Probability Distribution and Density

Way of characterizing random signals

Based on how instantaneous of signal are distributed

The probability that value of random signal is less
than or equal to a specified value

Random signal x(t) with min and max values shown in a
given window

Probability distribution function,
P(x) = Pr [x(t) x]
 Probability density of a random signal with maximum and minimum values

P(x) must have above form

x(t) is certain to be less than or equal to the maximum
value xmax (i.e. P(xmax) = 1) and it can never be less than
the minimum value xmin (i.e. P(xmin) = 0).

The probability that x(t) is between x + x and x is
P(x + x) P(x).
Probability Density

Probability density

Since p(x) = dP(x)/dx and in the general case P() = 1

while P() = 0 it is obvious that,

Total area under the probability density curve must

always be one.
Probability density function for Gaussian Signal

It is basically an exp(x2) curve centred on the mean

value , scaled in the x-direction in terms of the standard
deviation and in the y-direction so as to make the total
integral unity.

Statistical parameters Mean, Variance, Skewness and

Kurtosis can be obtained from the pdf.
PDF of Gaussian Signal (for zero mean value )
Statistical Parameters

Mean value:

First moment of pdf. Defines the centre of gravity as

area under pdf is unity.

For symmetric function like Gaussian distribution
mean value is the line of symmetry.

Variance:

Second moment about mean value (centred moment)

Similar to Moment of Inertia about mean value
Statistical Parameters
Skewness:

Third centred moment

Zero for symmetrical function and large for
unsymmetrical functions .

Kurtosis:

Fourth moment

Value is large for spiky signals
Fourier Analysis

Signals represented as sines and cosines

Fourier series applied to finite length signal

For any periodic signal g(t) having period T
g(t) = g(t + nT), Where n is an integer then,

0 = Fundamental angular frequency (rad/sec)

Fundamental frequency in Hz = 1 /T
ak and bk are coefficients
Fourier Analysis
Fourier Analysis

Division into sines and cosines depends on arbitary
allocation of time (t) =0.

Total component at frequency k = k 0 is given as ,

Which is alternately given as,

Amplitude is constant, phase depend on time t.

Fourier Analysis

Fourier series can also be written as

Interpreted as two rotating vectors, each of length

Ck/2, one rotating at angular frequency k with initial
phase k and the other rotating at angular frequency
k with initial phase k ,
Fourier Analysis

Sinusoid as sum of two rotating vectors

Fourier Analysis
Fourier Analysis

Equivalence of vector sum of positive and negative components and

projection on the real axis of a positive frequency component
Fourier Analysis

Fourier Integral Transform for signals other than
periodic.

Sampled time signals: Signals processed digitally are
sampled.

Discrete Fourier Transform: When sampled length is
finite, Fourier series can be used
Discrete Fourier Transform

In Matrix form forward DFT is written as
Gk = 1/N Wkn * gn
Where,

Gk : Vector of N frequency components,

gn : The N time samples g(n).

Wkn : A square matrix of unit vectors exp(j2kn/N)
with angular orientation depending on the frequency
index k (the rows) and time sample index n (the
columns).
 Gk = 1/N Wkn * gn

Matrix representation of the DFT (note the rotated real and imaginary axes)
Fast Fourier Transform (FFT)

An efficient algorithm to calculate DFT

In FFT, total number of complex operations is
reduced from N2 to N log2 N,

Saving by a factor of more than 100 for the typical
case where N = 1024 (= 210).
Convolution and Convolution Theorem

Fourier analysis converts convolution in one domain into a
multiplication in the other domain (as does the Laplace
transform).

Simplify the solution of many problems

Also useful in graphical illustrations of many relationships.

Convolution : operation by which the output (response) of
a linear system is obtained from the input (forcing
function) and the transfer properties of the physical
system, in the time domain represented by its impulse
response function.

The impulse response function (IRF) of a system is its
output when excited by a unit impulse (delta function) at
time zero.
Convolution and Convolution Theorem

When a forcing function f (t) is applied to a physical
system, the effect between time t and t + dt can be
considered as an impulse of value f (t) dt, giving an
impulse response starting at time t and scaled by the
strength of the impulse (i.e. proportional to f (t)).

The total response over time will thus be the sum of
all these scaled impulse responses initiated at different
times in the past, and can be represented by the
Duhamel integral
 Convolution and Convolution Theorem

The equation shows convolution of f(t) and h(t)

Represented as : x(t) = f(t) * h(t)

The operation is commutative, f(t) and h(t) can be
exchanged in Duhamels Integral

The convolution operation is quite complex and can be
seen to consist of four stages:
1. One function h() is reversed to h( ).
2. It is then displaced by an amount t to h(t ).
3. It is then multiplied by the other function f () to
give f ( ) h(t ).
4. This product is then integrated over the dummy
variable to give the total output at time t.
Convolution and Convolution Theorem

Fourier transform converts a convolution in one

domain into a product in the other domain.

