Aero-acoustics MACE 61131

Lecture Notes

6. Experimental
Techniques

Sampling Data! 3

Lecture Notes MACE 61131/1. Introduction 1/31

 The Fourier Transform! 3

Measurements in the Frequency Domain! 3

The Discrete Fourier Transform! 4

Resolution! 5

Windowing! 9

Aliasing! 13

Acoustic Testing Methods! 14

The Two microphone technique! 14

Derivation of Equations! 16

Calibration! 18

The swapped microphone technique! 18

The Calibration box technique! 19

Pistonphone! 21

Static Calibration! 22

Design Issues! 23

Choice of pipe radius! 23

Position of first microphone! 24

Microphone spacing! 25

The Moving microphone technique! 26

Measurement of waves! 26

Advantages and Disadvantages of the moving microphone technique! 29

Summary! 30

Lecture Notes MACE 61131/1. Introduction 2/31

6.1. Sampling Data
6.1.1. The Fourier Transform
All continuous functions of time can be expressed as functions of
frequency using the Fourier Transform:
�∞
F (w) = f (t)e−iωt dt (6.1)
−∞

This form for acoustic signals is more useful than expressions in terms
of time, because so much of the generation and response depends
upon the frequency of the wave. These include our ears (see figure
6.1), transmission across walls, sound-absorbing materials,
resonators, etc.

6.1.1.1. Measurements in the Frequency Domain

Figure 6.1 The frequency response of the ear

The Fourier Transform described by equation (6.1) is an ideal

function which can be applied exactly to any analytical function. In
practice, however, when we measure sound we will not have either a

Lecture Notes MACE 61131/1. Introduction 3/31

 continuous function of time or a full history from -∞ to ∞. In this,
section we discuss the implications of separating a signal into its
frequency components, as in figure 6.2, without full knowledge of the
signal. Unsurprisingly, two issues are raised. The first arises from the
need to do the measurement over a finite period and the second arises
from the need to sample the data. We start, however, by defining the
Discrete equivalent of equation (6.1).
3

-1

-2

-3

0
-50

-40

-30

-20

-10

10

20

30

40

50
3 3

2 2

1 1

0 0

-1 -1

-2 -2

-3 -3
=
0
-50

-40

-30

-20

-10

10

20

30

40

50
0
-50

-40

-30

-20

-10

10

20

30

40

50

-1

-2

-3
0
-50

-40

-30

-20

-10

10

20

30

40

50

Figure 6.2 A signal split into its component frequencies

6.1.2. The Discrete Fourier Transform

Our first job is to rewrite the Fourier transform defined in equation
(6.1). Noting that, since we only have a finite period for the signal, the
frequency becomes a discrete function rather than a continuous one.
Thus the angular frequency becomes:
ω → kω0

where k is an integer and ω0 is the fundamental angular frequency, 2π/

NTs. Similarly the time is also discrete and becomes:
t → nTs

Lecture Notes MACE 61131/1. Introduction 4/31

 Finally the integral now becomes a sum so we have:
�
→Σ

That just leaves the time step dt. Since this is just a constant it is
normally ignored (just as the total time is ignored in going from a
Fourier series to a Fourier transform). Thus our discrete Fourier
Transform is defined as
N
�
DFT(k) = fn e−ikω0 nT
0

Notice that the discrete Fourier Transform is dimensionally different

from the Fourier Transform, just as the terms in a Fourier series are
dimensionally different from the Fourier Transform.

6.1.3. Resolution

Figure 6.3 A random signal

The first issue we need to consider when regarding real signals is

the problem which arises from only being able to measure it for a finite
time. Figure 6.3 shows a random signal. What we need to know is
how does the Fourier Transform of the signal change by virtue of only
having a finite proportion of the signal, such as illustrated in figure 6.4.

Lecture Notes MACE 61131/1. Introduction 5/31

 Figure 6.4 A Window on a random signal

Figure 6.4 is exactly equivalent to multiplying the original signal in

figure 6.3 by a square pulse. This is illustrated in figure 6.5. To see
the effect of this, we first consider the Fourier Transform of a unit
rectangular pulse.

Figure 6.5 A rectangular windowing function

Lecture Notes MACE 61131/1. Introduction 6/31

 To find the Fourier Transform of a unit rectangular pulse beginning
at t = 0 and finishing at t = T we simply change the limits of the integral
from -∞ to ∞ to 0 to T and complete the integral:
�T � �T � �
−e−iωt −e−iωT + e0
e−iωt dt = =
iω 0 iω
0

� �
e−iω 2 � iω T � e−iω T2
T
T ωT
= e 2 −e−iω 2 = 2 sin
iω ω 2
� � ωT � �
T sin 2
= T e−iω 2 (6.2)
ωT /2

Equation (6.2) has introduced the factor T on both the numerator and
the denominator so that the term in parentheses has the same
argument for the sine function as on the denominator. The term in
parentheses is a spherical Bessel function usually referred to as a sinc
function in the context of spectrum analysis. It is plotted in figure 6.6.

