Vibration: Analysis and Monitoring

Analysis

Noise and vibration: Causes and effects

They are caused by particular processes where dynamic forces excite structures. Most noise and
vibration problem are related to resonance phenomena. Resonance appears when the dynamic
force excites the natural frequencies or mode of vibrations. In any given situation there are three
factors: the source, the path (how the energy is transmitted) and the receiver. Any of these may
contain the cause of the problem and must be investigated.

Signal vs System analysis

Signal analysis is the process of determining a response of the system, due to some general
unknown excitation (operational modal analysis), and presenting it in a manner easy to
interpret. By analysing the system under operating conditions, we can analyse the source of
vibration. Indeed by analysing the frequencies of the vibration, we can find what part of the
system is the cause of this vibration.

System analysis (experimental modal analysis) deals with techniques for determining the
inherent properties of a system. This can be done by excitation of the system with measurable
forces and measuring the response-force
ratio. For linear system this ratio is an
independent, inherent property which
remains the same whether the system is
excited or not. Once the source of vibration
has been located (by signal analysis), we can
concentrate on the system. The properties of
the transmission path, between the source
and receiver, represent the inherent dynamic
characteristics of the system. A first step is to
proceed a run-up/coast-down test, during
which the response is measured for different
speeds. The response is then plotted in function of the speed. This graph gives qualitative
indications on the resonances in the operating frequency range, since excitation frequency is
proportional to speed. If peaks are found in the plot, it is reasonable to conclude that they exist
 2

resonance frequencies in the system (only if the excitation force is known because without that,
the peaks can be the resonance of the force).

Modal analysis
Modes of vibration which lie within the frequency range of the operational dynamic forces
always represent potential problems. An important property of mode is that every force or
dynamic response of a structure can be reduced to a discrete set of modes. The modal
parameters are:

 Modal frequency
 Modal damping
 Modal shape

The modal parameters of all the modes constituted a complete description of the structure.
Modal analysis is the process of determining all the modal parameters which are then enough to
formulate a mathematical dynamic model. This last one is useful to understand and
communicate how structure behaves under dynamic loads, to use in data reduction and
smoothing techniques, to simulated and predict the response to assumed external forces and to
simulate changing dynamic characteristics, due to physical modifications.

The frequency response function

One very efficient model of linear system is the frequency domain model where the output
spectrum is express as the input spectrum weighted by the system descriptor

( ) ( ) ( )

The system descriptor is called the frequency response function. It represents the complex ratio
between output and input as a function of the frequency.

Resonance in the operational frequency range can be considered as weakness. The severity of
the resonance depends on the magnitude of the FRF between the point where the operational act
on the structure and the point where the response is observed.

Implementing the excitation

Excitation forces can be generated by many different devices divided in to classes: the attached
exciters (electromagnetic shakers, electrohydraulic shaker, eccentric rotating mass) or non-
attached exciters (hammer, large pendulum impactors). Note that acoustic excitation cannot be
used in modal analysis since we cannot control the direction of the excitation.
 3

The excitation force is usually measured by a piezoelectric transducer in which a fraction of the
force is transmitted to a piezoelectric element. The exact force exciting the structure can only be
measured if the force transducer is mounted directly on or as closed as possible to the structure.

The exciter must be attached directly to the structure so that the excitation force acts only on the
desired point and in the desired direction. The structure must be free to vibrate in all the five
other directions. A good attachment technique is to fix the exciter to the force transducer with a
slim push rod or a stinger. This gives a high axial stiffness but a low transverse and rotational
stiffness.

Response measurement
For response measurement, any of the motion parameters can be measured. The best choice of
transducer is the piezoelectric accelerometer. It offers:

 Good linearity
 Low weight
 Broad dynamic range
 Wide frequency range
 A strong construction and simple design
 High environmental resistance
 Low transverse sensitivity
 Simple mounting method

The velocity or displacement can be obtained by numerical integration. For optimal

accelerometer performance, the best mounting technique is to use a threated steel stud.
Tolerance for the mounting surface and recommended mounting torque are generally specified
 4

by the manufacturer. This method is not always possible. Other techniques exist (such as
magnetic mount or thin layer of beeswax). These alternative techniques can lower the frequency
range but it is generally not a problem for modal analysis. In a test where it is necessary to
obtain scaled shape mode, a driving point measurement is needed. A problem is then how to
excite the structure and measure the driving point response at the same place and in the same
direction. We can proceed by applying the excitation very closed to the transducer without
creating a significant error. If the structure is too small it is also possible to fix the driving point
and the force on opposite sides of the structure. An alternative is to use an impedance head (an
integrated force and response transducer).

Random excitation
The term random apply on the amplitude of the force which follows a Gaussian law. With this
type of excitations, individual time records in the analyser contain data with random amplitude
and phase at each frequency. On average, the spectrum is flat and continuous containing energy
at approximately the same level for all the frequencies. Due to the random, the structure is
excited to a large force range at each frequency. This gives a best linear approximation. The
excitation is random and continuous in time, but the record length is finite so leakage error may
occur. This error can be minimized by using a window like the Hanning one.
 5

Impact excitation
The waveform produced by an impact is a transient energy
transfer event. The spectrum has a periodic structure which is zero
at frequencies where n is an integer and T is the duration of the
transient.

The useful frequency range is from 0Hz to the frequency F where

the spectrum magnitude has decayed by 10 to 20dB. The duration
(and thus the shape of the spectrum) is determined by the mass
and stiffness of both the impactor and the structure.

Impact hammer are constructed by adding a force transducer to a

classical hammer and a stiffness controller element to the end of
the hammer. The true force applied on the structure is the
measured force multiply by the ratio

The advantages of hammer testing are the speed, no elaborated fixture are required, there is no
variable mass loading of the structure, it is portable and very suitable for measurement in the
field, and it is relatively non-expensive.

The transient window

The duration of the impact is generally small
compared to the duration of the record length.
Special consideration must therefore be taken for the
application of windows. The window to use is a
transient one which take the data unweighted during
the impact and set it to zero the rest of the time. This
window can contain soft transition at the beginning
and the end.

If the hammer is too heavy, the structure can rebound

and produce a double impact of the hammer. A
double hit cannot be used since the spectrum will
contain zero with a spacing where is the time
delay during the impacts.
 6

The double impact cannot be compensated by the window (or any other stuff like that). So the
data must be removed.

The response window

The response to an impact is a free decay of all the modes of vibration. Consider two typical
situations:

 A low damped structure giving sharp resonance that rings for a long time. If the record
length is smaller than the decay time, the measurement will introduce leakage error
resulting in the observed resonance frequency being too low and too broad.
 A heavily damped structure where the response decay very fast and is zero after a short
time. If the record length is longer than the decay time, there will be a poor signal to
noise ratio and the measurement will be contaminated by noise.

The exponential window will handle both situations equally well. It is a function

( )

which add decay to the response with the following effects:

 For lightly damped structure, the window force the response to decay in the record time
and thus the truncation effect is avoided.
 For heavily damped structure, the noise is attenuated by the window
 7

All structures exhibit modal behaviour

An FRF measured on any structures will show a response which is a series of peaks. Each peak
corresponds to a resonance of a single degree of freedom structure. If broader peaks are
analysed with increasing frequency resolution, they will give two or more frequency peaks. So
the structure behaves like it was a set of single degree of freedom structures. This is the principle
of modal analysis which permits to analyse a structure by analysing all its modes of vibration
separately.

Normal modes and complex modes

Modes can be divided in two categories:

 The normal modes are characterised by the fact that

all parts of the structure are moving either in phase
or in anti-phase with the others. The modal
displacements are then real and are positive or
negative. Normal modes can be thought as standing
waves.
 The complex modes can have any phase
relationship. The modal displacements are complex
and can have any phase. Complex mode shapes can
be thought as moving waves with no static node.
 8

The damping distribution of a structure determines if the modes are complex or normal. If the
damping is very light (or inexistent) or distributed in the same way as the stiffness, we can
expect normal mode shapes. If the damping is much localised, the mode shapes will be complex.

Modal coupling
Modal coupling is a general term to express how much a mode shape is influenced by the others
(for a given frequency). There are two types of coupling:

 Light coupling: On a lightly damped structure, the modes are well separated and distinct
(and then lightly coupled). Such structures are known as simple structure (the separation
in several modes is well working).
 Heavily coupled mode: For structures with heavy damping or high modal density, the
FRF does not display clearly distinctive modes. Then the response at any frequency is a
combination of several modes.

