Tools-1. Frequency Domain Analysis PDF

NVH Problem Diagnosis - Tools 1.
Frequency Domain Analysis page 1

Author: D. Griffiths Revised 01/17/2011 total pages 27
Frequency Domain Analysis

Contents

Introduction

1. Basic Concepts of the Frequency Spectrum

2. Definitions

3. Typical Frequency Spectra and Their Characteristics

4. Basic Problem#1 Aliasing

5. Basic Problem #2: Digitizing, Overloads and Underloads

6. Basic Problem#3 Leakage

7. Hann Window

8. Other Windows

9. Comparison of Window Parameters

10. Averaging and the Power Spectrum

11. Overlap Processing

12. Statistical Fluctuations

13. Effect of Frequency Resolution Power Spectral Density (PSD) 3
rd
Octaves

14. Cross-Spectra

15. Coherence

16. Transmissibility and Transfer Functions

17. Strategy and Checklist

Appendix: Alternative Conventions for Fourier Coefficients

NVH Problem Diagnosis - Tools 1. Frequency Domain Analysis page 2
Introduction

Most automotive problems are best analyzed in the frequency domain. There are several reasons for this
including:

a. Many problems are due to rotating components which produce sinusoidal forces at specific
frequencies
b. Human sensitivity to noise and vibration is strongly frequency dependent
c. Many problems are associated with resonances which occur at specific frequencies.

The main objectives of this document are to:

- Familiarize the engineer with basic concepts of frequency domain analysis
- Provide basic definitions
- Outline key measurement parameters
- Outline a strategy for accurate measurement of the frequency spectrum

1. Basic Concepts of the Frequency Spectrum

Joseph Fourier developed a system of representing any signal by a sum of sine waves. The amplitudes of
the sine waves are known collectively as the frequency spectrum. Later this method was extended to
signals that are sampled at discrete time intervals.

The standard procedure starts with a block of data with length T. It is sampled it at a sampling
frequency fs, the samples being separated by a time T=1/fs. The total number of samples per block is
the blocksize N=Tfs=T/T. The individual samples start at the beginning of the block and are labeled
n=0,1,..N-1. The general concept of time domain sampling is illustrated below for N=32, a(n) being the
value of the sample at t=nT.

Sampling Data Block Length T

Fourier analysis is then used to represent the signal a sum of cosine waves of the form Akcos(2fkt+k).
The frequency is given by fk=k/T . Ak is called the amplitude of the spectrum and k is the phase. The
term frequency spectrum is applied to a plot of the amplitudes of the sine waves versus frequency. For
NVH applications each spectral line represents the root mean square (RMS) level of the sine wave at
0 4 8 12 16 20 24 28 32
a
(
n
)

Time and Sample Number n
This point not in block
T
AT
the specified frequency i.e. Ak/2. This is illustrated schematically in the following plot. Most
spectrum analyzers show the envelope of the spectrum which is indicated by the dashed line.

Schematic of the Frequency Spectrum

Notice that the spectral lines are separated by f=1/T. In practice the maximum frequency for which the
spectrum is accurate is approximately fmax~0.8(N/2)f.

2. Definitions

This section covers most of the boring details, specifically the basic parameters and the mathematical
expressions defining the discrete Fourier transform.

Basic Parameters

N = Blocksize = Number of measurements per block (samples)
T = length of time record (seconds)
n = sample number from 0 to N-1
tn = time at which n-th sample is taken (from beginning of block), N nT t
n
/ =
fs = Sample rate = N/T
T= time between samples=T/N=1/fs
f = frequency resolution = 1/T
fmax = Maximum frequency which can be analyzed =Nyquist Frequency = fs/2=Nf/2
fk = Frequency of k-th component of the frequency spectrum, k=0 to N/2
a(tn) = n-th measurement in block
A(fk)= Real Fourier Amplitude (RMS) at frequency fk
(fk)= Phase angle at frequency fk
Ac(fk)=Complex Fourier Amplitude, )) ( exp( ) ( ) (
k k k C
f j f A f A | =

j = -1

Real NVH Fourier Expansion

The usual NVH convention is to write the Fourier expansion as a real expansion whose coefficients are
the RMS values of the equivalent cosine waves. This expansion has the form:

=
=
+ + + =
1 2 /
1
2 / 0
) 2 cos( ) ( 2 ) 1 )( ( ) ( ) (
N k
k
k n k k
n
N n
t f f A f A f A t a | t

In this expansion:
A
k
/2
f=1/T
f
k


- Each A represents the RMS amplitude of a cosine wave; there are (N/2+1) As
- The s are phase angles; there are (N/2-1) s
- There are a total of N frequency domain parameters
- The factor \2 is introduced to make the As from k=1 to k=(N/2-1) refer to RMS values
The factor 2 does not apply to the k=0 and k=N/2 terms because their amplitude is equal
to their RMS level e.g. the constant DC level equals its RMS level.
- The maximum frequency in the expansion is ) 2 /( 2 /
2 / max
T N f N f f
N
= A = =

The observant reader will note that based on this formula it would appear that the signal level at t=0
must equal the signal level at t=T, that is a(T)=a(tN)=a(0). Of course there are many signals for which
this is not true. However we have excluded the point n=N from the block so this restriction does not
hold. And in fact the formula gives an accurate and unique representation of the function at the discrete
points tn as long as N n < s 0 .