3
Secondary
2 peak spread

-1

-2

-3
0
-50

-40

-30

-20

-10

10

20

30

40

50

Figure 6.6 The response of a rectangular window alone

Lecture Notes MACE 61131/1. Introduction 7/31

 The sinc function shown in figure 6.6 has two important features. First,
instead of being a pure d.c. signal (i.e. with a peak purely at zero
frequency), as would be obtained for a constant unit amplitude, the
peak is spread over a range of frequencies. Secondly, secondary (and
indeed tertiary) peaks exist away from the initial peak. Both of these
features have important consequences when we consider measuring a
signal for a finite length of time. To see this, consider an ideal, purely
harmonic signal which exists only for a period 0 to T.

f (t) = sinω0 t

As for the rectangular pulse, we complete the integral in equation (6.1)

using a time interval of 0 to T to reflect the duration of the
measurement:
�T �T � �
eiω0 t − e−iω0 t
F (w) = sinω0 te−iωt dt = e−iωt dt
2i
0 0

�� � � ��T
ei(ω0 −ω)t e−i(ω0 +ω)t
= −
2(ω − ω0 ) 2(ω + ω0 ) 0
� � � �
(ω−ω0 )T (ω+ω0 )T
T
sin 2 T
sin 2
= −iei(ω0 −ω) 2 + ie−i(ω0 +ω) 2 (6.3)
ω − ω0 ω + ω0

spread

Secondary
peak

Figure 6.7 Rectangular windows applied to a harmonic signal

Lecture Notes MACE 61131/1. Introduction 8/31

 Equation (6.3) also contains the sinc function. This time, however, it is
shifted to frequencies ±ω0. Figure 6.7 shows three curves. The blue
curve is the ideal Fourier Transform for a harmonic signal for all time.
The black and red curves show plots of equation (6.3), with the red
curve being for a longer period, T. The effect of the finite
measurement time is therefore twofold: the frequencies are spread
out over a range (instead of being purely harmonic) and additional
peaks are created which could be wrongly interpreted as additional
frequency components. Both of these effects mean that the ability to
distinguish two closely spaced tones (i.e. the resolution) is reduced.
The only way to improve the resolution is to increase the measurement
period, as is evident from the red curve in figure 6.7. An example of
the Fourier transform of two distinct but closely spaced tones
measured for a finite time is given on the example sheet.

6.1.3.1. Windowing

0.75

0.5

0.25
0

3
0.5

1.5

2.5

Figure 6.8 A half cosine window

The resolution problem reported earlier cannot be reduced without

increasing the measurement period. Nevertheless, we can help to
distinguish real tones from artificial secondary peaks. We can do this
by realising that the secondary peaks in figures 6.6 and 6.7 come
about from the infinitely fast rise or fall at the start and end of the

Lecture Notes MACE 61131/1. Introduction 9/31

 measurement, just as rectangular pulse trains need an infinite number
of Fourier coefficients to fully represent the sudden rises. We can,
therefore, alter the secondary peaks by applying a smoothing function
called a window. At first we apply a simple half-sine pulse (as
illustrated in figure 6.8). Although this is simple, it does begin at zero
and end at zero so there will be no sudden rise or drop as there was
for the rectangular window. For our analysis, we use the same
harmonic signal:
f (t) = sinω0 t

Multiplying by the windowing function and taking the Fourier

Transform we have:
�T � �
πt
F (ω) = sin ω0 t sin e−iωt dt =
T
0

�T � �� �
eiω0 t − e−iω0 t eiπt/T − e−iπt/T
e−iωt dt
2i 2i
0

�T � �
1
=− ei(ω0 +π/T −ω)t
−e i(ω0 −π/T −ω)t −i(ω0 −π/T +ω)t
−e +e −i(ω0 +π/T +ω)t
dt
4
0