Complement of modal analysis

Introduction
The majority of structures can be made to resonate, i.e. to oscillate with excessive amplitude.
Resonant vibration is mainly caused by an interaction between the inertial and elastic properties
of the material within the structure. To better understand any vibration problems, the resonant
frequencies need to be identified and quantified. This is why we use modal analysis.

The modal model

Modes are inherent properties of a linear structure, and are determined by the material
properties (mass, damping, and stiffness), and boundary conditions of the structure. Each mode
is defined by a natural (modal or resonant) frequency, modal damping, and a mode shape (i.e.
the so-called “modal parameters”). If either the material properties or the boundary conditions
of a structure change, its modes will change. For instance, if mass is added to a structure, it will
vibrate differently.

Single degree of freedom

A single degree of freedom system is described in the time domain by

̈ ̇

Transforming to the Laplace domain yields

 9

( ) ( ) ( )

Where ( ) is the dynamic stiffness ( ) . The transfer function between

displacements and forces is the inverse of the dynamic stiffness

( )

The roots of the denominator of the transfer function are the poles of the system. In mechanical
structure, the damping coefficient is usually very small leading to a complex conjugated poles
pair

The frequency response function is obtained by replacing the Laplace variable by

( )
( )

Multiple degree of freedom

Multiple degree of freedom are described by

̈ ̇

where M, C and K are matrices. We can always define the transfer function

( ) [ ]

When the damping is small, the roots of this function are complex conjugated poles and .
So the transfer function can be written in a pole-
residue form

( ) ∑

It can be shown that the matrices are of rank 1

and then can be express as

with a vector representing the mode shape of

mode m. So we can conclude that the transfer
function of time invariant MDOF system is a sum
of SDOF transfer function. Then the full transfer
 10

function matrix is fully described by the modal parameters, i.e. the poles and the mode shapes
vectors.

Observability and controllability of modes

The DOFs where a mode shape vector equals zero are called nodal points or nodes. In practice,
this means that the force actuator should not be positioned in a nodal point of the modes of
interest. To reduce the risk of missing modes, the number of excitation points can be increased.
The same is true for the response measurements. The number of inputs (excitation points) is
typically in the order of 1 to 10, while the number of outputs (response measurements) can reach
more than 1000 points when using optical measurement equipment such as for instance a
scanning laser Doppler vibrometer.

Monitoring

Maintenance philosophies
The maintenance philosophies can be divided in several categories:

 Breakdown or run to failure maintenance

 Preventive or time-based maintenance
 Predictive or condition-based maintenance

Breakdown maintenance
The principle is to run the machinery to failure and only repair or replace damaged components
before or when the equipment comes to a complete stop. The disadvantage is that the
maintenance team works in unplanned crisis maintenance mode. Without doubt it is the worst
method of maintenance.

Preventive maintenance
The philosophy is to schedule maintenance activities at predetermined time intervals. Here the
replacement or the repairs are made before breakdown. This is a good approach for equipment
which does not run continuously. The main disadvantage is that the maintenance risk to be
done too early or too late.

Predictive maintenance
The philosophy is to scheduling maintenance activities only when a failure is detected. The
mechanical parts are periodically monitored and when a problem appears, maintenance
activities are scheduled. It reduces the need of large inventory of spares. The disadvantage is
 11

that the maintenance work may increase due to an incorrect assessment of the deterioration of
machines.

Principle of predictive maintenance

Predictive maintenance monitors mechanical condition, equipment efficiency and other
parameters and attempts to derive the approximate time of a functional failure. A
comprehensive predictive maintenance program utilizes a combination of the most cost-effective
tools to obtain the actual operating conditions of the equipment and plant systems.

Predictive maintenance techniques

There are numerous predictive maintenance techniques:

 Vibration monitoring: This is the most efficient technique to detect errors in rotary
machinery
 Acoustic emission: It can be used to continually detect locate and monitor in structures
and pipeline
 Oil analysis: Here, oil is analysed and the occurrence of certain of micro particle in it can
be linked to the conditions of bearings and gears.

Vibration analysis
It can identify improper maintenance or repair practices. This includes improper bearing
installation, shaft misalignment or imprecise rotor balancing.

Ultimately, vibration analysis can be used as part of an overall program to significantly improve
equipment reliability. This can include more precise alignment and balancing, better quality
installations and repairs, and continuously lowering the average vibration levels of equipment
in the plant.

Acceleration transducer
Accelerometers are the most popular transducer used for
rotary machinery applications. They are rugged, compact,
lightweight transducers with a wide frequency response
range. Accelerometers are extensively used in many
condition-monitoring applications. Components such as
rolling element bearings or gear sets generate high vibration
frequencies when defective. Machines with these
components should be monitored with accelerometers.
 12

Accelerometers are designed to be mounting on machine cases. This can provide continuous or
periodic sensing of absolute case motion in term of acceleration.

Accelerometers are inertial device which translate mechanical motion into an electric signal. The
signal is proportional to the vibration acceleration. The transducers use the piezoelectric effect.
Inertial measurement devices measure the acceleration relative to a mass using the third law of
Newton (action-reaction). The accelerometer is composed with a piezoelectric crystal and a
small mass normally enclosed in a protective metal scale. When it is subjected to vibration, the
mass exerts periodic forces on the piezoelectric crystal which is directly proportional to the
vibration acceleration. The charge output is measured in pico-coulomb per g (where g is the
gravitational acceleration). A charge amplifier converts this charge in a voltage output (mV/g).

It is important to know the different mounting methods for this vibration sensor. There are four
primary mounted methods: stud mounted, adhesive mounted, magnetic mounted and non-
mounted. Stud mounted provides the widest frequency response and the most secure and
reliable attachment. The three others reduce the upper frequency range of the sensor. In general,
inserting mounting piece (adhesive, magnet…) add a mounted resonance. This one is lower than
the transducer resonance and then reduces the frequency range of this last one.

The stud method is performed by screwing the sensor in a stud or a machined block. The
mounting location for the sensor must be clean and paint-free. The surface should be spot-faced
to achieve a smooth surface. The spot-faced diameter must be a little bit larger than the diameter
of the sensor. Any irregularities in the mounting surface will lead to error in the measurement.

The adhesive mounting provides a secure attachment without extensive machining. But in this
case, replacement or removal of the sensor becomes very hard. Here also the surface must be
clean.

The magnet mounting method is typically used for temporary measurement. The sensor could
be inadvertently moved and the multiple surface and material of the magnet can interfere with
high frequency signal.

Displacement probes – Eddy current

transducers – Proximity probes
Eddy current transducers are the preferred
vibration sensors for vibration monitoring on
journal bearing equipped rotary machines. They
are the only transducers which provide
 13

displacement of shaft or shaft-relative vibration measurement.

An eddy current sensor is composed with a probe, an extension cable and an

oscillator/demodulator. A high frequency radio signal is produced by the
oscillator/demodulator. This is send by the extension cable and radiated from the probe tip.
Eddy currents are generated on the surface of the shaft. The oscillator/demodulator
demodulates the signal and provides a modulated DC voltage where the DC portion is
proportional to the gap and the AC portion is proportional to the vibration. Thus Eddy current
transducer can be used both for radial vibration and distance measurement.

On small machinery, one Eddy current transducer per bearing is adequate. For bigger
machinery, is is recommended to use two transducer per bearing. The probes for this mounting
are applied 90° apart from each other.

Eddy current transducers are also sensitive to the shaft smoothness for radial vibration. A
smooth surface with a diameter approximately three times the diameter of the probes is needed.

Phase measurement system

A very important aspect of the vibration wave is the phase relationship. The phase difference
between two waveforms is utilized for many different applications like machinery defect
analysis. It must be noted that in vibration analysis, phase measurement are used in conjunction
with analyser. There are several phase analysers:

 Photocells
 Electro-magnetic and non-contact pick up

Photocells
A photocell detector responds to the reflectivity of
the target. One common way is to wrap the shaft
 14

with black tape and then stick a thin reflective across the tape (or paint a white line on the tape).
The purpose is to provide an abrupt change in the reflectivity of the target area of the photocell
during each rotation of the shaft. The photocell is somewhat similar in principle to the
stroboscope. A steady light source is transmitted to the device and the photo detector provides a
pulse each time light is reflected from a reflective surface.

