Vibration Principles

An Introduction to Spectrum Analysis

Summary
One of the most important tools used in the investigation of bearing vibration is
frequency spectrum analysis. Therefore, it is important that technicians and managers
understand the possibilities and limitations of this technique. This document explains in
detail, displacement, velocity, and acceleration, and the methods used to obtain these
measurements. Also included is the derivation of a frequency spectrum by Fourier
transform, the consequences of sampling, and the value found in finite measurements.

SKF Reliability Systems JMO2007

@ptitude Exchange SKF Employee
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United States Published 2002
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Vibration Principles
Table of contents

1. Introduction ............................................................................ 3
2. What is a Function?................................................................. 3
3. What is a Sine Wave? .............................................................. 4
4. Physical Domains .................................................................... 4
4.1. Displacement ....................................................................................... 5
4.2. Velocity ............................................................................................... 5
4.3. Acceleration......................................................................................... 9
5. Vibration ............................................................................... 10
5.1. LVDTs ................................................................................................10
5.2. Moving Coils .......................................................................................11
5.3. Piezoelectric Transducers......................................................................11
6. Why Spectra? ........................................................................ 12
7. The Fourier Theorem ............................................................. 15
8. The Fourier Transformation................................................... 17
9. The Shannon Sample Theorem .............................................. 21
10. Time Windowing ................................................................... 24
11. Transfer Function .................................................................. 27
12. Typical Vibration Spectrum ................................................... 32

Vibration Principles © 2011 SKF Group 2 (35)

 The most common graph is the Cartesian
1. Introduction grid. Two axes are drawn at right angles.
The vertical axis is called ordinate (y), and
One of the most important tools used in the
the horizontal axis is the abscissa (x). In
investigation of bearing vibration is
general, mathematicians use ‘y’ for the
frequency spectrum analysis. Therefore, it is
ordinate and ‘x’ for the abscissa (Figure 1).
important that technicians and managers
understand the possibilities and limitations
of this technique. This document explains in
detail, displacement, velocity, and
acceleration, and the methods used to
obtain these measurements. Also included is
the derivation of a frequency spectrum by
Fourier transform, the consequences of
sampling, and the value found in finite
measurements.

Figure 1. A function Represented in a Cartesian grid.

Understanding the methods and processes
behind the use of a particular tool can
greatly increase the effectiveness of that We can say 'y' is a function of 'x' and write it
tool. This is especially the case of spectrum as:
analysis, where the 'mystic flair' is
transformed into a well-understood, simple y = f (x)
process. The ability to clearly convey how (1)
this process is achieved lends a great
amount of respect to person and company. For each point of 'x' for which the function
exists, a corresponding point on 'y' can be
Spectrum analysis of displacement, velocity, found; thus, 'y' is a projection of 'x'.
and acceleration measurements allows one
to better understand what is occurring in the One of the simplest mathematical functions
system. These three measurement types are is a straight line:
the physical domains in which we can
describe static and dynamic situations. y=kx+d
Therefore, a basic understanding of (2)
measurement parameters is necessary. Here
we begin with the basic terminology.

2. What is a Function?
When one thing is dependant upon another,
and there is a correlation between the two
items, the relationship can possibly be
expressed in a graph or formula. This
graphical form is referred to as a function.
Figure 2. Straight Line as a Function.

© 2011 SKF Reliability Systems All Rights Reserved 3

In Figure 2, 'd' represents the initial
displacement on the ordinate and 'k'
indicates the slope of the line. When 'k'
equals one and 'd' is zero, we get a 45-
degree line through the zero-point of the
grid (the scale of the axes is must be the
same). Equation (2) describes a straight line
exactly in an infinite two-dimensional space.

Figure 3. Generation of a Sine Wave.

3. What is a Sine Wave? Let’s assume we have a radius 'r' rotating
counter-clockwise (Figure 3). Parallel light-
A sine wave is a special function of
rays come from the left side and generate a
tremendous importance in the investigation
bright and a dark field at the projection
of vibration.
surface 'y'. The borderline between dark and
bright moves up and down according to the
According to equation (3), a sine wave can
rotation of the radius. The current position
be written as a function:
of the borderline depends on the current
angle of the radius; therefore, it is a
y = A sin (x)
function of the angle.
(3)
On the abscissa we can draw the angles and
their corresponding borderline between
The projection on our ordinate depends on a
bright and dark. The curve from this is a
constant factor 'A' (amplitude), which is
sine wave.
multiplied by the sine-function, where 'x' is
an argument indicated by brackets. Any
In principle, a cosine wave is the same, with
calculator, table, or slide rule gives us the
its phase shifted 90 degrees. At the zero-
right value when we put in the argument 'x'.
point of the coordinates the cosine-wave
But, what is the foundation behind this?
starts at the positive maximum.
You may have seen ‘π’ in connection with
It becomes obvious, therefore, that a
sine-functions. Remember, the
function so strongly connected to rotation is
circumference of a circle with diameter one
important for describing bearing behavior.
equals π. When we describe a circle by its
radius we get 2π as the circumference when
the radius is one. This example indicates
4. Physical Domains
that a sine wave is somehow connected to a
circle. Actually, a sine wave is a projection of There are different types of transducer that
a point on a circle, as it is a function over can be used for the three physical domains
the angle. of displacement, velocity, and acceleration.
Based on their specific principle of operation,
their response is proportional to the physical
domain (Figure 4).

© 2011 SKF Reliability Systems All Rights Reserved 4


Figure 4. Three Major Types of Transducers.
Deciding which sensor principle to use
depends on the performance of the
measuring principles in the existing
frequency range, and the parameters you
are seeking.
4.1. Displacement
According to Figure 5, displacement tells us
the distance between two points, measured
in millimeter or inches.
The words distance and displacement are, in
principle, describing the same type
measurement; however, ‘distance’ describes
static situations like diameters or lengths
between two points.
© 2011 SKF Reliability Systems All Rights Reserved
Figure 5. Example of Displacement (Y) and Distance
(X).
The word ‘displacement’ describes the
deviations around a middle value, measured
in a unit of distance. The most common
displacement units when judging the quality
of bearing components are:
μm or μinch
4.2. Velocity
When we ask how fast somebody drove from
Madrid to Gothenburg, we divide the
distance 's' between the two towns, by the
time 't' the driver traveled. The result of this
division is an indication of the speed υ:
s
υ=
t
(4)
To normalize the result for comparison
reasons, we agree upon measurement units.
When the speedometer in a European car
5
displays 100 km per hour, a European Based upon the rules of calculus we can
citizen realizes the meaning according to the change equation (5) to:
map. An American citizen expresses the
value in miles per hour to understand how s
long it takes to arrive at the destination. υ=
t
Normally, neither drivers recognize the
(6)
importance of speed in relation to
unexpected events. Apart from speed limits,
Δ is now written as a ‘∂’, thus the equation
nobody drives 100 km/h in an urban area
is ‘∂s over ∂t’. Another way to express it is
where children are playing because each
to say, "distance s is differentiated over time
second his car moves 28 meters. This is an
t.”
example of choosing units to transmit the
necessary message.
When we referred to Figure 3, we let the
radius rotate to explain how a sine wave is
The result of the equation (4) is an average
generated. Now we can also ask how fast
value that does not tell us the speed in
our radius is rotating. As we did for
different parts of the route. To reach that
distances, we can define an angular speed
information we can divide the route into
by dividing a certain distance by a special
different parts indicated by delta (Δ). Thus,
time-unit.
the distance between Madrid and Toulouse is
Δs, and the driving time: Δt. This results in
After one rotation the same situations
a new velocity:
repeats itself, which allows us to choose one
rotation as a distance-unit. Since we are
s looking for angular speed, we can make life
υ=
t easier by choosing the radius-unit one.
(5) Therefore, our distance-unit becomes 2. The
symbol ω (omega) is used to describe
Since this is still an average velocity, we do angular speed:
not know if the driver exceeded the
maximum speed. We can solve this problem 2
by dividing the route into smaller parts. ω=
T
However, how small should Δs be? Taking
(7)
into account that the car cannot accelerate
infinitely fast, we find some suitable values
The time-unit 'T' is called periodic time. If
for Δs, but we will never be exact. When we
the periodic time is one second, we have
want to know the speed of the car at each
one rotation per second. In the case of 0.5
point of the route exactly, we have to
seconds periodic time, we have two
reduce Δs to an infinitely small distance.
rotations per second (one rotation is
This means that the route consists of an
repeated twice in our time-unit). We can
infinite number of different parts,
also say our rotation is twice as frequent as
independent of the length of the route.
our first example, which brings us to the
word ‘frequency’ (f). Frequency is measured
Mathematically we have to cross the border
in Hertz (Hz). From our example we can
to the so-called “infinitesimal calculus,”
derive the relation between periodic time
which sounds more complex than it is.
and frequency:

