Signal Processing and Diagnostics

Signal Processing and Diagnostics
Ji Tma
VSB-TU Ostrava, Czech Republic
2012
Outline of a course on Signal processing and diagnostics
Introduction to diagnostics and basic terms
Principles of signal processing (SP)
Transformations for signal processing (Fourier, FFT, etc)
Basic methods and algorithms for SP
Diagnostics principles and devices
Noise, sound and vibration measurement
Experimental measurement on Diagnostic Laboratory
2
Application areas for the knowledge gained in the course
of signal processing
Signal analysis
Sound and speech analysis
Noise and vibration signal analysis
Condition monitoring of machines during their technical life
Condition base maintenance (CBM)
Quality control of new products
Research in experimental dynamics
Design of new products emitting low level noise
Environmental noise and vibration control
3
Introduction
Lecture 1
4
Signal processing tools
Analysis of signals in the time and frequency domains
Frequency and order analysis
Averaging in the frequency and time domains
Envelope analysis
Phase demodulation

Methods
Plots
Time Domain
Time history
Orbit
Frequency Domain
Frequency spectra (2D)
Time-frequency analysis (3D - waterfall, spectral maps)
Trend
5
Hardware
Microphones
Accelerometers, sensors of velocity, proximity probe
Sensors for angular vibration measurements
Sensors for RPM measurements
Laser Doppler vibrometers

Sensors - transducers
Multipexers and A/D convertors
Signal analysers
Multifunction cards
Multifunction cards for dynamic measurements
Computers
Personal computers
Work station
6
Sensors - transducers
Microphones
Accelerometers
RPM measurements
for linear vibration
for torsional vibration
7
Signal analyzers
8
CPB (Constant Percentage Band) analyzers, alternatively designated as Real Time
or 1/3-octave or 1/1-octave analyzers. The principle is based on a bank of frequency
filters with the percentage bandwidth which is equal to a constant.
FFT analyzers. The principle is based on the Fast Fourier Transform
Vector analyzers. It is an analyzer for the measurement of the amplitude and
phase of the input signal at a single frequency within an intermediate frequency
bandwidth of a heterodyne receiver.
according to the principle of operation
according to the application area
Laboratory analyzers
Analysers for machine diagnostics
Analyzers for noise and vibration signals
Analyzers for electrical signals (frequency range of MHz)
List of types of signal analyzers
Signal types
Lecture 2
9
Signal types
Deterministic Random (stochastic)
Periodic Non-periodic Stationary Nonstationary
Sinusoidal Complex
periodic
(harmonic)
Almost
periodic
Transient Ergodic Non-ergodic Special classification
Deterministic signals are defined as a function of time while random signals can be defined
in terms of statistical properties.
Deterministick signals can be predicted as opposed to random signals whose instantaneous
values are not predictable only statistical properties can be predicted.
Deterministic vs. random signals
Periodic vs. non-periodic signals
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,5 1,0
Time [s]
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,5 1,0
Time [s]
10
11
Random variables, random signals
Names of random variables and processes: , (t), , (t),
{ } ( ) x x p x x x P A = A + s <
Probability that a random variable belongs to the interval of values greater than x and less
than x+x is proportional to the interval of the length x

t
1

t
Realization of the random
process
Realization of the random
variable in time t
x
Independent variable
1

Realization of random variables and processes: x, x(t), y, y(t),
The coefficient of proportionality is denoted as a probability density function (pdf).
( ) x p
( ) t x
1
( ) t x
2
( ) t x
3
( ) t
Probability density function
12
Statistical properties of a random variable
Continuous random variable < > Discrete random variable
( )
( )
+ < <
|
|
.
|
\
|
o

t o
=
x
x
x p ,
2
exp
2
1
2
2
Probability density function for
the normal (Gaussian) distribution
Gaussian function (bell curve)
Probability density function (pdf)
Two-dimensional pdf
13
Stationary and non-stationary signals
A stationary signal (or strict(ly) or strong(ly) stationary signal) is a stochastic process whose joint
probability distribution does not change when shifted in time or space. As a result, parameters such
as the mean and variance, if they exist, also do not change over time.
-3
-2
-1
0
1
2
3
0,0 0,5 1,0
Time [s]
-1
0
1
2
3
0,0 0,5 1,0
Time [s]
Stationary signal Non-stationary signal
( ) x p ( ) t x p ,
is not a function of time
is a function of time
x x
t
1
t
2
t
3
t
1
t
2
t
3
The basic properties of stationary continuous signals x(t) is as follows
( ) ( )
2 1 2 1 2 1 2 1
, , , , , t t x x p t t x x p =
( ) ( )
1 1 1
, x p t x p = one-dimensional probability density function (pdf)
two-dimensional pdf
14
Mean value, variance and correlation function of a random
variable
{ } ( )
}
+

= = x x p x d E
{ } ( )
}
+

= = x x p x M
k k
k
d E
{ } { } ( ) 0 E , E = A = A x x x x
( ) { } ( ) ( )
}
+

= = x x p x m
k k
k
d E
{ }
2
2
D m = o =
( ) ( ) ( ) { } ( ) ( ) ( )
} }
+

+

= =
2 1 2 1 2 1 2 2 1 1 2 2 1 1 2 1
d d , , , E , x x t t x x p t x t x t t t t R
xx
Centered variable .
Mean value ..
The nth central moment
The nth central moment
about a mean value
Variance ....
Correlation function ..
( )
2 1 2 1
, , , t t x x p The two-dimensional probability density function
Expected value (moment in physics)
For a random variable , it is introduced
For a random signal (process) (t), it is defined
( ) o = value Peak factor Crest
Crest factor ...
Standard deviation . { } = o D
( ) ( ) ( ) t = =
xx xx xx
R t t R t t R
1 2 2 1
,
If ( ) ( )
2 1 2 1 2 1 2 1
, , , , , t t x x p t t x x p = then
15
Stationary and ergodic signals
For example the parameters (mean and
variance) can be computed from values
corresponding to random signal
realizations in time t
1
.
If the realizations are sections of the long record and T
tends to zero or to the interval, which is small enough,
them the values across a group of identical processes can
be replaced by the samples of the time record.
time time
An ergodic process is one which conforms to the ergodic theorem. The theorem allows the time
average of a conforming process to equal the ensemble average. In practice this means that statistical
sampling can be performed at one instant across a group of identical processes or sampled over time
on a single process with no change in the measured result.
Random variables in time t
1
t
1
T+t
1
2T+t
1
3T+t
1

4 signal realizations
0 T t
1
-3
0
3
1
3
4
2
0 T 2T 3T 4T
-3
0
3
1
2
3
4
For ergodic signals, it is assumed that the mean value can be replaced by the time average
( ) ( )
} }
+
+
+

= =
2
2
d
1
lim d
T
T
T
t t x
T
x x p x x
16
Mean value and standard deviation (RMS) of a sine signal
( ) 0 d
1
0
= =
}
T
t t x
T
x
-1,2
-0,8
-0,4
0,0
0,4
0,8
1,2
0,0 0,5 1,0
Time [s]
( )
|
.
|
\
|
t
= t
T
A t x
2
cos
( ) ( )
2
d
2
2 cos 1
2
1
d
2
cos
1
d
1
2
0
2
0
2
0
2
2
A
t t
T
A
T
t t
T
A
T
t t x
T
T
T T
=
|
|
.
|
\
|
|
.
|
\
|
t
+ =
=
|
|
.
|
\
|
|
.
|
\
|
t
= = o
}
} }
2
A
= o
Sinusoidal signal
Mean value
The computation of RMS for the sinusoidal signal
results in the formula
The value of RMS is approximately equal to 70%
of the harmonic signal amplitude. The amplitude
of this signal is the 1.4-multiple of RMS.
0,0
0,2
0,4
0,6
0,8
1,0
1,2
0,0 0,5 1,0
Time [s]
x(t)
(x(t))
2
If the signal is harmonic (sinusoidal) than we can calculate
17
Signals with continuous and discrete time
The analog signal x(t) is a real or complex function of continuous time t. The other definition
points to the fact that signal contains information. But the white noise does not formally contain
any information which is in fact information.

Analog and digital signals

Process of converting of an analog signal into a digital signal is called digitalisation. There are
two issues in digitalisation

sampling
quantizing.

First, we notice sampling. Sampling of an analog signal produces a time series which is a
sequence of samples in the discrete time n. The sequence of samples may be denoted either as an
indexed variable or as a function of an integer number x(n).

x(t)
x
0
x
1
x
2
x
3
x
4
x
5
x
6
x
7
0 1 2 3 4 5 6 7 where T
S
is a sampling interval for the uniformly sampled data.
The sampling frequency (rate) f
S
in hertz or in samples per
second is the reciprocal value of the sampling interval.
T
S

( ) | |
T
x x x t x ,... , , : Sampling
2 1 0
( ) ( ) ( ) ( ) t x nT t x nT t t x
n
S n
n
S
o = o

t
n
Sampling may be considered as a mapping
The time continuous signal is related to the time series and can be
substituted by the following way
18
Stationary and ergodic signals continuous and discrete
time
For ergodic signals, it is assumed that
the mean value can be replaced by the
time average
( )
}
+
+
=
2
2
d
1
lim
T
T
T
t t x
T
x
( ) ( )
}
+
+
=
2
2
2
d
1
lim
T
T
T
t x t x
T
s
( ) ( ) ( )
}
+
+
t = t
2
2
d
1
lim
T
T
T
xx
t t x t x
T
R
The correlation function of ergodic signals
depends only on the lag t
Root Mean
Square =
RMS
Mean value
Standard
deviation
Auto-correlation
Cross-correlation
( )
=
=
1
0
1
N
i
i x
N
x
( ) ( )
=
=
1
0
2
1
N
i
x i x
N
s
( ) ( ) ( ) 2 ,..., 2 , 1 , 0 ,
1
1
0
= t t +
t
= t

t
=
N i y i x
N
R
N
i
xy
( ) ( ) ( ) 2 ,..., 2 , 1 , 0 ,
1
1
0
= t t +
t
= t

t
=
N i x i x
N
R
N
i
xx
For discrete ergodic random signals
(x(i), i = 0, 1, 2, ), the formulas take the form
( )
2 2 2
1
0
2
2
1
x RMS x i x
N
s
N
i
= =

=
( )
=
=
1
0
2
1
N
i
i x
N
RMS
Time and frequency domains
-2,0
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
2,0
0,0 0,2 0,4 0,6 0,8 1,0
U

Time [s]
6

H
z

9

H
z

2
3

H
z

3
7

H
z

0,0
0,1
0,2
0,3
0,4
0,5
0,6
0,7
0,8
0 20 40 60 80 100
R
M
S

Frequency [Hz]
The time signal Autospectrum
An example
The time domain The frequency domain
20
Signal to Noise Ratio and Nyquist-
Shannon Sampling Theorem
Lecture 3
21
Sampling a continuous signal to a discrete signal
Sampling
Continuous time t
x(t)
DAC
Discrete time n
ADC
x
n
Quantizing noise
0
1
-1 -0,5
p(x)
0,5
x
Quantizing
0
( )
12
1
d d
5 . 0
5 . 0
2 2 2
= = = o
} }
+
x x x x p x
Variance of the quantizing noise
1
2
3
4
5
6
Let x(t) be a continuous signal
Discrete time n
0 1 2 3 4 5 6 7
0 1 2 3 4 5 6 7
p(x) uniform probability
density
12
1
=
o
Standard deviation of the quantizing noise
Quantizing is a part of AD conversion resulting in the attribution
of values x(nT
S
) to specified discrete values x
n