Electromagnetic detector
To use this device, the shaft must have a notch, a depression, a key or a
keyway. A temporary method is to attach the key on the shaft with hard
tape to hold the key in place (not recommended for high speed shaft). In
electromagnet pick up sensor, the output voltage change to indicate that
the reference feature has passed. This output voltage is then compared to
the occurrence of maximum vibration amplitude to determine the phase
difference at different locations.

Orbits
Orbits are Lissajous patterns of time domain signals that are simultaneously plotted in the X–Y
coordinate plane of an oscilloscope or vibration analyzer. In this form of display, it is very
difficult to trace the start of the orbit as it appears to be an endless loop. In order for us to
determine the direction of rotation, a phase trigger is employed. The trigger will show the
direction of rotation by looking at the dot on the orbit as the starting point of 1xrpm and the
blank space as the end point.

Orbit analysis is the vibration measure of any rotor system in an X–Y plot. In most applications,
the unit of measurement is the displacement which is measured directly using proximity probes.
These types of measurements are relative vibration readings. Relative readings are considered
vibration measurements of the shaft with respect to the bearing housing. As the probes are
clamped firmly to the housing, there is no relative motion between the probe and the housing.
Thus, the orbit is achieved. With that in mind, orbit plots give a visual graph of the actual shaft
centerline movement inside the bearing housing.

Orbital analysis
To proceed to an orbital measurement, we measure the vibration in the x direction and in the y
direction. They give us

̈ ̈
 15

By defining

We find

̈( ) ( ) ( ) ( )

If , we find a forward orbit

But if it is not the case, we will find

( ) ( ) ( )

which is the equation of two orbits (one forward and one backward). So we see that all the
rotational vibration can be divided into two complex orbits with different amplitudes.

Complement of orbital analysis

Forward precession means that the shaft is vibrating
or whirling in the same direction as the rotation.
Backward precession means that the shaft vibrate in
the inverse direction than the rotation.

The orbit plot will be the same for the forward and a
backward precession except for the blank-bright
(forward) of bright-blank (reverse).
 16

Orbital frequency component

The presence of the phase reference on the orbital plot says us how many rotation take one
vibration of the shaft.

If we have only one mark on the Lissajou plot (on blank-bright),

we can determine the frequency of the vibration by counting
the number of loop in the plot. If we look at the figure on the
left, we see that the plot present one loop. So the frequency of
the vibration will be two times the frequency of rotation of the
shaft. If we have two loops, it will be three times and so on. We
see that the frequency is given by

where is the number of loops and is the number of

rotation. This is what happens when the loop are inside. But if the loops are outside, the
frequency of the vibration will be given by

So for two loops, we have a frequency which is one times the frequency
of rotation.

When there are several marks (blank-bright) on the plot, we cannot use
these formulae. But we can determine the frequency of vibration by
analysing the number of mark. Indeed, if there are several marks on the
plot, this means that the complete vibration takes several time of
rotation. If we analyse the figure of the right, we see that the vibration need two rotations of the
shaft to accomplish on complete vibration cycle. So the frequency of vibration will be half the
frequency of the shaft rotation. We can also see that external loop means that the backward
precession is dominant and that internal loops means that the forward precession is dominant.

Cascade plot
A cascade plot is a representation of three parameters: amplitude, frequency and machine
speeds. An FFT plot of amplitude vs frequency is recorded at specific machine speed intervals
(selected by the user). After collection of all the FFTs, they are cascaded one after another in a
form similar to a waterfall plot.
 17

It is important to note that waterfall plot is the FFT of the same location collected at different
time intervals. A cascade plot is a collection of FFT’s at different machine speeds and is taken
during transition of a machine speed. Cascade plot is a tool for transient analysis which forms
an essential diagnosis tool for critical machinery.

Full spectrum
The full spectrum is an additional diagnostic tool and is also called the spectrum of an orbit. It
shows the same information than an orbit but in another format. It helps to determine the
degree of ellipticity associated with the various machinery conditions along with the
precessional direction for all the frequency components present.

To obtain the full spectrum, the orthogonal X and Y transducer signals are fed into the direct
and quadrature parts of the FFT input. The positive and negative vibration components for each
frequency are obtained. Positive is defined to be forward precession.

One of the possible applications of full spectrum is analysis of the rotor run out caused by
mechanical, electrical or magnetic irregularities. Depending on the periodicity of such
irregularities observed by the X–Y proximity probes, different combinations of forward and
reverse components are observed. The method forms the basis for many useful machinery
diagnostics.

Cepstrum analysis
The one characteristic of vibration spectra common to anti-friction bearings or gears is that there
is a kind of harmonic series with the possibility of multiples of the fundamental bearing tones
and/or rotational rate (frequency of defect or harmonic, thus add/subtract 1xof the shaft). This
can be described as a common frequency spacing separating the peaks of signature groups.
 18

Cepstrum analysis is the name given to a range of techniques involving functions which can be
considered as the spectrum of a logarithmic spectrum. It converts a spectrum back to the time
domain, and hence has peaks corresponding to the period of the frequency spacings common in
the spectrum. These peaks can be used to find the bearing wear frequencies in the original
spectrum.

Spectrums from a rotating machine can

be quite complex, containing several
sets of harmonics from rotating parts
and there may be several sets of
sidebands due to frequency
modulations (changes in frequency
mostly due to torsional oscillations).
Because cepstrum has peaks
corresponding to the spacing of the
harmonics and sidebands, they can be
easily identified.

Significant peaks in the cepstrum

correspond to possible fundamental bearing frequencies. Using a set of embedded rules, an
expert system can automatically compare these frequencies to the peaks in the spectrum that are
not related to any machine fundamental forcing frequency. If a match is found, then the spectral
peak is considered to be a possible bearing tone, and it is passed to another part of the expert
system for rule based decision-making.

Through cepstrum analysis, the expert system has the advantage of detecting rolling contact
bearing wear without knowing exactly what type of bearings the machine uses. Cepstrum also
distinguishes bearing wear patterns from flow noise or cavitation.

Unbalanced
Vibration due to unbalance of a rotor is probably the most common machinery defect. It is
luckily also very easy to detect and rectify. The International Standards defines unbalance as:
That condition, which exists in a rotor when vibratory, force or motion is imparted to its bearings as a
result of centrifugal forces. It may also be defined as: The uneven distribution of mass about a rotor’s
rotating centreline.
 19

When they are apart, the rotors will be unbalanced. There are three types of unbalance that can
be encountered on machines, and these are:

 Static unbalance
 Couple unbalance
 Dynamic unbalance

Static unbalance
For all types of unbalance, the FFT spectrum will shows a 1xrpm predominant frequency of
vibration. The amplitude of this vibration will be proportional to the square of the rotational
speed.

Static unbalance will be in-phase and steady. If the pickup is moved from the vertical to the
horizontal direction, the phase will shift by 90°.

Couple unbalance
Couple unbalance tends to 180° out of phase on the same shaft. Note that almost a 180° phase
difference exist between two bearings in the horizontal plane (the same in the vertical one).

Misalignment
It is a major cause of machinery vibration. There are basically
two types of misalignment:

 Angular misalignment: the shaft centerlines of the two

shafts meet an angle to each other
 Parallel misalignment: the shaft centerlines are parallel
to each other and have an offset.

Angular misalignment
Angular misalignment subjects the driver and driven shaft to
axial vibration at 1xrpm frequency. A pure angular
misalignment is rare. Thus misalignment is rarely seen as
1xrpm peak. Typically there will be high axial vibration with both 1xrpm and 2xrpm.

A 180° phase difference is observed when measuring the axial phase on the bearing of the two
machines across the coupling.

Parallel misalignment
Parallel misalignment results in two hits per rotation and then a 2xrpm vibration in the radial
direction. Like for the angular one, pure parallel misalignment is rare. So we will observe both
 20

1xrpm and 2xrpm peaks in the spectrum of the vibration. If the 2xrpm is predominant, then the
parallel misalignment is predominant. If it is the 1xrpm, then it is the angular one.

When either parallel or angular misalignment becomes severe, it can generate high amplitude
peaks at higher harmonics. Or even a whole series of high frequency harmonics.

Looseness between machine to base plate

This problem is associated with loose pillow-block bolts, cracks in the frame structure or the
bearing pedestal. Higher harmonics are generated due to the rocking motion of the pillow block
with loose bolts.