� �T
ei(ω0 +π/T −ω)t ei(ω0 −π/T −ω)t
= −
4i(ω − ω0 − π/T ) 4i(ω − ω0 + π/T ) 0

� �T
e−i(ω0 −π/T +ω)t e−i(ω0 +π/T +ω)t
+ − +
4i(ω + ω0 − π/T ) 4i(ω + ω0 + π/T ) 0
� �
iT i(ω0 −ω)T /2 cos((ω − ω)T /2) cos((ω − ω)T /2)
= e −
4 T /2(ω − ω0 − π/T ) T /2(ω − ω0 + π/T )
� �
iT i(ω0 −ω)T /2 cos((ω + ω)T /2) cos((ω + ω)T /2)
+ e −
4 T /2(ω + ω0 + π/T ) T /2(ω + ω0 − π/T )
(6.4)

Equation (6.4) describes the effect of the half-sine window on the

frequency components of the signal. To see this effect, figure 6.9
shows a comparison between a rectangular window (from equation

Lecture Notes MACE 61131/1. Introduction 10/31

 (6.3)) and a half-sine window (from equation (6.4)). There are three
differences. First the amplitude is reduced because more of the data
has been ignored. This can obviously be catered for by applying a
scaling. Secondly, as expected, the secondary peak is reduced
because there is now no sudden change in behaviour. It should be
noted, however, that the window uses less rather than more data and
so the actual resolution cannot improve. This is seen in the third
difference, which is the broader nature of the main peak.
Nevertheless, applying a window function like this can help, in
comparison with a rectangular window, in distinguishing between real
peaks and secondary peaks resulting from the window function

Rectangular
window
Reduced
peak Half-sine
window
Increased
spreading
Reduced
secondary
peak

Figure 6.9 The effect of a half sine-window

Hann window Half-sine

0.5(1-cos(2πt/T) window

Hamming window
0.53836-0.46164cos(2πt/T)
zero
gradient

Figure 6.10 Sine, Hann and Hamming windows

Lecture Notes MACE 61131/1. Introduction 11/31

 Other notable windows are the Hann and Hamming windows
shown in figure 6.10. These have the advantage over the half-sine
window in that they also have zero gradient at the edges of the
window. Their response is shown in figure 6.11
Rectangular
window
Half-sine
window
Hann window
0.5(1-cos(2πt/T)

Hamming window
0.53836-0.46164cos(2πt/T)

Figure 6.11 The Effect of Sine, Hann and Hamming windows

The effect of the different windowing functions is difficult to see

from figure 6.11 because of the different amplitudes of the main peak.
To see the effect more clearly, figure 6.12 shows the effects of the
windowing functions when rescaled to give the same amplitude of the
main peak. The Hann and Hamming windows have the smallest
secondary peak (with the Hamming the smallest). However, these two
also have the broadest main peak (again the Hamming is the
broadest).

Figure 6.12 Rescaled effects of different window functions

Lecture Notes MACE 61131/1. Introduction 12/31

6.1.4. Aliasing

Figure 6.13 Two sampled signals

The second problem associated measuring data occurs because

of the need to sample the data. This results in discrete points, which
leads to the possibility of a different signal passing through exactly the
same points. As a simple example of this, figure 6.13 shows a green
signal with exactly two samples per wavelength. The red signal, which
has twice the frequency, also passes through exactly the same points.
This means that it would be impossible to distinguish the two signals.
Although one could assume that only the lower frequency is present,
there is another problem. If both frequencies exist then the higher
frequency tone will distort the reading, making it appear greater than it
really is. To see what frequencies distort the signal, consider a tone of
frequency 2π/T0 sampled with a period of Ts. The sampled signal then
appears as:

ei2πnTs /T0 = ei(2πnTs /T0 ±2nπ)

Ts ±T0
i2nπ( )
=e T0

which is the signal which would also be obtained by a wave of

frequency ω0 ± ωs. This is like mirroring the Fourier Transform about
the line ωs/2 as shown in figure 6.14

Distortion by aliasing is a common problem when sampling data.

It can be eliminated by using a low pass filter. For a perfect filter this
can be applied at ωs/2. However, real filters are not perfect and so a
filter is usually applied at around 80-90% of ωs/2. It is also important to

Lecture Notes MACE 61131/1. Introduction 13/31

 realise that the filter must be applied as an analogue filter before the
signal is sampled.

RORRIM MIRROR

Aliased component Actual signal

ω0 - ωs ωs/2 ω0 ω1
ωs - ω1
Figure 6.14 Aliasing

6.2. Acoustic Testing Methods

In this section we discuss the two major techniques used for
testing acoustic materials and devices, such as resonators. Both of
these techniques are used to deduce the response of system to
acoustic excitation. The first method decomposes the measurements
at two independent locations in a pipe into the forward and backward
travelling waves, whereas the second method deduces this information
from the measurements of the amplitude and position of the first
maximum and first minimum of a signal. Both rely on the signal being
of sufficiently low frequency for only plane waves to exist in the pipe.

