. 82 . 7. Figure 3. only one of nx.1. all three of nx. ny.8 m.1.1. It is observed that the set of numbers (nx.1. nz is > 0.0. two of nx. Tangential modes are twodimensional.⎛ nx ⎞ ⎛ n y ⎞ ⎛ nz ⎞ ⎜ ⎟ +⎜ ⎟ +⎜ ⎟ ⎜l ⎟ ⎜l ⎟ ⎜l ⎟ ⎝ x ⎠ ⎝ y ⎠ ⎝ z ⎠ The modes can be divided into three groups: Axial modes are onedimensional.0 m. 4. nz are > 0.1. ny. ny.7 m. The lines are isosound pressure amplitude curves. nx 0 1 0 1 0 1 0 1 1 2 0 2 1 2 0 ny 1 0 0 1 1 0 2 1 2 0 2 1 2 0 0 nz 0 0 1 0 1 1 0 1 0 0 1 0 1 1 2 fn (Hz) 25 30 36 39 43 47 49 53 58 60 61 65 68 70 72 Table 3.1. (2.1.0) is onedimensional and (2. nz are > 0. 3. c fn = 2 2 2 2 (3. Calculated natural frequencies at low frequencies using (3. nz) indicate the number of nodes (places with p = 0) along each coordinate axis.0) is twodimensional.1. Examples of room modes.5) Some examples are shown in Fig. Oblique modes are threedimensional. ny.
respectively.1. It should be noted that the modal points of the tangential and axial modes in Fig. A measured transfer function is shown in Fig. (4 π f 3 / 3) / 8 = π f 3 / 6. i.1. If these and the axial modes are also taken into account.5) shows that the natural frequencies of a rectangular room may be interpreted in a geometrical way.1. 3.2 Transfer function in a room The transfer function is the frequency response from a source position to a receiver position in a room. the width and the height. respectively.2. Therefore we count the tangential points only as halves and those on the axes only as quarters. representing the axial modes of the length. At low frequencies it is possible to identify the modes by their modal numbers. So. The natural frequencies of the onedimensional modes are marked on each of the axes.2. Each mode occupies a volume c3 / (8 lx ly lz) = c3 / (8 V).1.1. 3. It fluctuates very much with frequency and the maxima can be identified as the natural frequencies of the room. Figure 3. and the distance to each point from the origin is the natural frequency of that mode.6) is a good approximation for any room. the number of oblique modes below a certain frequency f is equal to the number of grid points inside the sphere with radius f.1. 3.1. 3.3 Density of natural frequencies A closer inspection of equation (3. At high frequencies the oblique modes dominate. The volume is 1/8 of the sphere with radius f. 3. The example in Fig. 83 .1. S = 2( lx ly + lx lz + ly lz) is the total area of the surfaces. the number of modes with natural frequencies below the frequency f is: 3 2 4π V ⎛ f ⎞ π S ⎛ f ⎞ L f (3.1.e. Transfer function in a rectangular room.1. The interesting observation is now that the points in the grid represent the oblique modes.1. not only for rectangular rooms.3.2 has the same room dimensions as was used for the calculations in Table 3. and the first term in (3. the number of oblique modes below f is approximately: π f 3 8V 4π V f 3 = N obl = 6 c3 3 c3 The tangential modes are found in the plane between two of the axes.6) N = ⎜ ⎟ + ⎜ ⎟ + 3 ⎝c⎠ 4 ⎝c⎠ 8 c V is the volume of the room.3 are located on the coordinate planes and axes. and L = 4 (lx + ly + lz) is the total length of all edges. So.3. A threedimensional frequency space is shown in Fig.1.
1.8) ≅ 4π 3 f 2 df c 84 . 3.3. The modal density is the average number of modes per hertz.4 this is compared to the actual modal density in a room.1. πS dN V L = 4π 3 f 2 + f + (3.1.Figure 3. For high frequencies it is sufficient to use the first term (oblique modes) for the modal density: dN V (3. in which each grid point represents a room mode.7) 2 df 8c c 2c In Fig. Frequencygrid.1.
2. in many cases the diffuse sound field can be a good and very practical approximation to the real sound field. It is assumed that the modal density is high enough.1. The diffuse sound field is an ideal sound field that does not exist in any room.Figure 3. However. based on energy balance considerations.1 The diffuse sound field In this chapter the acoustical behaviour of a room is treated from a statistical point of view. so the phase relations between individual reflections can be neglected. The diffuse sound field is defined as a sound field in which: The energy density is the same everywhere All directions of sound propagation occur with the same probability It is obvious that the direct sound field near a sound source is not included in the diffuse sound field. This means that the reflections in the room are assumed to be uncorrelated and their contribution can be added on an energy basis. so the influence of single modes in the room can be neglected.1. Neither are the special interference phenomena that are known to give increased energy density near the room boundaries and corners. 85 . Modal density as a function of frequency. 3.2 STATISTICAL ROOM ACOUSTICS 3. Actual number of modes per 10 Hz in a rectangular room and estimated by (3.4. It is also assumed that the reflection density is high enough.7).
I inc = 2 p diff (3.2. the incident sound power per unit area on the surface is 2 p diff cosθ I θ = I 1 cosθ = (3.2. see Fig. b: Diffuse incidence on a surface.2 Incident sound power on a surface In a plane propagating sound wave the relation between rms sound pressure p1 and sound intensity I1 is p12 = I 1 ⋅ ρ c In a diffuse sound field the rms sound pressure pdiff is the result of sound waves propagating in all directions.2) 4π ρ c where pdiff is the rms sound pressure in the diffuse sound field.2.1) pdiff = ∫ I1 ⋅ ρ c dψ = 4π ⋅ I 1 ⋅ ρ c ψ = 4π In the case of a plane wave with the angle of incidence θ relative to the normal of the surface. The integration covers the solid angle ψ = 2π. The total incident sound power per unit area is found by integration over all angles of incidence covering a half sphere in front of the surface.1. This is just the sound intensity in the plane propagating wave multiplied by the cosine. and all having the sound intensity I1. which is the projection of a unit area as seen from the angle of incidence.2.3) 86 .1. see Fig. 2 1 2π π / 2 pdiff I inc = ∫ Iθ d ψ = cos θ sin θ d θ d ϕ 4π ∫0 ∫0 ρ c ψ = 2π = 2 pdiff 1 ⋅ 2π ⋅ 4π ρc ∫ 1 0 sin θ d(sin θ ) = 2 1 pdiff 1 ⋅ ⋅ 2 ρc 2 4ρ c It is noted that this is four times less than in the case of a plane wave of normal incidence. By integration over a sphere with the solid angle ψ = 4π the rms sound pressure in the diffuse sound field is 2 (3. p1 pdiff θ Iθ Iinc a b Figure 3.2. a: Plane wave at oblique incidence on a surface. 3.3.2.2. 3.2.
The product of area and absorption coefficient of a surface material is the equivalent absorption area of that surface..e.4).6) 4ρ c If Pa is the sound power of a source in the room.. The energy absorbed in the room is the incident sound power per unit area (3.e.2. α = 1 is an open window. In general. the equivalent absorption area may also include sound absorption due to the air and due to persons or other objects in the room.3 Equivalent absorption area The absorption coefficient α is defined as the ratio of the nonreflected sound energy to the incident sound energy on a surface. or twice the potential energy.2.abs = I inc Sα m = I inc A = A (3. i.. The unit of A is m2. The total energy E is the energy density multiplied by the room volume V: p2 E = ( w pot + wkin ) V = 2w pot V = V (3.2.5) ρ c2 Here and in the following. and α = 1 means that all incident sound energy is absorbed in the surface. = Sα m i (3.2. i. since the time average of the two parts must be equal. The equivalent absorption area of a room is A = ∑ Siα i = S1α1 + S 2α 2 + ..4 Energy balance in a room The total acoustic energy in a room is the sum of potential energy and kinetic energy.2. the energy balance equation of the room is 87 .2.2.2). Definition of angles of incidence in a diffuse sound field. An example of a surface with absorption coefficient.2.Figure 3. p2 Pa . It can take values between 0 and 1.2. 3.3) multiplied by the total surface area and the mean absorption coefficient.4) where S is the total surface area of the room and αm is the mean absorption coefficient. the area of open windows giving the same amount of sound absorption as the actual surface. the equivalent absorption area (3. p denotes the rms sound pressure in the diffuse sound field (called pdiff in section 3.2. 3.
and the steady state sound pressure in the room is 4 Pa (3. and the rms sound pressure is now a function of time: A V d 2 (3. The approximation comes from neglecting the term with the constants and reference values ρ c Pref 1.3. ⎛ 4A ⎞ L p ≅ LW + 10 log⎜ 0 ⎟ (dB) (3. So.7b) is zero.2. 3. also shown in Fig.2. follows an exponential decay function.Pa − Pa . (3.2. and this is called the decay curve. and hence the sound energy.12) 88 . Sabine´s formula If the sound source is turned off after the sound pressure has reached the stationary value.2.2.204 ⋅ 343 ⋅ 10 −12 10 log = 10 log = 0. For some cases it is more convenient to express eq. the sound buildup in the room follows a similar exponential curve.8) p s2 = ρc A This equation shows that the sound power of a source can be determined by measuring the sound pressure generated by the source in a room.2.2. the first term in the energy balance equation (3.2. It also shows how the absorption area in a room has a direct influence on the sound pressure in the room.14 dB ≅ 0 dB 2 1 ⋅ (20 ⋅ 10 −6 ) 2 A0 pref Pa − A= p2 3.11) 2 where ps is the mean square sound pressure in the steady state and t = 0 is the time when the source is turned off.2. provided that the equivalent absorption area of the room is known. the absorbed power equals the power emitted from the source.10) p 2 (t ) + ( p (t )) = 0 4ρ c ρ c2 d t The solution to this equation can be written p (t ) = p e (3.5 Reverberation time. 2 2 s − cA t 4V If instead the source is turned on at the time t = 0.7a) V d ( p2 ) (3. see Fig. p (t ) = 2 cA − t ⎞ ⎛ ⎜1 − e 4 V ⎟ p ⎜ ⎟ ⎝ ⎠ 2 s (3.7b) 4ρ c ρ c2 d t With a constant sound source a steady state situation is reached after some time. On a logarithmic scale the decay is linear.2.2.abs = dE dt (3.8) in terms of the sound pressure level Lp and the sound power level LW . It is seen that the mean square sound pressure. 3.9) ⎝ A ⎠ where A0 = 1 m2 is a reference area.2.3. and the right side of the equation is zero.
Reverberation distance A reverberation room is a special room with long reverberation time and a good diffusion.2. Top: linear scale (sound pressure squared).3 V Pa = ⋅ (3. and the sound power level is calculated from 89 . However.2. 3. for t = T60 . 2 2 s 2 s −6 − cA T60 4V Note: Sabine’s formula is often written as T60 = 0.2.3 V (3. the reverberation time is 4V 55.3. He was the first to demonstrate that T60 is inversely proportional to the equivalent absorption area A.13) = T60 = 6 ⋅ ln(10) ⋅ cA cA This is Sabine’s formula named after Wallace C. Bottom: logarithmic scale (dB). this implies that V must be in m3 and A in m2. i.16 V/A. Buildup and decay of sound in a room.6 Stationary sound field in a room.2. who introduced the reverberation time concept around 1896. Sabine. The reverberation time T60 is defined as the time it takes for the sound energy in the room to decay to one millionth of the initial value. a 60 dB decay of the sound pressure level.e.2. and the results for stationary conditions (3. Hence.Figure 3. In such a room the diffuse sound field is a good approximation.8) and for sound decay (3.14) 4 ρ c c T60 The reverberation time and the average sound pressure level in the reverberation room are measured. p (t ) = p 10 = p e So.13) can be applied to measure the sound power of a sound source: p s2 55.2. Here the source is turned on a t = 0 and turned off at t = 1 s.
since the reverberation distance depends only on the equivalent absorption area A rrev = = 0. which is the fraction of the sound power emitted to the room after the first reflection. and in this socalled far field the stationary. the direct sound field dominates.18) 16π At a distance closer to the source than the reverberation distance.LW = L p + 10 log 2 p ref ⋅ 55. the sound pressure squared of direct sound in the distance r from the source is Pa 2 (3.2.αm). It is a descriptor of the amount of absorption in a room.2. only one surface is highly absorbing A lack of diffusing or sound scattering elements in the room The ratio of longest to shortest room dimension is higher than three The volume is very large.2. diffuse sound field may be a usable approximation. An expression for the combined direct and diffuse sound field can derived by simple addition of the squared sound pressures of the two sound fields. and this is called the direct field.g. To do this. At longer distances the reverberant sound field dominates. the sound power of the source should be reduced by a factor of (1 .19) ⎜ r2 ⎟ ⎝ ⎠ ⎛ 1 ⎞ 4 2 ptotal = Pa ⋅ ρ c ⎜ (3. In most ordinary rooms the diffuse sound field is not a good approximation. The sound power radiated by an omnidirectional source is the sound intensity at the distance r in a spherical sound field multiplied by the surface area of a sphere with radius r (3. However.3 ⋅ V Pref ⋅ 4 ρ c 2 ⋅ T60 (3.2. Each of the following conditions may indicate that the sound field is not diffuse An uneven distribution of sound absorption on the surfaces.14 A (3. it should be extracted from the energy balance equation. which was used to describe the diffuse sound field. So.2.16) Pa = I r ⋅ 4π r 2 Thus. since the direct sound is treated separately.2. the squared sound pressure in the total sound field is ⎛ r2 ⎞ 2 2 ptotal = p dir + p s2 (1 − α m ) = p s2 ⎜ rev + 1 − α m ⎟ (3.8) 4 Pa p s2 = ρc A The reverberation distance rrev is defined as the distance where pdir2 = ps2 when an omnidirectional point source is placed in a room. e.15) T V = L p + 10 log − 10 log 60 − 14 dB V0 t0 3 where V0 = 1 m and t0 = 1 s.17) pdir = ρc 4π r 2 The stationary sound is described by (3.2. say more than 5000 m3 A rather simple modification to the stationary sound field is to separate the direct sound.20) ⎜ 4π r 2 + A (1 − α m ) ⎟ ⎟ ⎝ ⎠ 90 .
Figure 3. In large rooms with medium or high sound absorption (say. In the case of a highly directive sound source like a trumpet (Q >> 1) the direct field can be extended to distances much longer than the reverberation distance. in large industrial halls the attenuation in dB per doubling of the distance may be a better descriptor than the reverberation time. In the latter situation the last term in (3. the squared sound pressure of the direct sound is Q ⋅ Pa 2 (3. αm < 0. the slope of the spatial decay curve may be taken as a measure of the degree of acoustic attenuation in a room. αm > 0.2) the sound pressure level will continue to decrease as a function of the distance.1) the sound pressure level in the far field will be approximately as predicted by the diffuse field theory. 4π r 2 (3. Instead.4. ⎛ r2 ⎞ 4A L p ≅ LW + 10 log 0 + 10 log ⎜ Q rev + 1 − α m ⎟ (dB) (3.Normal sound sources like a speaking person. i.21) Q=I⋅ Pa So. The directivity factor Q is the ratio of the intensity in a certain direction to the average intensity. because the diffuse field theory is not valid in such a room.2.23) raises the sound pressure level above the diffuse field value.2. So. 91 . In a reverberant room with little sound absorption (say.2.22) pdir = ρc 4π r 2 This leads to a general formula for the sound pressure level as a function of the distance from a sound source in room.23) ⎜ r2 ⎟ A ⎠ ⎝ where A0 = 1 m2. The source has a directivity factor of one. The parameter on the curves is A / (1 .2.2. Relative sound pressure level as a function of distance in a room with approximately diffuse sound field.e. a loudspeaker or a musical instrument radiate sound with different intensity in different directions.αm) in m2. the last term will be close to zero.
see Fig.3. only in rather reverberant rooms. So. The initial sound pressure is p0 and after n reflections the squared sound pressure is 2 2 (3.3. The ray representing a plane wave may start in any direction and it is assumed that the decay of energy in the ray is representative for the decay of energy in the room. and only the direction of sound energy propagation is treated in geometrical acoustics.8 ⋅ lm T60 = ≈ − c ⋅ ln(1 − α m ) c ⋅αm The last approximation is valid if αm < 0.1.1 Sound rays and a general reverberation formula In geometrical acoustics rays are used to describe the sound propagation.3.1) p 2 (t ) = p0 ⋅ (1 − α m ) n = p0 ⋅ e n ⋅ ln(1−α m ) The distance of the ray from one reflection to the next is li and the total distance traveled by the ray up to the time t is (3.e. The approximation comes from: ⎛ 1 ⎞ α2 α3 ⎟ =α + − ln(1 − α m ) = ln⎜ + +L ⎜ ⎟ 2 3 ⎝1 − αm ⎠ 10 −6 =e c ⋅ ln(1−α m ) ⋅ T60 lm ⇒ − 6 ⋅ ln(10) = (3. all of which are assumed to have the mean absorption coefficient αm. 3.αm).3 GEOMETRICAL ROOM ACOUSTICS 3. Figure 3.3. The concept of rays implies that the wavelength and the phase of the sound are neglected. A plane wave travelling as a ray from wall to wall in a room.3.2) ∑ li = c ⋅ t = n ⋅ l m i where lm is the mean free path.3) 6 When the squared sound pressure has dropped to 10 of the initial value. The sound decay shall now be studied by following a plane wave travelling as a ray from wall to wall. By each reflection the energy is reduced by a factor (1 . the time t is by definition the reverberation time T60: 2 2 0 c ⋅ ln(1−α m ) ⋅ t lm c ⋅ ln(1 − α m ) ⋅ T60 lm This leads to an interesting pair of general reverberation formulas: 13.1. i.3.4) 92 . the squared sound pressure is p (t ) = p ⋅ e (3.3.3.8 ⋅ lm 13.3. The energy of the wave is gradually decreased due to absorption at the surfaces. The room may have any shape.
which depends on the temperature and the relative humidity of the air.2.8) S i In the extreme case of an anechoic room (αm = 1) Eyring’s formula gives correctly a reverberation time of zero.3 ⋅ V (3.3.4): 1 α m = ⋅ ∑ S iα i (3.4) gives the Sabine formula (3. However. The onedimensional case is just the sound travelling back and forth between two parallel surfaces with the distance l = lm.3.5) lm = S where V is the volume and S is the total surface area.2.9) The general reverberation formula then becomes 13. If this attenuation is included in (3.3) it gives the same result as Sabine’s formula. giving the value T60 = 55.11) ≈ T60 = c (− S ⋅ ln(1 − α m ) + 4 mV ) c ( S ⋅ α m + 4 mV ) These two expressions are the Eyring and the Sabine formula.3.13) it is seen that the equivalent absorption area including air absorption is A = ∑ S iα i + 4 mV (3. with the air absorption included. whereas Sabine’s formula is obviously wrong.3.1.3.3. Similarly.3) the squared sound pressure in the decay is p (t ) = p ⋅ e ⋅e = p ⋅e (3. 93 .3 V/c S.2 Sound absorption in the air A sound wave travelling through the air is attenuated by a factor m.3.4.3. see Fig.8 ⋅ l m (3.3.12) 2 2 0 2 0 i c ⋅ ln(1−α m ) ⋅ t lm −mct ct ⋅( ln(1−α m ) − m⋅ lm ) lm Some typical values of m are found later in Table 3.13).2.With the assumption that all directions of sound propagation appear with the same probability. whereas insertion in the first part of (3. the mean free path in a twodimensional room can be derived. but in highly absorbing rooms Eyring’s formula is theoretically more correct. or structureborne sound in a plate.3 ⋅ V 55. it can be show (Kosten.6) lm = U where Sx is the area and U is the perimeter. in normal rooms with a mixture of different absorption coefficients it is recommended to use Sabine’s formula.8 ⋅ lm 13.3.5) in the last part of (3.3.3.10) ≈ T60 = c ( − ln(1 − α m ) + m ⋅ l m ) c (α m + m ⋅ lm ) In the threedimensional case with (3.4) leads to the socalled Eyring’s formula for reverberation time in a room: 55. Insertion of (3. The height or thickness must be small compared to the wavelength.2. 3.3 ⋅ V (3. The unit of the air attenuation factor is m . By comparison with (3.5) we then have 55. In practice the absorption coefficients are not the same for all surfaces and the mean absorption coefficient is calculated as in (3. In this case the mean free path is π Sx (2dimensional) (3.3. 1960) that the mean free path in a threedimensional room is 4V (3dimensional) (3. This could be the narrow air space in a double wall.7) T60 = − c ⋅ S ⋅ ln(1 − α m ) In a reverberant room (αm < 0.3.3. respectively.3. 3.