Oil whirl
Oil whirl is an oil film-excited vibration. It is known to occur on machines equipped with
pressure-lubricated journal bearings operating at high speeds (beyond their critical speed).
Consider a shaft rotating in a bearing at speed N. The bearing speed is zero. The oil film is
wedged between the shaft and the bearing and should ideally rotate at a speed of 0.5xrpm.
However, some frictional losses cause the oil film to rotate at 0.42–0.48xrpm.
 21

Under normal circumstances, the oil film pushes the rotor at an angle. An eccentric crescent-
shaped wedge is created that has sufficient pressure to keep the rotor in the lifted position.
Under normal conditions, the system is in equilibrium and there
are no vibrations.

Some conditions would tend to generate an oil film pressure in the

wedge much higher than required to just hold the shaft. These
conditions can cause an increase in bearing wear resulting in the
shaft to have lower eccentricity (the shaft center is close to bearing
center) causing a reduction in stiffness, oil pressure or a drop in oil
temperature. In these cases, the oil film would push the rotor to
another position in the shaft. The process continues over and over
and the shaft keeps getting pushed around within the bearing. This phenomenon is called oil
whirl. This whirl is inherently unstable since it increases centrifugal forces that will increase the
whirl forces.

Oil whirl can be minimized or eliminated by changing the oil velocity, lubrication pressure and
external pre-loads. Oil whirl instability occurs at 0.42–0.48xrpm and is often quite severe. It is
considered excessive when displacement amplitudes exceed 50% of the bearing clearances.

Oil whirl is basically a sub-synchronous fluid instability. When viewed in the orbit domain, it is
shown with the characteristic two dots. When viewed with an oscilloscope, the two dots do not
appear stationary, but seem to be rotating instead. This is because the frequency is marginally
less than 0.5x rpm. An oil whirl phenomenon generates a vibration precession, which is always
forward.

Oil whip
Oil whirl can be caused when the shaft has no oil support, and can become unstable when the
whirl frequency coincides with a critical speed. This special coincidence of shaft resonance
coupled with the oil whirl frequency results in a more severe form of oil whirl called oil whip.
Whirl speed will actually lock onto the rotor critical speed and will not disappear even if the
machine is brought to higher and higher speeds.

The oil whip phenomenon occurs when the rotor is passing through its critical speed. Oil whip
is a destructive bearing defect. The precession of vibration is in the forward direction in this
case, but some reverse 1xand sub-synchronous components are present due to anisotropy of the
bearing pedestal stiffness.
 22

The period of this self-excited defect may, or might not, be harmonically related to the rotating
speed of the shaft. When it is not harmonically related, the dots appear to be moving randomly.
When it is harmonically related they appear stationary.

Rolling element bearings

A rolling element bearing comprises of inner and outer races, a cage and rolling elements.
Defects can occur in any of the parts of the bearing and will cause high-frequency vibrations. In
fact, the severity of the wear keeps changing the vibration pattern. In most cases, it is possible to
identify the component of the bearing that is defective due to the specific vibration frequencies
that are excited. Raceways and rolling element defects are easily detected. However, the same
cannot be said for the defects that crop up in bearing cages. Though there are many techniques
available to detect where defects are occurring, there are no established techniques to predict
when the bearing defect will turn into a functional failure.

Acoustic
Fundamental concept of acoustic
When an external mechanical excitation is applied on a material, a liquid or a gas, vibrations are
induced in it. The molecules of the medium vibrate around an equilibrium position. If this
phenomenon occurs in a solid or a liquid, we talk about vibrations. The term sound is used if air
is the medium, as long as it can be perceived by the human ear. The reason for this sound
production is the vibration of the instruments which create over and under pressure in the air.
Note that the order of magnitude is really small compared to the atmospheric pressure. Sound
we propagate like a wave in the medium and it is then characterized by a amplitude, a
frequency and a wave velocity. The frequency range of the sound is from 20 Hz to 20 kHz.
Under the lower limit we talk about sub-sonic wave and over the upper one, we talk about
super-sonic wave. Sound wave are longitudinal wave (the particle displacement is parallel to the
wave propagation).
 23

Plane sound wave

A plane sound wave is an oscillating system of plane regions with alternating over and under
pressure. The wave front propagates in a direction normal to the flat planes. In a given point, the
density is depending of the time and of the position

( )

We define the following parameters: the fluctuating pressure (in a given point the pressure is
given by ), the particle displacement and the particles velocity (which is not the
velocity of the wave but the vibrating velocity of the particles). The one dimension equation of
the sound wave is

And the harmonic solution describing the wave propagation is

( ) ( ) ( )

Here we consider an amplitude independent of the time and the position, so this model is valid
only for undamped propagation. One can show that the damping is given by

and then for high frequencies, we need to take into account the damping

( )

We can also prove the following relationship

The speed of sound

The speed of sound is the velocity at which the wave front propagates into the medium. Then it
depends on the properties of this medium (especially density and elasticity). In a gas, the speed
of sound is given by

In a liquid it is given by

√
 24

where B is the compression or bulk modulus. And in a solid it is given by

The one dimensional wave equation

We define s as the condensation, i.e. the relative change in density

Here we do not consider the gravitation and so the density and the atmospheric pressure are
constant. The gas or fluid is assumed to be homogeneous and isotropic, i.e. there is no
dissipative force due to viscosity or heat conduction. If we apply the mass conservation
principle to a volume between x and x+dx, we find

( )

And by substituting the definition of the condensation, we find

( )( )

And then

Secondly we can use thermodynamic change of state (adiabatic)

( )

If we consider a very small condensation, we can assume that and then with the
definition of the speed we find

But we say that the movement of the particle is harmonic and then we find that
. The fundamental equation of dynamic gives us
 25

If this force is generated by a difference of pressure, we find

And then if we fill in p (given by previous equation), we refind the wave equation. In harmonic
wave, the displacement, the particle velocity and the pressure varies periodically in time and
space and all of them satisfy the wave equation. But in acoustic, we prefer to focus on the
pressure and then we use the following equation

And so the pressure will be given by ( ) ( ).

Acoustic impedance of a medium

The specific acoustic impedance of a medium is defined as the ratio of the sound pressure over
the particle velocity. It is then given by

[ ]

Where is the compression modulus of gases.

Spherical and cylindrical sound wave

Let us take a small spherical surface which all points move radially in a periodical way, with
same amplitude, frequency and phase, around equilibrium position: this is the monopole or
isotropic radiator. The surface will exert a periodic pressure on the fluid in contact with that
surface. Consequently the perturbation of fluid equilibrium will propagate radially in the shape
of spherical waves. The sound wave will be given by

( ) ( )

For a cylindrical sound wave, we have

( ) ( )
√

An important conclusion concerning the three simple types of sources is the following:

 For plane sound waves the sound pressure does not decrease with the distance.
.

 For spherical sound waves the sound pressure decreases linearly with increasing
.

distance.
 26

 For cylindrical sound waves the sound pressure decreases proportionally to the square
root of the distance.

The effective sound pressure

Suppose a given source producing sound, i.e. the quickly fluctuating air pressure makes our
eardrum to vibrate which causes through our ear and nervous system a sensation of sound. One
could think that if we have the impression of a constant sound level, intensity or loudness it
implies a constant sound pressure in time. Nothing appears to be less true. The sound perceived
with a constant loudness may be both a pure sine tone and a stochastic sound. A first
conclusion: the instantaneous value and the
algebraic mean value does not matter (this latter is
eventually zero). The human hearing system is
quite insensitive for sharp positive and negative
peaks, which may be cut off. In contrast, it seems to
be sensitive to the energy of sound waves. This led
to the consideration of the effective or Root-Mean-
Square (RMS) value of the sound pressure, over a
certain time interval, as a measure of intensity:

√ ∫ ( )

The dB-scale
The dB-scale allows us to describe the sound like it is perceived by the human hearing system.
In this way, the sound pressure level is defined as

with .

Superposition of two sound

When two sound are superposed, the resulting instantaneous pressure is given by the sum of
the two pressure ( ) ( ) ( ). But it is not interesting because the human don’t perceive
the instantaneous variations. A more interesting parameter is the effective value of this sound
pressure which is given by

√ ∫ ( ( ) ( )) √ ∫ ( ( ) ( ) ( ) ( ))
 27

Two case can be distinguished: the incoherent source (where ) and the coherent
source ( ). For the first one, we will have . Incoherent sources
can be two pure sine tones with different frequencies, two independent stochastic sounds or to
source with energy in different frequency band. In the case where we have n sources with the
same pressure level and are incoherent, the total effective pressure is or in dB-
scale . In the case of coherent sources, the pressure level depends on the
phase delay between the sources. There are two extreme cases: in phase and in opposition of
phase. In the first one we have or . In the second case, we
will have . Considered apart, the two sources are audible, but together no sound is
generated. This property is used in active sound control.