6.2.1. The Two microphone technique

The two (or four) microphone technique relies upon the
measurement of the acoustic pressure at two independent locations in
a pipe. If the frequency is sufficiently low that the sound waves remain
planar, then these measurements can be decomposed into the forward
a backward travelling waves. The basic arrangement is illustrated in
figure 6.15. A sample of material is attached at one end of the pipe
with a rigid backing. A sound wave is then sent along the pipe, and the
acoustic pressure is measured simultaneously at two independent
locations, as shown. This arrangement is sufficient to deduce the

Lecture Notes MACE 61131/1. Introduction 14/31

 acoustic behaviour of a rigidly-backed sample. To obtain a more
fundamental measurement we must also measure using a different
backing, such as a rigidly-backed cavity. An alternative is to apply the
two microphone technique twice, once either side of the sample. This
is illustrated in figure 6.16.

Data logger

Charge
amplifier

rigid back Microphones

sample

sample holder

Figure 6.15 The two microphone technique

The arrangement depicted in figure 6.16 gives a more

fundamental view of the acoustic response by measuring the
transmission across the sample as well as the reflection from it. It also
eliminates the need for a second experiment with a different backing.
Since it uses four microphones it is sometimes referred to as a four
microphone technique. However, since it simply applies the two
microphone technique separately to either side of the sample, it is
usually known as simply the two microphone technique. A photograph

Lecture Notes MACE 61131/1. Introduction 15/31

 of the two microphone technique used for a sample of material can be
found in figure 6.17.

Data logger

Charge
amplifier

rigid back Microphones

sample

sample holder

Figure 6.16 The four microphone technique

Figure 6.17 A photograph of the four microphone technique

6.2.1.1. Derivation of Equations

Figure 6.18 shows a pipe in which two plane waves, one left
travelling and one right travelling, are propagating. Two microphones,
at positions x1 and x2 respectively, measure the local sound pressure.
For harmonic waves these pressures can be expressed as:

p1 = p̂1 eiωt

and

p2 = p̂2 eiωt

Lecture Notes MACE 61131/1. Introduction 16/31

 Both pressures are the local superposition of the two waves. Since
both the waves are plane, harmonic waves, we can express the right-
travelling wave as:

A = Â1 ei(ωt−kx)

and the left travelling wave as:

B = B̂1 ei(ωt+kx)

x$ p$ x% p%

Aei!
" t"kx#

Bei!
" t!k x#

Figure 6.18 Measuring waves using the two microphone technique

Considering the superposition at position 1, and dropping the factor

eiωt as constant throughout we have:

p̂1 = Âe−ikx1 + B̂1 eikx1 (6.5)

Similarly at position 2 we have:

p̂2 = Âe−ikx2 + B̂1 eikx2 (6.6)

We wish to solve for the waves A and B so we simply solve the

simultaneous equations (6.5) and (6.6). First we eliminate B.
Multiplying equation (6.5) by e−ikx1
:

p̂1 e−ikx1 = Âe−2ikx1 + B̂1 (6.7)

Similarly multiplying equation (6.6) by e−ikx2 :

p̂2 e−ikx2 = Âe−2ikx2 + B̂1 (6.8)

Lecture Notes MACE 61131/1. Introduction 17/31

 The wave A is then simply found by subtracting equation (6.8) from
equation (6.7) and rearranging:
� �
p̂1 e−ikx1 − p̂2 e−ikx2 (6.9)
 =
e−2ikx1 − e−2ikx2

Similarly the left-travelling wave B is:

� �
p̂1 eikx1 − p̂2 eikx2
B̂ = (6.10)
e2ikx1 − e2ikx2

Equations (6.9) and (6.10) are extremely useful for determining the
acoustic response of materials and acoustic devices to acoustic
excitation. However, since it depends upon the simultaneous
measurement of two different signals by two microphones, it is
important that the microphones are calibrated both for their individual
responses and their relative responses.

6.2.1.2. Calibration
Two types of calibration are essential for the two microphone
technique: relative calibration and absolute calibration. Of these
relative calibration is the most important to ensure that the
measurements by any of the microphones is the same as it would
have been if measured using the other microphone. We consider this
type of calibration first. There are two techniques commonly used for
this: the swapped microphone technique and the calibration box
technique.