: Harris1966). The air temperature is 20 °C.3. (Ref.Figure 3. 3. i. This principle can be extended to higher order reflections. as known from optics. 3. Reflection in one surface (a) and in two surfaces (b). The air attenuation factor m as a function of the relative humidity.e. see Fig. the angle of reflection is equal to the angle of incidence.3. Figure 3.3. A is the source and R is the receiver. 94 . First order image sources are indicated by A’ and second order image sources by A’’.3.3.3.3 Sound reflections and image sources The direction of a sound reflection from a large plane surface follows the same geometrical law. This means that the reflected sound can be interpreted as sound coming from an image source behind the reflecting surface.2.
Rectangular room with a sound source and image sources. It is defined as a single sound reflection that is clearly audible as separate from the direct sound.3. Reflections from room surfaces outside the ellipse (as R2 on the figure) are delayed more than 50 ms and may cause an echo at the receiver point.4 Reflection density in a room The image source principle can easily be applied to higher order reflections in a rectangular room. i. The echoellipse in the longitudinal section of an auditorium. and each cell in the grid is an image room containing an image source.: Petersen 1984). (Ref. 3. L is the source and P the receiver. The socalled echoellipse is shown in Fig.5. 95 .4. Figure 3.3. Any point E on the ellipse represents a potential reflection with a delay of 50 ms. here shown in two dimensions. 3.3. Image sources located inside the circle with radius ct will contribute reflections up to time t.3. 3.5. Figure 3.3.Echo is a wellknown acoustic phenomenon. An infinite number of image rooms make a grid. the distance LE + EP = 17 m.e. The principle is shown for the twodimensional case in Fig.4. The human ear is able to hear a reflection as an echo if the time delay is approximately 50 ms.
This would be very unfortunate. 3.2 Reflection control In room with an audience it is very important to design the room surfaces with respect to the early reflections. especially at low frequencies in small rooms for speech. see also Table 3. whereas rooms for music should not give too much reflection directly from the ceiling. Figure 3.4. 3.1. 3. 3.4. A good acoustical design of a room implies that the transfer function should be as smooth as possible. see Fig.1 Choice of room dimensions The room dimensions determine the natural frequencies of a room. 96 .4.3.4.1.1.3.: Petersen 1984). music or acoustic measurements. it is striking to observe the analogy between reflection density in the time domain and modal density in the frequency domain. a) concave ceiling causing focusing and uneven sound distribution. but also to ensure a good distribution of reflections to the audience area.2 is clear that the room dimensions of a rectangular room should not be identical.8). With reference to Fig. If (3.3.4. see Fig. and by differentiation: dN c3 2 = 4π t (3.14) is compared to (3.4 ROOM ACOUSTICAL DESIGN 3. b) plane reflectors causing an even sound distribution.13) N (t ) = V The reflection density is then the number of reflections within a small time interval dt. The dimensions of such rooms should be designed after calculations of the normal modes below 100 Hz. so the higher order reflections are normally so dense in arrival time that it is impossible to distinguish separate reflections.1. (Ref. because in a cubic room many modes will have the same natural frequency. In such room the ceiling should rather give diffuse reflections. In rooms for speech the ceiling reflections are most important.If an impulse sound is emitted the number of reflections that will arrive within the time t can be calculated as the volume of a sphere with radius ct divided by the room volume V: 3 4 3 π (ct ) (3.14) dt V The reflection density increases with the time squared. but the side walls are important because lateral reflections contribute to the acoustic of a concert hall. 3.1. First of all in order to avoid problems with echo and focusing.1. Ceiling reflections in auditoriums.2. and thus there will be bigger gaps in the transfer function.
Typical values of absorption area A in m2 for persons.43 1.2.95 Frequency (Hz) 500 1000 0.02 0.2.3.4.1) ≅ T60 = cA A with volume V in m3 and A in m2.35 0. if relevant (3.4.3 V 0.4.03 0.4. Persons Standing.3.60 250 0.10 250 0.55 0.11 0.4.3 Calculation of reverberation time Sabine’s formula (3. c) inverse fan shape room.04 0. the absorption from persons or other items in the room should be included.01 0.03 0.2.98 0.06 0. Wall reflections in auditoriums.08 4000 1.04 0.12 0.4. 100 mm air space 125 0. The air attenuation can be taken from Table 3.08 0.25 0.4. 97 .2) A = ∑ S iα i + ∑ n j A j + 4 mV i j Here nj is the number of items.12 1.16 V (3.2. with overcoat Sitting musician with instrument 125 0. b) fan shape room.02 0. bare concrete Parquet floor on studs Needlepunch carpet Window glass Curtain draped to half its area.Figure 3.13) is the most well known and simple method for calculation of reverberation time in a room 55. The equivalent absorption area is calculated as in (3.59 0.06 1. normal clothing Standing. but in addition to absorption from surfaces and air.41 0.4.20 0. Frequency (Hz) 500 1000 0. Typical values of the absorption coefficient α for some common materials.25 2000 0. respectively.35 0.08 0. each contributing with an absorption area Aj.12 0.1. 3.47 1. a) rectangular room.4.04 0.12).14 0.1 and 3.10 0.08 2000 1.07 0.70 4000 0.24 0. Examples of absorption coefficients of common materials and absorption areas for persons are given in Table 3.07 0.16 0.17 0.18 0.91 1.13 1.70 Table 3.30 1.08 Table 3.02 0.65 Material Brick.
5 m 98 .3. but all other surfaces are acoustically hard (α = 0.Relative humidity (%) 40 50 60 70 80 1 kHz 0. A shorter reverberation time will appear in a room in which the first reflections are directed towards the most absorbing surface. l 13.4. 3. Examples of air attenuation factor m (m1) at a temperature of 20°C.0061 0.0023 0.4).0143 0.3.0024 0.0011 0.0237 0.4. Volume V = 5 ⋅ 10 ⋅ 20 = 1000 m3 Surface area S = 700 m2 Equivalent absorption area A = 200 ⋅ 0.0053 0.4.) lm = π Sx / U = π ⋅ 200 / 60 = 10.45 Mean free path (3dim.4.0056 0.7 m Mean free path (2dim.1).8 ⋅ lm (lm in m) (3. The ceiling has a high absorption coefficient (α = 0. 3.4. there is a possibility of prolonged decay in certain directions.0009 0.8). As an example the room in Fig.8 + 500 ⋅ 0.4.30 Mean absorption coefficient (height) αm = (0.4 Reverberation time in nondiffuse rooms In a room with the sound absorption unequally distributed on the surfaces the assumption of a diffuse sound field is not fulfilled.0072 0.3. The measured reverberation time may be either shorter or longer than predicted by Sabine’s formula.0051 8 kHz 0.0162 0. In order to give an idea of the problem it is possible to calculate the different reverberation times associated to onedimensional decays in each of the three main directions using the general reverberation formula (3.0133 Table 3.0021 0.3) T60 ≈ ≈ 0.0192 0. In a rectangular room without sound scattering surfaces or elements.0026 0. In an auditorium this is typically the floor with the audience.1 = 210 m2 Mean absorption coefficient αm = A / S = 210 / 700 = 0.0009 0.1 b.1) / 2 = 0.04 ⋅ m c ⋅αm αm Figure 3.0020 0.0010 0.0008 Frequency 2 kHz 4 kHz 0.8 + 0. 3. and thus Sabine’s formula will not be reliable.3 is considered. A rectangular room with indicated absorption coefficients. see Fig.) lm = 4 V / S = 4 ⋅ 1000 / 700 = 5.
e. and the reverberation time will be reduced.8 0.10 0. In the onedimensional case it is strictly not correct to use the arithmetic average of the absorption coefficients. and thus the measuring result depends on which part of the decay curve is considered for the evaluation of reverberation time.5 to 2. It could be furniture or machines on the floor or some diffusers on the walls.4 s.6 4. 3. it will come closer to the Sabine value.1) it is seen that the mean absorption coefficient should be calculated from (1 − α m ) = (1 − α1 )(1 − α 2 ) (3.9 s and independent of the frequency between 100 and 4000 Hz in order to obtain good acoustical conditions for speech. For music the reverberation time may be up to 50% longer at low frequencies (125 Hz) and somewhat shorter at high frequencies. In workshops with noise sources it is important to have a reverberation time as short as possible.7 10.4.5 Optimum reverberation time and acoustic regulation of rooms The optimum reverberation time depends of the activities in the room.The results are shown in Table 3.0 s and 0. introducing some sound scattering elements in the room can have a significant effect.4.4. if one of them is high. In a room with long reverberation time due to nondiffuse conditions and at least one soundabsorbing surface.2 s at mid frequencies (500 – 1000 Hz) with the longer values in the bigger halls. (horizontal) 1dim.30 0. In schools the classrooms should have a reverberation time between 0.4. 3. αm = 1 and hence the reverberation time is zero. (width) 1dim. i.10 0.7 5. The measured decay curve will be bent.6 and 0.4. (Sabine) 3dim. By inspection of (3. (length) 1dim.0 4. It is important to choose the room volume and the surface materials with such sound absorbing properties that the reverberation time can get the right value for the purpose. 99 . Note. and the reverberation time will be considerably longer than predicted from Sabine’s formula. 8. if one of the surfaces is reflective and the other is totally absorbing.3.5 20 10 5 Direction 3dim. This will make the sound field more diffuse.3. The real decay that is measured in the room will be a mixture of these different decays. A twodimensional reverberation in the horizontal plane between the walls has also been calculated (4. In other words: The sound absorption available in the room becomes more efficient when scattering elements are introduced to the room. respectively! lm (m) 5. (Eyring) 2dim.2 s). The latter is unavoidable in a big hall due to the air attenuation.4. Calculation of the onedimensional reverberation times of the rectangular room in Fig. The onedimensional decays are the extreme cases with the longest reverberation time being 20 times the shortest one.4) So. (height) αm 0. In concert halls the reverberation time should be around 1.4.2 8.4 Table 3.0 0.10 0.30 0.45 T60 (s) 0. Eyring’s formula is even worse.
3. 3.5 – 2. Optimum reverberation time at mid frequencies for various purposes in rooms with an audience.Use of room Cinema Rock concert Lecture Theatre Opera Symphony concert Choir concert Organ music Optimum reverberation time. So.4.2 1. If the sound in the room is sufficiently diffuse and a sufficient large number of modes are excited the decay curve is close to a straight line between the excitation level and the background level.7 1. The lower part of the decay curve is influenced by the background noise and the upper part may be influenced by the direct sound.4.8 – 1.4 – 1. the part of the decay curve used for evaluation begins 5 dB below the average stationary level and ends normally 35 dB below the same level. s (500 – 1000 Hz) 0.6 Measurement of reverberation time The reverberation time in a room can be measured with a noise signal or with an impulse. Figure 3.1 0.0 – 1.0 0. From the microphone the signal is led to a frequency filter.0 – 3.2 1. Typical decay curve measured with noise interrupted at the time t = 0.4.8 – 1.5.5 2. The traditional method uses white noise emitted by a loudspeaker and a microphone to measure the sound pressure level as a function of time after the source is switched off.7 – 2.4. which is either an octave filter of a onethird octave filter. which gives a steeper start of the curve.3 – 1. The evaluation range is thus 30 dB and the slope is determined by fitting a straight line or 100 .0 Table 3.4.2 1. The dynamic range is seldom more than around 50 dB and the whole range of the measured decay curve is not used. This gives a decay curve and a typical example is shown in Fig.4.
1964.W. The result is sometimes denoted T30 in order to make it clear that the actually used evaluation range is 30 dB. i. 101 . If the background noise is too high and a sufficient dynamic range is not available the reverberation time can instead be measured as T20. 3. Absorption of sound in air versus humidity and temperature. Acoustica 10. Collected papers on acoustics.e. The mean free path in room acoustics. pp 148159. i. Harris (1966). J. New York. W.and receiver positions. a room divided into sections with different reverberation times. because the subjective evaluation of the reverberation is related to the initial decay. SBIanvisning 137. One reason can be that it is a measurement at low frequencies in a small room and maybe only two or three modes are excited within the frequency band of the measurement. however. Inc. and in each position the decay is determined as an average of a number of excitations. Sometimes the decay curves are not nice and straight and it is difficult to measure a certain reverberation time. JASA 40. It might be possible to determine both of these reverberation times. In this case there may be interference between the modes causing very irregular decay curves. The reverberation time is measured in a number of source. White noise is a random noise signal and thus the measured decay curves are always a little different. Sabine (1922).5 REFERENCES C. Dover Publications.C. From the slope of the decay curve in dB per second is calculated the reverberation time. the upper part shows a short reverberation time and the lower part shows a longer reverberation time. C. which is the time for a 60 dB drop following the straight line. Rumakustik (in Danish).e. A typical example is a theatre with a reverberant stage house and a rather dead auditorium. Danish Building Research Institute.M. In this case the decay curve will be bent. the shorter one representing the initial decay is the most important one.automatically by calculating the slope of a linear regression line. Petersen 1984). Kosten (1960). pp 245250. Another difficult situation is coupled rooms. In this case the slope of the decay curve is evaluated between –5 dB and –25 dB below the excitation level.
.
primarily absorb low frequency sounds. general demands on reverberation control exist. these techniques are not always very reliable. In Section 1. For certain types of rooms. Therefore the last section in this chapter is devoted to examples on how sound absorbing materials can be applied in the design of such rooms.1(c). a method for measuring the absorption coefficient.1). Gade 4. the tube method. e.2.4 Sound absorbers and their application in room design Anders Chr. In this chapter we will give a basic introduction to the physical mechanisms involved in sound absorption and present some types of sound absorption materials well suited for . 4. according to Equation 3. people and machinery). Therefore.2.or specifically designed for . the absorption will normally depend on the direction of sound incidence1.2) mainly absorb middle and high frequencies.g. which provides the relevant diffuse field absorption coefficient: the reverberation room method.2 The room method for measurement of sound absorption.5.5.1) is a very basic part of room acoustical design which in turn calls for the availability of reliable data on the frequency dependant sound absorption characteristics of materials used for room surface cladding and for furnishing of rooms (such as furniture.4.2. was presented which reveals the absorption coefficient for a single angle of incidence (usually normal incidence as illustrated to the left in Figure 4.1 Introduction The reverberation time T60 as defined in Section 3.4.5 is the most important descriptor of the acoustics of a room.g. In order to obtain a well balanced T60 versus frequency for a given type of room it is therefore important to mix properly different types of materials when designing the room.2. However. calculating predictions of T60 (e. 4. On the other hand. windows and wooden floors on studs.2.sound absorption and reverberation control. Materials applied in rooms with a (more or less) diffuse sound field will be exposed to sound arriving from many different directions as illustrated in Fig. Figure 4. curtains and persons (see Table 3.1 – or as a function of angle of incidence can be measured using various techniques using separation in time or subtraction of incident and reflected sound pulses.2. 1 103 . Therefore we will start this chapter by presenting a method for measurement of sound absorption. From [1] The absorption for oblique incidence as illustrated in case (b) in Figure 4. However. In Table 3.4. The values indicate that some of these. The absorption properties will be described in terms of the sound absorption coefficient as defined in Section 1.2.1 absorption coefficients per octave band were listed for some materials generally found in rooms. such as schools and work rooms.1 Different conditions for sound incidence on a surface.
3.4. if the of different materials versus area available absorption material can be provided in smaller (measured in square feet). pieces and spread out over the room surfaces. Hereby it can be assumed that the sound field will fulfil the requirements for application of the Sabine reverberation equation. sound diffusing elements. the equation changes into: 0. Combining equations 4. with highly irregular or non parallel surfaces and/or suspended. 4.empty ⎥ ⎣ ⎦ The measurement is normally carried out in 1/1 or 1/3 octave bands from 100 to 5000 Hz. 104 . this phenomenon can be applied successfully in Fig. as seen in Figure 4.empty = (4.2 (disregarding air absorption) yields: 0.16 V T60. In this case equations 3.1. of the test sample: 0.1 and 4. these sometimes appear to have a absorption coefficient larger than 1.1) S Room α empty If now we place a test sample of a material with area Ssample (usually 10 m2) in the room. The phenomenon is probably due to diffraction of sound around the edges of the sample.16 V ⎡ 1 1 ⎤ (4.2.0. total surface area S and that αempty is the absorption coefficient of the room surfaces (which ideally should all be made from the same.2.2) T60. Assume the room has a volume V. 4.2 Absorption coefficients practice by providing increased absorption effect. From [1]. 4. sample = S sample α sample + ( S Room − S sample )α empty in which we have considered that an area. which dominates the behaviour in cases where the linear dimension of the sample approaches the wave length of the sound. Although a complication in documentation of absorption properties. if the absorption power should be related solely to the physical area of the sample.e. resonating and membrane absorbers respectively.3) − α sample = ⎢ ⎥ + α empty S sample ⎢ T60. of the room surface has now been covered by the sample. the effect is more pronounced at low frequencies.4.1 Typical behaviour of absorption versus frequency for Porous.16 V (4. If absorption measurements using the room method is carried out on small sized samples. Of course this is not logical. each with its own characteristic frequency dependency of the absorption coefficient as sketched in Fig. Fig.3. i. αsample.sample T60. 4.The measurement takes place in a reverberation room. acoustically hard material).2.3 Different types of sound absorbers In this section the three most common types of sound absorbing constructions will be described.1 and 3. Ssample.2 by eliminating S yields for the unknown absorption coefficient.