© 2011 SKF Reliability Systems All Rights Reserved 6


1 circle, in part or multiple.
f=
T
(8)
The equation for ω can now be written as:
ω=2πf
(9)
When we multiply ‘ω’ (angular speed) by
time ‘t’, we come back to angle ‘φ’.
Mathematically, T and t cancel each other
out:
2
φ=ωt= t=2π
T driver stands still.
(10)
when t = T
In equation (3) we expressed 'x' as an
abstract argument for our sine wave. In
Figure 3 the abscissa is also called 'x', but
scaled in ‘φ’. According to our example we
can now be more precise and name the
abscissa as an angle. Time is always an
important parameter when studying
technical problems. To emphasize this, the
abscissa’s name changes from ‘φ’ to ‘ωt’.
Figure 6. Representation of Replacing ωt at the
Abscissa to Arrive at Equation (11).
Equation (3) can now be written as:
y = A sin (ωt)
(11)
© 2011 SKF Reliability Systems All Rights Reserved
The amplitude 'A' can be the radius of our
Now we have the basics to analyze the
consequences of the differential equation
(6).
There are some special rules of differential
equations that are not be covered in this
discussion. Instead, we focus only on the
simplest ones.
In simple terms, the first derivation of a
function tells us the slope of the function at
each point. From equation (6), it is obvious
that velocity is zero when there is no
deviation ‘n’, distance, which means the
Refer to equation (2) and Figure 2, where
we abstractly defined the function of a
straight line. We can apply this straight line
to our driver when we assume he drove for
a time at a constant speed. Then we can
name the abscissa time and scale it in
hours. On the ordinate, draw the meter
indication (km) of the odometer in the car.
The constant 'd' in equation (2) becomes the
meter indication at the beginning of our
observation (km (start)). We can now write
equation (2) applied to this example as:
km = kt + km (start)
(12)
In equation (12) 'k' is the most important
factor when looking for the speed of a car.
When 'k' is zero, 'km' does not increase,
which means the car stands still. The greater
'k', the faster our meter counts, and the
faster the car. Thus, factor 'k' is the speed of
the car.
In equation (6) we saw that velocity is
derived from displacement, or distance over
7
time. Using the common parameters, s = A sin (ωt)
equation (12) appears as: (17)

s = kt + s0 Which vertical velocity ‘v’ do we get?

(13)
Assume we have a sinusoidal road and we
s0 is the constant factor and indicates the drive on this road with constant horizontal
position at the beginning of observation. speed as shown in Figure 7. At the top and
the bottom of our road we do no have any
Based on our example we know that: vertical velocity. The maximum value of
vertical velocity is at the sinusoidal road’s
s  points of highest slope. These points are
υ= = (k t + s0) = k obviously where the sine wave of the road
t t
crosses the abscissa.
(14)

The constant factor s0 no longer exists. The

factor multiplying the variable 't' (to which
we apply the derivational process) is our
desired result.

To make life easier there is an international

convention: derivations over time 't' can be
expressed by an apostrophe after, or a point
over, the variable in question:
Figure 7. Representative of Displacement (Black),
s and Vertical Velocity (Red).
υ= = s’
t
(15)
When we also analyze the situation outside
the extreme points, we find the wave for
Following the same rules, angular speed ‘ω’
vertical velocity (Figure 7). Once again, this
can be expressed as the derivation of the
is a sine wave shifted through 90 degrees.
angle (φ or ωt) over time, written as φ’ or
As previously mentioned, we call such a
∂φ/∂t:
function a cosine wave.

 Now that we know the derivation of a sine

φ’ = (ωt) = ω
t
(16)
The factor at the variable where we apply
the derivational process is the proper result.
But, what happens if our displacement
follows a sinusoidal function? Let’s replace
the abstract name 'y' in equation (11) by
displacement 's':
wave is a cosine-wave, we have found a
general rule. But, before we derive equation
(17) we must take into account that the
argument of the sine wave is ωt, which itself
has to be derived. This is called the inner
derivation. We saw previously that the result
is ω. Based on the fact that the function is
linear, we can combine the results of the
inner and the outer derivation via
multiplication, and come to the desired
result for the vertical velocity:

© 2011 SKF Reliability Systems All Rights Reserved 8

 s
υ= = ω A cos (ωt) (18)
t
The result of the inner derivation ω,
becomes a multiplication-factor. This is very
important for the relation between
displacement and velocity. Since ω equals
2πf, the frequency is involved in the answer.
It can be shown that the higher the
frequency of the sinusoidal road, the higher
the resulting vertical velocity.
This fact can easily be demonstrated with an
experiment. Assume two wagons are pulled
with the same horizontal speed over two
types of roads (Figure 8).
Figure 8. Example of Two Types of Velocities.
Though the road with the higher frequency
has lower amplitude, passengers pulled over
this road have much more trouble.
This example shows that frequency has a
tremendous importance for responses in the
different physical domains. For example,
waves on rings or rollers may cause big
quality problems in the application, even if
the displacement proportional amplitude is
below the diameter of an atom!
© 2011 SKF Reliability Systems All Rights Reserved
The most common units in measuring
velocity to judge bearing quality are:
μm/s or μinch/s
4.3. Acceleration
In principle we have the same relation
between acceleration and velocity as
between velocity and displacement.
Acceleration answers the question: how fast
does velocity change? Therefore, it describes
the slope of the velocity function. The same
rules we found when discussing velocity can
be applied to acceleration. To derive
acceleration from velocity the following
equation has to be used:

a= (19)
b
Since the velocity ‘v’ is derived from
displacement 's', acceleration is the second
order differentiation of displacement. This is
conventionally indicated in a formula as:
2s
a=
t 2
(20)
If we assume we have a velocity following a
sine wave function, we just replace ‘v’ by ‘a’,
and ‘s’ by ‘v’ in equation (18) to derive the
acceleration. The velocity is multiplied by ω,
and once again we have the frequency
dependence in the acceleration proportional
response. The higher the frequency, the
higher the acceleration even at the same
velocity level.
When we want to derive acceleration from
displacement, we apply the derivational
process to equation (18). From the inner
derivation of ωt, we get additional
9
multiplication with ω and end up with ω2.
Following the same rules we used in Figure 7
to derive velocity from displacement, we find
the slope of a cosine wave as an inverse
sine wave. Therefore, we arrive at:

2s
a= = -ω A sin (ωt)
t 2
(21)

Acceleration is very important for dynamic

mechanics because force is connected with
acceleration in the Newton equation:

F=ma
(22)

In this equation 'm' indicates mass, which is

normally measured in kg.

The most common acceleration units of Figure 9. The Principle Properties of Sensors in the
Three Physical Domains.
measurement are:

m/s2 or inch/ s2 The following explanation only gives an

impression of the principle of operation. For
more details please refer to basic literature
5. Vibration about measuring techniques. However, it is
not necessary to understand sensors in
We just discussed the three physical
detail to achieve good results.
domains in which we can measure, and the
difference between distance and
displacement. 5.1. LVDTs

Linear Variable Displacement Transducers

The deviation surrounding a middle value is
(LVDTs) are common for measuring
typical of vibration, regardless of the
displacement. A differential coil-system is
physical domain. This deviation represents a
connected to a precise sine wave generator.
wave that can be transmitted through solids
A magnetic cylinder connected to the tip
via transverse or longitudinal waves, and
influences the impedances of the coils.
hits our ears as a sound pressure wave.
Therefore, the impedance ratio is a function
of the sensor’s position. Since the
The physical domain selected for measuring
impedance ratio is responsible for the
depends only on the considered parameters
distribution of the bridge sine wave, the
and the performance of the sensor system in
amplitude of the output- voltage is also
the desired frequency area.
directly related to the position of the sensor.
Figure 9 lists common sensor-systems and
A simple rectifier or a more advanced
describes the principle of operation.
enveloping demodulator can prepare the

© 2011 SKF Reliability Systems All Rights Reserved 10

signal for further processes. Common bridge Before we can give an answer, we have to
frequencies are between 5 kHz and 20 kHz. be more precise. Do we want to judge the
quality in an application, do we want to
select the bearings after final assembly, or
5.2. Moving Coils
do we want to check the quality of the
The principle used in moving coil individual bearing parts?
transducers has been understood since the Based on the equations we previously
beginning of electrical techniques in derived, we can draw a graph (Figure 10).
generators and electric motors. Broadcasting
stations continue to use this concept in
microphones. The principle shows that
whenever a wire is moved in a magnetic
field (or the magnetic field changes in time)
a voltage is induced in that wire. The
amplitude of this voltage depends upon the
strength of the magnetic field and the length
of the wire (coil). When all other parameters
are constant, the voltage generated in this
kind of sensor is proportional to the speed of
the coil. Since the coil, in the case of a
moving coil transducer is connected to the Figure 10. Frequency Response of Displacement
(Blue), Acceleration (Yellow) Based Upon Constant
tip, we measure velocities with these Velocity (Red). The Two Vertical Lines on the Graph
sensors. are Referenced as f1 and f2.