A convolution in the time domain is transformed into
a product in the frequency domain,

By the same principle a product in the time domain is
transformed into a convolution in the frequency
domain.
 Drawbacks of FFT

All properties of DFT affect FFT.

Three stages in passing from the Fourier integral
transform to the DFT

Digitization of the time signal - can give rise to
Aliasing;

Truncation of the record to a finite length - can give rise
to leakage or window effects;

Third step results from discretely sampling the
spectrum - Can give rise to the picket fence effect (it is
as though the spectrum is viewed through the slits in a
picket fence)
 Aliasing can
happen Nyquist
Infinite continuous Theorem - Low
time signals :Fig. a, b, pass filter
c required

Energy at a
Fig. d-e: Signal is single frequency
truncated to length T is spread into
by multiplying it by a adjacent
finite (rectangular) frequencies -
window. Convolved leakage
with Fourier
Transform of window Fig. f, g:
Continuous
spectrum
Corresponds to
discretely
convolution with a
sampled,
train of delta
not necessarily
functions of spacing
sampled at
T in time domain
peaks, -picket
making the time
fence effect;
signal periodic
 Data Windows Continuous signals

Major function of the window is to reduce the effect of
the discontinuity which usually arises when a random
section of signal is made periodic.

It practically means minimizing the sidelobes in the
filter characteristic, both the highest and the
remaining ones.

To improve enhancement of discrete frequency
components with respect to broadband noise:
It is desirable to minimize the noise bandwidth of the
characteristic,
Attention must also be paid to minimizing the picket
fence effect.
Data Windows Continuous signals

Data windows for

continuous signals:
(a) rectangular;
(b) Hanning;
(c) KaiserBessel;
(d) flat top
Data Windows Continuous signals
 Signal Conditioning

Raw vibration signals always contain some contamination
(noise)

Some components may obscure other actual components that
comprise the important part of the signal

The most frequent signal conditioning operation is - filtering.

A-to-D signal conversion is the first step in data acquisition

Filtering performed computationally from the digitized signal.

Low-pass analog filtering should be inserted ahead of the A-to-D
converter to avoid aliasing,

Other frequently performed signal conditioning operations
include integration (i.e., to extract displacement from measured
acceleration or velocity signals, or velocity from measured
acceleration signals) and signal amplitude conversion.
 Filters: Types

Analog and Digital filters
Analog filters: Circuit of Transistors, capacitors,
Resistors, Inductors, etc.
Digital Filters: Use Digital Signal Processing

Passive and Active filters
Passive: Analog filter having passive elements like
resistors, capacitors. Needs no external power
Active: Analog filter having active elements like
transistors and operational amplifiers. Need external
power supply. Available in monolithic IC forms.

Based on frequencies allowed: low-pass, high-pass,
band-pass, notch, and tracking filters.
 Filters

Filtering is now routinely performed digitally after A-
to-D conversion,

The initial signal must first be passed through an
analog low-pass filter with cut-off frequency
sufficiently below the Nyquist frequency (1/2 sampling
rate) to eliminate aliasing

The analog filters cut-off frequency must be
substantially below the Nyquist frequency because no
analog filter has a perfect frequency cut-off, that is, it
has its roll-off above the cut-off frequency.
 Low pass filter

Most frequently employed signal conditioning operation in
handling machinery vibration measurement signals.

Frequency components above 10 times spin speed are
usually not of interest, be they noise or true signal.

The low-pass filter is intended to remove signal content
above the designated cut-off frequency and thus passes
through the remaining portion of the signal that is below
the designated cut-off frequency.

In using low pass filter - it is assumed that the original
analog signal has already been passed through an analog
low-pass filter to avoid aliasing.

Caution: Typical A-to-D expansion board for PCs does not
have an analog low-pass filter to avoid aliasing. However,
modern digital tape recorders do have it .
 Low pass filter

Low pass filter: frequency response

Ideal (upper) and Actual (Lower)
 High pass filter

The high-pass filter is the converse of the low-pass

filter

Removes signal content below the designated cut-off
frequency

It passes through the remaining portion of the signal
that is above the designated cut-off frequency.

Since routine rotating machinery vibration
assessments are usually not focused on frequency
components above 10 times spin speed, high-pass
filtering by itself is not often used in machinery
applications.
 High pass filter

High pass filter: frequency response

Ideal (upper) and Actual (Lower)
 Band pass filter

Band-pass filter: Combination of high and low-pass filtering.