Types of sound
Based on the frequency spectrum some types of sound can be distinguished: .

 Pure tone: A sound characterized by only one frequency and can only be generated,
approximately, by a tone generator
 Musical tone: It consists of a fundamental with overtones called partials. (Harmonics are
.

partials). The number and the nature of the harmonics define the so-called tone color.
 Chaotic or stochastic sound: (noise, hiss, etc.). It covers a wide frequency spectrum. In
.

acoustics a very detailed analysis will not be performed (cost) but an analysis in 1/1
octave band or 1/3 octave band. These are normalized.
 Impulse noise: is a type of sound which is of very short duration, mostly generated by an
.

impact. Sound can also be classified in other ways like considering the change of
amplitude in function of time.
 28

The acoustic intensity

The intensity is the sound energy that propagates in one direction and is incident on one square
meter per second (sound power per unit area). It is important to note that the sound intensity is
a vector. So it is possible to determine the preferential direction of the sound with it.

The sound intensity level is defined by

with .

The source power

The sound power is the acoustic power delivered by a sound source . The sound
power level is defined by (with ).

The human hearing system

 29

The human ear comport three parts: the outer, middle and inner ear.

The external ear contains the auricle (pinna). The use of both ears allows us to determine the
direction of the sound (measure of the phase delay). The sound perceive by the auricle is
translate into a vibration of the eardrum.

The middle ear contains the hearing ossicles: hammer (malleus), anvil (incus) and the stirrup
(stapes). The middle ear is connected to the nasal cavity by the Eustachian tube to have the same
pressure as outside. The ossicles are responsible for the transmission of the sound wave from the
eardrum to the oval window of the inner ear.

The inner ear contains the half circular channel which provides the balance and the cochlea
which accounts for the hearing function. The cochlea consists of a spiral shaped cavity that is
divided into two channels by the cochlear tube: the two channels are filled with a liquid, the
perilymph and are connected at the end of the spiral. The upper channel starts at the oval
window and the lower channel ends at the round window. When a sound wave arrives on the
eardrum, the ossicles pass the motion to the stirrup that compresses the oval window and
creates a pressure wave in the upper channel. This wave propagates farther in the upper channel
as the tone of the sound decreases. The transversal component of the wave exerts its force
directly on the cochlear duct where the organ of Corti is located. This is the actual organ that
serves to perceive sound. The membrane that separates the upper channel from the cochlear
duct is compressed and gives rise to a pressure wave in the liquid of the cochlea (the
endolymph), which in turn compresses the basilar membrane on which the organ of Corti is
situated. The organ of Corti consists of hair cells internally arranged in a row and externally
arranged in 3 to 5 rows. The hairs of the cells are in direct contact with a heavy membrane,
called the tectorial membrane. When the basilar membrane is compressed, the contact of the
hairs with the tectorial membrane will be lost. Every time the contact is broken or recovered, the
electrical potential of the cells is changed. The changes in the electrical potential are transmitted
to the brain via the fibers of the cochlear nerve. In the brain they are decoded and converted into
a perception of sound.

Measuring sound
In order to select a useful measuring technique we need to determine what the purpose of the
sound measurement is. A first important objective can be to determine a sound pollution
problem. To do this extend, measurements of the sound pressure level are usually sufficient. In
such a case the availability of a simple, portable measuring system is desirable. It the hindrance
is momentary, one wishes a swift registration of the peak levels. For a long observation, an
automatic averaging and statistical processing is advised and most of the times required by the
 30

legislator. A second objective can be the reduction of noise after the confirmation of nuisance.
One speaks of sound sanitation. Mostly, there is a need to examine the frequency spectrum of
the sound to accomplish this task. To this extend specific measuring devices are developed.
Measurements in frequency bands give the general picture of the composition whereas the
linear spectrum can be an important aid in localizing the source of the sound. A third objective
of sound measurements is to investigate if the norms and regulations concerning noise pollution
are not violated. In this case, the measuring parameters as well as the measuring conditions are
often prescribed.

The measurement microphone

A measurement system to measure sound pressure consists of a microphone, a signal amplifier,
a conditioning unit and a measuring device. The microphone converts the vibration of the air
into a mechanical vibration which is converted into an electric signal. This last one is amplified
(and sometime filtered) and then translated into the measuring device. The microphone is the
critical part of every measuring system. There are several types of microphones:

 Ceramic or piezoelectric microphone: They are based on the principle of piezoelectricity

(the piezoelectric crystal provides an electric signal when it is compressed). This type of
microphone is robust and not sensitive to moisture and other environmental impact.
They are also cheap and no external source is necessary.
 Condenser microphone: They use two
electrically charged plates with an air gap
in between. One of the plates is a thin
membrane which moves due to the
incoming of the sound pressure. The
moving of the membrane induces a
modification of the capacity of the
condenser created by the two plates. This
results in a variation of the voltage
difference between the two plates. This
last one is converted into an electric
signal. This type of microphone is relatively insensitive to temperature change.
 Electret microphone: An electret is a polymer film with an electrical charge bound to the
molecules. An electret microphone is made by applying an electret on a perforated metal
plate and shields it off on the front side with a plastic membrane on which a thin metal
coating is applied. Incoming sound wave alter the capacity of the condenser. This leads
 31

to an electrical current. The electret microphone does not need an external polarization
(which is the case of the condenser one).

Condenser measuring microphones exist in a number of standard sizes, the core number to
describe the size of the microphones is the diameter of the microphone in inch: one distinct the
1/8, 1/4, 1/2 and 1 inch microphones. If the wavelength of the sound to observe is about the size
of the diameter of the microphone, the sound pressure will be partially averaged out over the
surface of the membrane and the microphone loses a lot of its sensitivity. To measure high
frequency, a small microphone must be used.

The sound field where measurements take place

The sound field has influence on the measured sound pressure. To keep this influence as low as
possible, different types of microphone are designed:

 The free field microphone: The free field is defined as an area where no reflected
sound waves are present. This microphone will compensate the influence of the
microphone on the free field. The highest accuracy is obtained when pointing the
microphone to the source.
 Random incidence microphone: For measurements in a diffuse field another
microphone is developed that compensates incoming sound pressure from all directions.
 .

 Pressure microphone: A pressure microphone gives a constant frequency response of

the sound field, the way it exists, including the influence of the microphone itself (no
compensation is carried out). These microphones are useful when the sound pressure on
the side of a cavity is to be measured (e.g. exhaust systems).

Influence of the wind speed

It is known that wind in the vicinity of the microphone
produces additional noise and then pollutes the signal.
Thus in presence of wind, it is advised to use a wind
screen which is essentially transparent in the frequency
range of interest. The energy of the noise caused by the
wind is the highest at low frequencies. At those wind
speed, measurement outside are better postponed.

Division of sound in frequency bands

The sound pressure level of the human hearing band (20 Hz-20 kHz) can be measured in a
number of consecutive frequency intervals called frequency bands. Mostly one uses octave or
fractional of octave bands. The link between two consecutive octaves is that the middle
frequency of the second one is the double of the middle frequency of the first one. Third octave
 32

bands are obtained by divided the octave bands into three separated bands. The following
relationship is given for these bands

These give the center, the lower and the upper limit of the band.

Frequency weighting of microphone signal

Mainly, we will use an average of the different
frequency bands to minimize the cost of the
measurement. To correspond to the human
perception of the sound, one will weigh the desire
sound in the frequency range. In this goal, different
types of filters exist. The A-weighted sound level is
accomplished by adjusting the sound level in every
frequency bands to the frequency sensitivity of the
human ear for soft sound (40dB). The B and C
weighted sound level is defined in the same way but
for average and loud noise. The D weighted level is for very loud signal (e.g. for airplane).

The sonometer
Sound level meters are the basic equipment for direct measurements of the sound level, if one is
not interested in the frequency spectrum. It can be simple a instruments that can be held in the
palm of one’s hand and work on batteries. These instruments can easily be used on a site where
a noise problem might be present.