T
S
sampling interval
Quantization
Harmonic distortion typically characterized by a family of harmonic components
resulting from non-linearity in the analog signal conditioning
Cross-talk signals caused by inter-channel coupling
Spurious signals caused by various phenomena such as power supply imperfection,
clock circuits, bus communication and EMC coupling between circuits
ADC Resolution (given by the number of bits) and non-linearity
Aliasing originating from signal components of frequencies higher than the Nyquist
frequency
Digital output
Analog input
Quantizer
A
D Clipping level
Clipping level
Imperfections
22
Signal-to-Noise Ratio
( ) ( ) ( ) ( )
(
+ = =
|
|
.
|
\
|
=
|
|
.
|
\
|
=

6 log
2
1
2 log 1 20 6 2 log 20
2 2
log 20
Noise
Signal
log 20
1
1
m N S
m
m
RMS
RMS
dB
o
dB m N S
dB
76 , 1 02 , 6 + =
Number of ADC bits 4 8 10 12 14 16 18 20 22 24
Maximum of S/N dB 26 50 60 72 84 96 108 120 132 144
Standard deviation of quantizing noise
RMS value of a full-scale sine wave
m
2
1
2
m
Full scale
ADC resolution:
m number of ADC bits
Signal analyzers The newest models
|
|
.
|
\
|
=
PWR
PWR
dB
N S
Noise
Signal
log 10
Signal-to-Noise Ratio is defined as the ratio of a signal power to the noise power, which is corrupting
the signal
23
Historic overview of ADC specifications
Year ADC resolutions (bits) Signal to Noise Ratio
1970 10-12 60 dB
1980 14-16 70 dB
1990 16 80 dB
2000 24 100 dB
2005 24 110 dB
See [Bruel & Kjaer Technical Review No.1 2006]
A high quality noise and vibration transducers, including preamplifiers, can deliver
a practically noise-, spurious- and distortion-free signal over a dynamic range of 120 to 130 dB.
24
25
Effect of filtering on the S/N ratio
f
f
s
/2
BW
0
S(f)
s PWR
f BW
2
2 Noise

o =
2
o
( ) ( )
s
f
f f S df f S
s
2
2
0
2
2

o = = o
}
Total power:
BW bandwidth in Hz
Quantizing noise (white noise)
PSD spectrum of white noise
is independent on the frequency
s
f
2
2

o
The frequency filtration reduces the quantizing noise power and consequently RMS of noise.
Narrowing the bandwidth reduces
the quantizing noise power
The term improving
the S/N ratio
Low pass or band pass filtration improves the S/N ratio
( ) BW f
f BW
N S
S
PWR
S
PWR
PWR
PWR
dB
2 log 10
Signal
log 10
2
Signal
log 10
Noise
Signal
log 10
2 2
+
|
|
.
|
\
|
o
=
|
|
.
|
\
|
o
=
|
|
.
|
\
|
=

S/N ratio for full
frequency range
26
FFT noise floor
Process gain
N-point FFT
Effective number of bits (ENOB)
02 . 6
76 . 1 SINAD
ENOB

=
SINAD
FFT noise floor
m = 12 bits
N = 4096
74 dB = 6.02 m + 1.76 dB
SINAD is the abbreviation for Signal to Noise 'A'nd Distortion
-74 dB
Bin spacing
N
f
S
=
2
S
f
RMS quantization level
-107 dB
dB
0
20
40
60
80
100
120
ADC full scale
0 dB
|
.
|
\
|
=
2
log 10 dB 33
10
N
[ADC full scale] [FFT noise floor] [Process gain] = SINAD ENOB
The example assumes
that m = ENOB
Generaly m > ENOB
Effect of the background noise on the frequency spectrum
Autospectrum : 60 ; 80 ; 100 ; 120 SNR dB
0 dB
-180
-160
-140
-120
-100
-80
-60
-40
-20
0
20
0 100 200 300 400 500 600 700 800
Frequency [Hz]
R
M
S

d
B
/
r
e
f

1
N = 2048
60 S/N dB
80 S/N dB
100 S/N dB
120 S/N dB
Process gain ... 10 log(N/2) = 10 log(2048/2) = 30.1 dB
Signal 1 V
RMS
200 Hz
White Noise

0.001 V
RMS
0.0001 V
RMS
0.00001 V
RMS
0.000001 V
RMS
Ref level 1 V
RMS
Background Noise

Hz 2048 =
S
f
27
Electronic noise
Noise is a random process, characterized by power spectral density (PSD), which is measured in
watts per hertz (W/Hz). Since the power in a resistive element is proportional to the square of the
voltage across it, noise voltage (density) can be described by taking the square root of the noise
power density, resulting in volts per root hertz ( ). Hz V
-1.5
-1.0
-0.5
0.0
0.5
1.0
1.5
0 0.2 0.4 0.6 0.8 1
time
s
i
g
n
a
l

28
Nyquist-Shannon sampling theorem
Acording to the Nyquist-Shannon theorem, sampling the 10 Hz continuous sinusoidal signal
requires the sampling frequency, which is equal to 20 Hz at least
The sampling frequency, which is satisfying to the mentioned sampling theorem, results in the
fact that the sampled signal crosses the zero at the same frequency as the sampled signal
The sampling frequency 12 Hz causes that the sampled signal crosses the zero at the frequency
12 10 = 2 Hz. This phenomenon is called as an aliasing.
No-satisfactory sampling
Satisfactory sampling
29
Aliasing as a result of the improper sampling frequency
Nyquist
frequency
Sampling
frequency
2
0 S S
f f f > >
Shannon-Nyquist sampling theorem:
(Shannon-Kotelnik)
2
0 S
f f <
56 . 2
0 S
f f < BK signal analyzers:
f
f
s
0
aliasing
S(f)
f
s
/ 2
f
o
f
s
- f
o
symmetry
1
-80 dB

0

Analog
low pass filter

A/D

Sample rate
~ 80% of f
S
/2

S
f
MAX S
f f
S
f
MAX
f
2
S
f
MAX CUTOFF
f f =
MAX S STOP
f f f =
0
S
f
Stop band
(14-bit ADC)
Analog filter
(transient range)
Measurement range
Analog input Digital output
m bits
The analog low pass filter (antialiasing filter) faces
to the aliasing effect.
Let f
o
be satisfying
DC
1-2
Measuring chain
BNC connector
BNC connector for Microdot
Lemo 7-pin connector for mic
TNC connector for Microdot

Transducer
Signal
conditioning
Cables
Connector for signal analyzers
A/D
convertor
Direct (voltage input)
Microphone
Accelerometer
Digital data to
be analysed
Coaxial cables
Signal analyzers
Multifunction cards
A/D convertors
Front-end of the signal analyzer
PULSE with connectors
(BNC and Lemo)
Microdot
Multichannel analyzers
Multichannel signal analyzers have parallel inputs - each measuring channel has its
own A / D convertor. All the convertors are triggered simultaneously in the same
time moment.
Munction cards work with a multiplexor which connects signals to a A/D convertor.
There is only one A / D converter on the multifunction card. If the sample-and-hold
circuit is not used, then between the successively measured samples is a time lag.
Multifunction cards of low price are not equipped by an antialiasing filter except
cards for dynamic measurements (higher price).
Input analog signal
is sampled
Sample-and-hold circuit
AI analog
input
AO analog
output
C a control signal
Output voltage is maintained at a
constant value during conversion
Analog memory
31
Power spectrum
Power spectrum
32
S(f )
PWR = RMS
2

f
White noise
Energy of signal ..
x(t) signal with the arbitrary unit U (e.g. Pa, m/s, m/s
2
)
x(t)
2
signal power with the unit U
2
(e.g. Pa
2
, (m/s)
2
, (m/s
2
)
2
)
x(t)
2
energy U
2
s
t
(For instance, the electric power P = R i
2
is proportional to the square
of the current i as a function of time while R is only a scale factor)
}
+

= t d ... ...
2
=
n
2
... ...
f
frequency
( ) PWR d =
}
+

f f S
S(f )
f
( ) const = f S
Pink noise
S(f )
f
( )
2
1 f f S
0
0
0
(PoWeR)
( ) f S d
Quantizing noise can be considered as white noise
area
Spectrum shaping
33
Power spectrum
of input signal
f
S
xx
(f )
0
Frequency response
|G(f )|
2

S
yy
(f )
0
0
f
f
Power spectrum
of output signal
( ) ( ) ( ) f S f G f S
xx yy
2
=
( ) ( ) ( ) ( ) ( ) f G f G f G f G f G
*
2
= =
System
Input Output
Frequency response ( ) f G
It can be proved that
where
The transformation of the power spectrum of the input signal of a system to the output
signal power spectrum can be called a spectrum shaping
A linear dynamic system
defines relationship between an
amplitude and phase of a harmonic
signal at the input and its response
of the same single frequency at the
output of the given system.
Nyquist rate A/D convertors
Analog
input
Digital
output
ADC
DC
0 f
( ) f S
V
2
S
f
S
f
DC
0
( ) f S
V
2
S
f
S
f
f
2
S
f K
S
f K
Digital low pass filter
Quantizing noise 12 q
LSB 1 = q
DC
0
( ) f S
V
2
S
f
S
f
f
2
S
f K
S
f K
S
f
Effect of integration
and low pass filtration
ADC
S
f K
LP filter Dec
S
f

MOD
S
f K
LP filter Dec
S
f
Decimation
K-times
Area is proportional
to the noise power
2
o
( ) dB m N S
REF dB
76 , 1 02 , 6 + =
( ) ( ) K N S N S
REF
dB dB
log 10 + =
Frequency spectrum of quantizing noise and
methods of its decreasing
Frequency spectrum of quantizing noise
Sampling frequency is the same at the
input and output
Sampling frequency is greater at the
input than at the output
The use of modulation
Signal to noise ratio
Oversampled A/D convertors
34
Shaped spectrum of white noise
White noise
Improve S/N
Nyquist Rate convertor (succesive aproximation)
Transmitters on the principle of successive approximation . This
converter is assumed as the type for which the input (the first stage
of conversion) and output frequency (second stage of conversion) is
of the same rate.
DAC
Logic
U
U
0
= A U
R
0