Routinely employed in machinery vibration analyses.

A band-pass filter is designed to remove signal content outside a
designated frequency band,

It is a low-pass filter in series with a high-pass filter, where the
low-pass cut-off frequency is higher than the high-pass cut-off
frequency.

If filtering digitally, the original analog signal has to be first
passed through an appropriate analog low-pass filter to avoid
aliasing.

A band-pass filter centered at rotor speed is a standard operation
in rotor balancing,

Because, only the synchronous vibration component is processed
for rotor balancing purposes.
 Band pass filter

Band pass filter: frequency response

Ideal (upper) and Actual (Lower)
 Notch filter

The notch filter is the opposite of the band-pass filter.

Passes through all the signal content except that which
is within a specified bandwidth.

One application is magnetic bearings, which
inherently operate with displacement feedback
control, where a notch-type filter is frequently used to
filter out the once-per-rev bearing force components,
so they are not transmitted to the non rotating
structure of the machine, while the bearings continue
to provide static load support capacity and damping.
 Filters: Notch filter

Notch filter: frequency response

Ideal (upper) and Actual (Lower)
 Filters: Tracking filter

A tracking filter can employ the functionality of any of
the previously described filters,

It has the added feature that its cut-off frequency(s)
are made to track a specified signal component.

Schematic of two channel tracking filter

 Filters: Tracking filter

The center-band frequency of a band-pass filter to track

the once-per-rev frequency by the keyphaser signal
Amplitude Conventions
If a vibration signal is :
x = X sin t

Single Peak amplitude = X

Peak to Peak = 2 X

Average absolute value = 0.637 X

RMS = 0.707 X
 Modulators and Demodulators

Modulation: Modification of signals due to other
signals

May be done intentionally to maintain accuracy during
signal conditioning or transmission

Data signal (Modulating signals) is used to vary a
property (amplitude or frequency) of a carrier signal.

Carrier signal is modulated by data signal- AM or FM

Demodulation: Recovery of data signal by removing
carrier signal.
 Modulation and Demodulation

Amplitude modulation (AM): amplitude of periodic
carrier signal is varied according to amplitude of data
signal (modulating signal), frequency of carrier signal
is kept constant.

Frequency modulation (FM): Frequency of carrier
signal is varied in proportion to amplitude of data
signal (modulating signal) while keeping the
amplitude of carrier signal constant.

Pulse width Modulation (PWM)

Pulse frequency modulation (PFM)

Phase Modulation: Phase of carrier signal is modified
by data signal
Modulation of sinusoidal carrier signal

a) Data signal

b) Amplitude modulation

c) Frequency Modulation

d) Pulse width modulation

e) Pulse frequency modulation

Amplitude Modulation
xa(t) = x(t) xc(t)
Where,
xa(t) = Amplitude modulated signal
x(t) = data signal (modulating signal)
xc(t) = high frequency periodic carrier signal
xc(t) = ac cos(2 fc t) - Can be assumed like this

Carrier frequency (fc) to be 5 to 10 times the frequency

of highest frequency of interest (bandwidth) of data
signal
 Modulation Theorem

Frequency shifting theorem

If a signal is multiplied by a sinusoidal signal, Fourier
spectrum of the product signal is simply the spectrum
of original signal shifted through frequency of
sinusoidal signal.

Spectrum, Xa(f) of AM signal xa(t) can be obtained
from Fourier spectrum X(f) of the data signal x(t)
simply by shifting through the carrier frequency fc.
 Demodulation

Extracting original data signal from modulated signal.

It must be phase sensitive

Algebraic sign of data signal must be preserved and
determined.

Full wave demodulation output generated
continuously

Half wave demodulation No output for every
alternate half period of carrier signal

Carrier signal should be powerful and its frequency
should be very high
 Cepstrum Analysis

Earlier definition: The power spectrum of the logarithm of the
power spectrum

Power cepstrum was later redefined as the inverse FT of the log
power spectrum

Log function converts multiplicative relationship between the
forcing function and transfer function (from force to response)
into an additive one which remains in the cepstrum.

Gives rise to one of the major applications of the cepstrum. For
SIMO (single input, multiple output) systems, the addition in
the cepstrum corresponds to a convolution in the time domain of
the forcing function and impulse response function.