2. one can save material and just place a thin sheet (but still Fig.4. In other words. Figure 4. From [1]. with a rigid termination. 105 . Thus. so that friction takes place where the energy is primarily kinetic.g. In other words.3. If a porous sheet of a certain thickness is placed flush on a rigid surface and hit by an normal incidence sound wave a standing wave pattern will be created with pressure amplitude as indicated to the left in Fig. The absorption properties are caused by viscous friction between the moving air molecules in the sound waves and the often huge internal surface area of the structure whereby the (kinetic) sound energy is converted into heat. 4. the particle velocity and so the kinetic energy of the sound field will be high where the pressure amplitude (the potential energy) is low. As seen from Figure 1.3. there is a lower limiting frequency below which the absorption drops off because the material can no longer “reach” the region of high kinetic energy. Right: Absorption versus frequency of a thin. porous mortar in (unpainted !) brick walls and not least as a wide variety of dedicated sound absorbing products for suspended ceilings.1 Porous absorbers Porous absorbers are present in rooms in the form of textiles like curtains. Porous materials are characterized by having an open structure of e.3 Absorption coefficients for mineral wool (glasswool) with thickness as parameter (a) and with wall distance as parameter (b). carpets and furniture upholstery.g. as the absorber is not absorbing the potential energy anyway. On the other hand. the thickness of the porous layer need to be at least λ/4. 4. for a given thickness of the material.3. of fibres glued or woven together which is accessible by the air. by how densely a fabric is woven – try for yourself by blowing through clothing or curtains !).2.10 (c). air can be pressed through the material more or less easily depending on the flow resistance (determined e.2 Left: Standing wave pattern formed by an incident and a reflected sound wave in front of a porous material of a certain thickness flush mounted on a heavy and hard surface. for the absorber to be efficient (with normal incidence of the sound wave). porous sheet placed in front of a hard surface.3.
Fig. f0 = Fig. 4. which is determined by the depth. 4. if one tried to repaint them. Hereby the ceiling can absorb efficiently over a wide frequency range – as well as hide the installations. very often in the form of tiles which can be mounted in a suspended ceiling system. Mineral wool consists of thin fibres pressed and glued together.5) ⎢⎜ ⎟ + ⎜ ⎟ ⎥ m L m ⎢⎝ a ⎠ ⎝ b ⎠ ⎥ 12 (1 −ν 2 ) ⎣ ⎦ in which a and b are the dimensions of the plate (or the distance between studs supporting the plate). the porous properties and so the absorption normally disappears. Diffuse field incidence also causes the absorbers to be effective (α > 0. This system can resonate at frequency determined by the mass per unit area of the plate.with a suitable flow resistance) at a certain distance from the rigid wall (like a curtain in front of a window). m.3 shows how the absorption coefficient varies with frequency for mineral wool mats of different thickness (upper graph) and different distances to the rigid wall (lower graph). Mineral wool is used as porous sound absorbers. The fibres are made from melted glass (Glasswool) or stone (Rockwool) much like “Candy Floss”. but with diffuse field incidence this dip will not be very pronounced. and the spring function of the enclosed air.4 only apply if the plate is completely limp.2 Membrane absorbers A membrane absorber is characterized by consisting of a non porous sheet or panel placed at a certain distance from a hard backing whereby an air filled cavity is formed. h is the thickness while E and ν are the Young’s modulus and the Poisson ratio respectively. However. of the cavity: 1 ρ c2 (4.4.8) if just the thickness/distance is > λ/8. 4. 4. the resonance frequency is also determined by the plate stiffness and mode of plate vibration.3. In this case the resonance frequency can be described as follows: Eh3 + (4. L. Normally.4) 2π m L However. Such ceilings will often be placed below ventilation ducts and other technical installations. It is seen that more low frequencies are absorbed as the thickness or the wall distance increases.4 Different modes of vibration in a stiff plate. applying a thin sheet will cause the absorption to drop again at a higher frequency where the distance between sheet and hard wall equals λ/2.3. Mineral wool ceiling tiles are normally given a carefully controlled layer of special paint from the factory to make them look like normal (white) plaster ceilings as much as possible. whereby a large distance (typically between 20 cm and one metre) is ensured to the hard concrete deck behind. Equation 4. In the case of normal incidence.3.3. of which a few are illustrated in Fig. 1 fr = 2π ρc2 2 2 π 4 ⎡⎛ p ⎞ ⎛ q ⎞ ⎤ 2 106 . with p and q being integers determining the shape of the two dimensional oscillation pattern of the plate.
4. As expected it is seen that the thicker and heavier plate result in the lowest resonance frequency as expected from equations 4.g. Fig. From [1] Membrane absorbers are often found in rooms in the form of wooden floors on joists or as gypsum board or wood panel walls.3. The effect is a controlled low frequency T60 value as opposed to rooms made entirely from heavy concrete or masonry which causes the sound to be “dark” and blurred at low frequencies.5 show absorption versus frequency for two different thickness of plywood placed 45 mm from a hard backing – with and without mineral wool in the cavity. Fig.From this formula it is seen that a resonance frequency is determined completely by the stiffness if the depth of the cavity is infinitely deep – as is the case e. with a single pane window. Besides controlling low frequency reverberation. 107 .5 absorption versus frequency of membrane absorber for two different plate thicknesses and with and without mineral wool in the cavity. the panels also provide some diffusion of the sound.4 and 4. Fig. it is observed that the mineral wool inlay. Besides. 4. which increases the internal damping of the construction causes a significant improvement in the absorption around the resonance frequency and also causes the resonance frequency to become lower. 4.3.3.6 Example of membrane absorbers attached to the concrete side wall in the multi purpose hall (Kolding Teater).5.
3. An example of such a single resonator. For the resonance frequency of the panel we have: c P (4.3. the viscous damping can be significant if the hole/slit dimensions are small. as the frequency range of the absorption is normally very limited around the sharp resonance frequency. Also in this case. Regarding damping.is not confined to the physical length of the neck. it is important to adjust the damping to achieve optimal absorption. Like in the case of the membrane absorber.8 d.6 except for the opening area being replaced by the degree of perforation. P. Resonators like the build in “bottle” in the left side of Fig.6) f0 = 2π V ( l + δ ) with S being the area of the opening. is illustrated in Figure 4. and the physical proportions in this case often causes a much more useful frequency range of absorption.7. V being the enclosed volume. l the length of the neck and δ a correction to the neck length which is due to the fact that the oscillating air mass . but often the absorption can be optimised by placing a thin layer of mineral wool or glass felt (called vlies) in the cavity.often moving with very high velocity .3. we have for the end correction: δ ≈ 0.7. Resonating panels will often have a higher resonance frequency and absorb efficiently in a wider frequency range than the membrane absorbers. The resonance frequency (which can be experienced by blowing across the opening of a bottle) is given by: c S (4. From [1].7) f0 = 2π L ( l + δ ) which is almost identical with Equation 4. in the cavity provides the spring function. of the panel and the volume V being changed into the depth of the cavity L. However. 4. 108 .3.7 are not very practical.7 Single resonator (left) and resonating panel (right). then this construction can be regarded as a large number of single resonators put together. if a perforated panel is placed in front of a cavity as seen to the right in Fig. 4. called a Helmholz resonator. but some of the air outside both ends of the neck will be moving as well. the enclosed air Fig. If the holes are circular with diameter d. the mass can be oscillating air in an opening between a closed cavity and the open atmosphere.3 Resonator absorbers In stead of having a plate forming the mass of the resonating system. 4.4.3.
Perforated panels are found in the form of perforated gypsum board or steel plates (used e. The panel controls low frequency reverberation in a former power plant building made from heavy masonry converted into a concert hall (Værket. for suspended ceilings)2. echoes back to the stage placed more than 50 m away are avoided (Frihedshallen. or as panels made of wooden boards with slits between the individual boards as illustrated to the left in Figure 4.g. By making this wall absorbing. BR95) [2] contains demands on maximum T60 values in school class rooms.3.2.3. 4. whereas the Danish Working Environment Agency have issued rules for industrial buildings and offices [3]. Other possibilities are walls made from perforated tiles. The Danish Building Law (Bygningsreglementet af 1995. These current Danish rules are 2 It should be added that in many cases with perforated gypsum or steel plates used as suspended ceilings. day care institutions and apartment buildings.8 Resonating panel constructions in practice. which make use of the cavity already present in a double masonry wall as shown to the right in the same Figure. Randers). 4. 3. Right: Perforated bricks on the rear wall in a sports and multi purpose hall.4 Application of sound absorbers in room acoustic design The main purpose of introducing absorption for reverberation control in rooms is to reduce noise levels (see Fig. Left: Wooden boards separated by controlled gaps in front of a former window niche filled with mineral wool.4) and in some cases to increase intelligibility. Fig. the combinations of perforation and cavity depth causes the absorber to act more like a porous absorber but with reduced performance at high frequencies due to the panel shielding off the porous layer to some degree. Sønderborg). 109 .8.
4.4.briefly listed in Figure 4. sports arenas and restaurants .2 Examples of acoustic treatment mounted in ceiling in industrial halls. 4.1.4.4. The reason for this is that often calculation as well as measurement of T60 is often questionable in these rooms. 4. (The values listed for single person offices and corridors in office buildings just reflect common design practice. 110 . Fig. Recommendable values for other types of rooms – including auditoria and concert halls were listed in Table 3. acoustic concerns a generally included in modern design of these spaces as well. no rules exist for other public spaces like traffic terminals. Right: Vertical Mineral wool baffles.4.) As indicated in Figure 4. Fig.5.1 Listing of Danish rules regarding maximum values of reverberation time in buildings. Special standards exist for design of cinemas and studio control rooms and listening rooms. In most cases the ceiling is the most obvious surface to treat with absorption. Left: suspended ceiling of mineral wool tiles with integrated light fixtures. In Denmark.although the acoustic conditions in these places are often horrible. as it constitutes a large area which is normally available apart from a few light or ventilation fixtures and because here the often delicate absorption materials are not subject to mechanical damage.1 the rules for large industrial halls as well as open plan areas in offices and schools are specified in terms of a required minimum absorption area. However.
but also available wall areas must be used as illustrated by the mineral wool tiles to the right in Figure 4. sports halls etc. In many public places like traffic terminals. shops etc. In Figure 4. In speech the consonant sounds are often the weaker elements. Fig.4. but they contain most of the information. The vertical mineral wool baffles shown to the right can be a solution when the ceiling is already heavily occupied by technical installations.3. In high rooms.4 Schematic illustration of the influence of reverberation on the intelligibility of speech. Therefore.4.In Figure 4.. In rooms where practically all the absorption is placed in the ceiling. department stores. the reverberation time basically becomes a function of the room height as shown in Figure 4.4. the room acoustic absorption treatment is not only done with the purpose of reducing noise but also to ensure proper intelligibility of speech (often emitted through loudspeakers).4. This type of ceiling is often found in offices.3.4. Fig.2 are shown two examples of acoustic treatment of ceilings. 4. it is not always sufficient to place the absorption in the ceiling surface alone.4 is illustrated how a long room decay can cover (mask) the weak phonems illustrated schematically as vertical bars. 111 . To the left a normal suspended ceiling of mineral wool tiles with integrating lighting and ventilation.4.3 Simplified calculation of T60 in room with all absorption placed on the ceiling surface(left) shows the need for additional absorption on walls in tall rooms (right). 4. schools. a long reverberation can seriously deteriorate intelligibility..
4.5 Illustrations from the room acoustic simulation programme ODEON of a class room design with a partly absorbing (dark) and reflective (lighter grey) ceiling.4.5 illustrates such a case in which the ODEON programme was used to balance the application of absorbing and reflective part of the ceiling for a school project and to predict reverberation time and the intelligibility in terms of the Speech Transmission Index mentioned in Section 2. In these rooms also the design of the room geometry is important to ensure proper propagation of sound from the source to the listeners through reflection of the sound waves off non absorbing room surfaces.og Boligstyrelsen 1995. class rooms and theatres.ebst. And in order to support intelligibility.dk/sw5110. References [1] [2] Z.asp [3] 112 . 1994. Bygningsreglementet.1. Fig.4. november 1995. Maekawa and P.at. Fig.dk/pub_lydforhold/0/8/0 AT anvisning nr. 4. Akustik i arbejdsrum (Acoustics in work places) http://www.1.7. Thus. Bygge. these reflections must arrive not long (up to 40 ms) after the direct sound. Publications from the Danish National Agency for Enterprise and Construction (in Danish) can be found at: http://www. the room acoustic design not only consists of reverberation control by absorption treating of the room surfaces. Lord: Environmental and Architectural Acoustics. Even in normal sized class rooms this concern about supporting reflections may be applied by leaving a central part of the ceiling reflective (given that enough other surface areas can be found to provide the required reverberation control). E & FN Spon. London.In rooms dedicated for speech like auditoria.0.1.
Airborne sound transmission from source room (1) to receiving room (2) The most common case is the sound insulation between two rooms.1. and partly transmits into the receiving room. The sound power incident on the wall is.5.1) τ = P1 However. respectively.2.1.1 Definition A sound wave incident on a wall or any other surface separating two adjacent rooms partly reflects back to the source room. partly dissipates as heat within the material of the wall.2 Sound insulation between two rooms Figure 5. The power incident on the wall is P1 and the power transmitted into the receiving room is P2. The rooms are called the source room and the receiving room.1 THE SOUND TRANSMISSION LOSS 5.1.2) P2 τ Another name for the same term is the sound reduction index.6) 113 . An introduction to sound insulation Jens Holger Rindel 5. see eq.1. In the first room is a sound source that generates the average sound pressure p1. The sound transmission coefficient τ is defined as the ratio of transmitted to incident sound power P2 (5. the sound transmission coefficients are typically very small numbers. 5. (3. and it is more convenient to use the sound transmission loss R with the unit deciBel (dB). It is defined as P 1 = − 10 log τ (dB) R = 10 log 1 = 10 log (5. partly propagates to other connecting structures. With the assumption of diffuse sound fields in both rooms it is possible to derive a simple relation between the transmission loss and the sound pressure levels in the two rooms.1.1.
The results are averaged over two different source positions.3 V2 A2 = (5. see later in section 5. respectively. but the flanking constructions will also influence the result. If the sound intensity incident on the surfaces of the source room is denoted Iinc the total incident sound power on the partition is 114 .1. see eq.2) gives p2 S S R = 10 log 21 = L1 − L2 + 10 log (dB) (5. In addition to the two sound pressure levels it is also necessary to measure the reverberation time in the receiving room in order to calculate the absorption area.3) 4ρ c The area of the wall is S.5.7) (dB) R′ = L1 − L2 + 10 log A2 The apostrophe after the symbol indicates that flanking transmission can be assumed to influence the result.1.4 Multielement partitions and apertures A partition is often divided into elements with different sound insulation properties. In recent years the international standards for measurement of sound insulation have been revised and it is recommended to extend the frequency range down to 50 Hz and up to 5000 Hz. The sound pressure levels are measured as the average of a number of microphone positions or as the average from microphones slowly moving on a circular path.2. For measurements of sound insulation in buildings the apparent sound transmission loss is S (5. 5.6) c T2 Only under special laboratory conditions it is possible to measure the transmission loss of a wall without influence from other transmission paths.1. (3. More details are given in ISO 140 Part 3 and 4. (3.1.3 Measurement of sound insulation Sound insulation is measured in onethird octave bands covering the frequency range from 100 Hz to 3150 Hz.4. e. In the receiving room the average sound pressure p2 is generated from the sound power P2 radiated into the room. see eq.1. In recent years lightweight constructions have been more commonly used in new building technology. whereas heavy constructions have traditionally been used for sound insulation. This important result is the basis for transmission loss measurements.g. a wall with a door.1. Sabine’s equation is used for this.1. In a normal building the sound will not only be transmitted through the separating construction.4) p2 = ρc A2 Here A2 denotes the absorption area in the receiving room.p12 S (5.5) p2 A2 A2 Here L1 and L2 are the sound pressure levels in the source and receiving room. Each element is described by the area Si and the transmission coefficient τi .8) 4 P2 2 (5. Insertion in the definition (5.13) 55. P1 = I inc S = 5.1.2. One reason for this is that the low frequencies 50 – 100 Hz are very important for the subjective evaluation of the sound insulation properties of lightweight constructions.
5. The total sound power transmitted through the partition is P2 = ∑τ i =1 n i S i I inc Thus.1. It is seen that the relative area of the aperture defines an upper limit of the sound insulation that can be achieved.2. the transmission coefficient of the partition is P 1 n τ res = 2 = (5.P1 = ∑S i =1 n i I inc = S I inc The total area is called S. If also the area of the aperture Sap is very small compared to the total area.1 R1 + S ap ⎟ ≅ − 10 log⎜10 −0. As an approximation it can be assumed that the transmission coefficient of the aperture is 1. this leads to the following result for the resulting transmission loss of the wall with aperture: S ap ⎞ ⎛ ⎛1 ⎞ ⎟ Rres = − 10 log⎜ S1 10 −0.3 can illustrate the result.1.1.1 Ri ⎟ (5.9) ⎝ S i =1 ⎠ In the simple case on only two elements the graph in Fig.2 may be used.1.1.8) ∑τ i Si P S i =1 1 The same result can also be written in terms of the transmission losses Ri of each element ⎛1 n ⎞ Rres = − 10 log τ res = − 10 log⎜ ∑ S i 10 − 0.1. Graph for estimating the transmission loss of a multielement partition An aperture in a wall is a special example of an element with different transmission properties. 5.1 R1 + (5.10) ⎜ S ⎟ ⎝S ⎠ ⎝ ⎠ Fig. Figure 5. ( ) 115 .
1.Figure 5. Graph for estimating the transmission loss of a construction with an aperture 116 .3.
2.2. The surface of the material defines two transition planes where the sound waves change from one medium to another.2.1) (5.2) leads to: Z0 ( p1 − p 4 ) pi − p r = Zm (5.5. Thus the ratio of sound pressure to particle velocity in each of the plane propagating waves is: pi pt pr = = = Z0 = ρc ui ur ut (5. The material is characterised by the density ρm and the speed of longitudinal waves cL .2.3) p3 p1 p2 p4 = = = = Z m = ρ m cL u1 u2 u3 u4 Using (5. It is assumed that the medium on either side is air with the density ρ and the speed of sound c (also longitudinal waves).1 Sound transmission through a solid material The solid material is supposed to have the shape of a large plate with thickness h.2.2) ut = u 2 − u3 The characteristic impedance in the surrounding medium (air) is denoted Z0 and that in solid material is denoted Zm.2.2. reflected and transmitted sound waves The sound pressure is equal on either side of the two transition planes: p i + p r = p1 + p 4 pt = p 2 + p3 Also the particle velocity is equal on either side of the two transition planes: u i − u r = u1 − u 4 (5.1.2.2 SINGLE LEAF CONSTRUCTIONS 5.1.2. 5.4) Z0 ( p 2 − p3 ) pt = Zm Assuming propagation from one side of the material to the other without losses means that there is only a phase difference between the pressure at the two intersections: 117 .3) in (5. h pi p1 p2 pt pr p4 p3 Figure 5. Thick wall with incident. The symbols and notation are explained in Fig.