U
REF

I
1
R
2
R
k
R
2
D/A convertor
LSB MSB
The accuracy of a convertor
depends on the accuracy of
resistors, max 12-bit ENOB
(accuracy 0.002%)
bits
0 1 0
12
12
1
Comparator
Comparator
(ENOB Effective Number Of Bits)
U
+U
0
-U
0
bit 1
bit 0
Process of succesive aproximation
Example of a 4-bit A/D convertor
The most significant bit
The less significant bit
1000
or
0000 ?
1100
or
1000 ?
1010
or
1000 ?
1001
or
1000 ?
1st Comparison step by step : 2nd 3rd 4th bit
time
(convertor otput)
(clock)
(digital output)
Start Stop
Full scale
Input signal
DAC Output
Comparator result
LSB
Clock
MSB
(comparator output)
36
Flash convertors
Analog input
Reference
voltage
Accuracy 8 to 10 bits
1 GHz sampling
frequency and more
1
0
R
R
R
R
R
2
0
1
1
Digital output
Voltage divider
Input voltage is less than
the output voltage of voltage
divider
Input voltage is greated than
the output voltage of voltage
divider
Latch
Latch
Latch
Comparators
Latch is a flip-flop circuit with
two stable states, memory of a bit
37
Oversampled convertors
1-bit ADC comparator
Convertors based on principle of delta-sigma modulation
DAC
ADC
( )
}
dt ...
1-bit DAC
1
0

Analog
output
Digital output
+V
ref
-V
ref
1

Analog and
reference
inputs
Digital
input
1
1
1
Convertors of this type are referred to as single-bit, even if their accuracy reaches 24 bits
Block diagram of a first order digital delta sigma modulator
difference
Integrator
38
The second order modulator
1- bit ADC comparator
The second order modulator
DAC
ADC
( )
}
dt ...
1-bit DAC
1
0
0

Analog
output
Digital
output
+V
ref
-V
ref
1

Analog
input
Digital
input
( )
}
dt ...
( )
5 2 2 2
5K
E E
o t = o
2 = K
reduces noise by 2.5 bits
The converter of the type AD 1847 produced by Analog Devices Company contains
a 6th order modulator
16
39
Digital decimation and filtration process
Motorola Digital Signal Processors, Principles of Sigma-Delta Modulation for Analog-to-
Digital Converters by Sangil Park, Ph. D
Decimation ratio Number of bits
Sampling rate
16 samples is replaced by a sum of them
Block diagram of sigma delta convertotr of the first order
Digital decimation process
x(t)
y(t)
1bit
D/A
( )
}
dt ...

Digital
Decimation
Filter
y(n)
F
S 1 16
First order - loop
6,4 MHz
(1 bit)
100 kHz
(16 bit)
F
S
-> f
S

+
-
x(n)
Analog
Input
- loops
1 16 16
16 : 1
Comb Filter
4 : 1
FIR Filter
Digital
Output
6,4 MHz
(1 bit)
100 kHz
(16 bit)
400 kHz
(12 bit) Resolution
40
The resolution of a bitstream
at the sampling rate 6,4 MHz
is a Bit per a sample while
the resolution of a data
stream at the low sampling
rate 100 kHz is increased to
16 Bits per a sample.
y(n) is a bitstream at the
sampling frequency F
S

x(n) is a data stream at the
sampling frequency f
S

Signal diagram for conversion of a constant voltage
Block diagram of sigma delta convertor of the first order
x(t)
y(t)
1-bit
D/A
( )
}
dt ...

Digital
Decimation
Filter
1 16
6,4 MHz
(1 bit)
100 kHz
(16 bit)
F
S
-> f
S

+
-
x(n)
41
0
x(t)
Comparator output
Integrator output
1-bit D/A output
Total sum of 16 voltages: x(n) = 5 x (-V
ref
) + 11 x (+V
ref
) = 6 x (+V
ref
)

D Q
Clock
y(n)
D-type latch
0
1 1 1 0 1 1 0 1 1 0 1 1 0 1 1 0 1
Clock ....
Latch
Difference
y(n), D
Integrator
y(t)
Clock
Q
x(t) - y(t)

+V
ref
-V
ref
Signal diagram for conversion of a constant voltage
The Q output of the latch
of the D-type changes
state on the leading edge
of the clock
Time
Input
voltage
0
0
0
0
0
0
Comparator
Dynamic range limitations due to measurement
inaccuracies of the measuring chain elements
Limitation
Useful dynamic
range
of measurement
False
treshold
Microphone with
a preamplifier
A/D
convertor
Filtration
42
BK Dyn-X Technology - Single Range from 0 to 160 dB
Dyn-X
convertors
Low noise
sensor
Frequency
range of
analysis
6 kHz
Switched ranges
Measurement ranges (standard)
1E+01
1
0
3

d
B

1
1
2

d
B

1
1
9

d
B

1
2
0

d
B

1
2
7

d
B

1
2
7

d
B

1
2
8

d
B

1
2
8

d
B

1
6
2

d
B

1
6
1

d
B

1E+00
1E-01
1E-02
1E-03
1E-04
1E-05
1E-06
1E-07
1E-08
D
y
n
a
m
i
c

O
p
e
r
a
t
i
n
g

R
a
n
g
e

[
V
r
m
s
]

Narrowband 6 Hz
10 Vp
(Dyn-X)
High Quality
Transducer
Super
wide
range
analog
input
ADC 1
24 bit
ADC 2
24 bit
Real-time DSP
24 bit
24 bit

43
Summary of the basic properties of A/D convertors
Typ ADC Number of bits Antialiasing filter Delay
Succesive
approximation
max 12 (14) required
Low (depends on the bit
number)
Sigma-delta up to 24
not required (a part of
the converter)
Large (depends on LP
filter order)
Flash 8 (10) required negligible
Bandwidth of the analyzer = number of channels (inputs) x measurement bandwidth
Set of converters with synchronized A / D conversions forms a signal analyzer. The
main parameters of the analyzer in terms of number of used converters are
simultaneously measured channels and sampling frequency

Frequency range of measurements theoreticaly:

in comercial signal analyzer:

2
S
R
f
f =
S
f
56 , 2
S
R
f
f =
44
History
1980 - PRODERA Modal analysis systems of the France origin
Since the sixties, PRODERA has been manufacturing marketing world-wide complete
systems aimed at Modal Analysis. PRODERA was the first company in the world to use
microprocessors in its modal analysis systems.
The name of the computer was INTERTECHNIQUE IN-110.
f(t)
g(t)
z(t)
r
1878 Kelvins harmonic analyzer
45
Brel & Kjr signal analyzers for laboratory measurements
1996, PULSE,
multichannel analyzer
of the BK 3560 type
with an external DSP
and connection of a
front-endu to PC
using LAN.
1983 the first dual channel analyzers of the BK 2032 and 2034 type offers FFT and
Hilbert transform
1992, the first
multichannel
analyzer of the
BK 3550 type
End of 90
th
, PULSE,
multichannel analyzer of
the BK 3560 type
without an external DSP
and connection of a
front-end to PC using
LAN.
Present days,
PULSE
Before 1980 High Resolution (narrow band) Signal Analyzer BK 3031 and 2033
(single channel) with a desk-top calculator
46
SKF Portable signal analyzers for diagnostic practice
SKF Microlog Vibration
Spectrum Analyzer and
Data Collector
SKF Microlog
CMVA 65
Data Collector and
FFT analyser
Microlog GX series Microlog MX series Microlog CMXA 51-IS
SKF Microlog Analyzer
AX
Data Collector and FFT
analyser
Condition monitoring
47
Multifunction cards
8 analog inputs (14-bit, 48 kS/s)
2 analog outputs (12-bit, 150 S/s); 12 digital I/O; 32-bits
counter
Connection via USB
NI-DAQmx driver software and NI LabVIEW
SignalExpress LE interactive data-logging software
Low price multifunction cards NI USB-6009
Products of National Instruments, PCI a PXI.
A card for dynamic measurements, NI USB-4432
5 analog inputs (24bits - convertors, sampling 102.4 kS/s
per channel), AC a DC coupling, powering of acc (IEPE)
Input voltage 40 V,
48
49
Fourier Transform
Lecture 5
Joseph Fourier, Mmoire sur la propagation de la chaleur dans les corps solides. (1807)
( ) ...
2
5 cos
2
3 cos
2
cos +
t
' ' ' +
t
' ' +
t
' =
y
a
y
a
y
a y
( )
2
1 2 cos
y
k
t
+
( ) ( )
}
+
t
+ =
1
1
d
2
1 2 cos y
y
k y a
k
In these few lines, which are close to the modern formalism used in Fourier series, Fourier
revolutionized both mathematics and physics. [http://en.wikipedia.org/wiki/Fourier_series]
Multiplying both sides by and then integrating from y = -1 to y = +1 yields:
50
One-dimensional Fourier Transform
Signal processing
51
Spectral analysis in machine diagnostics
Centrifugal air compressor
Motor
Gear
Compressor
Frequency
Vibration and noise
spectra produced by
machines are composed
of tonal components,
which are originated by
gears, bearings, motors,
etc.
The amplitude of the
tonal components
reflects the technical
state of the mentioned
machine parts.
The spectral analysis is
a useful tool for
machine diagnostics.
Bearing
52
The one-dimensional Fourier Transform - summary
( ) ( ) ( )
}
+

e = e dt t j t x X exp
( ) ( ) ( )
}
+

e e e
t
= d t j X t x exp
2
1
( )
=
t =
1
0
2 exp
N
n
n k
N nk j x X
( )
=
t =
1
0
2 exp
1
N
k
k n
N nk j X
N
x
Continuous signal
Let x(t) be a function of time on the
infinite time interval
D
i
r
e
c
t

Discrete time
Let N be a length of a period of a discrete signal
(finite time interval)
I
n
v
e
r
s
e

1 ..., , 1 , 0 = N n
1 ..., , 1 , 0 = N k
( ) ( ) + e , , t t x ,... 2 , 1 , 0 , = =
+
n x x
N n n
The Discrete Fourier Transform (DFT) of N input
samples produces the same number of the output
complex values
A function x(t) has a direct and inverse
Fourier transform provided that
( )
}
+

dt t x
exists and there is a finite number of
discontinuities.
(
(
(
=
1
0
N
x
x
x
(
(
(
=
N
X
X
1
X
53
Fourier transform of a cosine signal
( ) ( ) ( ) + e e = , , cos
0
t t A t x ( ) ( ) ( ) ( ) ( ) ( )
0 0 0
d exp cos e + e o + e e o t = e e = e
}
+

A t t j t A X
Let x(t) be a cosine signal of the infinite duration () Dirac delta function,
+
X()
A(-
0
)
-
x(t)
+t -t
( ) ( ) ( )
( ) ( ) T T t t x
T T t t A t x
+ e =
+ e e =
, , 0
, , cos
0
( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) T T AT
t t j t A X
T
T
0 0
0
Sinc Sinc
d exp cos
e + e + e e =
= e e = e
}
+
x(t)
+t -t
For the cosine signal limited to the finite time interval
+T -T
A
-
0
+
0
+
X()
- -
0
+
0
A AT Sinc((-
0
)T)
Sinc(x) = sin(x)/x
54
Discrete Fourier transform of the sine burst signal
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
x
(
t
)

Time History : 100 Hz - 1/16 s
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
x
(
t
)

Time History : 100 Hz - 1/4 s
0
50
100
-400 -300 -200 -100 0 100 200 300 400
5
1
2

*

A
m
p
l

FFT : 100 Hz - 1/16 s
0
50
100
150
-400 -300 -200 -100 0 100 200 300 400
5
1
2

*

A
m
p
l

FFT : 100 Hz - 1/4 s
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
x
(
t
)