Parsevals Theorem

Parsevals theorem relates the mean square time domain signal (the average power) to the Fourier
amplitudes. The discrete version of Parsevals theorem states that:

=
=
=
=
=
2 /
0
2
1
0
2
) ( ) (
1
N k
k
k
N n
n
n
f A t a
N

This can be useful for checking the reasonableness of frequency domain results.

Complex NVH Expansion

The complex expansion is used because it makes for a more compact notation, and it makes it easier to
calculate theoretical formulas. The main reason for including this form here is that it is commonly used
in the literature and in software documentation. In particular we can easily prove that the expansion
gives a complete and unique description of time series. So this section is included for the theoretically
inclined, others may skip it.

The complex Fourier expansion is:

=
=
=
1
0
2
) ( ) (
N k
k
t f j
k C n
n k
e f A t a
t

The reverse transform is:

=
=
=
=

= =
1
0
2
1
0
2
) (
1
) (
1
) (
N n
n
N
nk
j
N n
n
t f j
n k C
e n a
N
e t a
N
f A
n k
t
t

Here we use ) (
k C
f A to denote a complex amplitude. The relationship between ) (
k C
f A and the
parameters of the real expansion is:

k
i
k k C
e f A f A
|
) ( ) ( =

The real and imaginary parts of the amplitude are then defined as:

) cos( ) ( )} ( Re{
k k k C
f A f A | =


) sin( ) ( )} ( Im{
k k k C
f A f A | =

Note that the frequency summation goes up to twice the frequency used in the previous real expansion.
However, because the as are real, the second half of the summation is related to the first half via:

k
i
k C k C k N C
e f A f A f A
|
= = ) ( )* ( ) (

This ensures that there are N independent frequency domain parameters. Also, in effect it implies that all
the information is contained in the range 0<f<fN/2.

To check the uniqueness and completeness of these transformations we can substitute the reverse
transform in the forward transform. With a little effort we find:

=
=

=
=
=
1
0
/ ) ' ( 2
1 '
0 '
) ' (
1
) (
N k
k
N n n k j
N n
n
e n a
N
n a
t

We then have to compute the sum:

) / ] ' [ 2 sin(
]) ' [ 2 sin(
) ' (
2
) ' ( 2
1
0
) ' (
2
N n n
n n
e
e
e S
N
n n
j
n n j
N k
k
N
n n k
j
= =

=
=
t
t
t
t
t

Now the value of S is:

S=N if n=n
S=0 otherwise (note: |n-n|<N)

Substituting this result into the equation for a(n) gives the required result:

) ( ) ( n a n a =

3. Typical Frequency Spectra and Their Characteristics

In this section we will illustrate the most typical spectra that an engineer might meet without considering
any of the technical problems that afflict measurement. These will be discussed later, sufficient unto the
day is the evil thereof.

Isolated Sine Wave

Sine Wave Amplitude A at Frequency fa Frequency Spectrum Plot fa=nf

f
a

A/2
Transducers are often calibrated using devices which output a known sinusoidal signal. For example, a
microphone may be calibrated by use of a piston-phone which produces a sound wave of known
frequency fa and known amplitude A. The microphone, and any associated amplifier, then produce a
sinusoidal voltage. Its amplitude is obtained by Fourier analysis thereby determining the overall
calibration factor. The plot illustrates the concept. The time signal consists of a single sine wave with
frequency fa and peak amplitude A. If the frequency fa corresponds precisely to a spectral line frequency
(fa=kf, k an integer) then the frequency spectrum plot shows it as an isolated spectral line with
amplitude A/2. The factor 1/2 arises from the NVH convention of showing the RMS (root mean square)
value of the amplitude.

If the frequency is not an exact integer multiple of f then the situation is more complicated since the
energy in the sine wave leaks into spectral lines close to fa. Techniques for dealing with this leakage
problem will be dealt with later.

Order Spectrum

An order spectrum represents a slightly more complicated version of sine wave analysis. NVH signals due
to operating equipment often consist of a set of sine waves whose frequencies are integer multiples of the
lowest frequency e.g. f1, f2=2f1, f3=3f1,... These forces are called order forces. An automobile engine, for
example, produces sinusoidal forces at frequencies which are multiples of the crankshaft rotation
frequency. Typically the order amplitudes (A1, A2, A3,.) decrease as the order increases. The time signal
plot below shows three order type sine waves with abitrary relative phases, and the signal that results
from adding these sine waves. The frequency spectrum plot indicates how the rms values of the orders are
to be plotted.

1
st
, 2
nd
& 3
rd
Order Sine Waves & Total Frequency Spectrum Plot (Fixed Speed)

Note that here we assume that the order frequencies are integer multiples of f that the order frequencies do
not change with time. Usually the order spectrum will be spread out by leakage and also by variations of
frequency with time. Also note that conventions vary with industry, for example tire manufacturers provide
peak order amplitudes rather than the usual NVH rms value.

It is a common practice to plot the spectra corresponding to multiple speeds on a single waterfall plot. A
simple example is shown
below.