5.3. Piezoelectric Transducers If we assume we have constant velocity over

the whole frequency range, the
In some crystals, charge can be drained
displacement proportional and the
away when pressure is applied. The effect
acceleration proportional response are
results from special triangular structures of
totally different. In figure 10, assume the
the molecules. For measuring acceleration,
first vertical line on the graph is f1 the
this type of transducer is based on the
second is f2, and where displacement and
seismic principle. One side of the crystal is
acceleration responses intersect is velocity.
connected to the object, and on the other
Where the frequencies f1 and f2 are really
side a mass is mounted without a connection
located depends on the sensitivities of the
to the sensor housing. According to equation
sensor systems.
(22) a force is applied to the crystal when
the transducer is accelerated. This force
Below f1, displacement should be used
generates a charge that is transferred into
whenever it is possible. Velocity is a
voltage in a capacitor. Since the mass is
compromise, but acceleration is the totally
constant when we are far enough below the
wrong decision. There is a frequency band
speed of light, the voltage we measure is
between f1 and f2 where velocity represents
proportional to acceleration.
the best solution, and displacement or
acceleration can be an alternative.
Now we can ask the question, which sensor-
Frequencies higher than f2 require
system should be used for noise problems?
acceleration as the proper physical domain,

© 2011 SKF Reliability Systems All Rights Reserved 11

and nobody displacement should not be
used in this area.

Therefore, for quality assessment of

bearings we have displacement proportional
and velocity proportional equipment for
roundness and waviness testing of individual
bearing parts. To judge the noise of
assembled bearings, velocity proportional
measuring equipment is in common use. The
procedure is laid down in a standard.
According to this standard, the bearing has
to run at 1800 RPM, and velocity
proportional RMS-values must be measured
in three filter bands (band A: 50 Hz–300 Hz,
band B: 300 Hz-1800 Hz, band C: 1800 Hz–
10,000 Hz). Some producers use
acceleration for this purpose regardless of
the international standards, and they also
come up with good quality results.
To monitor a bearing in its application, the
monitoring equipment has to respond to
pulses caused by a bearing defect. Since the
frequency response on pulses goes up to
very high frequencies, acceleration
proportional monitoring equipment is usually
used for this purpose.
There is an additional argument to assess
the noise of assembled bearing’s velocity-
proportionally. According to Figure 11, our
sound impression is proportional to the
velocity of a vibrating surface. Therefore, it
makes sense to select the same physical
domain in which the customer will judge the
noise level of a bearing.
Figure 11. The Human Sound Impression.
6. Why Spectra?
It is somewhat common to describe certain
parameters in relationship to other
parameters. For example, we often describe
speed in miles or kilometers per hour.
Angles and certain other parameters are
also addressed as functions of time or time
domain functions. Time domain functions
are very useful tools that describe a great
deal of information. However, when we use
time domain functions sometimes we don’t
receive the required information. For
example, when looking for the frequency
content of a signal, the time domain may
only help us at the dominant frequencies,
but often the needed information is at a less
dominant frequency.
Let us give some examples. When we make
a roundness measurement of an eccentric
oval as shown in Figure 12, the sensor
generates the signal in the linear chart.

© 2011 SKF Reliability Systems All Rights Reserved 12


Figure 12. Roundness Measurement of an Eccentric
Oval.
Without any further information beside the
linear chart we could say that the result is
dominated by eccentricity, but influenced by
some other parameter.
In our experiment we can project the vector
of eccentricity alone as we did during the
discussion of how a sine wave is generated.
Drawn over the angle we get a pure sine
wave within one rotation: the response of
eccentricity. This result is described as the
first order, as it has only one positive and
one negative part.
When we assume we have only ovality
without eccentricity, the sensor is moved
twice up and twice down during one
rotation. We can then say ovality is of the
second order, which means that two positive
and the negative parts are present during
one rotation. We can draw a representative
vector of ovality by rotating twice as fast as
the vector of eccentricity. The projection of
© 2011 SKF Reliability Systems All Rights Reserved
this vector over the angle is the response of
ovality.
In Figure 12 it is apparent that the ovality of
the measured part does not coincide with
eccentricity. The orientation of ovality is
22.5 degrees in front of eccentricity; that is,
the sine wave for ovality starts with a
phase-shift of 22.5 degrees or  .
8
When we add the values of the two sine
waves we have from our projection at each
angle-point, the result is a curve in the
linear chart. Therefore, the individual waves
tell us the content of pure sine waves in the
linear chart. Since the linear chart
represents the signal we received from the
sensor, we can never ask directly for the
content of pure sine waves. But, the answer
to this question becomes more and
more interesting when investigating pure-
tone problems.
To display the sine wave content of the
linear chart on another chart, the so-called
frequency chart or spectrum is much more
convenient, especially when a lot of
individual sine waves are present. For
example, with outer rings we have to take
into account 923 individual sine waves. If
this were the case, the result would be
nothing but a black bar in our time domain
chart.
The ordinate keeps the unit, in our case
displacement, but the unit at the abscissa
changes from angle to order number. We
call the lines at the order numbers
harmonics. When the linear chart depends
upon time, the spectra depend upon
frequency. Since this is the most common
case, we call the spectra 'frequency domain'
or ‘frequency chart', even when the abscissa
is not marked in frequency units.
For our simple example we can draw a
spectrum (Figure 13). In this example, only
the first and second order numbers exist.
13

Figure 13. Spectrum for the Example in Figure 12.
Order number zero represents the offset of
the sensor from its electrical zero point. This
offers the possibility of measuring also
diameters related to a master part. In an
electrical signal the value of the zero
harmonic tells the DC-content (direct
current).
In the next example let’s analyze the voice
of a vocalist. Two vocals ‘A’ and ‘E’ are sung
at the concert pitch ‘a’ (440 Hz). First we
display the amplified signal received from a
microphone on a digital storage oscilloscope
(Figure 14). The time signals are different,
but which is the vocal ‘A’ and which the
vocal ‘E’?
Figure 14. Time Signal of Vocal ‘A’ and Vocal ‘E’.
Vibration Principles
When we measure the time between two
peaks of the first chart we find a distance of
6.8 milliseconds. The corresponding
frequency (f = 1/t) is 147 Hz, which is only a
third of the desired concert pitch frequency.
Has our vocalist made a mistake?
In order to answer this question we should
look at the frequency domain as shown in
Figure 15. Instead of time at the abscissa we
have frequency, and the ordinate is marked
in voltage as before. To make the response
more understandable a logarithmic scale in
decibels (dB) is used. This type of scaling
also makes lower amplitudes visible that
would be hidden in a linear scale. The rule
for calculating the logarithmic scale in our
example is: (0dB = 5 Volt)
 amplitude 
response[dB] = 20 log  
 5 Volts 
 
(23)
Figure 15. Frequency Response of Vocal ‘A’ and Vocal
‘E’.
© 2011 SKF Group 14 (35)
The first chart displays a much higher
frequency content than the second one. Sing
an ‘A’ and an ‘E’ and ask yourself, which
chart represents which vocal. Do the ‘A’ or
‘E’ contain higher frequencies?
Back to the previous question if our vocalist
made a mistake. The answer is that he
didn't. The frequency line at 440 Hz is the
highest one in the second chart and one of
the highest in the first chart. The frequency
we found in the time domain exists at a third
of 440 Hz but it is not dominant by far.
algorithms used in a computer, but a basic
knowledge of the transform function helps
the analyst understand the possibilities and
limitations of frequency analysis.
The Fourier theorem tells us that each
periodic function can be represented as an
infinite number of pure sine waves. Figure
16 shows the first half of a square wave,
called im.

This example proves that the frequency

domain offers a lot of additional information
about the signal. Comparing the two spectra
you can imagine that automatic voice
recognition can be applied, supported by
further calculations.

This type of analysis is especially useful

when evaluating assembled equipment, and
of tremendous importance for quality
investigations when analyzing bearings and
rotating equipment.
Figure 16. Example of Fourier Synthesis and
Analysis.