2. First the case of a thin wall: Zm >> Z0 and kmh << 1 2 ⎛ ⎛ Z ⎞2 ⎞ ⎛ ⎞ ⎜1 + ⎜ m ⎟ sin 2 (k h) ⎟ ≅ 10 log⎜1 + ⎛ ω ρ m h ⎞ ⎟ ⎜ ⎟ R0 ≅ 10 log (5. From the above equations (5.2. Transmission loss at normal incidence of sound on a 600 mm thick concrete wall.2.2. They occur at frequencies where the thickness is equal to half a wavelength in the solid material.2.(5.1). (5. Two special cases can be studied. or a multiple of half wavelengths.2.2. 5. At high frequencies some dips can be observed in the transmission loss curve.8) 118 .5) can be derived the ratio between the sound pressures pi and pt and thus the transmission loss can be expressed by: 2 2 ⎛ ⎞ pi ⎜ cos 2 (k h) + 1 ⎛ Z 0 + Z m ⎞ sin 2 (k h) ⎟ (5.2.5) p 4 = p3 e − j k m h Here km = ω /cL is the angular wave number for longitudinal sound propagation in the solid material.7) m ⎜ ⎜ 2 Z0 ⎟ ⎟ ⎜ ⎜ 2ρc ⎟ ⎟ ⎝ ⎠ ⎠ ⎝ ⎠ ⎝ ⎝ ⎠ The other special case is a very thick wall: Zm >> Z0 and kmh >> 1 R0 ⎛ Z ≅ 10 log⎜ m ⎜ 2Z 0 ⎝ ⎞ ⎟ ⎟ ⎠ 2 ⎛ρ c ≅ 20 log⎜ m L ⎜ 2ρc ⎝ ⎞ ⎟ ⎟ ⎠ (5.2.6) ⎜ ⎟ R0 = 10 log = 10 log m m ⎜ ⎟ 4 ⎜ Zm Z0 ⎟ pt ⎝ ⎠ ⎝ ⎠ p2 = p1 e − j km h R0. dB Fig. the dips are very narrow and they are mainly of theoretical interest. However.4) and (5.
11) Zw = = jω m vn The separation impedance will be more complicated if the bending stiffness of the wall is also taken into account.2. The application of Newton’s second law (force = mass ⋅ acceleration) gives: dv (5.3).12) which leads to 119 . For wood it is 68 dB.7) to (5.2 The mass law pi vn vt = vn / cos θ θ pr pt Figure 5.2.10) Δ p = pi + p r − pt = m n = j ω m v n dt where vn is the velocity of the wall vibrations (in the direction normal to the wall). for concrete 80 dB and for steel 94 dB. see below. see Fig. (These numbers should be reduced by 5 dB in the case of random incidence instead of normal incidence. The result for a very thick wall (5.3.The crossover frequency from (5.3.2. Thin wall with sound pressures and particle velocities A thin wall with the mass per unit area m is considered.2.8) means that there is an upper limit on the sound insulation that can be achieved by a singleleaf construction.2.8) is the frequency fh at which kmh = 1: cL fh = (5.2.2.2. and this limit depends on the density of the material. The separation impedance Zw is introduced: Δp (5. Due to the continuity requirement the normal component of the velocity on both sides of the wall is: vn = ut cos θ = (ui − u r ) cos θ (5. 5. The particle velocities in the sound waves are called u with the same indices as the corresponding sound pressures.2.2. which will de derived in a different way in the next section. see section 5.2.9) 2π h This is the frequency at which the thickness is approximately one sixth of the longitudinal wavelength λL in the material: cL λL h = = 2π f 2π The result for the thin wall is the socalled mass law. 5.
in each direction the transmitted sound power is equal to the incident sound power multiplied by the transmission coefficient.23R0 ) (dB) (5.17) This is the theoretical result for random incidence.2.18) This is in good agreement with measuring results on real walls. In a diffuse sound field the incident sound power P1 on a surface is found by integration over the solid angle ψ = 2 π assuming the same sound intensity I1 in all directions.2.15) 5.2.2.2. in real life this is not true and it can be shown that the result is related to partitions of infinite size.2. 120 .vn cos θ The sound transmission loss Rθ at a certain angle of incidence θ is: pt = pi − p r = 2 Z0 (5.2.14) 1 (5.2.2.7).11) gives the important mass law of sound insulation: ⎛π f m ⎞ ωm R0 = 10 log 1 + j ≅ 20 log⎜ ⎟ ⎜ ρ c ⎟ (dB) 2 ρc ⎠ ⎝ Since m = ρmh this result is the same as derived above in (5.2.13) Rθ p = 10 log i pt Z cos θ = 10 log 1 + w 2 Z0 2 (dB) (5.14) In the special case of normal sound incidence (θ = 0) the insertion of (5. 2 (5.2. The principle is the same as used in section 3.3 Sound insulation at random incidence The transmission coefficient at the angle of incidence θ is from (5.16) τ (θ ) = 2 ⎛ωm⎞ 2 1+ ⎜ ⎜ 2 ρ c ⎟ cos θ ⎟ ⎝ ⎠ Random incidence means that the sound field on the source side of the partition is approximately a diffuse sound field. Since. the ratio between transmitted and incident power is: τ = P2 P1 π /2 = ∫ψ = 2π τ (θ ) I1S d ψ I1 S d ψ ∫ψ = 1 ∫ π /2 0 τ (θ ) cos θ sin θ d θ cos θ sin θ d θ 2 = 2π ∫ π /2 0 τ τ = 2 ∫ τ (θ ) cos θ sin θ d θ 0 = ∫ τ (θ ) d(cos 0 2 θ) = d(cos 2 θ ) ∫ 1 + (ω m 2 ρ c )2 cos 2 θ 0 1 ⎛ 2ρ c ⎞ 2 = ⎜ ⎜ ω m ⎟ ln 1 + (ω m 2 ρ c ) ⎟ ⎝ ⎠ ( ) R = − 10 log τ = R0 − 10 log(0.2. Taking the finite size into account the result is approximately: R ≅ R0 − 5 dB (5. However. and for typical values (R0 between 30 and 60 dB) it means that R is 8 to 11 dB lower than R0.
4 The critical frequency The bending stiffness per unit length of a plate with thickness h is: E h3 (5. The critical frequency is: c2 m (5. i. Thin wall with bending wave and indication of speed of propagation along the wall The coincidence leads to a significant dip in the sound transmission loss. The coincidence dip will be at a frequency higher than or equal to the critical frequency: 121 .2. If the bending wave speed happens to be equal to the phase speed of the incident sound wave.2.2.19) B = 12 (1 − ν 2 ) where E is Young’s modulus of the material and ν is Poisson’s ratio.5.20) m fc Here fc is introduced as the critical frequency. It is defined as the frequency at which the speed of bending waves equals the speed of sound in air. 5.2. see Fig. The speed of propagation of bending waves in a plate with bending stiffness per unit width B and mass per unit area m is (see section 6.3.3 for most rigid materials).4.2. cb = c. the phase speed is in general higher than c.e.21) fc = 2π B A sound wave with the angle of incidence θ propagates across the wall with the phase speed c / sin θ . (ν ≅ 0. this is called coincidence: cb = c / sin θ Figure 5.4.3): f B ω4 cb = = c (5.2.
In the frequency range below the critical frequency.2. f < fc: 2 R ≅ R0 + 20 log 1 − ( f f c ) − 5 dB (5.2.24) 5.2.15).11) is replaced by: ⎛ ⎛ f ⎞2 ⎞ Z w = jω m ⎜1 − ⎜ ⎟ sin 4 θ ⎟ (5. f ≥ fc: 2η f R ≅ R0 + 10 log (dB) π fc where η is the loss factor (see section 6.2.2.5 A general model of sound insulation of single constructions The general model of sound insulation is based on mass law as given in (5.27) ⎜ 2ρc ⎟ ⎝ ⎠ A sketch of the transmission loss as a function of frequency is shown in Fig.2.2.2.3). Sound insulation of a singleleaf construction. according to (5.5. However.2.8): ⎛ρ c ⎞ R ≤ 20 log⎜ m L ⎟ − 5 dB (5. 5.5 R. the following results are valid for sound insulation between rooms with approximately diffuse sound fields.26) The upper limit for sound insulation of a singleleaf construction is. dB fc Frequency (log) Figure 5. 122 .2.25) In the frequency range above the critical frequency.(5.14) leads to the sound transmission loss at a certain angle of incidence: 2 Rθ = R0 + 20 log cos θ + 20 log 1 − ( f f c ) sin 4 θ (dB) (5.2. (5.2.23) ⎜ ⎜ fc ⎟ ⎟ ⎝ ⎠ ⎝ ⎠ Insertion in the general equation (5. fc is the critical frequency and the upper limit is the dotted line.22) f co = f c sin 2 θ The separation impedance (5.2.2.2.
3.5.3.2.3.1. The separation impedance of the two plates is denoted Z1 and Z2. A double construction with indication of sound pressures and particle velocities A double construction with two plates in the distance d is considered. (5.10) the movement of each wall is: pi + p r − ( p1 − p 4 ) = Z 1v1 (5. 5.2) and (5.3. see Fig.1) p 2 + p3 − pt = Z 2 v 2 The velocity of each wall equals the particle velocity on either side: 1 ( pi − p r ) v1 = ui − u r = Z0 v1 v2 v2 = u1 − u 4 = u 2 − u3 = ut = = 1 ( p1 − p4 ) Z0 1 ( p 2 − p3 ) = Z0 (5.3) p4 = p3 e − j kd From the above equations (5.1). respectively. 5.1 Sound transmission through a double construction m1 pi p1 d p2 m2 pt pr p4 p3 v1 v2 Fig.3.2) 1 pt Z0 Assuming propagation from one side of the cavity to the other without losses means that there is only a phase difference between the pressure at the two intersections: − jk d p2 = p1 e (5.3. As for the single construction in (5.1.3 DOUBLE LEAF CONSTRUCTIONS 5.3) can be derived the ratio between the sound pressures pi and pt and thus the transmission loss can be expressed by: 123 .3.3.3.
4) (5. f0 = c 2π However.e. In the frequency range below the resonance frequency.3.3.9) found for the sound transmission through a solid material.R0 p = 10 log i pt 2 (5.3. f < f0: ⎛ ω (m1 + m 2 ) ⎞ ⎟ = R(1+ 2 ) R0 ≈ 20 log ⎜ (5.3.3.10) ⎜ ⎟ 2ρ c ⎝ ⎠ ρ⎛ 1 124 .3.7) and they are approximately kd = n π. i.3.3. The springlike behaviour of the air cavity changes from that of a simple spring below the crossover frequency to that of a transmission channel at higher frequencies.2 The massairmass resonance frequency The transmission loss is minimum when the last term is zero. m1 + m2 ρ c tg( kd ) = m1m2 ω For a cavity that is narrow compared to the wave length (kd << 1) we get: m1 + m2 ρ c ωd tg( kd ) ≈ kd = = c m1m2 ω The solution is the massairmass resonance frequency f0 = ω 0 / 2 π ⎛ Z + Z2 ⎞ ⎛ Z + Z 2 Z1 Z 2 ⎞ ⎟ cos(kd ) + j ⎜1 + 1 ⎟ sin(kd ) = 10 log ⎜1 + 1 + ⎜ ⎟ ⎜ 2Z 0 ⎠ 2Z 0 2 Z 02 ⎟ ⎝ ⎝ ⎠ If only the mass of each wall is taken into account the separation impedances are: Z1 = jω m1 2 (5. 5.8) ⎜m + m ⎟ ⎟ d⎝ 1 2 ⎠ If the depth d of the cavity is comparable to the wavelength there are many solutions to (5.5) yields: 2 2 ⎡⎛ ω (m + m ) ⎞ ⎤ ⎞ ⎛ ω (m1 + m2 ) ω 2 m1m2 1 2 R0 ≅ 10 log ⎢⎜ sin(kd ) ⎟ + ⎜ cos(kd ) − sin(kd ) ⎟ ⎥ (5. The dips in the sound insulation occur at frequencies at which the cavity depth equals one or more half wavelenghts: d = n λ /2.6) ⎜ ⎟ ⎜ ⎟ 2ρ c 2ρ c 2( ρ c) 2 ⎢⎝ ⎠ ⎝ ⎠ ⎥ ⎣ ⎦ This result will be discussed and simplified below.3 A general model of sound insulation of double constructions The result (5.3.3. Only. more important than these dips is the shift from low.to highfrequency behaviour of the air cavity. 5. in this case the transmission is through air.9) fd = 2π d This is quite similar to the result (5.3. but it is the frequency fd at which kd = 1: c (5.2.3.6) can be simplified in different way depending on the frequency range.5) Z 2 = j ω m2 Neglecting the smaller parts and inserting Z0 = ρ c together with (5. The crossover frequency has no particular physical meaning.7) 1 ⎞ ⎜ (5.
sin (kd) is replaced by its maximum value 1.11) ⎜ 2 ρ 2 c 3 ⎟ ≈ R1 + R2 + 20 log(2kd ) ⎠ ⎝ In this a much better sound insulation can be obtained.2. 125 .3. fc1 and fc2. dB f0 Frequency (log) fd fc1 fc2 Figure 5. A sketch of the transmission loss as a function of frequency is shown in Fig. 5. R.12) ⎜ 2 ( ρ c) 2 ⎟ ≈ R1 + R2 + 6 dB ⎠ ⎝ In this highfrequency range. and for f ≥ fd: ⎛ ω 2 m1 m 2 ⎞ ⎟ R0 ≈ 20 log ⎜ (5. f0 is the resonance frequency and fd is the crossover frequency of the cavity. respectively.This means that the construction behaves as a single construction with the mass per unit area (m1 + m2). R. and it depends on the product of the three parameters m1. f0 < f < fd: ⎛ ω 3 m1 m 2 d ⎞ ⎟ R0 ≈ 20 log ⎜ (5. Sound insulation of a doubleleaf construction.3. At frequencies above fd where the cavity is wide compared to the wavelength. dB f0 Frequency (log) fd Figure 5.3. In the frequency range above the resonance frequency.3. d is no longer an important parameter.2. Sound insulation of an asymmetric doubleleaf construction with two thin plates having different critical frequencies. m2 and d.3.3.
i = 10 log 1 (5.4.4. namely three possible paths for each of the four surrounding flanking constructions.1) R′ = 10 log = L1 − L2 + 10 log (dB) P2 + P3 A2 where P2 is the sound power transmitted through the partition wall to the receiver room and P3 is the sound power radiated to the receiver room from the flanking surfaces and other flanking paths: (5. and the apparent transmission loss is calculated from: ⎛ − 0 .1R ⎞ R′ = − 10 log⎜10 −0.i It is convenient to keep the incident sound power P1 on the partition wall as a reference for all the flanking transmission losses.1R + ∑ 10 F . When all relevant transmission paths are considered the sound insulation is described by the apparent sound transmission loss: P1 S (5. 5.4 FLANKING TRANSMISSION Fig. RF.3) PF . 5.1. 126 . see Fig.i i Each single flanking transmission path i can be characterised by the flanking transmission loss. the ceiling or the façade. The transmission of sound from a source room to a receiver room can be via flanking constructions like the floor.2) P3 = ∑ PF .4.1. In this way it is very simple to add all the contributions together.i : P (dB) RF .4. Direct transmission and three flanking transmission paths via the floor.4.i ⎟ (dB) (5.4.5.4) i ⎝ ⎠ In the typical case of horizontal transmission through a wall the will be 12 flanking paths.
or any other room in the building – the calibrated sound pressure level L2 is measured. the result is not bad as a rough estimate for the design of an enclosure.6 IMPACT SOUND INSULATION The noise generated from footsteps on floors is characterised by the impact noise level. The main data for the tapping machine are: • • • The noise is generated by steel hammers with a fall height of 40 mm Each steel hammer has a mass of 500 g The number of taps per second is 10.1) pencl = ρc αS The sound power incident on the inner surface of the enclosure is (still with the assumption of a diffuse sound field): 2 pencl S Pinc = (5.3) Pout = τ Pinc = Pa α The insertion loss of the enclosure is the difference in radiated sound power level without and with the enclosure: P α (5.e. The noise source is totally covered by an enclosure with surface area S. In the source room the tapping machine is placed on the floor in a number of positions.5.2) 4ρ c The sound power transmitted through the enclosure is then: τ (5. 5. especially in the case of lightweight floor constructions. It is measured according to ISO 140 Part 6 and 7 by a standardised tapping machine.5. i. and the enclosure is made from a plate with transmission loss R or transmission coefficient τ. The impact sound pressure level is the sound pressure level in dB re 20 μPa that would be measured if the absorption area is A0 = 10 m2: A (5. In the room below .4) that both transmission loss and absorption coefficient are important for an efficient reduction of noise by an enclosure.1) Ln = L2 + 10 log 2 (dB) A0 = 10 m 2 A0 The frequency range is the same as for airborne sound insulation. if a diffuse sound field is assumed: 4 Pa 2 (5. However. Especially the assumption of a diffuse sound field inside the enclosure is doubtful.5.5. absorption coefficient α on the inside.6. It is clearly seen from (5. 127 . the 16 onethird octave bands from 100 Hz to 3150 Hz. However.5 ENCLOSURES A noise source is supposed to radiate the sound power Pa.5. it is recommended to extend the frequency range down to 50 Hz.4) ΔL = 10 log a = 10 log = R + 10 log α (dB) Pout τ This result cannot be considered to be very accurate. The average sound pressure in the enclosure pencl can be estimated.5. The reverberation time in the receiving room must also be measured in order to calculate the absorption area A2.
128 .1.Fig.6. The reference curve is made from three straight lines with a slope of 9 dB per octave from 100 to 400 Hz. and 0 dB per octave from 1250 to 3150 Hz.7 SINGLENUMBER RATING OF SOUND INSULATION 5. for quick comparison of the sound insulation obtained with different constructions. and to specify requirements for sound insulation.7. 5.1 The weighted sound reduction index The singlenumber rating of sound insulation is practical for several purposes: • • • to characterise the measuring result of a building construction. Principle of measuring the impact sound pressure level from a floor to a receiving room (2) 5. 3 dB per octave from 400 to 1250 Hz. The weighted sound reduction index Rw is based on a standardised reference curve that is defined in onethird octaves in the frequency range 100 Hz – 3150 Hz.
The measured transmission loss is compared to the reference curve, and the sum of unfavourable deviations is calculated. An unfavourable deviation is the deviation between the reference curve and the measured curve if the measured sound insulation is lower than the value of the reference curve. The reference curve is shifted up or down in steps of 1 dB, and the correct position of the reference curve is found when the sum of unfavourable deviations is as large as possible, but do not exceed 32 dB. The value of the reference curve at 500 Hz is taken as the singlenumber value of the measuring result. The method is also shown in Fig. 5.7.1.
Frequency, Hz Fig. 5.7.1. Determination of the weighted sound reduction index. M is the measured curve, V1 is the reference curve in position 52 dB, and V2 is the shifted reference curve. The result is Rw = 60 dB.
5.7.2 The weighted impact sound pressure level The weighted impact sound pressure level Ln,w is very similar to the weighted sound reduction index. It is based on a standardised reference curve that is defined in onethird octaves in the frequency range 100 Hz – 3150 Hz. The reference curve is made from three straight lines with a slope of 0 dB per octave from 100 to 315 Hz, 3 dB per octave from 315 to 1000 Hz, and 9 dB per octave from 1000 to 3150 Hz.
The measured impact sound pressure level is compared to the reference curve, and the sum of unfavourable deviations is calculated. An unfavourable deviation is the deviation between the
Transmisson Loss, R, dB
129
reference curve and the measured curve if the measured impact sound pressure level is higher than the value of the reference curve. The reference curve is shifted up or down in steps of 1 dB, and the correct position of the reference curve is found when the sum of unfavourable deviations is as large as possible, but do not exceed 32 dB. The value of the reference curve at 500 Hz is taken as the singlenumber value of the measuring result. The method is also shown in Fig. 5.7.2.
Impact Sound Pressure Level, Ln,w, dB
Frequency, Hz Fig. 5.7.2. Determination of the weighted impact sound pressure level. M is the measured curve, V1 is the reference curve in position 60 dB, and V2 is the shifted reference curve. The result is Ln,w = 47 dB.