A
m
p
l
i
t
u
d
e
Time
1st Order
2nd Order
3rd Order
Total Signal
f
1
f
2
f
3

RMS
Amplitude

RPM
f
Frequency hz
ARPM
A(f)
f
n
=nRPM/60

Here RPM indicates the rotation speed of the equipment and we show only one n-th order. Often the data
corresponds to a continually changing RPM and this smears the spectral line. More advanced methods
can reduce this problem but are beyond the scope of this document.

Isolated Impulse

Impulsive forces are forces of short duration w such as the vertical force generated by a tire impacting a
tar strip. Another example would be the force applied by an impact hammer in an impact test. The plot
below shows the typical dependence of force on time.

In this case the time dependence of the force can be approximated quite well as a half-cosine. Specifically,
if the impulse has a maximum level F0, exists for a time length w and is centered on t = t

, then we can
approximate the impulse as:

(

=
w
t
F t F
t
cos ) (
0
if 2 / 2 / w t w + < < t t and zero otherwise.

In contrast to the sine wave case we now have a continuous frequency spectrum, that is the amplitude is
appreciable at most frequencies, but with zeros near to f=3/(2w), f=5/(2w) The spectra of short
pulses of length ~w typically have zeroes at frequencies which are multiples of approximately 1/w .

Spectrum Shape for Cosine Pulse

In this case the results depend somewhat on the parameters of our digital processing technique. If we take
N equally spaced measurements of a data record of length T, and if N is very large, then we have:

( )
( )
t t
t
t
k
f j
k
k
k
e
w f
w f
T
w
F f F
2
2 0
2 1
cos 2
) (

(
(
)
`
)
`
~

|F(f)|
f=3/(2w)
f

w
t
F(t)
The complex exponential factor gives the relative phasing or time delay of the sine waves, and this part
of the expression corresponds to when the impact occurred. The zero Hz or DC value gives the scale of
the spectrum:

(
)
`
~ =
T
w
F f F
k
t
2
) 0 (
0

This shows that the scale of the spectrum depends on T, the larger T the lower the spectrum. Methods for
dealing with this issue will be treated later.

Repetitive Impulses Order Spectrum

Engine firing provides a good example of a repetitive impulse, in this case repetitive torque impulses. We
now have a situation which is combines order effects with impulse effects. Suppose the firing frequency is
f1. We have an impulsive force F(t) which is repeated with a frequency f1. In this case the signal can be
represented by sine waves at frequencies f1, f2=2f1, f3=3f1,.. The amplitude of the sine waves will vary like
the amplitude of the spectrum of the single pulse.

Repetitive Cosine Pulse in Time Domain Order Spectrum Shape

Random Signals

Road noise due to a rough road surface can be considered to be a random signal. The main characteristics
of this random signal are:

- Its amplitude distribution (distribution of a(tn)is Gaussian
- It does not repeat itself with some characteristic time period
- The correlation between the signal at a time t and at a later time t+Q decreases as Q increases

The graph below shows the general idea of a random signal.

Amplitude

Random Signal: Amplitude vs. Time

w
t
F(t)
|F(f)|
f=3/(2w)
f
1/f
1

1/f
1

f
1

This particular signal was generated by adding 50 sine waves at frequencies 5, 10, 15.250 hz with
random phases. The amplitudes were arranged to peak at 15 hz and then fall-off slowly with frequency.
The spectrum is continuous as shown below. The time domain amplitude distribution is roughly Gaussian
as is shown by the frequency histogram (amplitude =0 represents values from +1 to -1, etc.).

Random Signal: Frequency Spectrum Random Signal: Time Domain Histogram

In the real world random inputs often excite resonances so that the output spectrum has resonance peaks.
For example, the noise in the interior of an automobile resulting from random road inputs will show
resonance peaks as shown below.

Road Noise Spectrum

The amplitude of this type of frequency spectrum depends on the frequency resolution. As a result
comparisons between spectrum require that all spectra have the same resolution. In the case of random
spectra the level is subject to statistical fluctuations which decrease as the quantity of data is increased.

1.E-04
1.E-03
1.E-02
1.E-01
1.E+00
20 120 220 320 420 520 620 720 820 920
S
P
L
P
a

r
m
s
Frequency Hz
0
0.2
0.4
0.6
0.8
1
1.2
1.4
1.6
0 50 100 150 200 250
P
e
a
k

A
m
p
l
i
t
u
d
e
Frequency (Hz)
0
20
40
60
80
100
120
140
-10 -8 -6 -4 -2 0 2 4 6 8
N
u
m
b
e
r

o
f

M
e
a
s
u
r
e
m
e
n
t
s
Time Domain Amplitude +-1
4. Basic Problem#1 Aliasing

Digital processing involves using a finite sampling rate (fs). Nyquist and Shannon showed that if we use
a finite sampling rate it is not possible to correctly treat signals at frequencies above fs/2. This
theoretical maximum is called the Nyquist frequency:

2 2
s
Ny
f
f
N
f = A =

Sine waves above this frequency are aliased (mapped) to frequencies in the range 0 to 2 /
S
f . Figure
1.4.1 shows the effect graphically. Suppose we have a sine wave at frequency f1. To determine the
corresponding aliased frequency we:

- Determine the ratio ) 2 / /(
1 s
f f A=
- Find A on the horizontal axis and draw a line to the saw-tooth shaped mapping curve
- Draw a horizontal line to the aliased value B
- The aliased frequency is then given by ) 2 / (
1 s Aliased
f B f =

Figure 4.1 : Aliasing of Frequencies above the Nyquist Frequency (fs/2)

For the case shown in the figure the frequency f lies in the range fs/2 to fs . As a result the aliased
frequency fa is related to the actual frequency f by:

f f
f
f f f
s
s
a
= = )
2
( 2
;
s
s
f f
f
s s
2

To give a numerical example consider the following case:

0
0.1
0.2
0.3
0.4
0.5
0.6
0 0.5 1 1.5 2 2.5 3 3.5 4
A
l
i
a
s
e
d

F
r
e
q
u
e
n
c
y
/
(
f
s
/
2
)
Actual Frequency/(fs/2)
Aliasing
A
B
- the sampling frequency is fs=16 Hz
- the sine wave has a frequency f=11 Hz, specifically ) 2 sin( ) ( ft t a t =
- the aliased sine wave appears to be at fa=fs-f=5Hz , specifically ) 2 sin( ) ( t f t a
a
t =

The negative sign arises from the choice of frequencies and is not important to our argument.

Figure 1.4.2 shows an 11 Hz sine wave sampled at 16 Hz and the equivalent aliased 5 Hz sine wave (a
small constant offset has been given to the aliased sine wave for clarity).

Figure 4.2 : 11 Hz sine wave sampled at 16 Hz aliased to 5 hz sine wave

Practical Solution to Aliasing Problem

To avoid problems due to aliasing, analyzers must filter out the frequencies above fNy = fs/2 before Fourier
analysis is performed. This is usually done automatically, but the user should be aware that this filtering
procedure involves a number of issues.

The typical procedure has four steps:

1. Filter incoming analogue signal with an analogue anti-aliasing filter set at a frequency sufficiently
high to cover all practical applications. This could be as high as 33 kHz for NVH applications
(human hearing can reach almost 20 kHz. 33 kHz may be required because sampling rates are
usually powers of 2).

2. Digitize signal at twice the analogue anti-aliasing frequency i.e. ~ 2
16
= 65,536 Hz.

3. Apply a digital low pass anti-aliasing filter appropriate to user chosen fNy = fs/2

-1.5
-1
-0.5
0
0.5
1
1.5
0 0.25 0.5 0.75 1
A
m
p
l
i
t
u
d
e
Time (seconds)
"-sin(2pi5t)" sin(2pi11t)
4. Decimate digital signal at fs i.e. take only samples occurring at fs.
Analyzers normally operate on a binary system so we have:

fs= . 128 Hz, 256 Hz, 512 Hz, 1024 Hz,

Practical Issues

Anti-aliasing filter are usually designed to produce a 3 dB attenuation at
s Ny
f f 4 . 8 . = , and this is usually
considered the maximum useable frequency, fmax. Some analyzers show the range 0-.5fs, some show the
range 0-.4fs.

The diagram below shows the typical attenuation of an anti-aliasing filter. It can readily be understood
that any data above .8fNy is not accurate, and in fact some level of error occurs at lower frequencies. It
should also be noted that the anti-aliasing filter will modify phase angles as a function of frequency, but
this is seldom of importance.

Typical Attenuation of an Anti-aliasing Filter

Typical analyzers use an 8-pole Butterworth filter as an anti-aliasing filter. An n-pole Butterworth filter
gives an attenuation of R dB:

{ }
n
c
f f R
2
) / ( 1 log 10 + =

The attenuation is thus 3 dB at the critical frequency fc independent of the order of the filter. The higher
the order of the filter the faster the roll-off at high frequency.
-70
-60
-50
-40
-30
-20
-10
0
0 0.2 0.4 0.6 0.8 1
f/fs
A
t
t
e
n
u
a
t
i
o
n

(
d
B
)
fny
fmax=.8fny
3db
Anti-aliasing Attenuation
(8th order Butterworth)
5. Basic Problem #2: Digitizing, Overloads and Underloads

After analogue anti-alias filtering, the signal is digitally sampled by an ADC (analog to digital converter)
at a fixed rate that depends on the anti-aliasing strategy. The ADC is designed to take a voltage in the
range Vmax and split it into a large number (M) of discrete values separated by the amplitude resolution
AV = 2Vmax/M . An adjustable gain G is applied before digitizing.

Overloads

If the gain G is set too high the ADC will be overloaded and the signal will be truncated when its
magnitude exceeds Vmax. The figure below illustrates this problem for a sine wave that overloads the
ADC.