7. The Fourier Theorem

When we use, for example only five sine
Jean Baptiste Joseph de Fourier (1768 -
waves i1 –i9, and add the amplitudes at each
1830), the famous French mathematician,
angular point, we end up with the curve
developed a basic theory for signal analysis.
named ∑iv. The result is not a square wave,
For a very long time this theory slumbered.
but it is similar. If we want to have a better
approximation, we only have to increase the
Supported by computers, the situation has
number of individual sine waves. In order to
now totally changed, the Fourier
reconstruct the sharp corners of the square
transformation, the most popular
wave, infinitely high frequencies of sine
implementation of this theory, can be used
waves are necessary. To put it another way,
for daily measurements and real-time
a square wave contains frequencies with
applications when the system is based on
very high values.
signal processing hardware.

We can prove this fact with a simple

The signal processing technique has a lot of
experiment. Switch on your radio and select
benefits and some limitations that have to
AM. Turn a power consumer on and off and
be known to obtain good results. It isn’t
you will hear a response in your radio. The
necessary to know the transformation

Vibration Principles © 2011 SKF Group 15 (35)

reason for this effect is the high frequency
content in the square wave current you
generated.
When we build up a function out of
individual sine waves, we call it Fourier
synthesis, and when we ask for the content
of individual sine waves in any function, we
call it Fourier analysis. Fourier synthesis is
used in generators and synthesizers, but
Fourier analysis is important for our
purposes.
Based on what we have derived until now we
can write the Fourier theorem in a formula:
 
f (t) = a k cos (ωk t)+ b k sin (ωk t)
k 0 k 1
(24)
In words we say that each periodic function
f(t) can be described by an infinite sum of
sine and cosine waves with a certain
amplitude and frequency. The index of
summation starts at zero, which represents
the offset in f(t) (DC - part ) because the
cosine of zero is always one (Figure 7).
Since the sine wave is zero when the
argument is zero we can start the
summation for the sine waves at k = 1.
The word “periodic” is important in the
Fourier theorem. Theoretically it means that
the signal f(t) must exist for an infinitely
long time. The reason for this requirement is
the response of transients, which is zero
only when the signal was generated an
infinitely long time ago. In practice, the
signal must be generated ‘long enough’ ago,
independent of the time frame.
The periodicity requirement is fulfilled
perfectly only in roundness testing when the
track measured is closed, as nothing is more
periodic than a closed circle (Figure 17).
Vibration Principles
Figure 17. Periodicity of a Closed Circle.
It is important to explain the development of
the Fourier theorem to establish the basis of
the theorem. If the theorem tells the truth
we can visualize it in a model. As in every
model the reality can only be approximated.
In Figure 18 we have a cube with a
horizontal plane in the middle. In this plane
the individual sine waves of our function are
drawn and represent reality. We can only
look at this cube from the left or front side.
In both cases we have amplitude in the
vertical direction, but the abscissa differs.
From the left side the abscissa represents
the time axis, and from the front side the
frequency axis. Therefore, the left side
represents the time domain and the front
side the frequency domain.
The response in the time domain is a
projection of the sum of the amplitudes of
each individual sine wave at each point in
time. Thus, we get our function f(t). In the
frequency domain we see individual lines
only at those points where a sine wave
exists. Since only one sine wave is included
in the summation, the response represents
the amplitude only of this individual wave.
© 2011 SKF Group 16 (35)

Figure 18. A Model for the Fourier Theorem.
Each projection represents only a part of the
whole information. The question as to which
projection plane offers better results
depends on what we are asking for. For
example, we never arrive at an answer
about the peak value in the frequency
domain.
When we stay in the time domain the
Fourier transformation changes our view to
the other plane. In computers Fast Fourier
Transformation (FFT) is usually
implemented. The development path back to
the domain is supported by the Inverse
Fourier Transformation (IFFT) equation.
It is important to note that Fourier
transformation does not generate new
information, as only other signal or function
behaviors are displayed.
Most measuring equipment measures in the
time domain (like an oscilloscope), but
equipment measuring directly in the
frequency domain also exist (like a
frequency indicator Figure 19).
Vibration Principles
a)
b)
Figure 19. Display of a Frequency Indicator
a) Frequency of the Mains is 0.25 Hz Too Low.
b) Frequency of the Mains is OK.
8. The Fourier
Transformation
Equation (24) describes the Fourier theorem,
but how do we get the Fourier coefficients ak
and bk? Without this information the whole
theory doesn’t tell us anything.
It is necessary to have a basic
understanding of integration methods to
calculate the desired coefficients. We start
with the basics to explain this very
important mathematical tool. To find the
area of a rectangle we often just multiply
width by height. With a right-angled triangle
it is nearly as simple.
Since this kind of triangle divides the area of
a rectangle into two equally spaced areas,
we just have to divide the area of the
rectangle by two.
The above-mentioned calculation of the area
of a right-angled triangle is exact, but we
can also approximate the result with a
different method. Let’s split the area of the
triangle into six rectangles (Figure 20) and
add up their individual areas.
© 2011 SKF Group 17 (35)

Figure 20. Area Approximation.
We can write the equation in Figure 20 more
mathematically:
6 lower limit from the result of the upper limit:
Atriangle = A
i 1
i (25)
It is obvious that the approximation
becomes more precise with an increased
number of rectangles. This especially true
with functions of higher complexity are
approximated, as it is necessary to decrease
the widths of the rectangles and to increase
the number of rectangles. Nevertheless, the
approximation isn’t exact unless it is
possible to decrease the width of our
rectangles to infinitely small extensions, and
add the infinite number of rectangles.
Once again, infinitesimal calculus offers a
solution to our problem. Integration allows
the building of an infinite sum of infinitely
narrow rectangles. The widths of the
rectangles are indicated by the prefix 'd',
and the symbol for summation in equation
(25) changes to the symbol for integration.
So the exact area of the triangle in Figure 20
can be written as:
Vibration Principles © 2011 SKF Group
t2
Atriangle =  f (t )t
t1
(26)
The right side of equation (26) is generally
written. To create a triangle, f(t) equals k
times t as demonstrated in Figure 2
(definition of a straight line). The
parameters t1 and t2 are called the limits of
integration.
According to the rules of integration we get
the following result:
t2 t2
t2
Atriangle =  ktt = k
2
(27)
t1 t1
Now we introduce the limits of integration
into the result, and subtract the result of the
(t 2 ) (t ) 2
Atriangle = k k 1 (28)
2 2
Since t1 is zero in our example, the second
term of equation (28) becomes zero.
Let us assume that k equals one and the
length of our triangle is four. This means for
our example t2 also equals four. When we
substitute this in equation (28), we get 42/2
= 8 for the area.
Conventionally for this equation we require
the triangle’s height. Since the slope of our
straight line is 45 degrees, the height is also
four. Therefore, the area of the rectangle
becomes 16, and the area of the triangle is
half, which is the same result as above.
In this simple example we might not see the
benefit of integration methods. However,
when we require the area below any function
only integration produces exact results
whenever a solution exists for the integral.
18 (35)

Note the important fact that the integral
calculates the area below a function. The
result is sign-sensitive, as areas lower than
the abscissa are subtracted from areas
above. Therefore, the integral of a sine wave
without an offset is always zero whenever
the limits of the distance of integration are a
multiple of 2π.
Now let's go back to the original question of
the Fourier coefficients ak and bk. First we
seek the offset in our function f(t).
state of the art. Note that phase information
φk, which may include important
information, is ignored!
In Figure 21 a periodic function is displayed
containing an offset. We get the height of
the offset with a rectangle that contains the
same area as the area below the function,
and take the sign into account. Therefore,
the area of the rectangle, A0 times T0 must
equal the integral of the function between
the limits zero and T0:
T0

A0 T0 =  f (t )t (32)
As mentioned previously, we get the offset 0

when the addition index k equals zero in

equation (24). The offset does not contain
any periodicity. To show this in more detail
we can rewrite equation (24) and extract the
offset, giving it the name A0. In order to
correct the cosine we let the addition start at
k=1 also for the cosine components:

 
f (t) =A0 +  a k cos (ωk t)+  b k sin (ωk t)
k 1 k 1
(29)
Figure 21. Offset of a Periodic Function - the Mean
Since the cosine function is shifted through Value is Above the Zero Level.
90 degrees relative to the sine function, the
vectors ak and bk can be added to the
Since we are interested only in A0 we get a
modulus using the Pythagorean theorem:
basic equation of a Fourier transformation:

Ak = (a k ) 2  (b k ) 2 (30) 1
T0

A0 =
T0  f (t )t
0
(33)

The cosine function and the sine function

can be written as one function that includes
But how can we get the amplitudes Ak of the
a phase-shift indication φk. Together with
periodic harmonics and the Fourier
(30) we can rewrite equation (29) as:
coefficients ak and bk? We don't know which
 harmonic components are included in our
f (t) =A0 + A
k 1
k cos (ωk t + φk) (31) function f(t), therefore we have to examine
the function from that point of view.