130
5.8 REQUIREMENTS FOR SOUND INSULATION
The Danish requirements for new buildings are laid down in “Bygningsreglement 1995” (BR95) and in “Bygningsreglement for småhuse 1998” (BRS 98). For dwellings in multistorey houses and for hotels the main requirements are: • • • The airborne sound insulation shall be R´w ≥ 52 dB in horizontal directions and R´w ≥ 53 dB in vertical directions. The impact sound pressure level shall be L´n,w ≤ 58 dB. Between rooms for common service or commercial use and dwellings the airborne sound insulation shall be R´w ≥ 60 dB and the impact sound pressure level shall be L´n,w ≤ 48 dB.
For rowhouses or semidetached houses the main requirements are: • • The airborne sound insulation shall be R´w ≥ 55 dB. The impact sound pressure level shall be L´n,w ≤ 53 dB.
In schools the main requirements are: • • • Between classrooms the airborne sound insulation shall be R´w ≥ 48 dB in horizontal directions and R´w ≥ 51 dB in vertical directions. The impact sound pressure level in classrooms shall be L´n,w ≤ 63 dB. From rooms for music or workshops to classrooms the airborne sound insulation shall be R´w ≥ 60 dB and the impact sound pressure level shall be L´n,w ≤ 53 dB.
The sound insulation of facades is not specified directly, but in buildings where then outdoor traffic noise exceeds LAeq, 24 ≥ 55 dB, the indoor noise in living rooms shall not exceed LAeq, 24 ≤ 30 dB.
5.9 REFERENCES
ISO 1403 (1995): Acoustics. Measurement of sound insulation in buildings and of building elements. Part 3: Laboratory measurements of airborne sound insulation of building elements. ISO 1404 (1998): Acoustics. Measurement of sound insulation in buildings and of building elements. Part 4: Field measurements of airborne sound insulation between rooms. ISO 1406 (1998): Acoustics. Measurement of sound insulation in buildings and of building elements. Part 6: Laboratory measurements of impact sound insulation of floors. ISO 1407 (1998): Acoustics. Measurement of sound insulation in buildings and of building elements. Part 7: Field measurements of impact sound insulation of floors.
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Rating of sound insulation in buildings and of building elements. 132 . BR95. ISO 7172 (1996): Acoustics. Bygningsreglement for småhuse (Building regulations for small houses. Bygningsreglement (Building regulations. Part 2: Impact sound insulation. Copenhagen. Copenhagen. Byggeog Boligstyrelsen. (1995). Byggeog Boligstyrelsen. Part 1: Airborne sound insulation.ISO 7171 (1996): Acoustics. in Danish). Rating of sound insulation in buildings and of building elements. in Danish). BRS 98. (1998).
Examples are musical sound from a string instrument or noise from a pump in a central heating system. which can occur in two planes. This is usually derived from a force balance of the mass element. in the case of a loudspeaker three degrees of freedom are required for describing the designed translational motion of the ‘piston cone’ and its unintentional rocking motions. referred to as phenomena of structureborne sound or vibroacoustics. as is the case 133 . If the response of the resonator primarily occurs in only one direction.6 MECHANICAL VIBRATION AND STRUCTUREBORNE SOUND Mogens Ohlrich 6. secondorder differential equations. This subject. Vibration of more complex systems requires more than one motion coordinate for a complete description. which can exist in the absence of external excitation. In general such motions will be governed by three coupled. Figure 6. and their interaction. For example. by using a special set of coordinates these equations can be uncoupled and solved independently. ie in a single motion coordinate. Vibration of simple resonant systems (resonators) is characterised by mass and stiffness properties and by some form of damping mechanism. After ref.1. The mathematical description of the vibration of such systems is governed by an ordinary secondorder differential equation. which dissipate vibrational energy.1 Examples of single degree of freedom resonators.1. [1]. The simplest description of dynamic behaviour applies to resonators that can be modelled as a (minimal) combination of discrete or ‘lumped’ elements. then the system is said to have a single degree of freedom (sdof). Figure 6.1 shows examples of sdofresonators. and eventually radiated into the fluid as audible sound. Solution of the equation shows that such systems have a single preferred ‘natural’ frequency of vibration. is important because sound or noise is very often generated directly by mechanical vibration of solid bodies or by waves transmitted in solid structures. However.1 INTRODUCTION Audio frequency vibration of mechanical systems and waves in solid structures form an integral part of engineering acoustics in describing the dynamic phenomena in solids and fluids.
The response of such a system is governed by a partial differential equation. Transient sources representing local impact are very common both as a single impact and in repetition. impacts in production machinery (punch presses. ie. If this is the case it is natural to threat the system as a continuous one. eg in wheel/surface contacts. door slamming). Figure 6. can occur when the wavelength of vibration in a solid structure is less than one of its typical dimensions. 6.1.1.2 Examples of sources that generate vibration and structureborne sound. because the response depends upon both time and a spatial position coordinate that specifies the location at which the response is to be determined. that is.1 SOURCES OF VIBRATION There are many types of excitation mechanisms that generate vibration and waves in solid structures. impulsive sources of vibration and noise in buildings (footfalls. piston slab). The local measurement quantity is either a motion (displacement. structural wave motion. Figure 6. in which case the excitation timehistory becomes periodic. Both types of variables are vectors. Such sources are associated with nature or they involve the employment of machines in the broadest sense. The accelerometer is based upon the piezoelectric principle with an output signal proportional to the acceleration a = a(t) of the vibrating surface.2 MEASUREMENT QUANTITIES Investigations of vibration in solid structures are usually carried out by measuring a local quantity at a specific position on the structure.2. eg caused by a sound field. ranging from a miniature loudspeaker in a hearing aid to a combustion engine of a truck. velocity or acceleration) or a force. and thus assigned to a certain orientation or direction. Other sources of vibration and noise are random variation of surface roughness. 134 . The hammer impact symbolizes a variety of excitation mechanisms such as musical percussion (drums. Accelerometers are available with different sensitivities. Vibratory motion is usually measured unidirectional with a small transducer of the accelerometertype that is fastened to the structure’s surface. The velocity v or displacement ξ of the vibratory motion is obtained by integration of the acceleration signal. Examples of sources of vibration are shown in Figure 6. forge hammers) and periodic impacts in combustion engines (valves. transient and continuous that includes time variation of either deterministic (periodic) or random nature.1. or distributed excitation of a structure. The sources can be classified by their temporal variations for which there are two types. such excitation is often of a harmonic (pure tone) nature. devices that do work. Vibration of different phase. Distribution of vibration over a larger area can be determined by measurements in a number of discrete positions. 6. xylophones).2b illustrates force excitation caused by an unbalanced rotating mass.for the sdofresonator.1. say.1.
1. which is often the case when vibration or waves have small amplitudes. In the discrete model the properties of system components are described by discrete (‘lumped’) quantities. The measurement is carried out by inserting the transducer between a source (eg a vibration exciter) and the measurement object.1. massless springs and dampers. System dynamics can therefore be described by linear differential equations. velocity v = v(t) 135 . one can use the very important superposition principle. These can be based either on a discrete model or on a continuous model. When this is the case.4a shows the basic lumped elements. the impedances or the mobilities. m is the mass and r is the damping constant of a viscous damper. where the physical properties are represented by ideal discrete elements of point masses. the quantity s represents the spring constant (stiffness). The properties of the elements are independent of time t. Figure 6.1. [kg] and [kg/s]. structural shape and forms of excitation. but is often based on intuition and practical experience.3. and there is a linear relation between forces Fi = Fi(t) and changes in. Symbolically. whereas only a brief summary will be given of wave motion in continuous structures (structureborne sound).3 LINEAR MECHANICAL SYSTEMS The dynamic properties of a physical system depend upon its mass and stiffness distribution and damping losses. massless springs and dampers. This means that the response contributions from independent excitations can be superimposed or summed as vectors. The physical properties of the continuous model are functions of the spatial coordinates. for example. represented by ideal masses.3 Lumped model of a physical system.1.A localised (point) force F = F(t) is mostly measured with a piezoelectric force transducer. see Figure 6. which produces an output proportional to the force. Herein we assume that systems considered are linear. the actual decision of the type of model is usually not strictly scientific. displacement ξ = ξ(t) . These properties are attempted described by mathematical models in the form of one or more differential equations of motion. the viscous damping is thought caused by motion of a piston in a fluidfilled cylinder. However. This arrangement is mostly used for measuring the dynamic properties of structures. respectively. 6. The viscous damper represents a velocity proportional resistance that results in energy losses. Figure 6. The choice between the two models depends upon a number of factors such as frequency range of interest. Dynamic properties of the system are therefore described by partial differential equations. The system is said to be linear if the dependent response variables are of first order. for translatory motion these quantities have units of [N/m]. In this note we shall focus mainly on the analysis of discrete models. respectively.
1.1a shows a model of a single degree of freedom system that is connected to a rigid foundation. In Figure 6. Note that both motion and force variables are vector quantities.2 SIMPLE MECHANICAL RESONATORS Figure 6. which is Newton’s second law of motion in its simplest form.1.and acceleration a = a(t) over the terminals of the elements. The system consists of a mass m . The viscous damping force is proportional to the velocity of the ‘deformation’ in the massless damper. (b) diagram which shows the forces acting on the mass m . and a velocity proportional viscous damper of damping constant r.(b) Excitation (action) and reaction by compression of spring. Figure 6. Figure 6. the motion variables are thus defined as positive in the xdirection. 6. 136 .4a.4b . a spring of spring constant s . Thus.4 (a) Forceresponserelations for ideal lumped elements.1. for the ideal spring there is proportionality between force and deformation according to Hooke’s law.1 (a) Viscously damped simple resonator driven by an external force F . Both quantities are defined as positive in the direction of the vector.2. as shown by the example in Figure 6. the positive force F required for accelerating the mass m is therefore F = ma .2.
a force which is also directed opposite to that of the motion of the mass and in proportion to its vibration velocity v = dξ /dt . ie d 2ξ Fi = ma = m 2 .6. F + Fs + Fr = F − sξ − rv . defined as s (6. (6.b) 2m 2mω 0 2 ms Moreover. in the horizontal plane in this example. So.2.1b it is seen that the total force Ff acting on the rigid foundation is equal to the sum of the spring force and the damping force.2.2.1 EQUATION OF MOTION FOR SIMPLE RESONATOR The system is assumed excited by a timevarying external force F = F(t) and it is understood that the system can vibrate only translatory. thus serves to accelerate the mass.2) dt that is. dξ Ff = r + sξ .2. If viscous damping is assumed as illustrated by the parallelcoupled dashpot in Figure 6. the reaction on the mass that is caused by the spring force.3) ∑ dt The equation of motion for the system therefore becomes d 2ξ dξ m 2 +r + sξ = F . this sum must be equal to the product of mass m and acceleration a = d2ξ/dt2 . The vector sum of forces that act on the mass. that is. Their definitions are respectively r r r δ = = and ζ = . (6. to and fro.2. (6.2. elastic spring force Fs = − sξ . according to Newton’s second law of motion.4a) dt dt This equation is often written in a reduced form as F d 2ξ r dξ 2 + + ω0ξ = . that is.4b) 2 m dt m dt where ω0 is the natural angular frequency in [rad/s] of the corresponding undamped system (r = 0). that is. The vibration response caused by the external force is uniquely defined by the instantaneous value ξ . and this is taken positive towards the righthand side.7) dt ω0 = 137 .4b) is often replaced either by 2δ or by 2ζω0 where δ is the damping coefficient and ζ is the nondimensional viscous damping ratio. (6.2.1) Thus. (6. The motion of the mass from its equilibrium position is denoted by the displacement ξ = ξ(t) .2. (6. This displacement of the mass results in a compression of the spring that produces a restoring.6a. acts in the opposite direction of the displacement imposed by the external force.2. from Figure 6. in the direction of the force. (6.1 then this element will exert a corresponding restoring damping force dξ Fr = − r .5) . m In the literature the fraction r/m in eq.2.2.2. (6.
2. (6.2.6.2.2.2.10b) follows from eq. the quantity ξstat represents the socalled static displacement.2 Time history of vibration builtup in the case of harmonic force excitation of a simple.2. damped resonator when ω < ω0 . since it is assumed that the initial builtup of vibration caused by ‘starting’ the force has completely decayed because of damping effects. (6. (6.2. we shall disregard the damping of the considered system by setting r = 0 . The vibration builtup response is succeeded by a stationary vibration at the angular frequency ω of the excitation.2.4) varies harmonically with time as F = F1cosωt with angular frequency ω.b.2. which describes the forced harmonic motion of the ξ = ξ1 cos ωt + solutions to the homogeneous equation ( ) 138 .2. After a certain builtup of vibration the mass will then also execute stationary.8) by substituting the assumed solution ξ = ξ1 cosωt : F 2 − ω 2 + ω 0 ξ1 cos ωt = 1 cos ωt . see Figure 6. (6. 6.4) reduces to F d 2ξ 2 + ω 0 ξ = 1 cos ωt .2 FORCED HARMONIC RESPONSE OF SIMPLE RESONATOR Let us assume that the excitation force F in eq.10 a.2. harmonic vibration with the same angular frequency ω.11) s The stationary part of the solution (6.2. (6. m which gives F F 1 1 1 ξ1 = 1 = 1 = ξ stat (6.2. (6. which is the compression or extension of the spring caused by the force F = F1cosωt when ω = 0: F ξ stat = 1 . Thus for harmonic excitation the equation of motion (6.1 Undamped system Initially.9). Herein we shall only deal with the stationary vibration of the system. Figure 6.2.2.8) 2 m dt The complete solution for ξ = ξ(t) of such a differential equation has the wellknown form (6.5).9) where the first term represents the stationary harmonic vibration and the second term represents the abovementioned phenomenon of vibration builtup or decay. The displacement amplitude ξ1 of the stationary vibration is obtained directly from eq. Furthermore.c) 2 2 2 2 2 m ω0 − ω s 1− ω ω0 1− ω 2 ω0 where eq.2.
13) F f = F f cos ωt = F1 cos ωt .3b the same quantity is shown as absolute value (modulus) and phase. This means that the excitation force is in equilibrium with the spring force.7) for r= 0.2. (b) the same response function plotted as modulus and phase. this simply means that the quantities ξ1 and F1 are inphase at low frequencies. the mass has a negligible influence. (6.3. which follows from eq. this excitation condition is called resonant excitation. for ω < ω0 where the system behaves springlike. that is.12) 1 (6. For excitation frequencies below the natural frequency of the system. and the frequency at which ω = ω0 is the resonance frequency. (6. Thus. Figure 6. 2 1− ω 2 ω0 The fraction 1/(1 − ω2/ω02) represents the variation of the vibration amplitude with respect to the excitation frequency ω and it is sometimes referred to as the response amplification factor. From the figure it can be seen that the vibration amplitude grows towards infinity when the excitation frequency ω approaches the undamped natural frequency ω0 of the system.2. is thus given by (6. 2 1− ω 2 ω0 This force ratio Ff /F1 has the same frequency variation as the motion ratio ξ1/ξstat shown in Figure 6. The disturbance force on the foundation thus follows directly by substituting the solution eq.3 shows the variation of this quantity ξ1/ξstat with angular frequency. ξ = ξ1 cos ωt = ξ stat 1 Figure 6. For this undamped case the force Ff that is transmitted to the foundation is caused by the spring force and is given by Ff = sξ . At ω = ω0 . which corresponds to a phase change of π radians. if the force on the foundation is to be reduced by vibration isolation it is required that natural frequency of the system is designed in such a way that ω0 << ω/√2 is fulfilled. which is transmitted unchanged to the foundation.2.2. For a set 139 .3 (a) Relative displacement response ξ1/ξstat for an undamped simple resonator.12) cos ωt . Physically. the response ξ1 is also seen to undergo a change in sign.resonator.2. whereas they are in antiphase for ω > ω0 where the response is lagging the harmonic force excitation by 180 degrees because of the system mass (inertia). that is for ω < ω0 .2. in Figure 6.2. this quantity also reveals the phase relation between the displacement response and excitation force.2.
2. and it is therefore necessary to take the real part of the mathematical solution when we want the time variation of the physical motion. harmonic vibration1 : m + iωr + s ξ1e iωt = F1e iωt F1 F1 ξ1 = = .14) By performing in eq.16a.2. (6. 2 m(ω 0 − ω 2 ) + iωr and Ff 2 = F1 ( s 2 + ω 2 r 2 ) 2 m 2 (ω 0 − ω 2 ) 2 + ω 2 r 2 2 . The solution of the equation of motion is assumed to be of the same form ξ(t) = Re{ξ1eiωt}.2.16c).16b) gives F1 ( s + iωr ) Ff = . velocity. 2 2 m(ω 0 − ω 2 ) + iωr ( s − ω m) + iωr 2 (− ω ) (6. However. (6. where ξ1 = ξ1eiφ is the complex amplitude of the harmonic displacement with φ being the phase angle between the displacement response and the driving force. 1 140 .2. we get ξ1 2 = F1 2 2 m 2 (ω 0 − ω 2 ) 2 + ω 2 r 2 . the problem is basically solved.2. when dealing with energy or power quantities.2.4a) substitutions of F(t) ≡ F1eiωt and ξ(t) ≡ ξ1eiωt result in the solution for the stationary. By using complex notation the harmonic excitation force F(t) = F1cosωt can be expressed as F(t) = Re{F1eiωt}.excitation frequency and system mass this is accomplished by selecting a ‘soft’ spring element with an appropriately small spring constant s . ξ1 is obtained by simply taking the squareroot of the expression (6.b) ) Here the symbol Re{··} is left out.1 then eq.2. (6.17) which by substituting eq. (6. (6. This does not result in any trouble as long as one is strictly dealing with field quantities (displacement. When damping losses are assumed to be of the viscous type as in Figure 6.) Furthermore.14).2.2.7) F f e iωt = (iωr + s )ξ1e iωt . since the squared modulus is given by ξ1ξ1* = ξ12 . one must only include the real part of the field quantity. (6.2.2. b) ⇔ Hereby. (6. The time variation eiωt is also often left out in the analyses.2. where F1 is the complex amplitude of the force.2. The force transmitted to the foundation follows similarly from eq. Physical quantities are of course always real. but it is of course to be recalled and taken into account when necessary. (6.15) (6. (If the time variation of the response is sought then this is obtained by substitution in eq.2.16c) Thus. 6.2. This yields ξ (t ) = Re{ξ1e iωt } = ξ1 cos(ωt + ϕ ) .2 Viscously damped system The influence of damping is now being considered.4) applies. force etc). (6.18a.
16b) to a real quantity this yields 2 m(ω 0 − ω 2 ) F1 (−ωr ) F1 ξ1 = 2 2 + i 2 2 (6.2.19b) . (b) Modulus and phase. Shown is the real and imaginary parts of the displacement response of the viscously damped resonator when this is driven by a harmonic force of constant amplitude F1 . this means that the force amplitude is assumed to be real. The squared modulus of the displacement is already given by eq. ie F1 = F1.19) is often written in the alternative ‘product form’ ξ1 = ξ1 e iϕ (6.2. (6. The solution (6.2. (6.2.2.2.16) for the complex displacement can also be written in terms of its real and imaginary parts ξ1 = ξ re + iξ im . Thus. The solution for the complex displacement response eq.4 Frequency variation of displacement ξ1 for a viscously damped simple resonator driven by a harmonic force of constant amplitude.2. Solution in product form. (a) Real and imaginary parts.20a) where the modulus ξ1 and phase angle φ as usual are determined from eq.16) or (6. Figure 6. (6.19b).19): ξ1 2 2 2 = ξ re + ξ im and tan ϕ = ξ im ξ re .2. whereas the phase angle is found directly from eq. (6.4a.2.2. The damping is seen to limit the displacement response in the frequency range around ω~ω0 where the response ξ1 is controlled largely by its imaginary part ξim . (6. ie − ωr tan φ = .19a) In the following we shall assume that the arbitrary phase of F1 is set equal to zero by a suitable choice of timereference (t = 0). by transforming the denominator in eq. (6. (6.2.2.2.Solution in sum form.16c).20b) 2 m(ω 0 − ω 2 ) 141 . 2 2 2 2 m (ω 0 − ω ) + ω r m (ω 0 − ω 2 ) 2 + ω 2 r 2 The frequency variations of this solution are sketched in Figure 6.