ADC Overload Truncated Sine Wave

It is not possible to completely avoid overloads for random signals. Random noise with an RMS value
of Vrms = o and a DC value of zero will have a probability density:

|
|
.
|
\
|
=
2
2
2
exp
2
1
) (
o t o
V
V P

The signal will be outside the range 2o about 4% of the time, and outside the range 3o about 3
times in a 1000. If we wished to have only one overload per 100 blocks of 1000 data points i.e. one
overload in 10,000 data samples, we would need to set the gain so that 4o=Vmax,

Random Signal Amplitude Probability Density P(V) vs. V/

The above plot shows a Gaussian P(V) plotted against V/, in effect as though =1.
-2
-1.5
-1
-0.5
0
0.5
1
1.5
2
V
/
V
m
a
x
Time (seconds)
Original Signal
Truncated Signal
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
-5 -4 -3 -2 -1 0 1 2 3 4 5
P
r
o
b
a
b
i
l
i
t
y

D
e
n
s
i
t
y

P
(
V
)
V/
Underloads

ADCs are usually classified by their bits Nb, the voltage range being split into M steps with:

b b
N N
M 2 1 2 ~ =

Nowadays it is common to use 16 bit ADCs with M=65,535.

The amplitude resolution of the ADC is AV:

1 2
2 2
max max
= = A
b
N
V
M
V
V

The accuracy with which the signal is measured clearly depends on the ratio of the amplitude of the
signal to AV, so the gain should be set as high as possible without producing unacceptable overloads.

The dynamic range (DR) of the ADC is formally defined as:

( ) ( )
b b
N N M
V
V
DR 02 . 6 2 log 20 log 20
2
log 20
max
~ ~ = |
.
|
\
|
A
=

The dynamic range gives the maximum difference in levels that can be accurately measured in a
frequency spectrum. In practice the range will be at least 10 dB smaller due to various factors such as
the safety factor required to avoid overload. The dynamic range is given in the table below for various
values of Nb.

Table 5. Dynamic Range (DR) vs. ADC Bits (Nb)

Nb 12 14 16 18 20
DR (dB) 72 84 96 108 120

NVH phenomena often have a frequency spectrum that falls off rapidly with frequency, for example
vehicle interior sound pressure may be roughly inversely proportional to the square of frequency (Af=1
hz). This implies a huge difference in dB level between 10 and 1000 hz:

2
100
) 100 ( ) (
|
|
.
|
\
|
~
f
P f P
rms rms
; 10<f<1000 hz

|
.
|
\
|
~
100
log 40 ) 100 ( ) (
f
SPL f SPL
dB

Consequently there is an 80 dB difference in sound pressure level between 10 hz and 1000 hz. This
implies that a 14 bit ADC barely has sufficient dynamic range to analyze a sound spectrum over this
range.

Apart from using a 16 bit ADC ,there are two other possibilities. In the case of sound measurements the
initial signal could be weighted with an analogue A-weighting filter which would suppress low
frequencies. An alternate is to use a method by which the frequency resolution is broadened at high
frequencies. This will be discussed later.

6. Basic Problem#3 Leakage

Up to now we have implicitly assumed that the signal is periodic in time T. If this is not the case then
some type of error will occur. The error is most obvious with sine and cosine waves. First consider a
non-leaky signal, e.g. a cosine wave with f=6Af and N=32 as shown in the figure 6.1

Figure 6.1 Cosine wave with f=6Af and N=32

The frequency spectrum consists of a single spike with amplitude 1/2 (the RMS level).

Figure 6.2 FFT of Cosine Signal

On the other hand, if we have a cosine wave with f=6.5Af and N=32, then the amplitude at the
beginning and end of the block is different as shown in figure 1.6.3:

Figure 6.3 : Cosine Wave with f=6.5Af and N=32

-1.5
-1
-0.5
0
0.5
1
1.5
0 4 8 12 16 20 24 28 32
A
m
p
l
i
t
u
d
e
Time in units of delta T
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Frequency/Delta F
A
m
p
l
i
t
u
d
e

(
M
a
g
n
i
t
u
d
e
)
-1.5
-1
-0.5
0
0.5
1
1.5
0 4 8 12 16 20 24 28 32
Time/delta T
a
(
t
)
The DFT does not have a spectral line at 6.5Af, so the energy must leak to other frequencies such as 6Af
and 7Af . This is shown in the figure below:

Figure 6.4 : Leakage of Signal at 6.5 Af , Blocksize N=32

Instead of a single line the spectrum is a line shape. This can make it difficult to compare the
amplitudes of sine waves with different frequencies. Note that leakage must occur whenever the signal
has:

(a) A different amplitude at the beginning and end of the block.
(b) A different slope at the beginning and end of the block.

The solution is to apply a window to the time series such that the amplitude is zero at the end of the
time record. This will then minimize the difference between leaky and non leaky cases. Naturally there
is a price to be paid. The details depend on the choice of window.
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
0.5
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Frequency/Delta F
A
m
p
l
i
t
u
d
e

(
M
a
g
n
i
t
u
d
e
)
7. Hann Window

The Hann window is the most commonly used window and we will use it to illustrate the general
concept. The window is called a Hann window because it was invented by the Austrian meteorologist
Julius von Hann. It is often called the Hanning window, but should not be confused with the Hamming
window which was invented by R. W. Hamming. The Hamming window is not used in NVH. The
standard definition of the Hann window used here is from Frederic J. Harriss article on windows in the
Proceedings of the IEEE (Vol. 66, No. 1, January 1978, page 60). Unfortunately some software packages
use slightly different non-standard definitions.