Time [s]
Time History : 10 Hz - 1 s
0
200
400
600
-400 -200 0 200 400
5
1
2

*

A
m
p
l

Frequency [Hz]
FFT : 100 Hz - 1 s
Example no.1 Discrete Fourier transform of a constant
15 ..., , 1 , 0 , 1 = = n x
n
( ) 0 16 2 exp
15
0
= t =
= n
k
nk j X
15 ..., , 0 = k

W
16
0
-2 /16
W
16
1
2 /16
1
( ) 16 0 exp
15
0
0
= =
= n
X
0,0
0,5
1,0
1,5
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
x
n
Index n
0
4
8
12
16
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
X
k
Index k
Let x
n
be a sequence of constant samples
55
Notice the fact that X
0
is equal to the N multiple of the sample value while X
k
is equal to zero for
k > 0 .
56
Discrete Fourier transform of the cosine signal
with the integer number of waves per record
( ) 1 ..., , 1 , 0 , 2 cos = + t = N n N Mn A x
n
( ) ( ) ( ) 2 exp exp cos jx jx x + =
( ) ( ) ( ) ( ) | | ( )
( ) ( ) ( ) ( ) ( ) ( ) | |. 2 exp 2 exp
2
2 exp 2 exp 2 exp
2
1
0
1
0
=
+ + t + + t =
= t + t + + t =
N
n
N
n
k
N n k M j N n k M j
A
N nk j N Mn j N Mn j
A
X
( ) ( ) ( ) ( ) 1 ..., , 1 , 0 , 2 2 exp 2 exp = + t + + t = N n N Mn j N Mn j A x
n
( ) ( ) M N M k X j
N
A X j
N
A X
k M N M
= = = =

, , 0 , exp
2
, exp
2
Let
It is assumed that N, M are integer numbers 2 0 N M s <
Using the formula the cosine function is changed into the exp function
After substituting into the discrete Fourier transform formula
we obtain
M is equal to the number of waves
per record of the length N. be a signal to be transformed.
Notice the fact that the absolute value of X
M
and X
N-M
is equal to the N / 2 multiple
of the harmonic signal amplitude while X
k
is equal to zero for k <> M, N - M .
57
Example no.2 Discrete Fourier transform of the periodic
signal with the integer number of waves per record
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
x
n
Index n
15 ..., , 0 = k
-16
-8
0
8
16
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Re(X
k
)
Index k
-16
-8
0
8
16
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Im(X
k
)
Index k
M = 3
N - M = 13
symmetry
( ) 15 ,..., 0 , 16 3 2 sin = t = n n x
n
( )
( )
=
= t
= t
=
13 , 3 0
13 , 2 exp 8
3 , 2 exp 8
k
k j
k j
X
k
Let
The Fourier transform is as follows
3 waves per record
The Fourier transform of a harmonic function having an integer number of waves per record for FFT
be a signal to be transformed.
{ } { } { } { } 0 , Im Im and Re Re > = =

M X X X X
M N M M N M
Notice the complex conjugate symmetry of the Fourier transform
periodic
extension
smooth
continuation
58
Example no.3 - Fourier transform of the periodic signal
with the fractional number of waves per record
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0 2 4 6 8 10 12 14 16
x
n
Index n
0
2
4
6
8
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
|X
k
|
Index k
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0 2 4 6 8 10 12 14 16 18 20 22 24 26 28 30 32 34 36 38 40 42 44 46 48
x
n
Index n
( ) 15 ,..., 0 , 16 5 . 3 2 sin = t = n n x
n
Notice the fractional number of waves per record (3.5)
which contains 16 samples.
Period of 16 samples for
evaluation FT
Peaks
are smeared
irregularities in the sequence of the adjacent periods
Let
Fourier transform of a harmonic function having a non-integer number of waves per record for FFT
be a signal.
3.5 waves per record
The fractional number of the waves in a record for computing FT causes smearing of the spectrum.
59
Positive and negative frequency
0 1 2 3 4 15 14 13 8
M N-1 N-M
0 1 2 3 4 -1 -2 -3 8
M -1 -M
-1 -3 -2 -4
-M
-5 -6 -8 -7
|X
N-M
|
|X
N-M
|
|X
M
|
|X
M
|
|X
-M
|
|X
k
|
|X
k
|
f
S

f
S

f
S
/2
f
S
/2
- f
S
/2
Negative
frequency
components
frequency
Positive
angle
Positive
frequency
components
index
frequency
index
frequency
index
Negative
angle
f
S
....... sampling frequency
f
S
/2 ... Nyquist frequency
N = 16, k = 0, 1, 2, ..., 15
symmetry
symmetry
Im
Re
( ) 15 ,..., 0 , 16 3 2 sin = = n n x
n
t
2-side spectrum
Let be the sinusoidal signal to be transformed by the use of the FT.
For a real signal the Fourier transform results in the spectrum with amplitude which is symmetric
about 8 (f
S
/2).
If the signal to be transformed is a complex, then the spectrum may be non-symmetric.
{ } { } { } { } 0 , Im Im or Re Re > = =

M X X X X
M N M M N M
60
( ) + e +
=
t j
P
e
A
X
2
A real harmonic signal as a sum of the complex-conjugate
phasor pair rotating in the opposite direction
N P
X X X + =
( ) + e
=
t j
N
e
A
X
2
Re
Im
( ) ( )
( ) ( ) ( ) ( )
N P
X X
t j t j
A t A t x + =
+ e + + e
= + e =
2
exp exp
cos
t time
A ... amplitude
angular velocity
initial phase
= t + phase
Complex
plane
Eulers formula
+
-
Phasors X
P
and X
N
are
complex conjugates
Notice the symmetry of phasors for real harmonic signals. It can be
concluded that, generally, the following can be applied to any real signal
( ) ( ) ( ) ( )
N P N P
X X X X Im Im and Re Re = =
61
( )
P
t j
P P
e A X
+ e +
=
A complex harmonic signal as a sum of the non-complex-
conjugate phasor pair rotating in the opposite direction
N P
X X X + =
( )
N
t j
N N
e A X
+ e
=
Re
Im
t time
A
P
and A
N
......................................... amplitude of phasors
.. angular velocity
P
and
N
.... initial phase of phasors
P
= t+
P
and
N
= -(t+
N
) phase of phasors
Complex
plane
If the phasors X
P
and X
N
are not complex conjugate phasors, then the signal is complex
+
-
Subscripts P and N indicate that the phasor is rotating in positive (P) or negative (N) direction.
Re
Im
P
X
N P
X X X + =
N
X
t
The locus of all endpoints of
the phasor X is an ellipse.
One-side and two-side spectrum
-1,5
0,0
1,5
-1,5 0,0 1,5
I
m
(
x
)

Re(x)
-1,5
0,0
1,5
0,0 1,0
R
e
(
x
)
,

I
m
(
x
)

Time [s]
Re(x)
Im(x)
0
10
20
30
40
50
60
70
-5 -4 -3 -2 -1 0 1 2 3 4 5
|
X
|

Frequency [Hz]
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,5 1,0
R
e
(
x
)

Time [s]
0
8
16
24
32
40
-5 -4 -3 -2 -1 0 1 2 3 4 5
|
X
|

Frequency [Hz]
0
8
16
24
32
40
0 1 2 3 4 5
|
X
|

Frequency [Hz]
Two-side Fourier transform One-side Fourier transform
non-symmetry
FT
FT
62
( ) 63 ..., , 1 , 0 , 64 2 sin = t = n n x
n
Let be a real signal to be transformed using the Fourier transform
( ) ( ) 63 ..., , 1 , 0 , 3 64 2 sin 64 2 sin = t + t + t = n n j n x
n
Let be a complex signal to be transformed
Time history Spectrum is symmetric about 0 Hz Symmetric part is omitted
Two-side Fourier transform
Time history Spectrum is non-symmetric Orbit plot
63
Inverse Fourier transform (IFT)
Im
Re
-
0
+
0
-3
0
+3
0
-5
0
+5
0
+
0
+3
0
+5
0
+

+t

+t

+t

+t

x
1
(t)

x
3
(t)

x
5
(t)

x(t)

0

-j
0
n t, n = 0, 1, ,15
j
1
-jt

j
Continuous time t
continuous rotation
Discrete time
nt stepwise rotation
N = 16 samples per record
T
0
= 2/T
The periodic signal as a sum of the phasor pairs which are rotating in the opposite direction
exp(-j
0
t)
exp(+j
0
t)
T
T
T
T
0
0
0
0
64
Inverse discrete Fourier transform (IDFT)
The complex conjugate of the complex number is given by the formula jb a X + =
jb a X =
*
( )
*
*
n n
x x = Let complex conjugate operation be applied two times then because yields
The inverse discrete Fourier transform may be written in the form
Im
Re
X
X*
Notice the fact that the inverse Fourier transform does not require a special algorithm.
The formula for calculation of the inverse Fourier transform
( ) . 1 ..., , 1 , 0 , 2 exp
1
1
0
= t =

=
N n N nk j X
N
x
N
k
k n
( ) ( ) ( ) . 1 ..., , 1 , 0 , 2 exp 2 exp
1
*
1
0
*
*
1
0
*
*
*
=
|
|
.
|
\
|
t =
|
|
.
|
\
|
|
.
|
\
|
t = =

=
=
N n N nk j
N
X
N nk j X
N
x x
N
k
k
N
k
k n n
. 1 ..., , 1 , 0 , DFT
*
*
=
|
|
.
|
\
|
)
`
= N n
N
X
x
k
n
65
Effect of integration and differentiation
on the amplitude of harmonic signals
( ) ( ) t D t d e = cos
( ) ( ) ( ) t
A
t t A t v e
e
= e =
}
sin d cos ( )
( ) ( )
( ) t D
t
t d
t v e e = = sin
d
d
( )
( ) ( ) ( ) ( )
( ) t D
t
t d
t
t v
t a e e = = = cos
d
d
d
d
2
2
2
( ) ( ) t A t a e = cos
( ) ( ) ( ) ( ) t
A
t t t A t t
A
t D e
e
= e = e
e
=
} } }
cos d d cos d sin
2
1
Displacement
Velocity
Acceleration
Filtration in the frequency domain
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Time History : Sine1+Sine2
0
100
200
300
400
500
600
0 200 400 600 800 1000
0,0
0,5
1,0
1,5
0 200 400 600 800 1000
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Time : Generator 2 : Sine1+Sine2 - 2
0
100
200
300
400
500
600
0 200 400 600 800 1000
Frequency [Hz]
FFT
IFFT
f
S
/2 f
S
FFT
( ) ( ) ( ) ( ) t x f X f X t x
F F