Note that this does not apply to MIMO (multiple input, multiple
output) systems, as each response is then a sum of convolutions.
 Cepstrum Analysis

C( ) = F-1 log (X( f ))

Where,
X( f ) = F [x(t)] = A( f ) exp( j( f ))
in terms of its amplitude and phase, so that
log (X( f )) = ln (A( f )) + j( f )
When X(f ) is complex as in this case, the cepstrum of
is known as the complex cepstrum,
ln (A( f )) is even and ( f ) is odd, the complex
cepstrum is real valued.
 Cepstrum Analysis
By comparison, the autocorrelation function can be
derived as the inverse transform of the power
spectrum,
Rxx( ) = F1 |X( f )|2 = F1 A2( f )
When the power spectrum is used to replace the
spectrum X(f ), the resulting cepstrum, known as the
power cepstrum or real cepstrum, it is given by
Cxx( ) = F1 [2 ln(A( f ))]
It is a scaled version of the complex cepstrum where
the phase of the spectrum is set to zero.
The term real cepstrum is sometimes used to mean
the inverse transform of the log amplitude, thus not
having the factor 2
 Cepstrum Analysis

Before calculating the complex cepstrum the phase

function ( f ) must be unwrapped to a continuous
function of frequency, and this is often difficult, so it is
easier to use the real cepstrum.
 Cepstrum Analysis: Terminology
Quefrency doamin Frequency domain
Cepstrum from Spectrum
Quefrency from Frequency
rahmonic from harmonic
lifter from filter
short-pass lifter from low-pass filter
long-pass lifter from high-pass filter
gamnitude from magnitude
saphe from phase
 Cepstrum Applications

Main application of cepstrum analysis in machine
condition monitoring is for signals containing families
of harmonics and sidebands (of uniform spacing)
where it is the whole family, rather than individual
frequency components, that characterizes the fault.

Most common for Gear and rolling bearing fault
analysis
Log power spectrum - Family of harmonics of the BPFO, with a spacing of
206 Hz. These harmonics are not even visible in the power spectrum, since
they are all below the20 dB line, which corresponds to 1% of full scale on the
linear (amplitude-squared) scale.
Autocorrelation function : Exhibits a beat between the two largest components
in the spectrum, which have no relation to the bearing fault.
Cepstrum is dominated by rahmonics corresponding to the BPFO
(spaced at 4.84 ms),
Cepstrum for missing blade in the Turbine
Removal of echo
using complex
cepstrum
Cepstrum: Practical Considerations

Remove noise from the spectrum. Harmonics immersed in noise
can not be detected by cepstrum.

Narrow bandwidth analysis ,zoom analysis , synchronus
averaging can be used to remove noise.

The result of the cepstrum is dependent on the filter
characteristic used in the original spectrum analysis

Choice of vibration parameter affects the cepstrum.

Dynamic range of measurement should be @ 80-100 dB.

The reference value for the logarithms should also be chosen
with care. (Approximately in the mid range of the spectrum
values)

Zoom spectrum is good to have good resolution of harmonic /
sideband components

Particular portion of spectrum can be selected for cepstrum
analysis to avoid unwanted low speed harmonics
Cepstrum: Practical Considerations

Factors affecting cepstrum

Cepstrum: Practical Considerations

Cepstrum analysis on a zoom spectrum

The left hand end of the spectrum not at zero frequency,
Two slightly shifted zoom spectra to obtain the cepstrum
encompassing the sidebands around the second harmonic of a
gearmesh frequency.
Cepstrum: Practical Considerations
1. The sideband families no longer pass through the
effective zero frequency at the left hand end of the
spectrum.
2. Rahmonics in the real cepstrum are no longer positive
peaks corresponding to the sideband spacings.
3. The quefrency corresponding to the 25 Hz spacing is
close to a zero crossing in both cases.
4. The quefrency corresponding to the 8.3 Hz spacing has
a positive peak in one case and a negative one in the
other.
Cepstrum: Practical Considerations
5. Problem can be solved by making use of Hilbert
transform principles,
6. The true spacing will be indicated by the peak in the
amplitude of the complex (analytic) signal obtained by
inverse transforming the one-sided log spectrum (zero
padded to replace the negative frequency
components).
7. Such a cepstrum is an analytic cepstrum, to
distinguish it from the complex cepstrum, which is
real.
Cepstrum: Practical Considerations

8. Exactly the same phenomenon will be encountered

whenever a sideband family does not pass through zero
frequency, even the genuine zero frequency.
9. For normal gears, the gearmesh frequencies are a
harmonic of both shaft speeds, and so modulation by
these shaft speeds gives sideband families that are also
harmonics and thus pass through zero frequency.
10. In planetary gears, not all modulation frequencies are
submultiples of the gearmesh frequency, so sideband
families do not necessarily pass through zero
frequency.
 References

Randall R B, Basic Signal Processing Techniques,
Vibration Based Condition Monitoring, John Wiley
and Sons, 2011

De Silva, C W, Vibration Testing Vibration
fundamentals and Practice, CRC Press, 2000