Window Shape

The Hann window is a function of time as shown below, with amplitude scaling. The signal is
multiplied by the window for the time interval T before Fourier analysis. This is demonstrated below for
a sine wave:

The result is zero at the ends of the time window and leakage variation very much reduced. The window is
defined mathematically as:

)
`
|
.
|
\
| A
=
)
`
|
.
|
\
| A
=
T
t n K
T
t n
K t w
n
t t 2 cos 1
2
sin ) (
2

Hann Window Amplitude Scaled
0
0.5
1
1.5
2
2.5
Time
A
m
p
l
i
t
u
d
e
T
Signal
-1.5
-1
-0.5
0
0.5
1
1.5
Time
A
m
p
l
i
t
u
d
e
Signal x Window
-2
-1.5
-1
-0.5
0
0.5
1
1.5
2
2.5
Time
A
m
p
l
i
t
u
d
e
This definition applies to a block as usually defined, that is { N n < s 0 }.

Leakage Peak Amplitude Variation

For a sine wave with fixed amplitude the variation in peak value of the Fourier transform amplitude as the
frequency varies from nf to (n+1/2)f is 15% or 1.4 dB. Without a window (rectangular window) the
variation would be 4 dB.

Effective Noise Bandwidth (Line Width)

The standard measure of the line width corresponding to a sine wave is the Effective Noise Bandwidth
(ENBW). For a Hann window ENBW=3f/2. Without a window (rectangular window) the width is f (see
figure 6.4).

Amplitude vs Energy Scaling

If we wish that the peak amplitude corresponding to a sine wave should give the RMS level of the sine
wave then we must set:

K(amplitude) = 2

If we wish to have the correct RMS level of a broad spectrum then we must set:

K(energy) = 1.63

Analyzers usually make automatic adjustments to the scaling factor according to whether the engineer
uses a single cursor (amplitude) or a double cursor with a request for rms level.

Engineers should be aware that data downloaded to spreadsheets may have either scaling.

8. Other Windows

Rectangular Window

The rectangular (box-car) window is used where leakage is not a problem. The shape is shown below.

Shape of Rectangular Window .

The mathematical formula for the window is:

w(tn)=1 0n<N

Rectangular Window
0
0.5
1
1.5
Time
A
m
p
l
i
t
u
d
e
T
Flat-top Window

The flat-top window is used for determining the amplitude of well separated sine waves. It is particularly
useful for calibration with a sine wave input. The shape is shown below.

Shape of Flat-top Window Compared to Hann

Mathematical Formula:

There is no clear standard for this window but the following formula appears to represent the commonly
used version.

) 8 cos( 0322 . ) 6 cos( * 388 . ) 4 cos( * 29 . 1 ) 2 cos( 93 . 1 1 ) (
T
t
T
t
T
t
T
t
t w
n n n n
n
t t t t + + =

Force and Exponential Windows

These specialized windows are only used for impact testing and will be treated separately.

9. Comparison of Window Parameters

The table below compares the main window parameters. Note energy scaling is not applicable to the flat-
top because it is only used for finding the amplitude of isolated sine waves.

Comparison Of Window Parameters
Rectangular Flat-Top Hann
Leakage-Peak Amplitude Variation 4 dB .016 dB 1.4 dB
Effective Noise Bandwidth (units of f) 1 3.77 1.5
Amplitude Scaling K= 1 1 2
Energy Scaling K= 1 N/A 1.63

10. Averaging and the Power Spectrum

It is common that the frequency spectrum is averaged over Q time records. A simple linear average
usually results in an almost zero answer because the phase angles vary randomly from record to record.
That being the case it is usual to determine the average of the square of the amplitude. This is known as
the power spectrum and we indicated it symbolically as > - < f A f A ( ) ( . It is defined by:

Flat-Top vs Hann (Amplitude Scaled)
Time
A
m
p
l
i
t
u
d
e
Flat-Top
Hann

=
=
=
=
= >= - <
Q q
q
Cq
Q q
q
Cq Cq
f A
Q
f A f A
Q
f A f A
1
2
1
| ) ( |
1
* ) ( ) (
1
) ( ) (

Here <> indicates an average, the index q refers to the time record (length T), and the ACs refer to
complex amplitudes. The symbol - indicates multiplication by the complex conjugate as per the equation.

Analyzers will display either the power spectrum or its square root which is usually referred to as the
amplitude (strictly speaking, rms amplitude). They will also display the power spectrum in db format:

) ( 10 ) ) ( ) ( ( 10
) ( ) (
10 K Log f A f A Log
K
f A f A
Log dB > - < =
(
> - <
=

The value of K depends on the quantity measured. It usually has the value K=1, but in the case of sound
pressure
4 . 9
10 = K , so the expression for sound pressure level (SPL) is given by:

94 ) ) ( ) ( ( 10 > - < = f A f A Log SPL

11. Overlap Processing

Data collection can usually be arranged so that the time records overlap. This is particularly advantageous
when processing random data such as road noise. In that case it is best to use a Hann window with a 50%
overlap. This tends to average out the effect of the Hann window, that is all data gets roughly the same
weighting.