IFFT Filter
( ) t x
( ) f X
( ) t x
F
( ) f X
F
symmetry
weighting
function
66
67
Effect of the plot scale on the diagram appearance
0,0001
0,0010
0,0100
0,1000
0 200 400 600 800 1000
Frequency [Hz]
Integration
Logarithmic scales
symmetry
( ) e e j G 1 =
0,0001
0,0010
0,0100
0,1000
1 10 100 1000
Frequency [Hz]
10
100
1000
10000
1 10 100 1000
Frequency [Hz]
0
1000
2000
3000
4000
0 200 400 600 800 1000
Frequency [Hz]
( ) e e j G = Differentiation
Linear scales
L
o
g
a
r
i
t
h
m
i
c

s
c
a
l
e
s

+20 dB/dec
-20 dB/dec
L
o
g
a
r
i
t
h
m
i
c

s
c
a
l
e

L
i
n
e
a
r

s
c
a
l
e

Nyquist frequency Nyquist frequency
The roll-off of the frequency response is -20 dB/dec for integration and +20 dB/dec for differentiation
Integration in the frequency domain
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
0
100
200
300
400
500
600
0 200 400 600 800 1000
0,0001
0,0010
0,0100
0,1000
0 200 400 600 800 1000
0,0
0,5
1,0
1,5
2,0
0 200 400 600 800 1000
Frequency [Hz]
FFT
IFFT
f
S
/2 f
S
FFT
( )
( ) ( )
( ) t x
f f j f X
f j f X
f f
f f
I
S S
S
t t
t t
>
<
2 2
2 2
: 2
: 2
IFFT
Integration
( ) t x
( ) f X
( )dt t x
}
( ) f X
I
symmetry
weighting
function
f t = e 2
f t 2 1 ( ) f f
S
t 2 1
( ) ( ) ( ) ( ) f X f f X f X f f X
I S I S
* *
, = =
( ) ( ) f X t x t 2
-5,E-03
-3,E-03
7,E-18
3,E-03
5,E-03
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
68
Differentiation in the frequency domain
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
0
100
200
300
400
500
600
0 200 400 600 800 1000
0
1000
2000
3000
4000
0 200 400 600 800 1000
-1000
-500
0
500
1000
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
0,E+00
5,E+04
1,E+05
2,E+05
2,E+05
3,E+05
3,E+05
0 200 400 600 800 1000
Frequency [Hz]
FFT
IFFT
f
S
/2 f
S
IFFT
Differentiation
( ) t x
( ) f X
( ) dt t dx
( ) f X
D
symmetry
weighting
function
f t = e 2
f t 2 ( ) f f
S
t 2
( ) ( ) ( ) ( ) f X f f X f X f f X
D S D S
* *
, = =
FFT
( )
( ) ( )
( ) t x
f X f f j
f X f j
f f
f f
I
S S
S
t t
t t
>
<
2 2
2 2
: 2
: 2
( ) ( ) f X t x t 2
69
70
Fast Fourier Transform
Lecture 6
James W. Cooley and John W. Tukey,
An algorithm for the machine calculation of complex Fourier series,
Math. Comput. 19, 297301 (1965).
71
Number of complex additions and multiplications
( ) ( ) ( ) N j N N j W
N
t t = t = 2 sin 2 cos 2 exp
( ) { } ( ) ( ) 1 ..., , 2 , 1 , 0 for , 2 exp
1
0
= t =
=
N k N ik j i x i x X
N
i
N
k
.
N k
N
k
N
W W
+
=
2 N k
N
k
N
W W
+
=
The N-point discrete Fourier transform require N
2
complex multiplications and additions (operation)
Periodicity properties of the W term in the FT formula
Twiddle factor
( ) N k j W
k
N
t = 2 exp
, then it can be proved that Let N be a power of 2 ( )
m
N 2 =
k
N
W
2 N k
N
W
+
2 N
steps
k
N
W
N k
N
W
+
N
steps
1
0
=
N
W
1
N
W
N t 2
See [Oppenheim & Schafer]
History Note - Milestones
DIT - Decimation in time
DIF - Decimation in frequency
This algorithm, including its recursive application, was invented around 1805 by Carl Friedrich
Gauss, who used it to interpolate the trajectories of the asteroids Pallas and Juno.
Computational methods for the Fast Fourier Transform
James W. Cooley and John W. Tukey,
An algorithm for the machine calculation of complex Fourier series,
Math. Comput. 19, 297301 (1965).
Runge in 1903 which essentially described the FFT. Danielson and Lanczos in 1942 recognized
certain symmetries and periodicities which reduced the number of operations.
FFTs for N = 2
m
became popular after J. W. Cooley of IBM and John W. Tukey of Princeton
published a paper:
The fast Fourier transform belongs to the 10 algorithms with the greatest influence
on the development and practice of science and engineering in the 20th century
72
See [Oppenheim & Schafer]
73
Principle
( ) 1 ,..., 1 , 0 , 2 exp
1
0
1
0
= = =

=
=
N k W x N kn j x X
kn
N
N
n
n
N
n
n k
t
Let N be a composite N = N
1
N
2
in the N-point discrete Fourier transform (DFT)
then we can rewrite the previous formula by letting indexes
1 2 1
N k k k + =
2 2 1
n N n n + =
We then have
1 ,..., 2 , 1 , 0 ,
1 2 1
1
0
1
0
2 2
2
2
2
2 1
1
1
1 1
1
2 2 1 1 2 1
= +
(
(
|
|
.
|
\
|
=

=
+ +
N N k k W W W x X
n k
N
N
n
n k
N
N
n
n k
N
n N n N k k
multiplies the result by the so-called twiddle factors W
N
k1n2
, and finally computes N
1
DFTs
of the size N
2
(the outer sum).
The term radix is used to describe N
1
and N
2
. If N is a power of two then the small DFT of the
radix is called a butterfly.
( ) ( ) ... and 1 ,..., 2 , 1 , 0 ,
1 2 2
1
0
1
1
1
1 1
1
2 2 1 2 2 1 1 2
= = = =

=
+ +
k N n W x x FT k Y
N
n
n k
N
n N n n N n N n
The algorithm computes N
2
DFTs of size N
1
(the inner sum),
This decomposition is then continued recursively.
( ) ( ) ( ) ( ) ... and 1 ,..., 2 , 1 , 0 ,
2 1 1
1
0
1 1 2
2 2
2
2
2
2 2 2 1 2 1
= = = =

=
+
k N n W k Y k Y FT k X
n k
N
N
n
n n N N k k
74
Decimation in time, the radix-2 DIT case
( )
( ) ( ) ( ) ( ) ( ). 2 2 exp 2 exp 2 2 exp
2 exp
1 2
0
1 2
1 2
0
2
1
0
N k j x N k j N k j x
N kn j x X
N N
N
n
n k
v t t + v t =
= t =

= v
+ v
= v
v
=
( ) 1 2 ..., , 1 , 0 for , 2 exp = t + = N k H N k j G X
k k k
, , ,
2
2 2
k
N
N k
N k N k k N k
W W H H G G = = =
+
+ +
( ) 1 2 ..., , 1 , 0 for , 2 exp
2
= t =
+
N k H N k j G X
k k N k
1 2 ..., , 1 , 0 for ,
2
=
=
+ =
+
N k
H W G X
H W G X
k
k
k N k
k
k
k k
Even-indexed samples Odd-indexed samples
Summary
Using periodicity properties we obtain the values X
k
,
where k > N/2-1
Let N-point DFT, where N = 2
m
, be decomposed
into
{ }
v v
=
2 2
FT x G
N
{ }
1 2 2
FT
+ v v
= x H
N
Two N/2-point DFTs:
( ) N k j W
k
N
t = 2 exp
Radix-2 DIT divides a DFT of size N into two
interleaved DFTs (hence the name "radix-2")
of size N/2 with each recursive stage.
The values X
k
, where k < N/2

Speeding up the calculation of FT is based on splitting the sequence into two parts.
75
Radix-2 butterfly diagram
a gain
Rules for creating signal-flow graph
Appearance of a butterfly
( ) ( )
( ) ( ). ,
, ,
1 1 0 1 1 1 0 1
1 1 0 0 1 1 0 0
C Y S X Y Y S Y C X X X
C Y S X Y Y S Y C X X X
+ = ' = '
+ + = ' + = '
Addition and multiplication of complex numbers
addition

y
x
y
x
1

y = x
1
+ x
2

x
2

y = a x
a
+
+
Signal diagrams
0 0
jY X +
1 1
jY X +
0 0
Y j X ' + '
1 1
Y j X ' + '
1
x
y x y =
x
y x a y =
a
1
x
2
x
y
1
a
2
a
b
( )
2 2 1 1
x a x a b y + =
76
Signal diagram for the 8-point DFT

x(0)
x(2)
x(4)
x(6)
DFT
N/2
x(1)
x(3)
x(5)
x(7)
DFT
N/2
X(0)
X(1)
X(2)
X(3)
X(4)
X(5)
X(6)
X(7)
G (0)
G(1)
G(2)
G(3)
H(0)
H(1)
H(2)
H(3)
-1
-1
-1
-1
W
0
W
1
W
2
W
3

x(0)
x(4)
x(2)
x(6)
DFT
N/4
G(0)
G(1)
G(2)
G(3)
GG(0)
GG(1)
HG(0)
HG(1) -1
-1
DFT
N/4
W
0
W
2

x(0)
x(4)
GG(0)
GG(1)
-1
( )
( ) . 2 1 2 exp
, 2 0 2 exp
4 0 4 0 1
4 0 4 0 0
x x x j x GG
x x x j x GG
= t =
+ = t + =
N-point DFT N
2
operations
Two N/2-point DFT
(N/2)
2
+ (N/2)
2
= N
2
/2 operations
( ) ( ) ( ) ( ) ( ). 2 2 exp 2 exp 2 2 exp
1 2
0
1 2
1 2
0
2
N k j x N k j N k j x X
N N
k
v t t + v t =

= v
+ v
= v
v
G
k
W
k
H
k

Three-stage algorithm
Number of operations:
1
st
stage: 1 x 8
2
nd
stage: 2 x 4
3
rd
stage: 4 x 2
Total: 24
1 operation =
1 complex multiplication
and 1 complex addition
X
k
= +
*
77
Visual Basic code for FFT, recursive DIT case
Public Type complex
re As Double: im As Double
End Type
Public Sub MyDITFFT(z() As complex, zz() As complex)
Dim N As Long, N2 As Long, k As Long, fi As Double, w As complex
Dim z0() As complex, z1() As complex, G() As complex, H() As complex
N = UBound(z) + 1
ReDim zz(N - 1) As complex
If N = 2 Then
zz(0).re = z(0).re + z(1).re: zz(0).im = z(0).im + z(1).im
zz(1).re = z(0).re - z(1).re: zz(1).im = z(0).im - z(1).im
Else
N2 = N / 2
ReDim z0(N2 - 1) As complex, z1(N2 - 1) As complex
ReDim G(N2 - 1) As complex, H(N2 - 1) As complex
For k = 0 To N2 - 1
z0(k) = z(2 * k): z1(k) = z(2 * k + 1) Even- and odd-indexed samples
Next k
MyDITFFT z0, G Recursive calling
MyDITFFT z1, H Recursive calling
For k = 0 To N2 - 1
fi = 2 * pi / N * k: w.re = Cos(fi): w.im = -Sin(fi)
zz(k).re = G(k).re + w.re * H(k).re - w.im * H(k).im
zz(k).im = G(k).im + w.re * H(k).im + w.im * H(k).re
zz(k + N2).re = G(k).re - w.re * H(k).re + w.im * H(k).im
zz(k + N2).im = G(k).im - w.re * H(k).im - w.im * H(k).re
Next k
End If
End Sub
Recursive calls
78
Complete signal diagram for the 8-point DFT

x(0) X(0)
x(4)
X(1)
x(2) X(2)
x(6) X(3)
x(1) X(4)
x(5) X(5)
x(3) X(6)
x(7) X(7)
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
W
N
0
W
N
0
W
N
0
W
N
0
W
N
0
W
N
0
W
N
2
W
N
2
W
N
2
W
N
1
W
N
3
W
N
0
1
st
STAGE 2
nd
STAGE 3
rd
STAGE
Reverse
bit order
input
Notice of the sequence sample index
1
st
stage: 4 x 2 additions and multiplications
2
nd
3
rd
79
Reverse order of bits
Initial index
order (Decimal
numbers)
Binary numbers Reversing bit
order in binary
numbers
Final index
order (Decimal
numbers)
0 000 000 0
1 001 100 4
2 010 010 2
3 011 110 6
4 100 001 1
5 101 101 5
6 110 011 3
7 111 111 7
Data reordering performs the index register of the Digital Signal Processors (DSP)
000
100
010
110
001
101
011
111
80
Pascal routine for FFT, non-recursive DIT case
PROCEDURE DIT(p,VAR f);