Spectra, as shown in Figure 15, display the Let’s make another experiment. Our function
amplitudes A0 and Ak, which is the current f(t) should be a pure sine wave with an

Vibration Principles © 2011 SKF Group 19 (35)

amplitude of one, and of the first order
within the periodic time T0. This sine wave
should be multiplied by sine waves of
different orders, starting also at order
number one. We get 'k' results of 'y'
according to:
yk = sin(ω0 t) sin(k ω0 t)
where 'k' should be a member of the group
of natural numbers. The results are shown in
Figure 22.
Figure 22. Multiplication of Sine Functions.
Vibration Principles
Looking at the areas outlined the multiplied
curves (thick lines) we see that the areas
above and below the abscissa are equal
(with the exception of the first chart).
Therefore, an integral over the periodic time
is zero. This is a proper indication that the
sine wave we currently use for multiplication
(34) does not exist in our function f(t). If the sine
wave exists, the integral results in a certain
value equal to the area of the gray rectangle
(as in the first chart of Figure 22). It is
obvious that the amplitude of the rectangle
is half as high as the amplitude of the sine
waves. In order to get the correct amplitude,
the result of the integral must be multiplied
by two.
It is now possible to write the equations for
ak and bk:
T0
2
ak =
T0  f (t ) cos(k t )t
0
0
T0
2
bk =
T0  f (t ) sin(k t )t
0
0 (35)
For each coefficient 'k' of interest we
multiply the function f(t) with a cosine and
sine function of the order 'k', calculate the
integral, and correct the result by the factor
2. As we did with the offset, we divide the
result by T0 in order to get only amplitude.
When a closed solution cannot be achieved a
large number of calculations are necessary
to calculate a spectrum. This is the normal
case in measuring technology. Therefore,
fast computers provide real-time facilities.
The operator gets the impression that the
spectrum on the screen is dynamically
driven by the signal. He doesn't recognize
that millions of complex multiplications are
done between each screen change.
© 2011 SKF Group 20 (35)
The concepts of calculations are optimized in
Fast Fourier Transformation, but in principle
the algorithm follows the rules we just
derived.
In literature you may find different equations
for Fourier transformation, but don't worry,
they give the same information in another
frame. To round off this section, the most
popular equations should be noted:
 an explanation for people with contact with
 f (t )e
 jt
F(jω) = t

 Fourier Transformation (FFT). Whenever the
1
 F ( j ) e
jt
F(jω) = 
2 
The prior pair of equations allows us to
calculate a spectrum out of a time-signal
(36) and the time signal out of a spectrum
(37), even if the signal is not periodic. In the
case of a non-periodic signal, the periodic
time can be seen as infinite; therefore, the
Fourier sum changes to the Fourier integral,
and the discrete spectrum becomes
continuous.
The exponential term is just a short-form of
writing according to the identity of Euler and
contains the multiplications with cosine and
sine functions:
ejwt = cos(ωt) + j sin (ωt)
(38)
Equations (36) and (37) are written using
the results of complex analysis. The prefix ‘j’
indicates the axis of imaginaries. The factor
1 convert the signals. The resolution of this
in equation (37) is a parameter of
2
integration.
Based on complex analysis, the calculation
of the Fourier coefficients F(jω), which
represents the spectrum, starts at minus
Vibration Principles
infinity as the lower limit of integration, as
complex vectors describe a real function.
The vector of imaginaries can be
compensated using a second vector of
imaginaries with the opposite spinning
direction. This is realized by minus infinity in
the lower limit of integration. Therefore, you
might sometimes see negative frequencies
that are physically nonsense, but caused by
the principle of calculation. This information
may be confusing, as it is referenced only as
(36) theses formulas.
Equations (36) and (37) are the basis of Fast
(37) number of samples is based on the power of
two, redundancies within the algorithm can
be used to optimize the process. For
example, an FFT based on 1024 samples is
about 100 times faster than a standard
Fourier transformation based on 1000
samples. In general terms: the higher the
number of samples, the greater the benefit
of the FFT algorithm. Therefore, you find a
lot of equipment, such as an angle encoder,
with increments based on the power of two.
9. The Shannon Sample
Theorem
In the previous discussion we spoke about
signals that are continuous in amplitude and
time. When we use computers, we digitalize
the signal and store the results in memory,
and then the calculation procedures can
start.
An analog to digital converter is used to
strategic part is of great importance for the
performance of signal conditioning and
information investigation; however, more
details go beyond the scope of this paper.
© 2011 SKF Group 21 (35)
In addition, the conversion takes time, and In Figure 24 the frequency of the signal is
we don't know the behavior of the signal in half the sampling frequency. We can imagine
the interim. Thus, we can end up with time- the dotted gray sine wave enveloping the
discrete samples. The consequences of this samples.
fact are of greater importance for daily
measurements and should be thoroughly
noted. Depending on the price and the
technology status this problem is the reason
for some insufficiencies and it is necessary
to be aware of it.

First assume that the sampling frequency is

higher than the frequency of the signal we
want to analyze (Figure 23).

Figure 24. The Sampling Problem. In this Example

the Sampling Frequency (Middle Curve) is Half the
Signal Frequency (Upper Curve):
Signal Frequency = (Sampling Frequency / 2).

The next test is done with a sampling

Figure 23. The Sampling Problem. In this Example
the Sampling Frequency (Middle Curve) is Higher
than the Signal Frequency (Upper Curve):
Signal Frequency < (Sampling Frequency / 2).
frequency as high as the frequency of the
signal (see Figure 25). If we draw a straight
line through the samples without any
knowledge about the original signal we
would say that we sampled an offset - a
totally wrong conclusion.

The lines in the samples chart represent the

amplitudes we get in the memory of the
computer. A longer line represents a higher
amplitude.

It is obvious that the samples reconstruct

the original sine wave. The higher the
sample frequency related to the frequency of
the signal, the better the approximation.

But what happens when the sampling
frequency becomes lower? The same
situation exists when the frequency of the
signal becomes higher at the same sampling
frequency.
Figure 25. The Sampling Problem. In this Example
the Sampling Frequency (Middle Curve) Equals the
Signal Frequency (Upper Curve):
Signal Frequency = Sampling Frequency.
Let's go a step further and assume the
frequency of the signal is higher than the
sampling frequency (Figure 26).

Vibration Principles © 2011 SKF Group 22 (35)