6.2.1.2.1. The swapped microphone technique

The swapped microphone technique relies upon taking a set of
measurements and then swapping the microphones over to obtain a
second set of measurements. Any relative errors between the
microphones are then eliminated by taking a geometric average of the
two sets of readings. To see this we start by taking the microphone
initially at location 1 as the reference microphone. On the first test
then we have readings
p̂1 = p̂1,ac and p̂2 = p̂2,ac rerr

where p̂1,ac and p̂2,ac are the calibrated readings (relative to the
reference microphone), and rerr is the relative error between the
second microphone and the reference microphone. If we then swap

Lecture Notes MACE 61131/1. Introduction 18/31

 the microphones over, the error moves to position 1, so the swapped
readings are
p̂1,s = p̂1 rerr

and
p̂2
p̂2,s = p2,ac =
rerr

We want to find the equivalent of equations (6.9) and (6.10) for the
calibrated signals. We can use the swapped measurements to find the
calibrated ratio between the signals at positions 1 and 2. Multiplying
the ratios from each of the two measurements we have
� �� � � �2 � � � �2
p̂1 p̂1,s p̂1,ac rerr p̂1,ac
= = = T12
2
(6.11)
p̂2 p̂2,s p̂2,ac rerr p̂2,ac

The calibrated signal at position 2 can then be found using the

calibrated ratio from equation (6.11):
p̂1
p̂�2 =
T12

We can then obtain the calibrated wave readings by using this

calibrated value in equations (6.9) and (6.10):
� �
p̂1 e−ikx1 − p̂�2 e−ikx2 (6.12)
Acal =
e−2ikx1 − e−2ikx2
� �
p̂1 eikx1 − p̂�2 eikx2
Bcal = (6.13)
e2ikx1 − e2ikx2

The swapped microphone technique is commonly used to improve the

readings. However, it relies upon having to repeat the entire
experiment which can be very time consuming. This problem is
exacerbated when three or more microphones are used to give
redundancy. As a result, the calibration box technique, which requires
calibration just once, is more common.

6.2.1.2.2. The Calibration box technique

The alternative technique to the swapped microphone technique
is to use a calibration box (an example of which is shown in figure
6.19). This is used to measure a transfer function between the two

Lecture Notes MACE 61131/1. Introduction 19/31

 microphones. Unlike the swapped microphone technique the
calibration only has to be performed once. It can also be easily
extended for any number of microphones. The basic technique is
illustrated in figure 6.20. The basic idea is for the microphones to be
placed as close as possible to each other. This allows up to two
microphones to be calibrated relative to one reference microphone at
once. Optical access is also desirable, as seen in figure 6.19, since
this makes it easier to locate the microphones at the same place.

Figure 6.19 A calibration box

The calibration box technique is a fairly robust way to calibrate the

microphones provided the frequency is not too high (in which case the
distance between the microphones may be significant in comparison
with the wavelength). At such high frequencies, however, the diameter
of the microphones also becomes an issue since the microphones will
give an average reading of the pressure across the face.

All measurements require relative calibration, as described.

When we are interested in reflection, transmission of absorption
coefficients, relative calibration is sufficient because we only need to
know the complex ratio between the left and right-travelling waves.

Lecture Notes MACE 61131/1. Introduction 20/31

 Sometimes, however, such as for non-linear measurements, we also
need to know the overall amplitude. This requires us to also calibrate
the amplitude. We describe this next.

Data
logger

microphones

Calibration
unit

Signal
generator

200

Figure 4: Illustration of the calibration box technique

Figure 6.20 Using a calibration box

6.2.1.2.3. Pistonphone
Adaptor

Pistonphone
7
microphone
Data logger

Figure 5: Illustration of the amplitude calibration technique using a pistonphone.

1.5.3
Figure 6.21 Using a pistonphone
Absolute amplitude calibration
Both the previous two techniques calibrate the microphones relative to each other. In this
The
sectionstandard
we consider thefortwo calibrating the acoustic
techniques for calibrating the amplitude. amplitude measure by a
The standard is rather
microphone
poor and is rather poor because it relates to only oneas frequency.
relates to only one frequency. The official way to calibrate is using a pistonphone
illustrated in figure 5. This device has a small diaphragm which acts as a loudspeaker which
The official
oscillates atway
a knownto calibrate
frequency (120Hz) andis a using a pistonphone
known amplitude (around 115dB, butas illustrated in
specified
for each unit). Each microphone fits snuggly in the device (using an adaptor for different
figuresize6.21.
microphones).This device
The output has
is logged a small
and compared to thediaphragm which
reference. The accuracy of the acts as a
loudspeaker
amplitude which oscillates at a known frequency (120Hz) and a
generated by the pistonphone depends upon the atmospheric pressure
be measured. A calibration chart is also included with a pistonphone. A photograph of a
which should

pistonphone kit is shown in figure 6.