Overlapping Records

12. Statistical Fluctuations

The power spectrum of a random signal will have statistical fluctuations. These fluctuations can be
reduced by increasing the number of statistically independent time records. If Q is the number of
independent records, then the fluctuations will be of order :

Q
f A f A > - <
~ A
) ( ) (

T
T
T
T
Overlap
Thus it pays to take a large number of independent averages to reduce statistical fluctuations. Also one
should note that if the records overlap by 50% then the number of independent records is the total
number of records.

13. Effect of Frequency Resolution Power Spectral Density (PSD) 3
rd
Octaves

The power spectral density (PSD) is defined in terms of the power spectrum as:

f
f A f A
f PSD
A
> - <
=
) ( ) (
) (

In effect we have divided the Fourier coefficients by .

Why bother?

Effect of Frequency Resolution on a Continuous Spectrum

Suppose we have band limited random noise with a continuous and more or less flat distribution. If we
change f to f/4 then we have four times as many spectral lines in the band. So from Parsevals theorem
the auto-power spectrum level must be reduced by a factor of 4, i.e. 6dB, as shown in figure 13.1 (the
Fourier coefficients are reduced by a factor of 2 or 6dB).

Effect of Frequency Resolution on the Power Spectrum of a Continuous Spectrum

If we used the power spectral density then the level would be unaffected by the change in frequency
resolution, and for this reason the PSD is often used to represent continuous spectra. In effect the PSD
gives the power spectrum level with a 1 Hz resolution.

Effect of Frequency Spectrum on a Non-Leaky Line Spectrum

If all the energy is concentrated at the frequency which is a multiple of f (no leakage) then changing the
frequency resolution has no effect on the power spectrum level (this assumes the usual convention that
f=2
p
, p an integer which may be positive, negative or zero). On the other hand the PSD level will change if
the frequency resolution is changed.

Power Spectrum
Frequency
6 dB
Bandwidth
Spectrum Level Resolution f
Spectrum Level Resolution f/4
General Case

The typical spectrum is liable to be a mixture of the above cases and the engineer must use judgment as to
whether the power spectrum or PSD gives the best representation of the spectrum.

Third Octave Spectra

Before digital signal processing was available it was common to analyze acoustic spectra with analogue
band-pass filters. The 3
rd
octave filters became the most popular because their output could be converted
fairly easily to loudness measures and a standard set was developed. The table below gives the nominal
frequency ranges for the preferred 3
rd
octave bands. Frequencies outside the range shown are found by
scaling frequencies inside the range by factors of 10.

Table: Preferred 3
rd
Octave Bands

Lower Limit 28 36 45 56 71 89 112 141 178 224 280 360 450
Band Center 31.5 40 50 63 80 100 125 160 200 250 315 400 500
Upper Limit 36 45 56 71 89 112 141 178 224 280 360 450 560

Third octave bands increase in width with frequency and are particularly useful for spectra that fall
rapidly with frequency e.g. the road noise spectra shown earlier.

To simulate the analogue procedure we introduce weighting functions W(k,i) which simulate the
weighting effect og the i-th filter at frequency fk. The power in each 1/3rd octave P(1/3,i) is then:

> - < E = ) ( ) ( ) , ( ) , 3 / 1 (
k k k
f A f A i k W i P

We should remark that it cannot be guaranteed that the analyzers of different manufacturers use exactly
the same Ws. The original frequency resolution may also have a slight effect on results.

14. Cross-Spectra

If we want to determine the relationship of two signals A and B we must first calculate the cross-
spectrum:

=
=
>= - <
Q q
q
Cq Cq
f B f A
Q
f B f A
1
* ) ( ) (
1
) ( ) (

Using purely real notation we find that the real and imaginary parts of the cross-spectrum are:

)] ( ) ( cos[ | ) ( || ) ( |
1
} ) ( ) ( Re{
1
f f f B f A
Q
f B f A
qB qA q q
Q q
q
| | = > - <

=
=

)] ( ) ( [ sin | ) ( || ) ( |
1
} ) ( ) ( Im{
1
f f s f B f A
Q
f B f A
qB qA q q
Q q
q
| | = > - <

=
=

From this we can obtain the magnitude and phase of the CPS

2 2
}) ) ( ) ( (Im{ }) ) ( ) ( (Re{ | ) ( ) ( | > - < + > - < = > - < f B f A f B f A f B f A

{ }
{ }
|
|
.
|
\
|
> - <
> - <
=
) ( ) ( Re
) ( ) ( Im
arctan ) (
f B f A
f B f A
f _

15. Coherence

The coherence
2
AB
indicates the degree of correlation between signals A and B:

> - >< - <
> - <
=
) ( ) ( ) ( ) (
) ( ) (
) (
2
,
2
f B f B f A f A
f B f A
f
C C C C
C C
B A

If the signals are perfectly correlated then:

) ( ) ( ) ( f B f K f A =

Here K is a constant which may be complex and which depends only on frequency. The symbol
indicates complex multiplication.

In the case of perfect correlation the coherence will be unity:

1 ) ( ,
2
= f B A

If the signals are totally uncorrelated the coherence is zero.