A typical device consists of a microphone, a preamplifier, weighting networks, an amplifier, an

RMS rectifier and a meter that displays the sound level in dB. A switch allows to choose
between an A-, B-, C-, D-weighting or no weighting at all. Finally the rectified signal is
converted in dB and send to an analog or digital display instrument.

According to their performance, three types of sonometer are specified by standard commission:
type 1 (precision device), type 2 (device for general use) and type 3 (inspection device). A
laboratory reference device is called a device of type 0.
 33

Calibration of the measurement systems

A microphone is usually accompanied with a calibration chart upon delivery. On this map the
frequency sensitivity is mapped. There exist numerous
calibration methods for microphones or measuring systems
as a whole. One has recorded that when measuring sound
pressure levels, the best results are obtained when a
pistonphone is used for the calibration of the system.
Pistonphone (piston calibrator) consists of an engine that
moves a few pistons back and forth. The calibration is done
by placing the junction piece of the pistonphone over the
microphone and switching on the device (are you sure??).

Sound exposure level

The sound exposure level is used to characterize a single event, both in time and sound level.
The SEL is defined by

(∫ )

with T the time measurement in second. The SEL can also be measured by an integrating sound
level meter. A dosimeter or noise exposure meter is an instrument that is designed to measure
the accumulated noise exposure of workers in an industrial environment (like dosimeters exist
for radiation). The dosimeter is a compact device with an integrating sound level meter that can
be worn by workers during their normal activities at work. Usually the dosimeter has an
internal memory to track the sound exposure of several workers. Apart from the sound
exposure levels in dBA, the percentage of the allowed level and the peak level is displayed. Also
the data and duration of the measuring period are registered.

The intensity meter

To measure the intensity, we need to measure both the sound pressure and the particle velocity.
Indeed, the sound intensity is given by

where is the energy of the sound. But if the sound pressure can be measure simply with a
microphone, it is most difficult to measure the particle velocity. This vectorial quantity can be
measure by a derived quantity
 34

In reality, the derivative of the pressure is computed by discretization.

( ) ∫( )

where A and B are two neighboring location. This discretized equation is used to calculate the
sound intensity with the aid of the intensity meter. This measuring device consists of two
microphones spaced out over a fixed distance with a so called spacer.

There are a few issues that require attention when using an intensity meter however:

 For a fixed distance between the microphones, the intensity meter has a rather limited
frequency range where the measurements are valid. At low frequencies the value
derived from the theory is still correct, but noise present in the measurement will result
in an incorrect measurement.
 The direction of the probe is of great importance. To measure the power of the source,
the intensity probe has to be held perpetual on the defined surface at all times (this is not
the case for an intensity meter that is not directional).
 ..

 The cost of an intensity meter is much higher than that of a sonometer. The calibration of
the probe is also a lot more devious (the two microphones must be matched perfectly.

Measures in an anechoic half-space

An anechoic half-space is a room that is confined by a hard surface that reflects the sound
waves. Along all other walls there is in principle no material border or the sound waves are
completely absorbed (and not reflected). This anechoic half-space is realized outside on a hard
concrete surface, or inside in an acoustical dead room with a hard floor. The measuring points, n
in number (6 or 8) are placed at a distance far enough from the source to guarantee that the
measurements are conducted in the far field. In practice this means that the measuring points
are placed at a distance of at least three wavelengths from the source. Moreover the measuring
points spaced out over the half-space (on a half spherical surface) in such a fashion that every
point is concerned with an equal part of the surface. If the machine radiates non-omnidirectional
sound, the various microphones will not measure equal sound pressure. One will therefore take
into account the spatial average effective pressure in the expression of the power W.
 35

Measurement in a full anechoic room

The more fundamental measurements are conducted in an acoustic dead room, where the
microphones are placed over a complete spherical surface surrounding the source. In this case
the sound level is given by:

The comparison method

In the comparison method one uses a reference source which is normalized. The test is
conducted in a so called acoustic hard room, since one does not usually have access to an
acoustic dead (anechoic) room. The reference source can generate for example a power .
Firstly, one has to install multiple microphones spaced out in the room and not to close to the
machine (in the far field). One measures the sound pressure levels and the spatial averages
of the sound radiated by the unknown source. Secondly, one then replaces the machine
with the reference source by the tested source and one measure in an identical fashion
(microphones on identical positions). Then, the unknown power can be calculated from:

( )

Measure the power with an intensity meter

Due to the fact that the measuring locations for the described comparison methods are located in
the far field, that method is not applicable when different objects are placed close to each other.
In this case one can use an intensity meter to measure the sound power. The following
procedure must be used to do this:

 Define a random surface that includes the sources.

 Measure the intensity in n discrete points of the surface
 Calculate the power level:

Instead of discrete measuring locations an alternative method can be used where the intensity
meter scans in a continuous fashion over the defined surface.
 36

Sound absorption
All materials have the property of sound absorption. It often happens that too much sound is
reflected and then we can hear echoes. One can decide to recover some part of the wall with
materials which better absorb sound to decrease the echoes.

Acoustic transmission between two media

In practice it never happens that an acoustic wave propagates in only one medium. It always
moves from one medium to others. If we consider a plane sound wave which impinges on a
surface between two media ( ). A part of the wave will be reflected and another will be
transmitted (absorb) . We can write

If we use the non-dimensional coefficients and , we find that . There

are two conditions which must be fulfilled at the interface between the two media ( ). The
first one is that the sound pressure of one side must be equal to the one of the other side

( ) ( ) ( )

The second one is that the normal velocity of the particle in the medium of one side must be
equal to the particle velocity of the other side

( ) ( ) ( )

We can rewrite this last one with the acoustic impedance

( ( ) ( )) ( )

By combining with the other formula derived a little bit earlier, we find

( ) ( )
( ) ( )

The coefficient of absorption will be given by the definition of the acoustic impedance

( )
( ) ( )

We can see that if the two impedances are equal, the sound wave will be totally absorbed by the
second medium ( ). But if there is a big difference between the two impedances, then the
sound wave will be totally reflected whatever which impedance is the big one. But it does not
 37

mean that the two cases ( ) are the same. Indeed, if we assume that the sound
pressure has the following form

( ) ( ) ( )

It is a combination of an incident and a reflected wave. In the first case, we will have and
then ( ) . In the second case, we have and then ( )
. So we see in the figures (supra) that the two waveforms will have a different
form at the interface.

So to conclude, we can say that a good absorbing material must have acoustic impedance which
is closed to the acoustic impedance of the air. But it is not easy to find material like that.
However it exists other solutions: plate on an air layer, Helmholtz resonator, porous material…

Plate on an air layer

A plate on an air layer belongs to the category of resonant absorption means. One fixes plate
(with wooden slats or profiled irons) at a distance of some centimeters in front of a hard wall.
The plate and the air layer, between it and the wall, constitute a mass-spring system. Thus its
model will be

̈ ̇ ̇ ∫

This last expression using for a harmonic solution gives

And by definition of the impedance ( ), we find

 38

One can show that the stiffness of the air layer is given by a function of the atmospheric pressure

and the distance to the wall: . Then the resonant frequency of the system is √ .

When a sound wave hits the panel, it vibrates at the frequency of the sound wave. If this
frequency is in the vicinity of the eigenfrequency of the panel, this one will vibrate strongly in
the air. Then a lot of friction losses will occur which cause the loss of vibrational energy, i.e. the
sound energy is absorbed. We can adapt the distance to the wall and the mass of the panel to
correspond to the need. We can also modify the damping coefficient if needed.

Helmholtz resonator
By applying a large number of holes in a panel, its properties are change. It will not only be the
panel which will vibrate but also the masses of air in the holes. These small masses will resonate
on the air spring behind it. It is the so-called Helmholtz resonator. The realization of the noise
absorption will follow the same principle as for the plate without hole. The formulations will
stay, but the mass does not have the same signification in this case. In practice, the resonant
frequency will be seven to ten times higher than for the simple plate. Moreover, the bandwidth
in which the sound can be absorbed is bigger.