Figure 26. The Sampling Problem. In this Example
the Sampling Frequency (Middle Curve) is Lower
than the Signal Frequency (Upper Curve):
Signal Frequency > (Sampling Frequency / 2).
Now the envelope over the samples is
interpreted as a sine wave with a lower
frequency than the frequency of the signal.
We call this alias-frequency. In German it is
said, "lattice-fence-effect" because the
samples are comparable with the response
we might get when we look through a lattice
fence. The reality might be misinterpreted.
We can derive from these examples that the
sample frequency has to be at least twice as
high as the highest sine-wave frequency
included in the signal. The Shannon sample
theorem does not provide any other
information. The effect of this limitation is
important, as square wave includes
components of infinitely high frequencies
that we can never sample.
In order to exclude alias-frequencies from
the results it is necessary to ensure
frequencies higher than half of the sampling
frequency are included in the signal. To fulfill
this demand the signal has to be low-pass-
filtered by an anti-aliasing filter.
We know how the filter characteristic should
look: in pass-band the signal should not be
influenced in any respect, and in stop-band
the damping ratio should be infinite. The
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border between pass-band and stop-band
must be a sharp corner (at least).
You cannot design a filter to fulfill this
demand. There are always residual pass-
band ripples and a phase-shift influence on
the signal. The situation is worse close to
the border because of the slope of the filter.
At the end the damping behavior in the
stop-band is also far from perfect.
For these reasons, some people speak of a
400-line spectrum even when 512 lines are
calculated. The residual lines might be
disturbed too much.
But if we do not use a filter, we have the
worst-case situation. No one takes the risk
of making a wrong decision because of a
frequency response that does not exist in
reality. Without filtering, an increasing
analogue frequency is mirrored at the
Shannon border.
Assume we have a sampling frequency of 40
kHz, which is nearly the sampling rate of a
CD-system. When we sample an analogue
frequency of 19 kHz the frequency response
is at 19 kHz (where it should be). But, when
we sample an unfiltered analogue frequency
of 21 kHz, the frequency response is again
at 19 kHz, which is totally wrong. An
analogue frequency of 22 kHz has its
response at 18 kHz, and it seems the
response is mirrored at the Shannon border.
If we continue in this way, the response is
also mirrored at 0 Hz. Therefore, an
analogue frequency of 41 kHz is seen at 1
kHz in our spectrum.
Without limitations the frequency response
is captured within 0 Hz and the Shannon
border, even if the analogue frequency goes
up to infinitely high values.
© 2011 SKF Group 23 (35)
In some precise laboratory sampling
equipment the filter is a major cost.
However, the disadvantages caused by an
anti-aliasing filter are fewer than the
problems we get without it.
10. Time Windowing
Finally, we must discuss an additional
limitation when using computers for signal
investigations. Since memory space is
always limited, we can make only a short
snapshot of the signal. The higher the
sampling rate, the faster the memory is
filled. But the Fourier theorem needs
periodic signals. In most applications it is a
matter of luck if we can sample the signal
exactly within the periodic time of the signal.
Therefore, we have to be aware of a gap
between the sample at the beginning and
the sample at the end of the array.
The crystal used in the sampling equipment
causes the unusual timescale in Figure 27. A
power of two crystals operated at 1.048576
MHz generates sampling pulses with a
frequency of 32.768 kHz after a scaling unit.
Therefore, the pulse repetition time is
30.517 μs. Since the equipment is designed
to take 1024 (210) samples, we aim for a
time frame of 31.25 ms.
The corresponding spectrum is shown in
Figure 28. We see a very dominating line at
640 Hz with an amplitude of nearly 5 Volt.
At 1600 Hz the second sine wave we fed into
the system is visible with a very low
amplitude of -58 decibels. According to
equation (23) the amplitude of this sine
wave is only 6.3 millivolts, which means
nothing is visible in the time domain of
Figure 27. There are also some lines below -
60 dB visible at higher frequencies that are
caused by uncertainties in the sampling unit.

In order to explain this problem and to

derive possible solutions, let us once again
look at an example. First we simulate the
ideal situation: two sine waves of different
order numbers and amplitudes are sampled
with a multiple of their periodic time.
Therefore, there is no gap between the
borders of the time window. The gap, visible
in Figure 27, is caused by the displaying
method. 0 2.048 4.096 6.144 8.192 10.240
12.288 14.366 Figure 28. Frequency Response of
the Two Sine Waves without a Gap at the Border of
the Time Window.

This is the ideal case. Now let's change the

frequency of the dominant sine wave so we
get a gap between the samples at the
borders of our time window. The time signal
is shown in Figure 29, the sine wave ends at
the bottom of the right-hand border, which
opens a gap as high as the amplitude.
Figure 27. Two Sine Waves without a Gap at the
Borders of the Time Window.

Vibration Principles © 2011 SKF Group 24 (35)


Figure 29. Two Sine Waves with a Gap at the Borders
of the Time Window.
Hopefully you are shocked by the result
shown in Figure 30. We can no longer see
the second sine wave, as it is hidden within
the response caused by the dominant sine
wave.
Figure 30. Frequency Response of the Two Sine
Waves with a Gap at the Borders of the Time Window
Using a Rectangle Window in the Time Domain.
The amplitude of the first sine wave is lower
than in the case without a gap: the missing
part of the amplitude seems to have leaked
out and spread over the whole spectrum.
Therefore, this is called leakage effect.
The spectrum does not represent reality, as
we only fed two pure sine waves. Why are
so many disturbing spectral lines present?
To answer this question we have to go back
to the basics previously discussed. The
Fourier theorem needs periodicity to achieve
correct results. The array we sampled is the
maximum space of periodicity and the
transformation algorithm assumes that the
Vibration Principles
content of this array is periodically continued
in both directions. Therefore, the gap is
interpreted as a content of the signal. As we
found during the discussion of square wave
frequency, the place of discontinuity also
causes frequencies up to infinite values. So
the gap is the reason for the leakage effect,
but how can we avoid the disturbing
response?
The easiest reaction is a repeated
measurement. We may have more luck and
reduce the space of the gap. The situation
may be improved by averaging the spectra.
But what can we do when this does not
satisfy us?
Let’s analyze a sampled array in more detail.
The array represents only an interval of
observation that is just a very small part of
reality. We can assume we got the array by
multiplying the infinite periodic signal by 1
during our sample time. Outside the interval
of observation the multiplication factor is 0.
The multiplication function opens a window
of observation for us; outside of which, we
don't know what's going on. Since the
window looks like a rectangle it is called a
rectangle window. To have the signal
multiplied by a rectangle window is just a
special approach that does not influence out
samples. Figure 31 displays the situation.
Figure 31. Rectangular Window: Below the Interval
of Observation.
© 2011 SKF Group 25 (35)
Obviously the problem is caused by the gap.
If we could force the gap down to zero at
both borders of the array we would certainly
influence the signal and the response, and
maybe we can reduce the response of the
leakage effect.

Let’s use a smooth windowing function that

multiplies the signal by zero at both borders. Figure 33. Frequency Response using Hanning
Within the interval of observation the Window.
function should continuously increase and
decrease the amplitude. To correct the
The influence of the leakage effect is
influence of damping at the borders we
dramatically reduced in Figure 33. The low
amplify the signal in the center by two.
second sine wave is no longer hidden in the
spectrum. But, the first sine wave is not a
The smooth function should be an inverse
single line as it was in Figure 28, there are
cosine with a positive offset of one. Such
still some neighboring harmonics that should
function exactly fulfils the above mentioned
not be present when a perfect sine wave is
demand. This is called a Hann or Hanning
sampled. To see the disturbing influence and
window. The effect on the signal within the
the very good results of this windowing
interval of observation is shown in Figure 32.
method let’s apply this window to the
measurement without a gap (Figure 34).

0 2.048 4.096 6.144 8.192 10.240

Figure 32. Hanning Window.
Now apply this window to our example and
look to the response after a Fourier
transformation (Figure 33).
12.288 13.336
Figure 34. Frequency Response of Two Sine Waves
without a Gap at the Borders of the Time Window
using a Hanning Window with the Time Domain.
It is apparent that relatively high
neighboring harmonics are artificially
generated, and influence selective RMS
values. In addition, the neighboring
harmonics reduce the spectral selectivity.
So, which kind of windowing should be used
to reduce errors? The Hanning window
described is the most popular one; although
you also find the Hamming, Kaiser Bessel,

Vibration Principles © 2011 SKF Group 26 (35)

Gauá, Blackman windows, etc. Your window the knowledge you have gained, you y(t) is
selection depends on the measurement a function of x(t):
problem.
y(t) = f{x(t)}

11. Transfer Function The black box named h(t) keeps the laws of
how x(t) is projected to y(t). If there is no
When the topic of bearings in applications
connection between the source x(t) and the
arises, it often leads to a discussion of
destination y(t), y(t) is zero, and therefore
immediate contact with the machine
h(t) is zero. If y(t) is directly connected to
structure. Even if we bearings existed with
x(t) without any disturbances, y(t) is the
ideal individual parts, they would behave like
same as x(t), and therefore h(t) is one.
a generator for vibrations. Only in the
theoretical case of zero radial clearance and
Unfortunately, it is not so simple that we can
ideal stiff bearing components, could
just multiply the generator function x(t) by
bearings run without generating any noise.
h(t) to get the answer function y(t). Since
the signal generated by x(t) can be
It is essential to realize the behavior of a
transferred in any way to the destination
structure to understand what's going on in
y(t), we call h(t) the transfer function in our
an application. The transfer function is the
current case represented in the time
best tool for this understanding. Again, a
domain.
tool becomes more powerful when you
understand the background. For that reason
If we want to be exact the situation becomes
please try to follow the subsequent way of
rather complicated, as we must allow our
thinking.
black box to cover any possible network.
When there are parts included that are
We can abstract the situation in the time
capable of storing energy, the situation
domain as shown in Figure 35. Due to our
becomes tricky. Once the system is charged
strong technical and emotional connection
with energy, the behavior can be influenced
with the time domain we start our expedition
for an infinitely long period of time.
here.