Although not strictly an official method for calibrating microphones, an equally reliable
Lecture Notes
method is illustrated in figureMACE
7. Here 61131/1.
an industrial Introduction
vacuum cleaner and a duster is used to 21/31
produce a pressure field. All transducers (such a microphones) are located in the space and
the dc component of the resulting signal is compared to the pressure tapping. This method
is particular useful for pressure transducers such as kulites which are used for both mean (dc)
pressure measurements and unsteady measurements. It assumes that the response is linear
 known amplitude (around 115dB, but specified for each unit). Each
microphone fits snugly in the device (using an adaptor for different size
microphones). The output is logged and compared to the reference.
The accuracy of the amplitude generated by the pistonphone depends
upon the atmospheric pressure which should also be measured and
compared to the calibration chart included with a pistonphone.

Pistonphone

microphone Pressure
adaptor gauge
Figure 6: Photograph of a pistonphone.

Figure 6.22 A pistonphone

A photograph of a pistonphone kit is shown in figure 6.22. The

9
official standard calibrates at just one frequency (usually 120Hz or
1kHz). The microphones must then be assumed to have a linear
response for all other frequencies.

6.2.1.2.4. Static Calibration

Along with the dynamic calibration, it is also useful to apply static
calibration. This can be achieved using the rig illustrated in figure
6.23.

Lecture Notes MACE 61131/1. Introduction 22/31

 pressure
tapping

duster microphones

data logger
Vacuum
cleaner

Figure 7: Illustration of the dc amplitude calibration technique.

Figure 6.23 Static Calibration
the microphone spacing) and the assumption that the acoustic waves in the pipe are planar.
ThisHere
latter condition affects the frequency
an industrial vacuum range which can be studied,
cleaner and a the duct/pipe
duster dimension
is used to
and the position of the first microphone from the end of the pipe. We look at both of these
produce
conditions a constant
in turn pressure
in the next two sections. field. All transducers (such a
microphones) are located in the space and the dc component of the
1.6.1 Microphone spacing
resulting signal is compared to the pressure tapping. This method is
In order to obtain equations for the forward and backward-propagating waves in the pipe,
particular usefulto solve
it was necessary for pressure transducers
the set of simultaneous equations such
1 and 2.as kulites
Like which are
all simultaneous
used for both
equations, they aremean (d.c.)
only soluble pressure
provided measurements
the matrix which represents them is and unsteady
non-singular.
Writing equations 1 and 2 as a matrix equation we have:
measurements. It assumes that the response is linear (as the
pistonphone method does). P�However, = AW � care should be taken (16) with
certain
where dedicated microphones where there is a drop off at low
� � � � � �
frequency. � p̂1 e−ikx eikx
1 1
� Â
P = A= −ikx2 ikx2 W = (17)
p̂2 e e B̂
In system
The commonis then with
singularall experimental
when the determinant oftechniques,
matrix A is zero. the two
microphone
I.e. when:
technique has limited �
applications.
�
In particular it is limited by the
� � � e−ikx eikx �
need to invert the
�A� = matrix
� � �
� −ikx (a � condition
1
�
=0=e
1
ik(x −x )which
− e−ik(x −xaffects
2 )
=1 the microphone
2 1
� e eikx �
2 2

spacing) and the assumption that the acoustic waves in the pipe are
planar. This latter condition affects 10 the frequency range which can be

studied, the duct/pipe dimension and the position of the first

microphone from the end of the pipe. In the next section we look at
how to design an experimental rig to avoid these issues.

6.2.1.3. Design Issues

6.2.1.3.1. Choice of pipe radius
The radius of the pipe used with the two microphone technique is
limited by the need for the waves to remain planar. To achieve this we
require that all high-order modes by cut-off in the pipe. In chapter 3

Lecture Notes MACE 61131/1. Introduction 23/31

 we found that the condition for all radial and circumferential modes to
be cut-off was:
2
jmn ω2
> 2
r02 c̄

We can drive the pipe with axisymmetric waves and so the condition
reduces to:
3.83c̄
r0 < (6.14)
2πfmax

As an example, suppose we wish to design an experiment to be used

to measure the response up to frequencies up to 1kHz. From equation
(6.14) we then have:
3.83c̄
r0 < = 209.7mm
2π1000

If we chose exactly this radius, all the high-order modes would, indeed,
be cut-off. Unfortunately, however, at frequencies near 1kHz the high-
order modes would only just be cut-off. As a result they would decay,
but the decay would be slow. This would require us to measure waves
some distance from the end to ensure that all these modes had
decayed sufficiently. As a result a safety margin would normally be
taken to ensure rapid decay. Thus a pipe radius of 50mm would be a
good compromise.