LOCAL Bp,Np,Np',P,b,k,BaseT,BaseB,top,bot;
BEGIN {DIT}
{initialise pass parameters}
Bp:=1<<(p-1);{No. of blocks}
Np:=2; {No. of points in each block}
{perform p passes}
FOR P:=0 TO (p-1) DO
BEGIN {pass loop}
Np':=Np>>1; {No. of butterflies}
BaseT:=0; {Reset even base index}
FOR b:=0 TO (Bp-1) DO
BEGIN {block loop}
BaseB:=BaseT+Np'; {calc odd base index}
FOR k:=0 TO (Np'-1) DO
BEGIN {butterfly loop}
top:=f[BaseT+k];
bot:=f[BaseB+k]*T(Np,k); {twiddle the odd n results}
f[BaseT+k]:= top+bot; {top subset}
f[BaseB+k]:= top-bot; {bottom subset}
END; {butterfly loop}
BaseT:=BaseT+Np; {start of next block}
END; {block loop}
{calc parameters for next pass}
Bp:=Bp>>1; {half as many blocks}
Np:=Np<<1; {twice as many points in each block}
END; {pass loop}
END; {DIT}
http://www.engineeringproductivitytools.c
om/stuff/T0001/PT04.HTM#Head317
{Perform in place DIT of 2^p points (=size
of f) N.B. The input array f is in bit
reversed order!
So all the 'even' input samples are in the
'top' half, all the 'odd' input samples are in
the 'bottom' half..etc (recursively). }
81
Decimation in frequency, the radix-2 DIF case
1 2 ..., , 1 , 0 , 2 , = = N k X
k
( ) ( ) ( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ). 2 2 exp
2 2 2 exp 2 2 exp
2 2 exp 2 2 exp
1 2
0
2
1 2
0
2
1 2
0
1
0
1
0
2
N n j x x
N N n j x N n j x
N n j x N n j x X
N
n
N n n
N
n
N n
N
n
n
N
n
n
N
n
n
t + =
= + t + t =
= t = t =
=
+
=
+
( ) ( )
( ) ( ) ( )
( ) ( ) ( ) ( ). 2 2 exp 2 exp
2 2 exp 2 exp
1 2 2 exp
1 2
0
2
1
0
1
0
1 2
N n j N n j x x
N n j N n j x
N n j x X
N
n
N n n
N
n
n
N
n
n
t t =
= t t =
= + t =
=
+
=
+
1 2 ..., , 1 , 0 , 1 2 , = + = N k X
k
Even-indexed output
Odd-indexed output
( )
1 2 ..., , 1 , 0
,
,
2
2
=
=
= +
+
+
N n
s W x x
r x x
n
n
N n n
n N n n
( ) ( )
( ) ( ). 2 2 exp
, 2 2 exp
1 2
0
1 2
1 2
0
2
N n j s X
N n j r X
N
n
n
N
n
n
t =
t =
=
+
Resulting formulas
Let
be used for substitution then
82
8-point DFT using the DIF method
( ) ( ) 2 2 exp
1 2
0
1 2
N n j s X
N
n
n
t =

=
+

x(0)
x(2)
x(4)
x(6)
DFT
N/2
x(1)
x(3)
x(5)
x(7)
DFT
N/2
X(0)
X(2)
X(4)
X(6)
X(1)
X(3)
X(5)
X(7)
r(0)
r(1)
r(2)
r(3)
s(0)
s(1)
s(2)
s(3)
-1
-1
-1
-1
W
0
W
1
W
2
W
3
( ) ( ) 2 2 exp
1 2
0
2
N n j r X
N
n
n
t =

n N n n
r x x = +
+ 2
( )
n
n
N n n
s W x x =
+ 2
R
e
v
e
r
s
e

b
i
t

o
r
d
e
r

o
u
t
p
u
t

R
e
v
e
r
s
e

b
i
t

o
r
d
e
r

i
n
p
u
t

83
Visual Basic code for FFT, recursive DIF case
Public Type complex
re As Double: im As Double
End Type
Public Sub MyDIFFFT(z() As complex, zz() As complex)
Dim N As Long, N2 As Long, k As Long, fi As Double, w As complex
Dim r() As complex, s() As complex, G() As complex, H() As complex
N = UBound(z) + 1
ReDim zz(N - 1) As complex
If N = 2 Then
zz(0).re = z(0).re + z(1).re: zz(0).im = z(0).im + z(1).im
zz(1).re = z(0).re - z(1).re: zz(1).im = z(0).im - z(1).im
Else
N2 = N / 2
ReDim z0(N2 - 1) As complex, z1(N2 - 1) As complex, G(N2 - 1) As complex
ReDim H(N2 - 1) As complex, r(N2 - 1) As complex, s(N2 - 1) As complex
For k = 0 To N2 - 1
fi = 2 * pi / N * k: w.re = Cos(fi): w.im = -Sin(fi)
z0(k) = z(2 * k): z1(k) = z(2 * k + 1) Even- and odd-indexed samples
r(k).re = z(k).re + z(k + N2).re: r(k).im = z(k).im + z(k + N2).im
s(k).re = (z(k).re - z(k + N2).re) * w.re - (z(k).im - z(k + N2).im) * w.im
s(k).im = (z(k).re - z(k + N2).re) * w.im - (z(k).im - z(k + N2).im) * w.re
Next k
MyDIFFFT r, G Recursive calling
MyDIFFFT s, H Recursive calling
For k = 0 To N2 - 1
zz(k) = G(2 * k): zz(k) = H(2 * k + 1) Even- and odd-indexed output values
Next k
End If
End Sub
Recursive calls
84
Pascal routine for FFT, non-recursive DIF case
PROCEDURE DIF(p,VAR f);
LOCAL Bp,Np,Np',P,b,n,BaseE,BaseO,e,o;
BEGIN {DIF}
{initialise pass parameters}
Bp:=1; {No. of blocks}
Np:=1<<p; {No. of points in each block}
{perform p passes}
FOR P:=0 TO (p-1) DO
BEGIN {pass loop}
Np':=Np>>1; {No. of butterflies}
BaseE:=0; {Reset even base index}
FOR b:=0 TO (Bp-1) DO
BEGIN {block loop}
BaseO:=BaseE+Np'; {calc odd base index}
FOR n:=0 TO (Np'-1) DO
BEGIN {butterfly loop}
e:= f[BaseE+n]+f[BaseO+n];
o:=(f[BaseE+n]-f[BaseO+n])*T(Np,n);
f[BaseE+n]:=e;
f[BaseO+n]:=o;
END; {butterfly loop}
BaseE:=BaseE+Np; {start of next block}
END; {block loop}
{calc parameters for next pass}
Bp:=Bp<<1; {twice as many blocks}
Np:=Np>>1; {half as many points in each block}
END; {pass loop}
END; {DIF}

See [http://www.engineeringproductivitytools.com/stuff/T0001/PT03.HTM#Head134]
{Perform in place DIF of 2^p points (=size of f)}
85
Comparison of the computional operation number with
the use of DFT and FFT
The Discrete Fourier Transform, where N is an arbitrary number, requires N
2
complex
multiplications and additions
The Cooley-Tukey algorithm for the Fast Fourier Transform, based on N = 2
m
, requires only
Nm = N log
2
N complex multiplications and additions. It is said that the algorithm is performed in
O(N log N) time.

An example

Let N be the teth power of two (N = 1024 = 2
10
)

DFT approximately 1 000 000 operations

FFT 10 240 operations

Each stage of the FFT algorithm, which is divided into two algorithms applied to half the number
of points, requires (N/2)
2
+ (N/2)
2
= N
2
/2 complex multiplications and additions.
N-point
Other FFT algorithms
radix-4
N = N
1
N
2
with coprime numbers N
1
and N
2
(they have no common factor other than 1)

86
FFT in Matlab
We can find out in the Manlab function references that the FFT functions (fft, fft2, fftn, ifft, ifft2,
ifftn) are based on a library called FFTW [3], [4]. To compute an N-point DFT when N is composite
(that is when N = N
1
N
2
), the FFTW (the fastest Fourier transform in the West http://www.fftw.org)
library decomposes the problem using the Cooley-Tukey algorithm [1], which first computes N
1

transforms of size N
2
, and then computes N
2
transforms of size N
1
. The decomposition is applied
recursively to both the N
1
- and N
2
-point DFTs until the problem can be solved using one of several
machine-generated fixed-size "codelets". The codelets in turn use several algorithms in combination,
including a variation of Cooley-Tukey [5], a prime factor algorithm [6], and a split-radix algorithm
[2]. The particular factorization of N is chosen heuristically.
When N is a prime number, the FFTW library first decomposes an N-point problem into three
(N-1)-point problems using Rader's algorithm [7]. It then uses the Cooley-Tukey decomposition
described above to compute the (N-1)-point DFTs.
The execution time for fft depends on the length of the transform. It is the fastest for powers of two.
It is almost so fast - as for the length - that it equals to powers of two if the prime factorization of the
length has only small prime factors. It is typically several times slower compared to the previous
case if the length is a prime or has large prime factors.
[1] Cooley, J. W. and J. W. Tukey, "An Algorithm for the Machine Computation of the Complex Fourier Series,"
Mathematics of Computation, Vol. 19, April 1965, pp. 297-301.
[2] Duhamel, P. and M. Vetterli, "Fast Fourier Transforms: A Tutorial Review and a State of the Art," Signal
Processing, Vol. 19, April 1990, pp. 259-299.
[3] Frigo, M. and S. G. Johnson, "FFTW: An Adaptive Software Architecture for the FFT," Proceedings of the
International Conference on Acoustics, Speech, and Signal Processing, Vol. 3, 1998, pp. 1381-1384
[4] Oppenheim, A. V. and R. W. Schafer, Discrete-Time Signal Processing, Prentice-Hall, 1989, p. 611 and 619.
87
Authors of the Fastest Fourier Transform in the West
Matteo Frigo received his Ph. D. in 1999 from the Dept. of Electrical
Engineering and Computer Science at the Massachusetts Institute of
Technology (MIT). Besides FFTW, his research interests include the theory
and implementation of Cilk (a multithreaded system for parallel
programming), cache-oblivious algorithms, and software radios.
Joint recipient, with Steven G. Johnson, of the 1999 J. H. Wilkinson Prize for
Numerical Software, in recognition of their work on FFTW.
Steven G. Johnson joined the faculty of Applied Mathematics at MIT in 2004.
He received his Ph. D. in 2001 from the Dept. of Physics at MIT, where he also
received undergraduate degrees in computer science and mathematics. His recent
work, besides FFTW, has focused on the theory of photonic crystals:
electromagnetism in nano-structured media.
See [http://www.fftw.org/]
88
FFTW speed
0
1
2
3
4
5
6
7
8
9
10
11
12
512 576 640 704 768 832 896 960 1024
Number of points for calculation FFT
R
e
l
a
t
i
v
e