In general:

1 ) ( 0 ,
2
s s f B A

16. Transmissibility and Transfer Functions

Suppose the signals are correlated via:

) ( ) ( ) ( f B f K f A =

In this case we might like to know the value of K. We can do this by noting that:

> - < >= - < ) ( ) ( ) ( ) ( ) ( f B f B f K f B f A

From this we find that:

> - <
> - <
=
) ( ) (
) ( ) (
) (
f B f B
f B f A
f K

Transfer Functions

If B is a force and A is a response, such as acceleration or sound pressure, then K is called a transfer
function.

Transmissibility

If A and B are similar signals, but A is in some sense due to B, then K is a transmissibility. For example, if B
is sound in the engine compartment, and A is sound in the vehicle interior, then K would be an acoustic
transmissibility.

Coherence

We have assumed that A and B are linearly related. We can check this via the coherence. Ideally this
should have the value unity. More complicated cases will be reviewed in the section on coherence
methods.

Alternate Calculations

In some cases it may be better to calculate K via:

> - < >= - < ) ( ) ( ) ( ) ( ) ( f B f A f K f A f A

> - <
> - <
=
) ( ) (
) ( ) (
) (
f B f A
f A f A
f K

In most cases the input power-spectrum is more uniform than the output power-spectrum which may
have zeroes. Therefore the first procedure is usually preferred.

If the input spectrum is low or has zeroes then considerable expertise may be required. It may be that
there is a serious problem with the experimental set-up.

16. Strategy and Checklist

This section outlines a strategy for accurate frequency domain measurements. It is in the form of a
checklist, but the engineer should recognize that some iteration may be required, and that if one
parameter is changed it may require the modification of other parameters.

1. Plan

If you fail to plan then you plan to fail.

Develop a brief plan outlining key parameters and issues:

a. Type of signal: sinusoidal, sweeping sinusoidal, random, impulsive,.
b. Type of analysis: power spectrum, transfer function,
c. Type of Fourier transform used e.g. single sided rms
d. Maximum expected frequency
e. Expected level DC problems?
f. Required frequency resolution
g. Leakage issues - type of window
h. Averaging? Type? How many averages? Statistical fluctuations a problem?
i. Data quality coherence

These issues are expanded below.

2. Type of Signal (Section 3.)

What type of signal is to be measured:

- sine wave or multi-order signal?
- transient impulse or repetitive impulse?
- random?

3. Frequency Range and Aliasing (section 4.)

What is the maximum frequency range required for accurate measurement of the frequency spectrum
(power spectrum) or transfer function?

Set up the analyzer so that the required anti-aliasing does not affect results.

It may be advisable to perform trial studies with a wider than expected frequency range so that
important effects just above the expected range are not missed.

4. Front End Parameters (section 5)

DC-Offsets: use AC coupling if DC offsets are a problem.

Over & Under-loads: adjust the gain to avoid under-loads and overloads.

Decide on free-run (e.g. for random signals) or triggered data acquisition (e.g. for transient signals).

5. Frequency Resolution

Choose a frequency resolution sufficient to show required detail in frequency spectrum or transfer
function. Note the effect of frequency resolution on the length of the time record.

6. Leakage Window (sections 6, 7, 8 and 9)

Given the input signal and frequency resolution, will leakage be a problem?

Choose a suitable window.

7. Averaging (section 10)

What type of averaging should be employed synchronous or non-sysnchronous?
Are statistical fluctuations a problem?
How many data records must be averaged?
Should data blocks be overlapped and by how much?
What is the total time required?

8. Spectrum Normalization Fourier and Window Conventions (sections 16, 2, 7 and appendix B)

Is the absolute level of the spectrum important?
Which is more appropriate, power spectrum or power spectral density?
Should the data be presented in 1/3
rd
octave format?

Do we want the RMS spectrum or some other convention (section 2 and appendix B)?
How should any window be normalized (section 7)?

9. Transfer Functions or Transmissibility (sections 16, 15, 14)

Is the coherence satisfactory?
Does the input spectrum have pronounced minima which could cause problems?

Appendix: Alternative Conventions for Fourier Coefficients

The conventions used here are those that are employed by analyzers used in the automotive industry
the NVH convention. Other conventions are sometimes used, for example the Fourier coefficients may
be normalized so that the transform equations are symmetric between between the time and frequency
domains. Also NVH analyzers sometimes give the user a variety of choices that can lead to confusion.
The most important choices are outlined below:

Single Sided RMS

This is the standard noise and vibration option where the coefficients represent the rms
amplitudes of the sine and cosine waves (except DC and Nyquist frequency terms) and these are
the values the software displays to the user. (But note that it is typical that software
documentation uses the complex expansion defined without the sqrt(2) factors).

Double Sided RMS

The coefficients are reduced by a factor of 1/sqrt(2)

Single Sided Peak

Here the coefficients represent the peak amplitude of the sine and cosine waves, so the
coefficients are sqrt(2) times the single sided rms coefficients

Double Sided Peak

The coefficients will be the same as for the single sided rms.

Warning Note

When windows are applied, the analyzer software will make corrections that can be optimized for the
line spectra of repetitive signals or for the continuous spectra of random signals. Analyzers frequently
change the correction depending on whether a single cursor is used to find the amplitude at one
frequency, or a double cursor is used to find the rms over a frequency range. This can lead to confusion,
particularly when the signal is downloaded to a spreadsheet so that only a single correction factor can be
applied.

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