6.2.1.3.2. Position of first microphone

Having chosen our pipe radius, we must now ensure that at the
highest frequency (when the modes are the nearest to being cut-on) all
the modes have decayed by a large amount, typically 20dB, by the first
microphone. The slowest decay will, of course, be for the lowest of
these modes. The first radial mode has the following wavenumber:
�
3.832 4π 2 fmax
2
γ01 = i −
r02 c̄2

Taking a distance of L to the first microphone, the decay by the time

the first radial mode from the end reaches the first microphone will be:
r !
„r « 4π 2 fmax
2 2
r0
3.832 4π 2 fmax
2
− 3.832 − L
− 2 − c̄2
L c̄ 2 r0
eγ01 L
=e r0
=e

Lecture Notes MACE 61131/1. Introduction 24/31

 For a 20dB decay we then require:
r !
4π 2 fmax
2 2
r0
− 3.832 − c̄2
L
r0
e = 0.1

A condition for the distance L can then be obtained by taking the

natural logarithm of both sides:
L ln(0.1)
= −� (6.15)
r0 4π 2 fmax
2 r02
3.832 − c̄2

Following our previous example with a 50mm radius pipe taking

measurements up to 1kHz we require:
L ln(0.1) ln(0.1)
= −� = −� = 0.61
r0 4π 2 fmax
2 r02
3.832 4π 2 (1000)2 (0.05)2
3.832 − c̄2
− (344)2

I.e. L = 31mm

6.2.1.3.3. Microphone spacing

Our next limitation is the need for the two microphone locations to
be independent so that we can solve the simultaneous equations. A
simple way to find the condition is to write the equations in matrix form
so that we can make sure the determinant is not zero. The equations
for the pressures at the two microphones are written in matrix form as:
� � � �
A p̂1
A = (6.16)
B p̂2
� �
e−ikx1 eikx1
where A =
e−ikx2 eikx2

For matrix equation (6.16) to have a solution we require the

determinant of the matrix to be non-zero. I.e.
� −ikx �
� e 1
eikx1 �
� −ikx � �= 0
� e 2
eikx2 �

Expanding then gives

eik(x2 −x1 ) − e−ik(x2 −x1 ) �= 0

which can be expressed as

Lecture Notes MACE 61131/1. Introduction 25/31

 sin(k(x2 − x1 )) �= 0

In other words the matrix becomes singular, and the two microphone
technique fails to be useful, when the distance between the two
microphones is an integer multiple of a half wavelength. In practice, a
working range of between 5 and 95\% of a half wavelength is usually
specified. However, this is not strictly correct since it could be greater
than a half wavelength so long as a half wavelength is itself ignored. If
the higher range is taken, then this leaves gaps in the solutions unless
multiple pairs of microphones are used to provide redundancy.

6.2.2. The Moving microphone technique

microphone sample

loudspeaker

Figure 6.24 The moving microphone technique

The traditional way to measure the acoustic response of

components is to use the moving microphone technique, as illustrated
in figure 6.24. This consists of a sample holder (where a rigid back or
a cavity can be attached) attached to a pipe. Acoustic waves are
generated using a loudspeaker through which is mounted a
microphone on a rod to measure the acoustic pressure inside the pipe.
The microphone is free to move along the pipe to measure the
acoustic pressure at more than one position.

6.2.2.1. Measurement of waves

To understand how this can be used to measure the sound field, it
is useful to reverse the direction of the diagram shown in figure 6.24 so
that waves moving away from the sample are right-travelling and those
moving towards it are left travelling. These are shown in figure 6.25

Lecture Notes MACE 61131/1. Introduction 26/31

 pmax pmin

xmax xmin
Figure 6.25 Using the moving microphone technique

We now need to find the waves from the measurements of the

maximum, minimum and position of the first maximum. If it is possible
to measure the phase difference between the two positions then we
can calculate the waves A and B completely. In reality, this is often
difficult to do. When it is not possible we can still determine the waves
A and B but are unable to determine which one is which. This is not
usually a restriction since we are usually measuring something with a
loss (other through transmission or absorption) and so we would
normally know which one is the bigger of the two. First we derive the
solution assuming we do measure the phase difference between the
signals at the two positions. For harmonic propagating waves we
have:

A = Âeiω(t−x/c̄)

B = B̂eiω(t+x/c̄)

The signals at the two positions will be the local superposition of these
two waves. Hence, writing the measured signals in terms of the two
waves yields:

pmax = Âeiω(t−xmax /c̄) + B̂eiω(t+xmax /c̄) (6.17)

pmin = Âeiω(t−xmin /c̄) + B̂eiω(t+xmin /c̄) (6.18)