Porous acoustic absorbing material

Consider a porous material which consists of fibers (diameter 2 to 20 μm) that are random
oriented, and bonded to each other at their contact points using a resin. The porosity of material
used in sound absorption is very large: the ratio of pore volume to total volume amounts to
about 95 to 98%. The pores are all in connection to each other, and one may therefore say that
the air in these pores participates completely to the sound movement if a wave is incident.
Viscous friction losses in the air occur due to the vibrating movement of the air and furthermore
also impulse losses due to the constrictions, dilations and turns along the many fibers. These
losses mainly occur at higher frequencies. Moreover, the air will be alternately compressed and
relaxed and therefore experience temperature fluctuations which will give rise to heat exchange
with the fibers. This creates thermal losses, and these tend to occur at lower frequencies. The
result of all occurring losses is that the compressibility modulus and the propagation velocity
are complex quantities in porous media. From the complex nature of follows that the wave is
damped.
 39

The art of sound absorption consists of adapting as good as possible the value of impedance of
the absorption means to the impedance of the air, within the desired frequency range. In the
selection of a porous acoustic material, the following factors are taken into account:

 The narrower the pores are, the stronger will be the air friction, thus the stronger the
damping, i.e. the absorption. We can measure it by the specific air resistivity (resistivity
to an air flow through the porous material).
 The thickness of the plate is also important. Indeed, in order to get a damped wave, it
must penetrate a sufficient distance in the porous material. We can see that a plate with a
relatively low resistivity and relatively thick lead to good absorption results.
 The frequency of sound is important since the particle velocity depends on it. The
friction losses increase with the speed of particles, i.e. with the frequency.
 The pore structure plays a role. This parameter can be described by the tortuosity.

Measuring acoustic absorption

To measure the absorption coefficient of a material, we will passé by another concept: the
reverberation time. It represents the time needed in order that the sound energy level decreases
60dB.

Consider a space with volume V, and a wall surface S. We consider a sound source in this space,
in which the walls have a mean absorption . When the source is disabled the sound intensity is
. Then the sound dies out, i.e. the sound intensity decrease due to absorption and reflection.
After reflection, the sound intensity is ( ) and after reflections it will be
( ) . The number of reflections is given by

So we have

( ) ( ( ))

So, by the definition of the reverberation time, we have

( ( ))

It gives ( )
. In reality, the different wall of the room will have different absorption
coefficients. Thus we define the mean absorption coefficient ̅ ∑ ∑ . And then, we find
the model of Eyring-Norris
 40

∑ ( ̅)

By approximation of this model, we can find the modal of Sabin which is simpler

We can use this formula to determine the absorption coefficient of a room. Indeed if we measure
the reverberation time of the room, we can find the total absorption A and then determine the
coefficient of absorption by ̅ ∑
.

Favorable reverberation times depend on the types and usage of the rooms: for a furnished
living room (0.5s), for a cinema or lecture hall (0.7-1s), for a theater (0.9-1.3s) or for a music hall
(1.7-2.3s). A too short reverberation time leads to a dry sound i.e. that does not reverberate
because the sound is immediately absorbed. But for some application, the reverberation is
needed for the subjective appreciation of the sound.

Measuring the absorption in a reverberation room

If one needs to find the absorption coefficient of an object, it is necessary to have a reverberation
room at its disposal. This is a room with very hard walls. First, the total absorption of the
reverberation room itself is determined during a first measurement using the formula of Sabine.
Next the object is introduced and a second reverberation time is measure. Then the absorption of
the object is given by

( )

Measuring absorption in a Kundt tube

It is a cylindrical tube with the specimen mounted at one end. At the other end, the speaker is
mounted. In the tube, there is a rod on which a microphone is fixed. Then the sound wave is
measured while moving the rod through the tube. Two major parameters are recorded: the
maximal and minimal pressure. These parameters are used to compute the standing wave ratio

One can show that the reflection factor can be computed from this ratio
 41

And then the absorption coefficient is given by

.

It must be note that this method gives only the

absorption coefficient for a perpendicular
incidence. We note also that the measurements
are only valid for limited frequency range. There is a lower limit which is determined by the
length of the tube and an upper limit given by the thickness of the tube.

The direct and diffuse sound field

The acoustic power produced by a sound source is fully absorbed by the walls of the room and
then

∑ ̅∑ ̅

For a diffuse field, one can show that the pressure level is given

In this way, one can compute the sound level of a given source by knowing the absorption of the
room and the source power.

Two different sound fields exist, with a different propagation behavior. The diffuse field and the
direct field. On the first one, the pressure is uniform throughout the field. This field is only valid
for a certain distance from the source. Close to the source, a direct field is present. In the direct
field, the pressure is given by

Because the two fields occur together in the space, we must take the
two contribution into account and thus

( )

Or ( ).

In conclusion, we can say that at a certain distance from the source, the sound field is amplified
because of the reflections which are due to the partial absorption of the wall.
 42

The separation between the direct free field and the diffuse one is given by the value of the
radius which verify

or √ .

Sound insulation
We must distinguish the airborne sound and the contact sound. The first one is a pressure wave
generated by a source in a room and transmitted to another one through a wall. A contact sound
is a sound transmitted in a room by a contact with the wall of this room (step on the floor).

Measuring sound insulation

We can describe the airborne sound insulation between to rooms by

We note that not only the separation wall play a role in the transmission, but also all the other
boundary surface of this two adjacent rooms. This mode of transmission through all the other
boundaries is called the flanking transmission. Thus we must distinguish the sound insulation
due to the separation dispositive called the sound reduction index and the total sound
insulation between two rooms which is called normalized level difference. It is obvious that
in practice, this last one is enough, but to measure the
quality of a partition construction, the flanking effect
must be avoided. Then a laboratory-only
measurement must be performed. Such a laboratory
is composed with two independent rooms where the
transmission of the sound between the two rooms is
only possible through the structure under test.

The sound reduction index will be given by the loss

of intensity between two rooms

But it is really difficult to measure the sound intensity in the second room because the walls will
contribute to the sound level in the room. However we can use the mean intensity and come
back the real one by the relation of Sabin ̅ . So we have
 43

Or .

Measurement of an impact sound

An impact device is used. It has five small steel
hammers. At impact of the hammer, we can
compute the normalized sound pressure in the room

where .

Airborne sound insulation of a wall

We consider a plane sound wave incident on a wall. We assume that the wall does not absorb
sound, and the wall has no stiffness and no damping. The wall is characterized by its mass per
unit area. On the source side, one can write that the velocity of the particle must be zero because
the wall does not move and does not absorb sound. We see that if the wall does not absorb
sound, the sound pressure on the wall is doubled and thus applying the formula of the dynamic,
we find

In practice, we have because the pressure on the transmission side is negligible

compared to the one of the incidence side. If we assume a harmonic wave, the velocity of the
particle will be (by integrating)

We can state that the velocity of the incident side is transmitted to the transmission side and
then say . And we know that so

So we see that and then the insulation is given by

 44

This simplify law is the so-called mass-frequency law. We see that doubling the mass or the
frequency implies a doubled insulation. One can show that for a non-perpendicular sound
wave, the insulation is measured by

( )

This model is valid for an angle between zero and 78°. In practice, the sound will be incident
from different directions. One can show that in this case, the insulation value of the wall is five
decibels less than the normal incidence. Thus for the air we have

Effect of the wall stiffness

In the previous statement, we consider only the mass per unit area and we neglect the effect of
the whole mass and the stiffness. But if these two parameters are considered, they can lead to
some negative consequences: resonance phenomena can occur (and then the wall becomes
transparent for the sound wave). If we consider and the sound pressure at the right hand
side and at the left hand side of the wall, they are given by

And

The equation of motion is

̈ ̇ ( ) ( )

If we consider harmonic wave, we find

( )

Or in function of the velocity

[ ( ) ( )]

And knowing that , we find finally

 45

( )
( )
( )

So we see that three cases can be distinguished: if the frequency is far lower than the
eigenfrequency of the wall, the insulation of the wall is determined principally by the wall
stiffness; if the frequency is far bigger than the eigenfrequency, the mass is the determining
factor of the insulation; and finally, if the frequency is the wall eigenfrequency, then the wall
becomes transparent for the sound.

The coincidence effect

At higher frequencies higher order bending waves will occur in the wall. When the wavelength
of these bending waves coincides with the wavelength of the acoustic
waves after projection on the wall a so-called coincidence phenomenon
will occur. The projection of the acoustic wavelength on the wall is called
the wave trace and is given by . One can show that the bending
wave in a clamped wall is governed by

where represent the bending stiffness of the wall. By

considering a harmonic wave and introducing it in the equation, we can
find the velocity of the bending wave

√ √
 46

By definition, the coincidence will appear when or when . Thus we

find

We note that a critical frequency can be computed. It is the lower limit at which a coincidence
can occur. We find it by taking the maximal possible angle: and then

for the air with √ ( ) which is the quasi-longitudinal wave velocity of the wall.