23).Ω2 (6. (6.23a. 142 . (6.22) ( ) { } By substituting this as well as the dimensionless viscous damping ratio ζ into eqs.2. where v1 = iωξ1 .20): v(t ) = dξ (t ) = − ω ξ1 sin(ωt + ϕ ) . which is close to the natural frequency of the system.2.2. For simple resonators the frequency ratio Ω is readily used as frequency parameter Ω = ω /ω0 .Note that the phase angle becomes φ = −π/2 at resonant excitation. The vibration velocity v(t) of the resonator is often of interest and this follows simply by taking the time derivative of the displacement response.4b shows how the modulus and phase of the displacement varies with frequency for harmonic force excitation.16) or (6.2.2. Amplitude and phase characteristics for this ratio between transmitted force and driving force are shown in Figure 6. (6.b) here. Nondimensional form. dt (6. 1 .5b. are shown logarithmically in Figure 6.2.20b) we obtain the general expressions for the displacement ratio ξ1/ξstat and for the phase angle φ : ξ1 1 = 2 2 2 ξ stat (1 − Ω ) + 4Ω 2ζ 2 2 and tan ϕ = − 2Ωζ .16c) and (6.2. with respect to the complex amplitudes a differentiation is simply archived by a multiplication with iω . ⎭ ⎩ dt So. it is recalled that the static displacement is ξstat = F1/s .18).20c) Figure 6. the actual physical time variation of the vibration response follows from eq.2.2.14) ξ (t ) = Re{ξ1e iωt } = Re{ ξ1 e iϕ e iωt } = ξ1 cos(ωt + ϕ ) ⇔ ξ (t ) = F1 2 m 2 (ω 0 − ω 2 ) 2 + ω 2 r 2 cos(ωt + ϕ ) . (6. eq. As previously.2. evidently integration is performed by a division by iω.2. Similar expressions for the force ratio Ff / F1 are obtained by substituting the nondimensional parameters in eq. Amplitude and phase characteristics for the displacement ratio (6. (6. This type of graph is the most commonly used form of presentation for frequency response functions.21) or ⎫ ⎧d v(t ) = Re⎨ ξ1e iωt ⎬ = Re v1e iωt .5a for different values of damping ratio ζ . It is often convenient to introduce nondimensional parameters that enable solutions for a class of systems to be presented in a general form.2.2.2. (6. the acceleration a(t) of the motion is obtained similarly by the time derivative of velocity or by the second derivative of displacement. It is clearly seen that the damping has a dominant influence on the response in the frequency range Ω ~ 1. Moreover.
From ref. However.2. that is.2. (6. The actual value of Ωr is determined by differentiating eq.2. when 2ζ 2 << 1 .05) the resonance frequency will nearly coincide with the natural frequency ω0 of the undamped system. say.25) Figure 6.24a) Ωr ≅ 1 − ζ 2 .23a) with respect to Ω and by setting the obtained expression equal to zero. This gives the value Ω ≡ Ωr = ⇔ 1 − 2ζ 2 (6.23a) we get ξ max 2 ξ stat 2 = 1 . (6. The maximum displacement thus occurs at an angular frequency.24a) in (6. and (b) Force ratio Ff /F1 . when the damping is small (ζ << 0.2.2. Ωr = ωr /ω0 where ωr is the resonance frequency. the maximum displacement thus becomes 143 .2.2.In forced harmonic vibration the displacement response of the system reaches its maximum value ξmax at. [2].5 Amplitude and phase characteristics for: (a) Displacement ratio ξ1/ξstat . 4ζ (1 − ζ 2 ) 2 (6. ωr ≅ ω0 .24b) in the last approximate expression use have been made of the truncated series: (1−x)½ ≅ 1 − x/2 provided that x << 1. By substituting eq. (6. which is slightly lower than the angular natural frequency of the undamped system.
2. Characteristic properties.16c). the combined system) are therefore characterised as being: • • • Stiffness controlled for Ω << 1 . Similarly. where ξ1 ≈ Damping controlled at Ω ≅ 1 . ω0r 2ζ (6. the vibration velocity of the system can be shown to take its maximum value vmax at ω = ω0 . Since v = ω ξ this yields F 1 ξ max ≅ ω 0ξ stat = 1 .28) ωr F1 mω 2 These asymptotic values for the displacement response ξ1 follow directly from eq. (6. Similar relations can be determined for velocity and acceleration. 144 .ξ max ≅ ξ stat F 1 = 1 .27) r 2ζ Relations for maximum acceleration can be derived in the same manner. The dynamic properties of the resonator (ie. (6. the modulus and phase of the frequency response functions for displacement and velocity.2. Finally. Figure 6.6 actually represent the dynamic properties of the individual elements s. at Ω =1 .26) The displacement at resonance is thus equal to ξstat divided by 2ζ . (6. A unit force excitation is assumed.2. are sketched in loglog format in Figure 6. m and r under the action of the force F1.6. As apparent from previous discussions the dynamic properties of the resonator are predominantly springlike at low frequencies (Ω << 1) and predominantly masslike at high frequencies (Ω >> 1) .2. where ξ1 ≈ F1 s F1 . the asymptotes shown in Figure 6.2. . . that is.2. respectively.6 Logarithmic plots of the frequency response functions of a simple resonator represented as displacement and velocity.2. where ξ1 ≈ Mass controlled for Ω >> 1 .
2. ie req = req (ω ) = sη / ω and ζ eq = ζ eq (ω ) = ηω 0 /(2ω ) .30) with (6. (6.2. It is therefore customary to assign specific names and symbols to the various types of frequency response functions. (6. 6. this is defined as ξ e iωt (6.15) shows that sη corresponds to ωr . By using the loss factor the equation of motion for a single massspring resonator becomes d 2ξ m 2 + s (1 + iη )ξ = F1 e iωt .3 Structurally damped systems So far we have only considered damping of the viscous type. like the displacement of a spring. A second type is structural damping.31a. For harmonic motion this can be represented by a complex stiffness s = s(1 + iη) where η is the damping loss factor and s is the real part of the complex spring constant. the loss factor of a parallel combination of an ideal spring and a viscous damper of constant r may be expressed as η = rω/s . because the equation of motion can be formulated initially without regard to damping and finally the spring constant is replaced by its complex value s = s(1 + iη) .2.3. 6. (6.15). F1e 145 .2.2. This broad characterisation by the term ‘frequency response’ is often imprecise because the response quantity can be either displacement or one of its time derivatives: velocity and acceleration. ξ1 = = (6. similar to eq.29) dt which.3 FREQUENCY RESPONSE FUNCTIONS The frequency response of a system is defined as the ratio of complex amplitudes of two quantities representing the response to a certain excitation. where ξ1 = ξ1eiφ is the complex amplitude: F1 F1 .2. has the solution ξ(t) = Re{ξ1eiωt}.2. The equivalent damping ‘constant’ req for a structurally damped spring thus becomes frequency dependent. When the system response is characterised by its displacement the complex frequency response is called the receptance H(ω) . Now.31b) becomes ζeq = η/2 at resonance.1 Receptance So far we have been dealing with ratios of response over force.b) Alternatively. and so does the equivalent damping ratio ζeq . This relation may be used as an approximation for other frequencies that are close to resonance. which is proportional to changes in elastic deformation. (6. comparing eq.2.2.32) H (ω ) = H (ω ) e iϕ (ω ) = 1 iωt . The loss factor thus defines the phase lag (hysteresis) between harmonic driving force and spring displacement. Such structural damping is therefore appropriately modelled by assigning the inherent losses to the spring element.2. (6.30) 2 2 ( s − ω m ) + i sη m(ω 0 − ω 2 ) + isη This ‘complex stiffness’ approach is very convenient. So.6.2.2. Note also that the equivalent damping ratio eq.
36) Y (ω ) = Y (ω ) e iθ (ω ) = 1 iωt .2 Mobility and Impedance The velocity response is often of interest in vibroacoustics.2.2.2. (6. its amplitude spectrum H(ω) and phase spectrum φ(ω) can be determined from Im{H (ω )} 2 . (6.c) .32) states that ξ1eiωt = H(ω)F1eiωt .39) Y (ω ) 146 .b. which means that the time variation of the displacement for harmonic excitation is ξ (t ) = Re{H (ω ) F1 e iωt } = H (ω ) F1 cos(ωt + ϕ (ω ) ) .2. 6. for instance. Thus iω 1 1 Ys (ω ) = . implies that the quantity is a continuous function of ω .35). Z (ω ) = (6.34) where the force amplitude is assumed to be real.where the notation with angular frequency dependence.2. this is called dynamic stiffness [3]. of course. (6.33) H (ω ) = H (ω ) H ∗ (ω ) and tan ϕ (ω ) = Re{H (ω )} The definition eq. It is sometimes useful to use the reciprocal of the receptance function. a very simple relation between mobility and receptance since the complex velocity amplitude is v1 = iωξ1 . damper r and mass m .35a. Receptances of the discrete elements: spring s.3. F1e There is. ie Y (ω ) = iωH (ω ) . Yr (ω ) = and Ym (ω ) = .2. (6.2.2. follow respectively from the fundamental relations between harmonic force and the associated motion for such elements −1 1 1 H s (ω ) = and H m (ω ) = 2 . (6. free end.38) s r iωm The reciprocal of a mobility function is named the impedance Z(ω) 1 .2. The complex ratio between response velocity and driving force is called the mobility Y(ω) (or sometimes admittance) and is defined as v e iωt (6. (6. because the radiated sound power from a vibrating structure is proportional to its surface velocity.2. H r (ω ) = s ω m iωr Since the ideal spring and damper are massless it is assumed in the definition of their receptances that one of their terminals is blocked and that a harmonic force drives the other.37) The mobilities of the ideal components are therefore easily determined either from the fundamental relations or directly from eq. H(ω). (6.
in machinery and in certain cases also in buildings.7 Motion excitation of a damped simple resonator.2.2. is it not necessary to use q explicitly.1 Definition of frequency response functions R/F and F/R . in this case by the motion differences .2. where F is the force and R is the response that represents either displacement. . In all these examples and in vibration isolation of delicate equipment from disturbing environments. because the motion of the system is uniquely described by a socalled generalized coordinate q = q(t) .2. for example.7. in transportation of any kind. Table 6.1 together with corresponding functions that involve acceleration response. Response Name of frequency response function quantity R R/F F/R Displacement ξ Receptance H(ω) Dynamic stiffness S(ω) Velocity v Mobility Y(ω) Impedance Z(ω) Acceleration a Acceleration A(ω) Apparent mass M(ω)  6. Figure 6. Thus we want to find the imposed/generated motion ξ = ξ(t) of the mass. the apparent mass.2. but despite of this the system has only one degree of freedom. .4 FORCED VIBRATION CAUSED BY MOTION EXCITATION Vibratory disturbances like motion excitation is very common and occurs. The quantities q and q describe. There are two motion coordinates. The latter is called accelerance and its reciprocal.2. the compression (or elongation) of the spring and the velocity difference over the damper.These different frequency response functions are summarized in Table 6. From eq. .2.2. (6. respectively. The accelerance is sometimes used because acceleration is the response quantity that is usually measured directly. velocity or acceleration.7 readily can be written down. Since the total force on the mass in Figure 6.40) q = ξ f −ξ and q = ξ f − ξ .3) follows directly 147 . (6. the ‘foundation’ has a given motion ξf = ξf (t) as shown in Figure 6.
(6.42) simply is interpreted as a special ‘forcing function’.2. .2. A treatment of free vibration of such systems and an analysis of more complicated multidegree of freedom systems is outside the scope of this introductory note on discrete systems.. m ξ + r ξ + sξ = r ξ f + sξ f .4a). (6. shells.43) ξ1 = ξ f . This finishes the analysis of simple sdof mechanical resonators.42) can be assumed to be ξ ≡ ξ1eiωt . rods.2.3 VIBRATION AND WAVES IN CONTINUOUS SYSTEMS Distributed solid structures become ‘dynamically elastic’ and exhibit wavetype vibratory behaviour as the frequency is increased to an extent.41) (6. a brief introduction will be given to vibration and wave motion in continuous systems. It is seen that there is a clear analogy between this expression and eq. In the audible frequency range this is the case for basic engineering components. The most important wave types in structures are considered to be (a) longitudinal waves. such as edges. 6. although wave conversion between different types generally occurs at most structural discontinuities.2. Only systems of one and two dimensions will be considered here. if the righthandside of eq. The frequency variation of ξ1 /ξf is therefore exactly identical to that of Ff /F1 shown in Figure 6. (b) shear or torsional waves and (c) bending waves.2. 148 . or less than.18a). the solution to eq. . plates. (6. Although discrete models can be used for analysing wave motion at the lower frequencies. Usually each wave type is treated separately. . Thus. membranes.. which are also called flexural waves. because most engineering structures have at least one dimension. it becomes expedient to use wavetype analysis in problems where the wavelength is short.2. In the following an introduction of these waves in plane structures will be given. corners and cross sectional changes. which is small in comparison with the relevant structural wavelength of vibration.42) This gives the equation of motion .1. see Figure 6.6]. where the wavelength become comparable to.∑F i . beams. . = s (ξ f − ξ ) + r (ξ f − ξ ) = m ξ . (6. (6.1).2. pipes etc.5b. Equations of motion that describe different wave types and vibroacoustic phenomena have been formulated for many types of continuous structures [4. In motion excitation the ratio between displacements is thus identical to the ratio between forces in the case of force excitation (Figure 6. 2 m(ω 0 − ω 2 ) + iωr This expression has the same form as eq. the physical dimensions of the structure.2. such as strings. We will therefore proceed with a brief introduction of continuous structures. by substituting these quantities we obtain the solution for the complex amplitude of the displacement ξ = Re {ξ1eiωt } s + iωr (6.2.3.5. In the case of steadystate harmonic motion excitation ξf eiωt .
After ref. For longitudinal waves the operator is given by −ES d2/dx2 . (6. If we assume purely harmonic excitation and harmonic wave motion u = u(x)eiωt this reads L{ u ( x)} − ω 2 m′u ( x) = F ′( x) . the wave speed cl2 of a freely propagation longitudinal wave in the beam is 149 . although there is also a small lateral deformation normal to the structural surface.1 Different wave types: (a) Longitudinal wave (the lateral deformations are exaggerated). This is given by ∂u F = − ES .3. 6. t).3. However. m' is its mass per unit length and F '(x) is an external force excitation per unit length.1 LONGITUDINAL WAVES Longitudinal waves in onedimensional structures like rods and beams are compressiontype waves that are similar to plane sound waves in a fluid. the longitudinal displacement in the wave motion will be denoted by u = u(x. Two field variables are required for describing the longitudinal wave motion. these are the already mentioned displacement u = u(x)eiωt – or its timederivative.1) where L{····} is a differential operator that describes the force gradient in the beam. [7].Figure 6. where E in [N/m2] is Young’s modulus of elasticity of the beam material and S is the cross sectional area of the beam. (6.2) ∂x Moreover. It should also be mentioned that the impedance of longitudinal waves in solids generally is very high. where x represents its spatial dependence. this deformation is generally so small that it can be neglected as a radiator of sound to the surrounding fluid.3.3. The equation of motion for longitudinal waves in an undamped beam can be written in a compact form. (b) Torsional wave and (c) Bending wave. The local structural deformation in connection with longitudinal wave motion is primarily in the direction of wave propagation. the velocity v = iωu(x)eiωt = v(x)eiωt – and the internal force F = F(x)eiωt associated with the wave motion.
but only 27 times higher than the impedance in water.3.1. Table 6. and for rubberlike materials ν ≈ 0. 6. a magnitude higher than for sound in air.3) where ρ is the material mass density.3. Shear waves are of importance in plates that are builtup of several layers of material with different properties.1 Material properties and wave speeds (phase speeds) for solid structures.5 . eg sandwich honeycomb panels. For common solid material ν ≈ 0. 150 .4) where ν is Poisson’s ratio. the mass density for metals is seen to be up to 7000 times higher than for air.cl 2 = E ρ . for example. Moreover. homogenous plate is slightly higher (by about 5%): cl1 = E . This means that the characteristic impedance (ρcl) for compression waves in solid structures is much higher than for air.2 SHEAR WAVES In this wave type only shear deformations occur.3. A listing of material properties and wave speeds are given in Table 6. The corresponding wave speed in a flat. Furthermore. Note that the wave speed in metals is about 3000 to 5000 m/s. index 2 on cl2 indicates that the structure has two surfaces that are small compared with the wavelength of the motion. [8]. that is.3. the direction of the ‘particle’ motion is perpendicular to the direction of propagation. but no volume changes. (6. which is a material constant that expresses the ratio between deformations in the lateral and lengthwise directions of the structure. ρ ( 1 −ν 2 ) (6. the characteristic impedance for steel is 105 times higher than in air. After ref.3 .3.
These waves may therefore be important for the wave transmission over large distances (eg in buildings and ships) and in wave conversion to bending waves. Bending waves do therefore play a dominant role in sound radiation from structures. and four field variables are thus required for describing the bending wave motion.3. (6.7) where the differential operator L{···} that describes the shear force gradient in the beam now takes the form B d4/dx4 . From the righthandside of this equation it is clear that there is a unique relation between Young’s modulus E and the shear modulus G. being air or water.3 BENDING WAVES Bending waves in beams and plates are characterised by the motion being perpendicular to both the direction of propagation.6) . The equation of motion for bending waves in an undamped beam can be written in the previous compact form. 2(1 + ν ) Shear waves in rods are called torsional waves. the wave speed cb of a freely propagation bending wave in the beam is 151 . we get L{w( x) } − ω 2 m′ w( x) = F ′( x) .3. If the rod has a circular cross section then the wave speed is as given by eq.1c. which is the first spatial derivative of w . and the surface of the structure. the internal shear force Fy = Fy(x)eiωt and the internal bending moment Mz = Mz(x)eiωt . otherwise the wave speed will be lower. If we again assume purely harmonic excitation and harmonic wave motion w=w(x)eiωt .5) ρ ρ 2(1 + ν ) where G is the shear modulus of the material.3. but with the transverse displacement of the bending wave motion being denoted by w=w(x.3. the transverse displacement w = w(x)eiωt and the angular displacement β = β(x)eiωt . which is the dominant wave type when it comes to sound radiation to the surrounding fluid media.3.3. and that bending waves are easily generated. Two force variables are associated with the wave motion. (6. the details shall not be given here.8) and ∂x ∂x Moreover. B is the bending stiffness of the beam. t). The wave speed cs for shear waves in a plate is given by G E cs = = .The equation of motion for shear waves is governed by a second order partial differential equation [5] of a general form similar to that of longitudinal waves. This type of wave motion that involves twisting of the cross section of the rod was shown in Figure 6. ie E G = (6. ie dw /dx .5).1b. these are given by ∂3w ∂2w Fy = B 3 Mz = − B 2 . The reasons for this are that the wave motion has a good ‘match’ to the adjacent air. There are two motion variables. (6. see Figure 6. (6.3. because of their low characteristic impedance. though. Here. The operator is of fourth order. The two wave types discussed so far have high characteristic impedances.3. 6. m' is its mass per unit length and F'(x) is an external force excitation per unit length.