Equations (6.17) and (6.18) are simultaneous equations for the two
unknowns. Eliminating B gives:
� �
Âeiωt
e−2iωxmax /c̄
−e−2iωxmin /c̄
= pmax e−iωxmax /c̄ − pmin e−iωxmin /c̄
(6.19)

Lecture Notes MACE 61131/1. Introduction 27/31

 Equation (6.19) can be rearranged to give the wave A. However the
positions of the two measurements have been specifically chosen to
be the first maximum and first minimum respectively. Since the
maxima occur when the waves are exactly in phase, and the minima
occur when the waves are exactly out of phase, adjacent maxima and
minima must be a quarter wavelength apart. Thus:
ωxmax ωxmin π
= +
c̄ c̄ 2

and

−2Âeiω(t−2xmin /c̄) = (−pmin − ipmax ) e−iωxmin /c̄ (6.20)

If we now rearrange equation (6.20) we obtain an expression for wave

A:
� �
pmin + ipmax
Âe =
iωt
eiωxmin /c̄ (6.21)
2

Similarly B is given by:

� �
pmin − ipmax (6.22)
B̂eiωt
= e−iωxmin /c̄
2

Equations (6.21) and (6.22) rely on a measurement of the phase of the

pressure measurements. This can be achieved by measuring the
acoustic pressure in a continuous experiment, i.e. allowing the signal
(and the data logger) to continue whilst the microphone is moved. If
this is not done, then we can still obtain the two waves, we just cannot
distinguish which one is which. To do this we note that at the peak the
two waves are in phase, and at the trough the two waves are out of
phase. Thus:
|pmax | = |A| + |B| (6.23)

|pmin | = |A| −| B| (6.24)

or
|pmin | = |B| −| A| (6.25)

If |B| > |A| (for an absorbing device, so we solve equations (6.23) and
(6.25) rather than (6.23) and (6.24)) we have:

Lecture Notes MACE 61131/1. Introduction 28/31

 |pmax | −| pmin | (6.26)
|A| =
2

|pmax | + |pmin | (6.27)

|B| =
2

Noting that the maxima of A and B occur at xmax, we then have:

A = |A|eiω(t+xmax /c̄) (6.28)

B = |B|eiω(t−xmax /c̄) (6.29)

6.2.2.2. Advantages and Disadvantages of the moving microphone

technique
Unlike the two microphone technique, the moving microphone
technique involves only one microphone. This has both advantages
and disadvantages. The advantages include:

• Only one microphone and so easier to calibrate.

The disadvantages include:

• It is difficult to gain an accurate measurement of the microphone

position with a movable microphone.

• Measurements need to be taken for two different cavities to

completely characterize a sample.

• The moving microphone technique use a measurement of the

acoustic minimum in the pipe. By definition, this is the smallest
pressure in the pipe, where the experimental errors are greatest.

Lecture Notes MACE 61131/1. Introduction 29/31

6.3. Summary

Fourier Transform
�∞
F (w) = f (t)e−iωt dt
−∞

Discrete Fourier Transform

N
�
DFT(k) = fn e−ikω0 nT
0

Window functions

# Half-sine:# # # # sin(πt/NT)

# Hann Window:# # # 0.5(1-cos(2πt/NT)

# Hamming Window:# # 0.53836-0.46164cos(2πt/T)

Aliasing:# More than one signal can pass through measured points

# # # Eliminate by low-pass filtering at half the sample frequency.

The Two microphone technique

� �
p̂1 e−ikx1 − p̂2 e−ikx2
 =
e−2ikx1 − e−2ikx2

Lecture Notes MACE 61131/1. Introduction 30/31

 � �
p̂1 eikx1 − p̂2 eikx2
B̂ =
e2ikx1 − e2ikx2

Maximum pipe radius:

3.83c̄
r0 <
2πfmax

Minimum distance to first microphone:

L ln(0.1)
= −�
r0 4π 2 fmax
2 r02
3.832 − c̄2

Microphone spacing condition:## sin(k(x2 − x1 )) �= 0

The Moving microphone technique

With phase information:# #

� �
p min + ipmax
Âeiωt = eiωxmin /c̄
2
� �
pmin − ipmax
B̂eiωt
= e−iωxmin /c̄
2

With no phase information:##

|pmax | −| pmin |
|A| =
2

|pmax | + |pmin |
|B| =
2

A = |A|eiω(t+xmax /c̄)

B = |B|eiω(t−xmax /c̄)

Lecture Notes MACE 61131/1. Introduction 31/31