In mechanics, mass and spring are the

x(t) h(t typical storage of energy. In electronics the

factors are capacity and inductivity. Mass

y(t) and spring or capacity and inductivity

Figure 35. Transmission Chain in the Time Domain.
On the left side we have the signal source.
For our example it is the generator, and
named x(t).
On the right side we have the destination,
which might be a microphone, a piezoelectric
crystal, or ears. We call this y(t). Based on
represent an oscillating circuit. Once
activated, the injected energy is transferred
from potential energy to kinetic energy, and
back at the so-called resonance frequency. If
the system ran without any friction or other
losses, the oscillation would never end. In
electronics the energy is transferred from
the electrical field of the capacitor to the
magnetic field of the inductivity, and vice
versa.

Vibration Principles © 2011 SKF Group 27 (35)

With friction or resistances the activated
oscillation decreases in amplitude as soon as
the foreign energy source is switched off.
Since that process is natural, the descending
function is exponential. As you know, the
exponential function approaches zero
asymptotically, which means oscillation
becomes smaller but it never reaches zero!
Technically, we may say that we can forget
such an influence from the past, but
mathematically we are not able to do that.
For example, you might have heard that a
specially equipped satellite measured the
very low oscillating thermic background
radiation in space. This can only be
explained by the big bang, and this event
happened quite along time ago.
Based on this way of thinking, events from
an infinite period of time ago can have an
influence on our black-box h(t); therefore,
our view y(t) to the generator x(t) might be
influenced and disturbed. In order to be
exact we have to sum up all influences from
an infinite long time ago until the present.
This statement is represented in the
following equation:
t
y (t )  

x( )h(t   )d
(40)
This integral is asymmetric since the variable
of integration starts at minus infinity and
ends at present time t. Equation (40) is so
important that it received its own name. The
"convolution integral" can be written as:
y(t) = x (t)* h(t)
(41)
We say x(t) is convoluted by h(t), and that
an infinite asymmetric integral is behind.
Vibration Principles
But how does a 100% correct equation help
us when we can't solve it to get proper
results? We have to create some restrictions
to make the equation more practical. We
always have to be aware that our results
might be affected by the restrictions!
First, we make the system causal by
postulating that before a certain event
nothing happened to our system. Thus, all
reactions of our system can be traced back
to a cause known in principle. We definitely
exclude events like the big bang.
Per definition, we say that our black box is
totally devoid of energy for all times lower
than zero. Therefore, the black box cannot
create influences from the past. When we
apply that to equation (40), we get an
interesting change.
As soon as the variable of integration τ is
bigger than present time t, h (t - τ) becomes
zero because of our previous definition. This
also means that the integral is zero from
that point on. Therefore, the result of the
integral does not change when we extend
the upper limit of integration from t to
infinity as is done in equation (42).

y (t )  

x( )h(t   )d
(42)
The integral is now symmetric, which offers
the possibility of further mathematical
operations.
Let’s apply the Fourier Transformation to
both sides of equation (42) using first the
notation of equation (41). A linear equation
does not change when we apply the same
operation to both sides.
F{y(t)} = F{x(t)*h(t)
© 2011 SKF Group 28 (35)
On the left side we get Y(f). The allows us to look closer at the generator
Transformation of the convolution leads to a (e.g. a bearing running in an application).
double infinite integral: Further techniques like Cepstrum Analysis
allow us to change the multiplication to
Y ( f )  F{x(t ) * h(t )} addition so the transfer function influence is
 looser; however, such methods are beyond
Y ( f )   [ x(t ) * h(t )]e  jt dt (44) the scope of this paper.

  Note that whenever we apply a spectrum
  x( )h(t   )e
 jt
Y( f )  ddt analyzer to a machine we have to take the
   fact that results are influenced by the
structure of the machine into account. This
When we substitute the variable q for t -'t, is represented by the transfer function.
we get an additional exponential function
based on q (t = q + 't): Very often, it is interesting to know how a
  machine behaves over frequency (i.e. the
  x( ) * h(q)e
 j (  q )
(f) ddq transfer function). Wherever resonance
  areas exist, a very weak generator is also
(45)
  able to create an unacceptable response. So
  x( )h(q)e
 jt  jq
(f) e ddq bearings should never generate frequencies
  in such areas, which of course can be
influenced by bearing design. However,
When we assume we have a linear system, application engineers require extremely
which is an additional restriction, the close contact with the customers' designers.
formula can be manipulated in this way: It will become more and more the case that
bearings are produced for well-defined
 
applications.
 x( )e d  h(q)e dq
 j  jq
Y( f )  (46) To achieve the transfer function, we just
 
rearrange equation (47):

When carefully looking at equation ( 46), we

see that each individual integral represents a
Fourier Transformation. The first integral
shows the Fourier Transformation of the
generator function x(t), which is X(f). The
second integral represents the Fourier
Transformation of h(t), which is H(f). So we
can write equation (46) much more easily:
Y(f) = X(f) H(f)
(47)
Y( f )
H( f ) 
X(f )
(48)
We get the transfer function as the complex
quotient of the output spectrum over the
input spectrum of our black box. Thus, we
just need a two channel spectrum-analyzer
and a proper test-pulse generator, and we
can derive the transfer function.

H(f) we simply call the transfer function. In
the frequency domain, the generator
function is just multiplied by the transfer
function. This is a looser contact than the
convolution integral. This big advantage
This statement should be demonstrated by
an electronic example that the author of this
paper performed with some of his students.
The principle is identical for a mechanical
structure.

Vibration Principles © 2011 SKF Group 29 (35)