6. Such dependence results in complicated sound radiation properties for plates and builtup structures. eq.9) cb = ω ⎜ ⎟ . or by eq.4 INPUT MOBILITY OF INFINITE SYSTEMS Finally.3. The corresponding point impedances are the reciprocal of the given point mobilities. is given by 1 2 1 cb ≅ 1.4.10) where f is the frequency (in Hz) and cl2 is given by eq.3.8 cl 2 h f . (6.10). In the case of a semiinfinite (s∞) beam driven axially at the end.3. 6. (6. (6. this special phenomenon is called dispersion. The input mobility of a semiinfinite plate driven normal to its surface and at the end (edge) is 152 .1 Beam or rod Longitudinal vibration.8 cl1 h f . m′ cb (6. in this brief introduction it is appropriate to list some input mobilities for point force excitation. (6.3).3.13) where m' is mass per unit length and cb is given by eq.3). (6.3. The phase speed of bending waves in a thin homogeneous beam with a rectangular crosssection and of thickness h in the direction of the motion.3. Bending vibration.4). the phase speed in a thin homogeneous plate of thickness h is given by cb ≅ 1.4.13).3.4 ⎛B⎞ (6.14) 4m′ cb Note that this is four times lower than the input mobility of the semiinfinite beam.9). The input mobility of a semiinfinite beam driven at the end is 1− i Ys ∞ = . (6.11) where cl1 is given by eq.3. input mobilities relating translational velocity v eiωt to translational force F eiωt . ⎝ m′ ⎠ which is seen to depend upon frequency. The input mobility of an infinite beam driven in the ‘middle’ is given by 1− i Y∞ = . 6. (6. (6. both at the same point and in the same coordinate (direction).3.3.3. Or more specifically. Moreover. the input mobility is 1 Ys ∞ = . (6.2 Plate Bending vibration. provided that the beam is of rectangular crosssection and is vibrating in the direction in which the beam thickness h is measured.3.3.3.3.12) m′ cl 2 where m' is mass per unit length and cl2 is given by eq. The wave speed or phase speed is furthermore noticed to depend upon the bending stiffness and the mass per unit length. (6.
This is generally accomplished by ‘disconnecting’ the transmission paths between the two systems. and for a homogeneous plate of thickness h the bending stiffness B' is E h3 .1a. than both the source structure and the receiving structure.3. From ref. The isolation principle is depicted in Figure 4. are given in ref. in the case of a harmonically driven simple source of mass m resting on a spring s attached to 153 . Thus. provided that the plate is undamped as is assumed here. metal springs or combinations thereof. (b) Mobilities of isolator.15).17) .1 (6.4.4 VIBRATION ISOLATION AND POWER TRANSMISSION Vibration isolation is one of the most effective ways of reducing the transmission of audio frequency vibration from a disturbing source (machine. apparatus.1b shows an example of measured mobilities of a rubber isolator. Ys ∞ = 6.3. dynamically soft). is purely real. eq. [9]. [5]. etc) to a connected receiver structure.1 (a) Vibration isolated diesel engine on elastic ship foundation. Y∞ = 8 B′ m′′ Other point mobilities relating angular velocity to moment excitation.16) B′ = 12(1 − v 2 ) It is noted that this input mobility. engine and ship foundation. (6.15) .5 B′ m′′ where m'' is the mass per unit area. In practice vibration isolation is done by inserting resilient mechanical connections or rubber elements that are much more compliant (ie. engine source and elastic receiver. The principle of vibration isolation has already been described in Chapter 6. Such vibration isolators have springlike properties and are often made of vulcanised rubber elements. and Figure 4. as well as cross mobilities.3. The input mobility of an infinite plate driven in the ‘middle’ is also real and is given by: 1 (6.3. Figure 6. (6.2. 3.
4.4.1) . length (height) d and made from a material of Young’s modulus E .3) 2 Note that the dynamic stiffness of rubberlike material generally differs from this value of static spring constant or stiffness s. ξ0 where g (= 9.004 m < ξ0 < 0. These two relations enable the determination of the static stiffness of the spring. eg see ref. ⇔ f0 = 0 ≅ [Hz] . This can be accomplished by reducing s.4. It was mentioned previously that the vibration isolation can be improved by reducing ω0. which corresponds to 8 > f0 > 5 Hz.2 Static deflection of spring. which in the unloaded condition has the length d . but this results in a more laterally unstable arrangement. that the simple oscillator model is a coarse simplification of the reality. As a compromise for a number of practical source cases it is therefore often ‘common’ to choose values in the approximate range of 0. (6. that is to say.4.81m/s ) is the gravitational acceleration.1 ESTIMATION OF SPRING STIFFNESS AND NATURAL FREQUENCY It is often easy2 to determine the important quantities (m. then the static spring constant s can be calculated from s = ES/d . ie mg s = (6.an idealised rigid foundation. by increasing the static deflection ξ0 .4.01 m. 2 154 . where an extended rigid body on springs will have six degrees of freedom and thus six natural frequencies. The designed natural frequency of the system can therefore be determined by a very simple formula: ω s g 0. Figure 6. This will be treated in more details in Section 6. s and ω 0 = s / m ) for uncritical arrangements of simple machinery sources that are mounted on vibration isolators (springs). it was found that vibration isolation is achieved when the angular natural frequency ω0 of the system is somewhat lower than the frequency component ω of the excitation force.4. [2]. 6. (6.2) ξ0 m 2π ξ0 If the spring element is slender and rodlike with a cross sectional area S . Usually the mass m of the machine is known. For a vertically loaded spring the static force F0 = mg from the mass results in a static deflection (compression) of the spring of magnitude ξ0 = F0/s . however.4. ) It should be recalled.5 ω0 = = [rad/s] .
because of the force reaction (−F) on the source. For reference we shall initially address the situation where the vibration source is rigidly connected to the receiving structure.5) vR = v free − YS F . Since per definition vR = YRF. and it is assumed that source and receiver are connected via a single motion coordinate (or terminal).4. In the analysis that follows we assume harmonic vibration vfree ≡ vfree eiωt .4. see Figure 6. (b) Source coupled rigidly to receiving structure. Figure 6.3a . we find directly for the rigid coupled system: F = (YS + YR ) v free −1 and 155 vR = YR (YS + YR ) v free . which is characterised by the input mobility YR . −1 (6.4.4. (6.b) .4) where v 2 is the timeaverage meansquare value of the free velocity vfree= vfree(t). This loading of the source causes the free velocity to change to vR . (c) Reaction forces on systems. It is reasonable to expect that the dynamic properties of the source and receiver will effect the vibration isolation that is achievable in practice. ie (6. is useful when comparing different vibratory sources.6. The source is now being connected to a receiving structure. These source quantities are suitably combined into a single descriptor [10] called the terminal source strength Jterm : J term = v2 free YS . with units of power [W].4.4. This free source strength Jterm.2 TRANSMISSION OF POWER IN RIGIDLY COUPLED SYSTEMS In contrast to the idealised model of a simple source on a rigid foundation we shall now examine the more realistic case of source and foundation or receiving structure of finite mobilities or impedances. First consider the source in a free uncoupled state in which the vibration activity of the source can be characterised by its free terminal velocity vfree and its ability to transmit power by its terminal mobility YS .3 Systems with a single coupling coordinate: (a) Free vibration source.6a.
7) By substituting the expressions from eq. 6. (6. (6. 2 (6.4. (6.The force and velocity at the coupling point have hereby been determined for this case of rigid coupling. because the spring is assumed massless. [10. 156 . Thus. eq. 11].3 VIBRATION ISOLATED SOURCE The effect of a vibration isolator is now considered. The power that is transmitted to the receiving structure is given by the wellknown relations: P = 1 2 ∗ Re FvR { }= 1 2 F Re{ YR } = 2 1 2 vR Re{ Z R }.4 (a) Block diagram of vibration isolation of a source with a single coupling coordinate.4. (6.4.4.4).4. + YR YS + 2 cos θ where CP = YS YR YS + YR 2 YS YR (6. Introducing the terminal source strength Jterm  .4.4a.9) cos ϕ R = cos ϕ R .4. (b) Diagram that shows the forces on the system elements.10) and φR is the phase angle of the receiver mobility and θ = φR − φS is the phase difference between receiver and source mobilities. The source is connected to the receiver via a vibration isolator as schematically shown in Figure 6. For further details see ref. For simplicity it is assumed that the isolator can be modelled as an ideal spring with a spring constant s . This takes values in the interval: 0 ≤ θ ≤ π . this implies that the force on the lefthandside of the spring Figure 6.4.8) For further evaluation of the transmitted power this can be written in a convenient alternative form. The power coupling factor is noted to be symmetric with respect to the logarithm of the mobility ratio YR /YS  .6) herein yields P = 1 2 v free 2 Re{ YR } YS + YR 2 = 1 2 vR 2 Re{ YR } YR 2 . and a power coupling factor CP yields P = J term C P .4.
18) Such a large value of inequality is not easily accomplished over the broad audible frequency range. of course. eq. (6. F1 = FR . gives the transmitted power to the receiver 2 Re{ YR } .7b).4.16) These results for the vibrationisolated source have to be compared with those for the rigid coupled case in order to realistically evaluate the influence of the vibration isolator.4. this gives ′ P′ = J term C P .4.13b) thus give Eiso = vR v′ R 2 2 iω / s = 1+ YS + YR 2 .4.11a. also called its insertion loss. (6. which also limit the effectiveness.13a. In the case of a symmetric vibration isolator. (6. together with eq.17) with the actual mobility of the isolator YI .4. iω ⎠ ⎝ iω which. (6.4. Eqs. because lightly damped resonance in elastic source and receiving structures will occur and limit the effectiveness of the isolator.b) and v1 = v free − YS F1 v′ = YR F1 . (6.4. where ′ CP = YS YR iω s + YS + YR 2 (6. at high frequencies the mass of the isolator can no longer be ignored and resonance occur in the isolator itself.12) and v′ = YR (iω s + YS + YR ) v free . This is defined as the ratio between the squared magnitudes of the receiver velocities before and after the installation of the vibration isolator or for that matter .6b) and (6.4. such modal behaviour can be accounted for in a prediction by replacing iω/s in eq.15) cos ϕ R . much more mobile or compliant) than the sum of the source and receiver mobilities. and similar to before given by (6.b) R −1 By substituting F1 into the general relation. Furthermore. [12].17) From this equation it is evident that a high effectiveness (ie. (6.14) P′ = 12 v free 2 iω s + YS + YR So.4.as the ratio of the corresponding injected powers.11) give F1 = (iω s + YS + YR ) v free −1 (6.4. The velocities are different. This influence is most suitably described by the effectiveness Eiso = Eiso(ω) of the vibration isolator. (6.4. large number) requires that the isolator mobility iω/s ≡ YI is much higher (ie. 157 .4. YI = iω/s >> YS + YR . R The force and the velocities are related according to Hooke’s law as v′ ⎞ ⎛v F1 = s ⎜ 1 − R ⎟ . see also ref. (6. that is.is identical to the force on the receiver. (6.4.
the definition of the isolator effectiveness in eq. eq. which is resiliently mounted on an elastic foundation. being the mass m . By comparison it is seen that eq. (6.4.(6.1a).18). Also shown are two course estimations based upon.4. where the dash refers to the case with the source resiliently connected to the receiver. so that 1 + η2 ≈ 1. Hence.5 Effectiveness of vibration isolation 10 log Eiso of a multicoupled machinery source on an elastic receiving structure. It is here assumed that the system is structurally damped and that this is accounted for by taken the spring stiffness to be complex s = s(1+iη).4. similar to the systems in Figure 6.4.4. whether this is moving or not. note that these isolators have a much higher mobility than the isolator example shown in Figure 6.1 or 6.2.4.5b. Figure 6.4. respectively.4.2. This follows from the fact that velocities and forces are related via the receiver mobility.5 shows an example of measured and predicted values of the isolation effectiveness for a complicated vibration source (the diesel engine in Figure 6.1b. Example 6.6a) and (6.2.4.4.4. The undamped natural frequency of the resonator is ω 0 = s / m .15) gives Eiso = m 1 − ω s (1 + iη ) 2 2 ≅ ⎛ω ⎞ ⎛ω ⎞ 1 − ⎜ ⎟ + iη ⎜ ⎟ ⎜ω ⎟ ⎜ω ⎟ ⎝ 0⎠ ⎝ 0⎠ 2 2 2 . (6. The effectiveness is seen to be rather good. (6.At first. and a simple monocoupled model. because Eiso might as well be defined as the ratios of forces acting on the receiver.19) is equal to the reciprocal of the results for Ff2 / F12 in Figure 6. has the mobility YS = (iωm)–1 and the mobility of the receiver in the form of a rigid foundation is YR = 0 . which is connected to a rigid foundation. This can also be deduced from eq. where m is its mass and s is the spring stiffness. The source.19) in the last approximation it is assumed that η << 1. where measured isolator mobility and average point mobilities of source and receiver have Figure 6.4.17).13a).1 The isolation effectiveness Eiso is to be determined for a harmonically driven massspring resonator.19)). for the general elastic receiver the effectiveness also reads Eiso = FR2 / FR' 2 .2. we obtain exactly eq.4. if the damping constant r is replaced by the equivalent constant req for a structurally damped spring req = sη/ω . So. (6. a simple massspringmass model (LFprediction of resemblance to eq. (6.4. by substituting the derived expressions for the corresponding forces. Substituting these into eq. (6.17) does not seem to apply to the ideal case of a rigid (immoveable) foundation that was assumed in Chapter 6. However. 158 . respectively. this is not so. about 25 dB on average.2. (6. The source is mounted on ten multidirectional isolators.
4. Despite of the coarse simplifications in these models. When such isolators are used it is therefore necessary to insert the dynamic stiffness value sdyn instead of s in the equation for the natural frequency. or more specifically their ratio Edyn /E . eg eq. the appropriate dynamic stiffness sdyn becomes sdyn = Edyn S/d .4. 6. Figure 6. Here. for a slender rubber isolator (of static stiffness given by eq.1 presents a coarse guide that shows approximate.17). (6. empirical values for the relation between rubber hardness.been used in eq.4. Thus. a 105 m tall building structure is mounted on large. An important parameter is the rubber hardness.4. (6. (6.3)).4. 6.4.4.20) .1 that the dynamic stiffness of rubberlike material generally differs from the static spring stiffness s determined by static measurement. At the fundamental natural frequency of the system vibration amplification is observed and in the frequency range above 90 Hz the effectiveness is seen to become very small at certain frequencies. which is usually characterised in °Shore A of hardness. static Young’s modulus E and dynamic Young’ modulus Edyn .4. on which the rubber pads and building structure rest.6. their static deformation is simply too small – in other words – the stiffness of the isolators is too high. The typical hardnessrange of commercial rubber isolators is from about 40°Shore A (for soft isolators) to 80°Shore A .4.6 that these thermal expansion devises are not very useful as vibration isolators. These correspond to the natural frequencies of the foundation columns (≈ ‘source’).1 Rubberlike materials The dynamic stiffness of rubber isolators depends upon a number parameters. a reasonable agreement with measurement is found in the frequency range up to 800 Hz.2a).6 Isolation effectiveness of rubber expansion devises that support a tall building.4.4. Another example of predicted isolation effectiveness is shown in Figure 6. It is apparent from Figure 6. Calculations were carried out in order to estimate their isolation effectiveness against structureborne sound transmission from disturbing underground rail traffic. 159 (6. flat rubber pads that allow thermal expansion or contraction of the huge building. Table 6. which is rather hard.4.4 DESIGN CONSIDERATIONS FOR RESILIENT ELEMENTS It was mentioned in Section 6.
ie on eq.4. The results apply to natural rubber. that this effect of course is accounted for when estimations are based on a static loaddeflection test.4 60 4.4. Furthermore. corrected dynamic stiffness.4.4.0 2.2 50 2.4.5 1. Figure 6. static Young’s modulus E and dynamic Young’s modulus Edyn. Rubber hardness Static Young’s modulus E Ratio: Edyn /E °Shore A 106 N/m2 40 1. This can be characterised by an area ratio (or shape factor) RS =Sconst /Sfree . 6. Generally.1 Approximate values for the relation between rubber hardness. (6. because the stiffness of rubber isolators also depends upon another important parameter.2  The stiffness expressed by eq.1).7 shows the stiffness correction factor Cs to be used for a given area ratio RS .5 1. (6.0 1. Thus. and Sfree is the total free surface area of the isolator. which is basically the compactness of the isolator. Figure 6. sdyn is to be multiplied with Cs to give the actual.18) therefore has to be corrected for the ‘bulkiness’ of the rubber isolator.) Table 6.7 Stiffness correction Cs to be used as a function of the area ratio RS of the vibration isolator. 160 . [13]. After ref. in which the area Sconstr represents the total constrained or loaded area of the isolator. the static and dynamic elastic properties for most engineering materials are found to be practical identical.4.8 70 6.4. (Note. For a given elastic material this means that its Young’s modulus E ≅ Edyn and its shear modulus G ≅ Gdyn .2 Metal and other elastic solids As oppose to the rubberlike materials. the stiffness of a short rubber block is found to be much higher than the stiffness of a long slender sample. this is generally not the final estimate.However.
However.4. The most common resilient element of metal is probably the helical spring.21) 8 n D3 where G is the shear modulus of the material. Other types of resilient elements are leaf springs. s = (6. Thereby the spring stiffness of the resilient element will increase by a factor of four. to become s = E S h2/L3 . where it supports the source to be isolated. The static and low frequency stiffness in the axial direction of the spring is G d4 .4.since ν ≈ 0. d is the diameter of the coil and n is the number of coils or windings.3 for most solid materials. and basically they are designed to achieve a specified small stiffness at low frequencies. However. 161 . Usually they are extended. we have E ≈ 3G . which is often made of harden steel. This clearly illustrates the importance of the boundary conditions at mounting positions. s= (6. at mid and high frequencies such a resilient element can support different wave types. This will diminish the isolator effectiveness. usually the source will be bolted to the beam and this will hinder angular motion at its ‘free’ end. continuous components with distributed mass and stiffness.22) 4 L3 in which E is Young’s modulus and L is the length of the beam. For a beam with constant thickness h and constant rectangular crosssection S the spring stiffness is E S h2 . D is the average diameter of the spring. which may be thin metal beams or plates. and resonances will occur in the resilient element because it is of finite size. unless damping and/or rubber elements are incorporated into the final design of the resilient element. Resilient elements of metal may take many different forms. One example is a socalled cantilever beam. which is rigidly builtin at the receiverend and is completely free at the other end.