Insulation of double walls

The following practical guidelines can be used for double walls:

1. By doubling the wall mass, the insulation increase by 5dB following the previous laws.
The air cavity also influence the insulation:
a. For a distance from 2 to 4cm, the sound insulation increase with 4dB
b. For a distance from 5 to 10 cm, the sound insulation raises with 9dB
2. The experimental frequency law has a slope of 6 to 8dB per octave
3. In the cavity between the two walls, an air resonance can occur. This can lead to an
important reduction of the sound insulation at the resonance frequency
4. Each panel has its own resonance frequency for which it is highly transmissive for
sound.
5. For low excitation, the two panels vibrate as a whole.
6. The two panels form a system of two masses with a spring in between
7. The air gap between the two walls has an infinite range of eigenfrequencies at which the
system becomes transparent for the sound.
8. Each panel has its own critical coincidence frequency. It is preferred to take the thickness
of the two walls different in such a way that the two frequencies do not coincide.
 47

Noise control

General procedure
A general procedure to control the sound is the following:

 Determine active and passive component:

An active noise component is a component which
produces noise. A passive noise component conducts
the noise produced by an active component.
 Determine if we have airborne, liquid-borne or
structural borne source
Aerodynamic noise: sound generated by oscillation
and friction of air molecules
Hydrodynamic noise: sound produced by oscillation
and friction of a liquid flow
Structural noise: sound produce by a mechanical vibration
 Identify the transmission path
 Identify the radiating surface
 Identify the primary contribution
 Use the design rules

There are different techniques to determine the sound source: with an intensity meter or with an
acoustic camera.

Reducing noise at the level of the source

Aerodynamic source
The source of aerodynamic sound can be turbulences, vortices or shocks and pulsations.

Vortices can appear because of bodies in the flow. They generate a pure tonal component (which
can also be generated by cavities). In channels, noise can be generated by sharp corner or valves.
Turbulent noises can also appear because of shear stresses that exist when there is a gradient in
the air pressure. The design rules for turbulent noise are the following:

1. Reduce the workload

2. Reduce the pressure drop
3. Reduce the outlet flow rate
 48

4. Minimize the tip speed motor

5. Avoid obstacles in the flow or adapt them
6. Do not point the flow outlet at the panel
7. Improve the geometry of the flow
8. Use special nozzle

Shocks are generated by a rapid discharge of a

compressed medium in an area of low pressure.
This happens, for example, when opening and
closing a valve in a pump. A single shock produces a broadband noise, but periodic shocks
result in a tonal noise. The noise generated by this phenomenon can be reduced by either
slowing down the pressure variation or reducing the pressure difference.

Hydrodynamic noise
Sources of hydrodynamic noise can also generate turbulence, vortices pulsation and shocks.
Therefore, the design rules are the same as in previous section.

Furthermore a peculiar effect, named cavitation, can also be produced. Cavitation occurs when
the static pressure is lower than the vapor pressure. Cavitation bubbles are created which
implode during re-compression, so high pressures can arise. This can occur for instance in
valves and pumps. Cavitation can be avoided by reducing the pressure drop per stage.

Some design rules for hydrodynamic noises:

1. Reduce the pressure drop

2. Reduce the flow rate
3. Increase the static operating pressure
4. Improve the geometry to counter cavitation
5. Keep the suction duct short
6. Position the reservoir higher than the position
of the pump

Structural noises
Impact noise is one of the most dominant noise sources in many machines. The most important
parameters of impact noise are the mass and velocity of the impact bodies and the duration of
the impact. The frequency spectrum of one single impact shows that this is a broadband noise.
Repeated impacts generate also harmonic noise. Some design rules can be found to reduce them:

1. Increase the time of the impact

2. Decrease the speed of the impact
 49

3. Minimize the mass of the impact body

4. Increase the mass of the solid body
5. Avoid loose parts with varying load

The gearing noise is a special form of noises which occur for example in the gearboxes.
Important parameters are the contact period, the time variation of the force during contact and
the stiffness of the teeth. Defects in the teeth may cause extra force variations and thus more
noise. Some measures can help to decrease this type of sound:

1. Increase the contact time

2. Use helicoidal gear
3. Increase the number of teeth
4. Improve the quality of the transmission
5. Use plastic gear for small load

Rolling noise is the result of the roughness or the irregularity of the contact surfaces. Rolling
noise occurs in roller and ball bearings, belts, rail and road vehicles. The rolling noise also
depends on the flexibility of the contact surfaces. The frequency content of rolling noise is
mainly broadband. Some measures:

1. Provide a smooth roll surface

2. Use suitable lubrication
3. Use precision bearing
4. Minimize tolerance of the housing of bearing
5. Increase the flexibility of the contact area

Acceleration of a mass leads to forces that can produce noise. Inertia forces can be caused by
oscillating masses or by (non-balanced) rotating parts. Design rules:

1. Balance rotors or use dynamic balancing

2. Minimize accelerating mass
3. Increase the uniformity of motion

Mechanisms where friction causes a so-called stick-slip phenomenon, are potential noise
sources. The variation of force leads to impact noise that can excite the resonances of the
structure. Friction noise occurs. The phenomenon is dependent on the materials and lubrication.
In principle, friction noise is broadband, but often due to the resonances strong tonal
components can occur. Design rules:

1. Control friction by suitable selection of material

2. Use suitable lubrication
 50

3. Increase the damping of the structure

Tackling noise transmission

Aerodynamic noise
The aerodynamic noises are transmitted by the air. There are several ways to control this
transfer:

 Acoustic casing: Absorption material can be placed on the inside to reduce the noise
production. The casing must be completely sealed (scellé). We use heavy material for the
outer wall and absorbing material for the inside. We must avoid rigid connection to the
machines (and then use flexible one, optionally with damping).
 Acoustic screen: Screens can be installed near to parts producing a large noise. However
their efficiency is lower than casing and depends on the direction and the distance.
 Noise muffler: Absorption muffler consist of a channel filled with a porous material.
Another type is the reflection mufflers which muffle the noise by reflection of the noise at
a change of the cross section
area. Some guidelines for the
use of muffler:
o Use absorption muffler
for broadband noises
and reflection muffler
for low frequency
noises.
o Avoid speed bigger
than 20 m/s in the
muffler
o Use pneumatic expansion mufflers for the exhaust of compressed air
 Noise absorption

Hydrodynamic noises
Transmission of hydrodynamic noise takes place in pipes and tubes. Noise control can be done
at the inlet of the system, in the system or at the outlet. The means is both reflection and
absorption. Reflection is obtained at the end of the system due to changes in the cross-sectional
area or by changing the rigidity of the wall by transition of pipes to tubes. Absorption of
Hydrodynamic noise is provided by accumulators. The design rules for the control of liquid-
born noise are:
 51

1. Use a combination of pipe and tube

2. Use dampers

Structural noises
The transmission of structure-borne noise from sources to radiating surfaces can be influenced
by changing the mass, stiffness and damping of the structure. The selected strategy depends on
a number of factors:

 If an increase of mass is possible, one near the region of excitation is enough

 In case of force excitation, we add impedance (mass) but in case of speed excitation,
adding mass is a non-sense.
 For a narrow band excitation, it is advisable to redistribute the stiffness or mass of the
system (the addition of damping is also effective). But this has no sense for a broadband
excitation.
 At low frequency, vibration isolation is the only solution. For the middle frequencies,
numerous solutions can be chosen: adding mass at the excitation point, increasing the
structural damping, isolate the source, reflection at discontinuities… For the high
frequencies, there are also several possibilities: increase the mass or the stiffness,
isolation of the source, discontinuities in excitation with extra damping…

Radiation noise
Air-borne noise can be radiated through outlet openings (e.g. the end of a tube). Usually the
noise has directivity in the direction along the axis of the tube. The opening can be adjusted to
reduce the noise in this direction. The design rules are in this case:

1. Put the opening in the right place and point it in the right direction
2. Use a damper or a screen at the opening

Structure-borne noise radiation depends on the size, shape, flexibility, mass and damping.
Regarding radiation, it is desirable to design the areas which are loaded, as compact as possible.
Design rules for structure-borne noise radiation:

1. Reduce the radiation surface

2. Use lids with low radiation efficiency:
a. Thin plate instead of thick plate
b. Perforated plates
c. Panel with damping material
52
53