s
p
e
e
d
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) N N N N N N N N N N N N
2 2 2 2 2 2
log 2 log 1 2 log 2 log 2 2 log 2 log : 2 ~ + = + =
The fastest FFT algorithm is for the record length that is a power-of-two. Comparison of the relative
computational speeds for a large range of log
2
(N) related to speed for N = 256 is shown in left figure.
0,01
0,10
1,00
10,00
100,00
1000,00
10000,00
1 3 5 7 9 11 13 15 17
k = log
2
(N) = log
2
(2
k
)
R
e
l
a
t
i
v
e

s
p
e
e
d
N = 2
8
= 256
Nlog
2
(N)
Computer
experiment
s
The FFTW algorithm is designed for any record length including non-power-of-two. The relative
computational speeds for the record lengths in between N = 512 and 1024 are shown in right figure.
The fact that doubling the record length results in doubling the computational time can be proved by
89
Zero padding
As it was noted some efficient FFT algorithms require the record length that is a power-of-two.
When applying these algorithms to non-power-of-two length signal records, you have to zero
pad or truncate the record to a power-of-two length. The record of the length M is padded with
zeros to the length N as follows
{ } ( ) ( ) ( ) { } 1 ..., , 1 , 0 , 2 exp 2 exp
1
0
1
0
= = t = t =

=
=
N k x G M N M k i j x N ik j y y F
i N M k
M
i
i
N
i
i i k
+ =
=
=
1 ..., , 1 , , 0
1 ..., , 1 , 0 ,
N M M i
M i x
y
i
i
The Fourier transform of the record y
i
, i = 0, 1, 2, , N -1 is related to the transform of the record
x
i
, i = 0, 1, 2, , M -1 according to the equation
If kM is an integer multiple of N then { } { }
i N M k i k
x G y F and are identical.
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
0
5
10
15
20
0 2 4 6 8 10 12 14 16
Frequency [Hz]
0
5
10
15
20
0 2 4 6 8 10 12 14 16
Frequency [Hz]
32 samples, 3 Hz signal
Hanning window (M = 32)
Magnitude of the FT transform
32-point record (N = 32)
Padded with zeros to128-point
record (N = 128)
Example
90
ZOOM FFT
The frequency resolution of the FFT spectrum in the baseband range is given by the formula
f = 1/T, where T = N / f
S
, N is the length of the input record for calculation FFT and f
S
=1/ t
is the sampling frequency. If the finer resolution over a limited portion of the spectrum is required
then the so-called Zoom-FFT may be applied.
Zooming-in on the mentioned portion of the frequency spectrum is defined by the central frequency
f
C
and the bandwidth F, which is containing the number of lines as the normal baseband spectrum.
The method, called Real-time zoom, is based on three consecutive steps.
( ) ... , 2 , 1 , 0 , 2 exp = A t = i t i f j x y
C i i
1 ,..., 2 , 1 , 0 , 2 exp
1
,
2
exp
1
1
0
1
0
=
|
|
.
|
\
|
|
|
.
|
\
|
t =
|
.
|
\
|
t
=

=
=
N i
f
f
N
k
i j F
N
y i k
N
j F
N
x
N
k
S
C
k i
N
k
k i
1) The shift of the frequency origin to the center of the zoom band is the first step of the algorithm.
The real samples x
i
, i = 0, 1, 2, are multiplied by a rotating unit vector exp(-j2 f
C
t )
The record of the real samples x
i
, i = 0, 1, 2, is transferred into the record of the complex samples
y
i
, i = 0, 1, 2, The index k corresponding to the center of the zoom band is given by the formula
k = N f
C
/ f
S
.
Multiplication by the rotating vector may cause aliasing in the negative frequency band.
91
ZOOM FFT contd
3) The last step is computing the FFT spectrum. Because the input record is composed of the
complex values, the two-side spectrum is computed and presented.
F
f
C
f
N
f
S
-f
N
-f
S
0 -f
N
-f
C
-f
N
+f
C

aliasing
2) To prevent aliasing the real and imaginary part of the complex record is low pass filtered. The
filter cut-off frequency is equal to F/2. The zoom factor is a ratio f
S
/F = (f
N
/F/2). The reduced
frequency range makes it possible to reduce the sampling frequency by a factor of zoom. The
sequence of the complex samples y
i
, i = 0, 1, 2, is decimated by the mentioned zoom factor.
Decimation is performed in cascaded octave steps, what required the zoom factor to be powers of 2.
For the required number of spectral lines it is needed record the corresponding number of samples.
f
N
= f
S
/2 Nyquist frequency
See [Randall]
In addition to the method described above, there is another method, called Non-destructive zoom,
for which the FFT calculation of a long record is split into repeated calculation of shorter records.
Overlap-add method in convolution
92

=

+
=

= = =
1
0 0
M
m
m n m
m
m n m n n n
x h x h x h y
The convolution of the input sequence with the finite impulse response h
m
, m = 0 to M -1 is a formula

where h
m
= 0 for m outside the interval [0, M -1]. For an arbitrary segment length, it yields
=
=
+
otherwise , 0
1 ..., , 1 , 0 , L n x
x
kL n
n
k
=
k
n
k
n
x x

= = = =
k
n
k
k
n
k
n
k
n
k
n n n n
y x h x h x h y
where
k
y
n
is zero outside the interval [0, L+M - 2] of the index n. For any parameter
where the upperleft index k dentes the segment order. The convolution takes the form

1 + > M L N
it is equivalent to the N-point circular convolution in the interval [0, N -1]. The advantage is that the
circular convolution can be computed very efficiently as follows, according to the circular convolution
theorem.
where FFT and IFFT refer to the fast Fourier transform and inverse fast Fourier transform,
respectively, evaluated over N = 2L discrete points (
k
x
n
is padded by L zeros and h
n
is padded by 2L-M
zeros). Long signals can be break into smaller segments for easier processing. FFT convolution uses
the overlap-add method combined with the Fast Fourier transform, which makes it easy to compute
convolution. The Fourier transforms of the input signal and the filter impulse response are multiplied
together.
{ } { } { }
n n
k
n
k
h x FFT y FFT IFFT =
Circular convolution
93
For a periodic function x
T
(t), with period T, the convolution with another function, h(t), is also
periodic, and can be expressed in terms of integration over a finite interval as follows
http://en.wikipedia.org/wiki/Circular_convolution
For discrete sequences x
N, n
= x
N, n+N

+
=
+
=

+
=

|
.
|
\
|
= =
m k
kN m n N m
m
m n N m n n N
x h x h h x
, , ,
( ) ( ) ( ) ( ) ( ) ( )
} }
+ +

= =
T t
t
T T T T
t t h t x t t h t x t h t x
0
0
d d
where t
0
is an arbitrary parameter and h
T
(t) is a periodic extension of h(t) defined by
( ) ( ) ( )

+
=
+
=
+ = =
k k
T
kT t h kT t h t h
When x
T
(t) is expressed as the periodic extension of another function, x, this convolution is
sometimes referred to as a circular convolution of functions h and x.
Overlap-add algorithm
94
1st segment: N = 2L samples in total, L samples of
0
x
n
padded by L
zeros
2nd segment: L samples of
1
x
n
padded by L zeros
L
0
x
n
0
L
0
+
1
x
n

n
x
0
x
n
= x
n

1st segment: N = 2L samples in total, L+M -1 samples equal to y
i

and L -M+1 zeros
2nd segment
L
+
n
y
Input
Convolution of a segment-by-segment
0
x
n
0
Output is produced piecewise by an overlap-add algorithm

h
n

0
0
L
M -1
M
0
y
n

0
0
y
n

0
1
y
n

2L-M
=
=
=
L+M -1 samples of
0
y
n
padded
by L-M+1 zeros
N=2L
L L
1
x
n
= x
n+L

2
x
n
= x
n+2L

Convolution with the use of FFT
95
Circular convolution of segment-by-segment
0
x
n
0
h
n

0
0
M
0
Y
n

2L-M
For filter orders larger than about 64 points, FFT convolution is faster than standard convolution,
while producing exactly the same result.
0
X
n

H
n

FFT
FFT
N = 2L
=
2L
Multiplication element-by-element
0
0
y
n

2L
L L
+
IFFT
{ } { } { }
n n
k
n
k
h x FFT y FFT IFFT =
Generally
FFT Convolution, example

0
2000
4000
0

8

16

24

- 200
0
200
0

2

4

6

200
- 200
0
200
0

2

4

6

0
2000
4000
0

8

16

24

0

2000

4000

0

8

16

24

- 200

0

200

0

2

4

6

- 200

0
200

0

2

4

6

- 200
0
200
0

2

4

6

Segments

FFT

Filter

IFFT

Low

Pass
Low

Pass

+

+

=

=

(FFT Convolution for smoothing
discontinuities of consecutive
records)
Rev

0
2000

4000
0

8

16

24

- 200
0

0

2

4 6
Rev

Ord
Rev
Rev
Rev Rev Ord
96
97
Autospectrum - Cross-Spectrum
Windowing - Averaging
Lecture 7
98
Computation of autospectrum
Time History : Generator 1 : Sine1 - 1
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
FFT : Generator 1 : Sine1 - 1
0
200
400
600
0 100 200 300 400 500
Frequency [Hz]
5
1
2

*

A
m
p
l
Autospectrum : Generator 1 : Sine1 - 1
0,0
0,2
0,4
0,6
0,8
0 100 200 300 400
Frequency [Hz]
R
M
S
A*N/2,
(A*N/2)*2/N/2 = A /2
A, amplitude
N = 1024
The RMS autospectrum requires
change of the spectrum scale
Root Mean Square
A harmonic signal to be analyzed
The magnitude of spectrum
compoment as a result of FFT
99
Autospectrum scales
Let the Fourier transfor of a signal
RMS: Root Mean Square
RMS spectrum
Autospectrum
( )
( )
( ) ( )
=
=
=
. 1 2 ..., , 2 , 1 , 2 2
, 0 , 0
N k N k X
k N X
k RMS
( )
( )
( ) ( ) ( )
=
=
=
. 1 2 ..., , 2 , 1 , 2 2
, 0 , 0
2
2
N k N k X
k N X
k PWR
( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
=
=
=
. 1 2 ..., , 2 , 1 , 2
, 0 , 0 0
*
*
N k N k X N k X
k N X N X
k PWR
( )
( ) ( )
( ) ( )
=
=
=
. 1 2 ..., , 2 , 1 , 2
, 0 , 0
2
2
N k N k X T
k N X T
k PSD
( ) ( ) { } 1 2 ..., , 2 , 1 , 0 , = = N k n x FT k X
( ) 1 ..., , 2 , 1 , 0 , = N n n x
sampling frequency
S S
T f 1 =
PWR = RMS
2
: Power
Power spectrum
PSD = PWR / f :
Power Spectral Density
(diagnostics)
(acoustics)
(random signals)
Unit: U, e.g. [m/s
2
]
Unit: U
2
, e.g. [(m/s
2
)
2
], [Pa
2
]