E. 5 REFERENCES 1. pp. M. London 1985. Heckl and F. W. Vibration of one. M. Proceedings of Nordic Acoustical Meeting. Berlin 1988. Berlin 1996. Jacobsen: Isolation of structural vibration from machinery. Crede: Shock and vibration handbook. Ohlrich: Vibrational source strength as a prerequisite for response prediction by SEA. 4. McGrawHill. VDI 2062 Blatt 2 Vibration isolation: Resilient elements (In German). D. Mondot and B. 5. Mead: Passive vibration control. Cremer und M. Chapter 64 in ‘Encyclopedia of acoustics’ (ed.and twodimensional continuous systems by M. pp. Chapter 20 in ‘Modern methods in analytical acoustics: Lecture Notes’ (ed. Heckl and E. A.. Ohlrich and F. 11. on CDROM. 2nd ed. M. Crocker). 13. 507518. 2nd ed. Heckl. E. 7. M. London Ltd. Harris and C. D. Stockholm. 2000. J. NOVEM 2000. Dietrich: Highfrequency vibration isolation. 1987.12. M. Journal of Sound and Vibration 114. 224241. J. Ungar: Structureborne sound. C. Heckl. 1982. Fahy: Sound and structural vibration. Ungar and C. 1966. 12. New York 1976. Springer Verlag. Conf. 9. F. J. John Wiley & Sons. Crighton. Skudrzyk: Simple and complex vibratory systems. Journal of Sound and Vibration 4(2). Springer Verlag. Ney York 1997. E. M. Proceedings of Intern. 1976. 8. E. M. L. 10. E.J. P. E. 309312. 1994. NAM82. SpringerVerlag. G. Ffowcs Williams. Leppington). L. Cremer . pp. G. John Wiley & Sons. Chichester 1999. on Noise & Vibration Predesign and Characterisation using Energy Methods. Petersson: Characterisation of structureborne sound sources: The source descriptor and the coupling function. 3. J. Lyon. 162 .6. 2. Resonators by M. 6. Academic Press. Penn State University Press 1968. Dowling. 2nd ed. pp. Heckl: Körperschall.
mass [kg]. total length of edges [m]. insertion loss [dimensionless] frequency [Hz] resonance frequency [Hz] critical frequency [Hz] force [N] shear modulus [N/m2] distance [m]. bending stiffness [Nm2] bending stiffness per unit width [Nm] speed of sound [m/s] speed of bending waves [m/s] speed of longitudinal waves [m/s] power coupling factor [dimensionless] length [m] directivity [dimensionless] directivity index [dB] total acoustic energy [J]. Young’s modulus of elasticity [N/ m2] vibration isolation effectiveness. acceleration [m/s2] equivalent absorption area [m2]. mass per unit area [kg/m2] mass per unit length [kg/m] 163 . length [m] Aweighted sound pressure level [dB re pref] equivalent Aweighted sound pressure level [dB re pref] sound exposure level [dB re pref] Cweighted sound pressure level [dB re pref] equivalent sound pressure level [dB re pref] sound intensity level [dB re Iref] impact sound pressure level [dB re pref] sound pressure level [dB re pref] sound power level [dB re Pref] sound pressure level measured without frequency weighting [dB re pref] air attenuation factor [m1]. plate thickness [m] receptance [m/N] Struve function sound intensity [W/m2] reference sound intensity [W/m2] component of sound intensity [W/m2] Bessel fuction terminal source strength [W] wavenumber [m1] stiffness constant [N/m] adiabatic bulk modulus [N/m2] length [m] mean free path [m] loudness level [phone]. accelerance [m/Ns2] reference area [m2] bending stiffness per unit length [Nm].LIST OF SYMBOLS a A A0 B B´ c cb cL CP d D DI E Eiso f f0 fc F G h H H1 I Iref Ix Jm Jterm k K Ks l lm L LA LAeq LAE LC Leq LI Ln Lp LW LZ m m´ radius of sphere [m].
number of modes [dimensionless] sound pressure [Pa] instantaneous Aweighted sound pressure [Pa] reference sound pressure [Pa] rms value of sound pressure [Pa] static pressure [Pa] power [W] sound power [W] reference sound power [W] volume velocity associated with a fictive surface [m3/s]. z Cartesian coordinates [m] Za acoustic impedance [kg m4s1] acoustic radiation impedance [kg m4s1] Za. reflection factor [dimensionless]. transmission loss [dB] R0 transmission loss at normal incidence [dB] s standing wave ratio [dimensionless].mass per unit area [kg/m2] mass [kg] natural number [dimensionless] loudness [sone]. averaging time [s] T60 reverberation time [s] u longitudinal displacement [m] u particle velocity [m/s] ux component of the particle velocity [m/s] U velocity [m/s] v velocity [m/s] V volume [m3] w transverse displacement [m] wkin kinetic energy density [J/m3] wpot potential energy density [J/m3] x. r mechanical radiation impedance [kg/s] Zw separation impedance [kg m2s1] Y mobility (mechanical admittance) [s/kg] m´´ M n N p pA(t) pref prms p0 P Pa Pref q Q r α αm β γ δ ΔL absorption coefficient [dimensionless] mean absorption coefficient [dimensionless] angular displacement [radian] ratio of specific heats [dimensionless] damping coefficient [s1]. y. end correction [m] insertion loss [dB] 164 . cross sectional area [m2] t time [s] T absolute temperature [K]. r Zm mechanical impedance [kg/s] Zm. damping constant of viscous damper [kg/s] rrev reverberation distance in a room [m] R gas constant [m2s2K1]. spring constant [N/m] S surface area [m2]. generalised coordinate [m] volume velocity of source [m3/s]. directivity factor [dimensionless] radial distance in spherical coordinate system [m].
transmission coefficient [dimensionless] phase angle [radian].ΔV ζ η θ λ ν ξ ρ τ φ ω Ω ^ volume displacement [m3] viscous damping ratio [dimensionless] loss factor [dimensionless] polar angle in spherical coordinate system [dimensionless] wavelength [m] Poisson’s ratio [dimensionless] displacement [m] density [kgm3] time constant [s]. azimuth angle in spherical coordinate system [radian] angular frequency [radian/s] frequency ratio [dimensionless] indicates complex representation of a harmonic variable 165 .
166 .
15 Constructive interference. 63 167 . 23. 16 Bark. 6. 9 Cartesian coordinate system. 142. 58. 28. 134. 103 Absorption. 1. 36. 97. 15. 71 Beam. 2. 87. 29 Angular displacement. 6. 87. 57 Coincidence. 3 Cavity. 14 Crest factor. 135 Damping loss factor. 33 Consonant (intelligibility) 111 Constant percentage filters. 9.INDEX Absorption area. 145 Damping force. 39. 72 Basilar membrane. 23. 148 Bel. 93. 29 Acoustic properties of materials. 29 Acoustic impedance. 151 Bending waves on structures. 147 Acceleration. 110 Absorption coefficient. 5. 147 Apparent sound transmission loss. 35 Accelerance. 76 Air attenuation factor. 9. 75 dB HL. 62. 5 Antinode see Node Antiphase. 121 Combinations of monopoles. 151 Angular frequency. 27 Afilter see Aweighting AI principle. 4 Cancellation of sound. 109 Dantale. 106 Cfilter see Cweighting Characteristic impedance see Impedance Cochlea. 70 Damping coefficient. 59. 3. 18. 6 Audiogram. 52 Complex exponential representation. 2 Conservation of sound energy. 29. 20 Backward masking. 66. 98 Amplitude. 82 Background noise. 137 Damping constant. 145 Compliance see Stiffness Condenser microphone. 53. 121 Crossover frequency. 62 Averaging time see Integration time Aweighting. 148 Converging waves. 147 Accelerometer. 43 Boundary conditions. 69. 30. 151 Bending stiffness. 134 Acoustic filters. 42 Bandpass filters. 119. 21 Conservation of mass. 137 Danish Building Law. 3 Adiabatic process. 52 Complex stiffness. 29. 29 Adiabatic bulk modulus. 70 Axial modes. 124 Cweighting. 121 of beam. 41 see also Quadrature Aperture. 26 Critical band. 114. 29. 18 Bending moment. 137. 72 Critical frequency. 41 Complex amplitude. sound field in. 52 Analogous electrical circuit. 71. 109 Danish Working Environment Agency. 69. 3 Admittance. 115 Apparent mass. 20 Continuous structure. 2 see also Wave types Bessel function. 127 Audible frequency range. 66. effect of. 35 Acoustic twoport. correction for. 67 Baffle. 40.
71 Frequency response of microphone. 4. 137. 143 Diverging waves. 32 Energy of a signal. 114. 27. 22 Engine exhaust system. 15 Filter bank analysers. 17 Field variables. sound field in Energy balance equation. 48. 95 Echoellipse. 53. 70 Dwellings (reverberation control) 110 Dynamic stiffness. 3 Destructive interference. 2. 99 Far field. 18 Density of the medium. 24 Equivalent viscous damping ratio. 142. 134. 17 Dfilter see Dweighting Diatonic scale. 141. 61. 24 Equivalent rectangular bandwidth. 6 Excursion of a loudspeaker membrane. 52 168 . 100 Decibels. 90 Far field approximation. 14 Double construction. 16 Equation of motion for simple resonator. 48 Dispersion. 23. 41 Dipole strength. 64. 159 Echo. 10 Gas constant. 5 Frequency analysis. 15. 123 Dweighting. 103 Dipole. 9. 90 Directivity. 25 see also Intermittent noise Focusing. 74 Forward masking. 145 ERB. 32 potential. 71 Frequency weighting filters. 139. 95 Electret microphone. 127 Flexural waves see Wave types Fluctuating noise. 6. 96 Force. 147 Displacement ratio. 15 Frequency discrimination. 15 Flanking transmission. 152 Displacement. 137 Equivalent integration time. 138. 63 Freefield correction. 135. 72 Equivalent sound pressure level. 98. 22 Frequency selectivity. 142 Displacement response. 47 Exponential averaging see Time averaging Eyring’s formula. 155 Frequency. 146. 2 Diffuse sound field. 17 Deterministic signal. 38 Equally tempered scale. 87 Energy density in sound field kinetic. 138. 2.Decade. 149 Diffraction. 38 Harmonic sound field. 138 continuous structures. 16 Differentiation with respect to time. 135 Formant. 72 Euler’s equation of motion. 23 Free terminal velocity. 91 Directivity index. 15. 43 Directivity factor. 43 FFT analysers. 3 Gauss’s theorem. 140 Force transducer. 20 Detection of a pure tone in noise. 35. 147 Ground effect. 53. 127 Flanking transmission loss. 41 Direct field. 23 Fundamental frequency. 148 Equilibrium position. 149 Filter. 93. 21 Enclosure see Cavity. 85. 16 Decay curve. 33 Generalised coordinate. 67 Free field. 5. 22 Freefield method see Sound power determination Freefield microphones.
96 Modes. 69 Loudspeakers. 76. 23 Linearised wave equation. 87. 67. 37 Motion excitation. 55. 63. 46 specific acoustic. 31 Hooke’s law. 27. 56. 84. 135 Mass density. 10 Hearing level. 10. 67. 69 Mass. 92 Mean square value. 60. 64. 62 Hearing threshold. 59 Helmholtz equation. 157 Instantaneous energy density. 147 MTF. 7 Helmholtz resonator. sound in. 7. 136 see also Mechanical oscillator Mechanical systems. 55. 57. 29 Incident sound intensity. 36 Incident sound power. 152 Modal density. 65. 109 Intensity see Sound intensity Interface between two fluids. 71 Input impedance. 155 see also Mechanical admittance Mobility. 55. 55. 28 Mechanical resonators. 28. 34 Logarithmic frequency scale. 61. 37. 28 characteristic. 135 Membrane absorber. 136 Image sources. 64. 29. 32 Integration time. 11 Interference effects. 1 see also Wave types Loss factor. 152. 38. 64. 29 Masking. 35. 9. 4 Locally plane waves. 29 Input point mobility see Mobility Insertion loss. 81 Monopole. 31 mechanical. 66. 136 Lumped parameter models. 146. 15. 28 see also Mobility Mechanical oscillator. 128 Impedance acoustic. 58 Mobility. 69 Loudness level. 120 Material properties. 106 Middle ear. 2. 128. 58. 25 see also Time averaging Intelligibility. input for semiinfinite or infinite beam or rod. 150 see also Acoustic properties of materials Mean absorption coefficient. 2 Linearity. 146 radiation. 17 Mechanical admittance. 57. 65. 59. 161 Junctions between coupled pipes. 17. 68. 135. 42. 62. 16 169 . 94 Impact sound pressure level. 26 Inverse distance law. 55. 2 Inner ear. 150 Mass law. 16 Longitudinal waves. 29 Kinetic energy see Energy density Levels. 4. 99 Mean free path. 69 Helicotrema. 18 Linear averaging see Time averaging Linear frequency weighting. 56. 86 Incoherent signals see Uncorrelated signals Industry (reverberation control) 110 Independent sources. 76 Musical tones. 32 Instantaneous sound intensity. 65.Harmonics. 38 Intermittent noise. 157. 152 plate. 122 Loudness. 14. 20 Isolation effectiveness. 20 Inhomogeneous medium. 47 Lumped elements. 20. 4 Liquids. 63.
113 Reference sound intensity. 23 Random noise. 18 Reference velocity. 83 Oblique modes. 153. 17 see also White noise Piston in a baffle. 71 Pulsating sphere. 2 Oscillating sphere see Dipole Outdoor sound propagation see Ground effect Overtones see Harmonics Parseval’s formula. 148 Point dipole. 150. 137 Natural frequency. 3 Rayleigh’s integral. 9. 20 Reflection. 106. 5 Onethird octave bands. 17 Rapid Speech Transmission Index. radiation from. 37 Puretone source see Sinusoidal source Quadrature. 42 Pitch. 41 Point source see Monopole Poisson’s ratio. 15 Newton’s second law of motion. 23 Pressure node see Node Psychoacoustics. 156 Power transmission. 10. 161 Porous absorber. 105 Potential energy see Energy density Power coupling factor. 112 Office spaces (reverberation control) 110 Omnidirectionality. 16 Normal ambient conditions. 82 Noise see Random noise Noise event. 111 Pink noise. 6. 137 Node. 64 Phone scale. 2 Particle velocity. 37 Random errors see Statistical uncertainty Random incidence microphone. 4. 21 Reference sound pressure. 27 Nominal centre frequencies. 71 Partitioning into frequency bands. 77 RASTI see Rapid Speech Transmission Index Ratio of specific heats. 82 Octave bands. 21 Reference sound power. 23. 94 170 . 55. 3 Peak level. 156 Receptance. 17 Pascal. 26 Phase. 3 Number of modes. 81 see also Resonance frequency Nearfield characteristics. 42 Reactive sound field. 52 Phase speed see Wave speed Phon. 16 Orders of magnitude of perturbations.Natural angular frequency. Monopole Onedimensional wave equation. 38 Reduction index. 17 Partial masking. 146 Reciprocity principle. 28. 145. 5. 2. 43 see also Directivity. 136. 16 Plane waves. 15 see also Antiphase Radiation impedance see Impedance Radiation of sound. 67 Partials see Harmonics Particle displacement. 15 Receiving structure. 2. 4 Plate. 8. 155 Pressure microphone. 68 Phonems. 16. 63. 103 ODEON programme.
3. 160 Rubber isolator. 103 Sound exposure level. 38. 148 Root mean square value. 17 171 . 36 Refraction. 73 Source strength. 41. 76. 147 Sinusoidal source. 16 Sensitivity of auditory system. 9. 35 Sound power level. 30. 9. 4 Spherical coordinate system. 153 Sources of vibration. 75 Speech spectrum. 89. 109 Rigid surface. 76. 159 Sabine’s formula. 136. 20 Sound level meter. 106. 55. 10 see also Standing wave tube Standing wave tube. 97. 98. 90 Reverberation room. 6 Solution in product form. 134. 13 Spherical sound waves. 23 Separation impedance. 36 Standing waves. 124. 29 Simple source see Monopole Simultaneous masking. 103 Reverberation time. 64 Single degree of freedom system. 2. 112 Speed of sound. 108 Reverberation distance. 151 Shear waves see Wave types Sign convention. 2. 68 Sound absorption. 33 Sound power determination. 154 see also Stiffness Standing wave pattern. 122 Shadow see Diffraction Shear modulus. 135. 27 SII. 3 Static stiffness. 76 Speech intelligibility index. 9 Standing wave ratio. 31. 99 Scattering. 21 Sound power. 74. 89. 15 Rod. 161 Resonance. 76 Silencers. 142 Son. 38 Rms value see Root mean square value Rms sound pressure. 154 Stationary signals. 99 Schools (reverberation control) 110 SEL see Sound exposure level Semitone. 119. 26 Sound intensity. 68 Sone scale. 17 Speech intelligibility. 28. 18 Sound pressure. 96 Reflection factor. 141 nondimensional form. 112 Speech level. 1 Sound reflection see Reflection Source spectrum. 81 Stapes. 139 see also Natural frequency Resonant excitation. 97. 13 see also Monopole Spherical symmetry. 75. 75. 28. 8. 5. 103. 30 Resonance frequency. 159. 76 Speech Transmission Index. 2 Resilient element. 12 see also Monopole SPL see Sound pressure level Spring constant. 151 Shear force. reflection from. 21 Sound pressure level. 6.Reflection density. 79. 56 Static pressure. 10. 153 Specific acoustic impedance see Impedance Spectral density. 159. 138 Resonator absorber. 141 sum form. 1. 32 Sound intensity in a plane wave. 155 Source structure. 74. 136. 89. 98. 15 Rubber hardness. 34 Sound intensity level.
133. 133 Structural damping. 105 Viscously damped system. 18 Undamped simple resonator. 5 Wave speed for longitudinal waves. 100 Young’s modulus of elasticity. 19 Wavelength.Statistical models of sound fields. Dipole Vibration isolation. 24 Time derivative see Differentiation with respect to time Time integration. 156 Vibration isolator. 139. 151 Twoport. structural longitudinal waves. 17. 53 see also Time averaging Time weighting see Time averaging Timeaveraged energy density. 145 Struve function. 25 Time constant. 147 Vibrating sphere see Pulsating sphere.160 172 . 53. 33 Timeaveraged sound intensity. 52 Suspended ceiling. 149 shear waves. 20. 29 Typical values of sound power levels. 153. 140 Voiced. 73 Volume acceleration. 1 Transverse displacement in beams. 17. 160 Stochastic signals see Random noise Structureborne sound. 142. 5 Wavenumber. 148. 19 Uncorrelated signals. 3 Temporal integration. 3 Volume velocity. 106. 138. 21. 82 Tapping machine. 153 Vibroacoustics. 130 Weighted sound reduction index. 69 Thick wall. 2 Temperature. 150 bending waves. 113. 33 Transfer function. 142 Transversal waves. 129 White noise. 136 Viscous damping ratio. 134. 114 Transmitted force. 151 Weighted impact sound pressure level. influence on the speed of sound. 74 Velocity. 113 Transmission loss. 11 Transmission coefficient. 139 Undamped system. 152 Wave types. 148. 47 Volume displacement. 28. 128 Temperature fluctuations in sound field. 54 Time averaging exponential. 46 Subwoofer see Loudspeakers Sum of harmonic signals. 143 Unvoiced. 148 Viscous damper. 137. 32 Typical values of sound pressure levels. 118 Time average of a product. 83 Transmission between fluids. 28 see also Spring constant Stiffness correction. 109 Tangential modes. 144. 25 STI see Speech Transmission Index Stiffness.149. 37 Water. 142 Viscous friction. 11. 24 linear. 31 Statistical uncertainty in measurements. 27. 135. 4. 137. 148 shear or torsional waves. 150 bending or flexural waves.
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