Figure 36. Test Circuit.
The test circuit in Figure 36 represents the
black box, for which we want to know the
transfer function H(f). As a representative of
a typical quadripole network, the two
terminals on the left side are the input-
connectors for the generator signal x(t). The
connectors on the right side offer the output
signal y(t). The input signal is conducted
parallel to channel 1 of a two channel
transient recorder and the output signal is
connected to channel 2.
A test-pulse generator triggers the transient
recorder a certain time before the test-pulse
loads the circuit. The transient recorder
takes 1024 samples for each channel. The
pre-trigger time is adjusted so the test-pulse
occurs in the middle of the sampling
window. Figure 37 shows the measurement
of the two channels.
Fi
gure 37. Test-Pulse at the Input ( Channel 1) and the
Output (Channel 2) of the Test Circuit.
Vibration Principles
There are very few measurements different
from the straight line over the whole time
window. No one can say anything about the
content of the black box based on the time
signal. We are able to figure out only the
information that the signal is damped.
The next step is the transformation to the
frequency domain via the Fourier
Transformation. The results are shown in
Figure 38. The selected time-window of
31.25 milliseconds, together with the 1024
samples, leads to a frequency range of
16384 Hz, represented by 512 harmonics.
As usual, only the amplitudes of the
individual harmonics are plotted although
the phase is also important. When
calculating the transfer function H(f) we
have to make a complex division, taking
both amplitudes and phases into account.
Figure 38. Spectra for the Pulses of Figure 37.
Figure 39 displays the result of the division
of channel 2 over channel 1 in the so-called
Bode diagram. The locus diagram is another
possibility for displaying the results.
It is always fascinating that the two short
pulses in Figure 37 keep such complex
information, as shown in Figure 39. The
traditional way of analysis is to apply all
sinusoidal frequencies of interest and
measure both amplitude and phase at each
step. When the measurement is done
automatically, we call it a wobble. In
© 2011 SKF Group 30 (35)
mechanical systems a shaker normally
applies the sinusoidal vibrations. It is
obvious that traditional measurement
methods take much more time than the new
computational features.
Figure 39. Transfer Function of the Test Circuit in
Figure 36, Represented in the Bode Diagram.
Let’s briefly interpret the result in Figure 39,
related to the diagram in Figure 36. At the
frequency zero, which means under direct-
current conditions, the capacitor represents
an infinitely high resistance. The circuit is
just a real voltage divider consisting of the
two resistances consisting of the upper left
resistance and 470 Ohms resistance in
Figure 39. Thus, the log-magnitude of the
Bode diagram starts at approximately -10
dB, and the phase at zero degrees. At rising
frequencies the capacitor starts to conduct
more current. Due to this parallel path, the
output voltage decreases. The dominant
impedance of the capacitor causes negative
phase.
At around 10 kHz we approach a point
where the rising inductive impedance is as
high as the decreasing capacitive
impedance. This is the point of serial
resonance. At this point the resistance of the
coil and the capacitor theoretically becomes
zero, due to the exchange of magnetic and
electrical energy in the rhythm of the
activating source. The residual resistance RL
Vibration Principles
is therefore connected parallel to the
resistance with 470 Ohms. Thus, the output
voltage drops down to -30 dB.
Above that resonance frequency the
impedance of the coil becomes higher.
Therefore, the output voltage again
increases. The phase is now dominated by
the impedance of the coil, which makes it
positive. Above 14 kHz the necessary anti-
aliasing filter disturbs the measurement
A software package could automatically
analyze the Bode diagram and give a
diagram representing the black box
identically. This can be very helpful when
structures should be optimized.
When we look carefully at the spectra in
Figure 38 we see that the spectrum of
channel 2 is similar to the log-magnitude of
the Bode diagram. This is because we used a
pulse similar to the dirac impulse. Such a
pulse creates a white-noise characteristic
that can be seen in the spectrum of channel
1. For such a white-noise characteristic we
divide Y(f) according to equation (48),
always by the same value (ideally one when
a dirac-impulse is applied). Y(f) then
represents the transfer function H(f), so it
becomes unnecessary to perform the
complex division and use a two-channel
spectrum analyzer.
On a mechanical system such an impulse
can easily be applied with a hammer blow. A
piezo transducer leads the response of the
structure to a spectrum-analyzer, and the
spectrum displayed represents the relative
transfer function. In this case the transfer
function is relative since the response
depends on the load of the hammer blow.
With a second sensor mounted on the
hammer and a two-channel analyzer, the
absolute transfer function can be evaluated.
© 2011 SKF Group 31 (35)
Sometimes it is enough to see dangerous
resonance areas.
Let us now list some basic examples
according to transfer functions. Sensor
transfer functions are important for
measuring techniques. For example, a
piezoelectric transducer shows a very clear
resonance area based on the elastic crystal
and seismic mass. Vibrations of very low
amplitudes at the resonance frequency
create a high response in the sensor, which
cause wrong measurements. The same
problem exists on piezo-based micro
positioning systems. Figure 40 shows an
example of the frequency response or the
transfer function of a piezo transducer.
Figure 40. Transfer Function of a Piezo
Transducer.
Another well-known example is the critical
speed of generators or turbines. When a
high amount of mass is rotating, the rotation
itself is a dangerous generator of vibration.
It damages the machine when the frequency
caused by the speed of rotation is similar to
the maximum of the transfer function of the
machine. Moreover, the transfer function can
have some maxima. As long as the machine
is running at a speed below critical, this
problem does not exist, but this is not the
case with generators and turbines. This
equipment normally runs at speeds above
Vibration Principles
critical. In order to avoid damage operators
ensure the machines accelerate / decelerate
fast through the critical phase.
The designers of machines running at
speeds above critical must optimize the
transfer function of the whole construction in
order to increase the life span of the
equipment and its individual parts.
12. Typical Vibration
Spectrum
Interpreting a spectrum needs some
experience and a good knowledge of the
process or machine being analyzed. The
following examples give some basic hints
and explanations about different conditions
and their possible conclusions.
In the elementary case just one line rises
out of the spectral swamp. By "swamp" we
mean those parts of the spectrum where all
lines have similar amplitudes. These areas
can be caused by machine noise, shocks
applied during the measurement, or even by
the machine. In very quiet conditions the
swamp can be caused by the measuring
system, due to the border of resolution.
Advanced, and more expensive measuring
equipment might allow a closer look into
that area.
If one single line rises out of the spectrum a
pure tone is indicated. We then know that a
pure sinusoidal vibration without any
disturbances is the reason for that response.
In Figure 41 the harmonic at the left side of
the spectrum at low frequencies is of that
order.
© 2011 SKF Group 32 (35)
 A typical effect caused by bearings is
transient vibrations, which occurs when
passing rolling elements come in contact
with a defect. The local structure is activated
by the pulse applied by the rolling element
and vibrates at its resonance frequency. The
vibration decreases enveloped by an e-
function due to the damping effects. The
vibration starts again when the next rolling
element comes into contact. This effect is
shown in chart a of Figure 43. The
corresponding spectrum, shown in chart b of
Figure 41. Spectrum with a Pure Tone at Low
Frequencies. Figure 43, consists of a periodic pattern of
harmonics. The distance between the lines is
The "hills" on the right side of the spectrum
equivalent to the reciprocal of the frequency
can have different causes. There might be
of balls passing.
some unstable processes at higher
frequencies in the machine possibly
Since the energy applied is spread over the
including effects caused by modulation. It
whole spectrum, the individual lines carry
can also be the normal noise swamp of the
just a small part of it. Therefore, these lines
machine amplified by resonance areas of the
are normally hidden in the spectrum. Special
structure (transfer function). Tests at
techniques like enveloping, SEE-technology,
different rotational speeds, and a
or cepstrum analysis are proper tools to
comparison with the transfer function are
analyze the spectrum according to these
necessary when further investigation is
effects.
requested.
To distinguish between responses related to
rotational speed, and vibration caused by
other influences (e.g. transformers) a
waterfall plot can be very helpful. The
spectra at different rotational speeds are
plotted in a three dimensional graph. In
Figure 42 the fixed-frequency component is
clearly visible.

Figure 42. Waterfall Spectra Help Distinguish Figure 43. The Situation when Rolling Elements
Between Rotationally Related Responses and Come into Contact with e.g. a Local Defect
Vibrations Independent of Rotational Speed. Periodically a) Time Domain b) Frequency Domain.

Vibration Principles © 2011 SKF Group 33 (35)

When the series of lines is visible in some
areas of the spectrum, the zoom-FFT can
also give further details of the sources of
vibrations. Such a spectrum is shown in
Figure 44. One reason for different
amplitudes of the periodic lines is again the
transfer function of the machine.
With some spectrum analyzers it is possible
to highlight multiples of the current cursor
position in the spectrum. This feature makes
it possible to find and distinguish different
sources of repetitive series of lines. Then, it
is also easier to find sources of vibrations
that are independent of producers of such
patterns. In Figure 44 there is one such line
at approximately 3.4 kHz. Without zooming
techniques this response could not be
figured out in a normal spectrum.
Figure 44. Zoom Spectrum.
Series of lines are also produced as soon as
vibrations with discontinuity points (ie.
square waves, saw-tooth, or signals of
triangular shape) are applied. The spectrum
for a square wave with amplitude A, and
frequency f is given in equation (49). The
formula for other spectra can be looked up
in any formula book.
Vibration Principles
4A 1 1
f (t )  [cos(t )  cos(3t )  cos(5t )
 3 5
1
 cos(7t )  ...]
7
(49)
A graphic interpretation of equation (49)
was already shown in Figure 16. It clearly
indicates that at higher frequencies the
amplitudes become smaller and smaller, but
again, they are never zero.
A transfer function between a discontinuity
vibration generator and the measuring
sensor can dramatically change the
response. The amplitude of the ground wave
can be much smaller than harmonics at
higher frequencies. In order to find the right
source of that vibration it is essential to
isolate the ground wave. A search based on
other harmonics leads to wrong conclusions.
But when a ground harmonic of the order
mentioned can be traced back to a rotational
cause, the series of lines tell that the
influence is not sinusoidal, there are
shocking elements included. This information
might help improve the situation.
Another important effect is modulation.
Amplitude, frequency, and phase modulation
are possible. We briefly discuss amplitude
modulation, as it is of greater importance in
vibration testing.
© 2011 SKF Group 34 (35)
 Figure 46. Spectrum of a Carrier (fc) Modulated by a
Low Frequency Sinusoidal Signal fs.
Figure 45. Amplitude-Modulated Signal in Time
Domain.
Normally, the carrier frequency shows up
Figure 45 shows a typical amplitude
higher in the spectrum so the modulation
modulated signal in the time domain. The
effect is visible, and a distinction from other
amplitude of a signal at higher frequency is
causes is possible. It is also normally the
influenced by a lower frequency signal. The
case that the low frequency signal is band-
low frequency does not show up at the
limited and shows higher amplitudes at
expected low frequency order, but it can be
lower frequencies. Therefore, we speak of
seen on the left and right side of the higher
side bands or side band modulation. The
frequency harmonic. The higher frequency,
lower side band is mirrored and has an
which is modulated, is called the carrier in
inverse frequency order. This principle is
our example fc. The frequency distance from
shown in Figure 47.
the carrier to the side harmonics is exactly
the frequencies of the low frequency signal fs
, in both directions (Figure 46).

In the case of non-sinusoid allow frequency

signals, for example square waves, the
corresponding series of lines are spread over
the spectrum in both directions from the
carrier frequency fc. In that case, it is wrong
to look for a ground wave for the visible
series of lines.

Figure 47. Side Band Modulation.

As an example of the modulation effect think

of a shaft loaded with varying torque. A
common example of that is a combustion
machine. The varying torque modulates the
gear-meshing frequencies in the connected
gearbox and the phenomena mentioned are
visible in the spectrum.

Vibration Principles © 2011 SKF Group 35 (35)