Unit: U
2
/Hz, e.g. [(m/s
2
)
2
/Hz]
( ) T NT N f f
N f k f
S S
S k
1 1
1
= = =
=
frequency of components
( ) k PSD Unit V/ Hz (Noise in electronic circuits)
be denoted by
100
PWR, RMS, and PSD, dB scale
( ) ( )

=
=
=
=
A = = =
1 2
0
1 2
0
N k
k
N k
k
f k PSD k PWR PWR RMS
PSD
0
0
f
f
S
/ 2
PWR = RMS
2

f
Power Spectral Density
f = 1 / T The frequency difference
f T NT N f f
S S
A = = = = 1 1
1
According to the Parsevals theorem
( ) ( ) ( )

=
=
=
=
A = = =
1 2
0
1 2
0
2
N k
k
N k
k
f k PSD k PWR RMS PWR
sound velocity acceleration force voltage
ref 2x10
-5
Pa

1x10
-6
mm/s

1x10
-6
m/s
2
1x10
-6
N

1 V
Reference values
dB scale
|
|
.
|
\
|
=
ref
y
ref log 20 dB
101
Electrical power
Power is a product of quantities as follows:
voltage x electric current, force x velocity
Let
( ) ( ) ( ) ( )
0
2 , cos , cos e = t = e + e = e = k T t I t i t U t u
The instantaneous power is as
follows
The mean value of the power
per a period T
( ) ( ) ( ) ( ) ( )
( ) =
= = + + e = =
} }
cos
cos
2 2
cos 2 cos
2
1
0 0
RMS RMS
T T
I U
I U
dt t
T
UI
dt t p
T
P
( ) k U
( ) k I
*
( ) ( ) ( ) = cos
*
RMS RMS R
I U k I k U
( ) k I
( ) k I
I
( ) k I
R
( ) ( ) ( ) = sin
*
RMS RMS I
I U k I k U
Real part (active power W)
Imaginary part (reactive power VAR)
Let
(apparent power VA)
( ) ( ) ( ) ( ) ( )
U I U RMS U RMS
j I k I j U k U = = = exp , exp
RMS RMS
I U
( ) k U ( ) k I
e
e
Im
Re
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) + + e = + e = = cos cos
2
cos 2 cos t
UI
t UI t i t u t p
Complex power
be position vectors at t = 0
Phasors in complex plane
be electrical voltage and current.
102
Mechanical power
F(t) force
v(t)
( ) k F
( ) k v
*
( ) k v
( ) ( ) k v k F
*
( ) k F
( ) k v
*
( ) ( ) k d j k v e =
( ) k d
Energy dissipation
( ) ( ) ( ) ( )
0
2 , cos , cos e = t = e + e = e = k T t V t v t F t F
velocity
Vectors
d(t)
( ) ( ) k v k F
*
( ) k F
( ) k v
*
( )
( )
e
=
j
k a
k v
( ) k a
( ) ( ) k v k F
*
acceleration
a(t)
F(t) force
F(t) force
( ) ( ) ( ) ( )... , , , t d t v t a t F
( ) ( ) ( ) ( )... , , , k d k v k a k F
vectors, harmonic (sine, cosine) functions of time
Phasors (position vectors in complex plane) at time t = 0 and rotating
at the angular frequency
0
e = e k
Phasors
Differentiate between vectors (e.g. force, velocity and displacement defined by both the magnitude
and direction) and phasors (rotating position vectors in the complex plane creating a harmonic
signal and determining the phase of this signal).
displacement
active power
103
Cross-spectrum
( ) ( )
( ) ( )
=
=
=
1 2 ..., , 2 , 1 , 2
0 , 0 0
2
*
2
*
N k N k Y k X T
k N Y X T
CROSS
k
( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
=
=
=
. 1 2 ..., , 2 , 1 , 2
, 0 , 0 0
*
*
N k N k X N k X
k N X N X
k PWR
( ) k X
( ) k X
*
( ) ( ) k X k X
*
( ) k X
( ) k Y
*
( ) ( ) k Y k X
R
*
( ) k Y
( ) k Y
I
( ) k Y
R
( ) ( ) k Y k X
I
*
Power as a real value (equivalent P = R I
2
)
Power as a complex value (equivalent P = U I
*
)
Real part
Imaginary part (reactive power)
(active power)
Real part
(active power)
Let
( ) ( ) { } ( ) ( ) { } 1 2 ..., , 2 , 1 , 0 , , = = = N k n y FT k Y n x FT k X
( ) ( ) 1 ..., , 2 , 1 , 0 , , = N n n y n x sampling frequency
S S
T f 1 = be two signals.
The Fourier transform of these signals is as follows
T NT N f f N f k f
S S S k
1 1 ,
1
= = = =
frequency of components
( )
( )
( )
=
=
=
. 1 2 ..., , 2 , 1 , 2
, 0 , 0
2
2
N k N k X
k N X
k PWR
or
104
Weighting of a cosine signal with the use of the rectangular
time window
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) e - e
t
= O O e O
t
= e = e
} }
+

+

T T w
W X W X t t j t w t x X
2
1
d
2
1
d exp
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( )
0 0 0 0
2 d
2
1
e + e + e e = O O e e + O o + e O o t
t
= e
}
+

T T T w
W W A W A X
( ) ( )
( )
( ) 2 Sinc
2
2 sin
d exp
2
2
T T
T
T
T t t j W
T
T
T
e =
e
e
= e = e
}
+
( ) ( ) + e = t A t x cos
( )
+ e
+ e
=
2 , 2 , 0
2 , 2 , 1
T T t
T T t
t w
( ) ( ) ( ) ( ) ( ) ( )
0 0 0
d exp cos e + e o + e e o t = e e = e
}
+

A t t j t A X
( ) ( ) ( ) ( ) ( ) ( ) 2 Sinc 2 Sinc 2
0 0
T T TA X
w
e + e + e e = e
Convolution of the Fourier transforms in the frequency domain

( ) ( ) t w t x
t + t
2 T 2 T +
The Fourier transform of the continuous harmonic signal
The Fourier transform of the rectangular time window as the continuous time signal
- 0,4

- 0,2

0

0,2

0,4

0,6

0,8

1

1,2

- 30

- 20

- 10

0 10

20

30

( ) 2 Sinc T e
2 T e
Sinc function
105
Rectangular (uniform) time window

e A
T
t
= e A
2
0
e
0
e
e A = e k
0
0
e
e A
,... 3 , 2 , 1 , 0 = k
Continuous frequency
e A = e k
0
Period T
| | s rad e
| | Hz f
0
0
Angular frequency
Frequency
Zeros
Discrete frequency
T
t
= e A
2
Discrete frequency
e A = e k
0
0
e
e A = e k
0
T
f
1
= A
Zeros
e A
Frequency
step (k = 1)
Frequency
step (k = 1)
Frequency axis
The rectangular window is suitable only for signals composed from harmonics of the 1/T frequency
?
( ) ( ) 2 Sinc T T W
T
e = e
relative error of up to
% 100
2
t
106
Overview of the basic time window types
0,0
0,5
1,0
1,5
2,0
2,5
0,00 7,81 15,63
Time [ms]
0,0
0,2
0,4
0,6
0,8
1344 1472 1600 1728 1856
Frequency [Hz]
R
M
S

-1
0
1
2
3
0,00 7,81 15,63
Time [ms]
0,0
0,2
0,4
0,6
0,8
1344 1472 1600 1728 1856
Frequency [Hz]
R
M
S
-2
0
2
4
6
0,00 7,81 15,63
Time [ms]
0,0
0,2
0,4
0,6
0,8
1344 1472 1600 1728 1856
Frequency [Hz]
R
M
S
( ) ( )
127 ..., , 1 , 0
, 128 * * 2 cos 1
=
t =
n
n n w
( ) ( )
( )
( )
( )
127 ..., , 1 , 0
, 128 * * 8 cos 0322 , 0
128 * * 6 cos 388 , 0
128 * * 4 cos 29 , 1
128 * * 2 cos 98 , 1 1
=
t +
+ t
t +
+ t =
n
n
n
n
n n w
Hanning window
Kaiser-Bessel window
Flat Top window
Flat Top
( ) ( )
( )
( )
127 ..., , 1 , 0
, 128 * * 6 cos 00305 , 0
128 * * 4 cos 244 , 1
128 * * 2 cos 29 , 1 1
=
t
t +
+ t =
n
n
n
n n w
Time domain Frequency domain
Calibration
procedure
Signals with
unknown
frequency
spectrum
The use of the
time windows
:
107
Hanning time window
Hanning : Sine1 - 50 Hz ; Sine1 - 50,5 Hz
-2,5
-2,0
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
2,0
2,5
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Sine1 - 50 Hz Sine1 - 50,5 Hz
Hanning : Sine1 - 50 Hz ; Sine1 - 50,5 Hz
0,0001
0,0010
0,0100
0,1000
1,0000
35 40 45 50 55 60 65
Frequency [Hz]
R
M
S

[
U
]
The Hanning window applied to the harmonic signal of the unity amplitude
108
Flat top time window
Flat Top : Sine1 - 50 Hz ; Sine1 - 50,5 Hz
-6
-4
-2
0
2
4
6
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Flat Top : Sine1 - 50 Hz ; Sine1 - 50,5 Hz
0,0001
0,0010
0,0100
0,1000
1,0000
35 40 45 50 55 60 65
Frequency [Hz]
R
M
S
Flat top
The Flat Top window applied to the harmonic signal of the unity amplitude
109
Time window bandwidth
Autospect - Hanning : Sine 50 Hz ; Sine 50,5 Hz
0,0
0,1
0,2
0,3
0,4
0,5
0,6
40 45 50 55 60
Frequency [Hz]
P
S
D

[
U
]
^
2
/
H
z
Autospect - Rect : Sine 50 Hz ; Sine 50,5 Hz
0,0
0,1
0,2
0,3
0,4
0,5
0,6
40 45 50 55 60
Frequency [Hz]
P
S
D

[
U
]
^
2
/
H
z
( ) ( ) PWR f k PSD BW f k PSD BW PWR
N k
k
N k
k

=
=
=
=
A = A =
1 2
0
1 2
0
Sine 50 Hz:
Sine 50.5 Hz:
Hanning
( )
=
=
A
1 2
0
N k
k
f k PSD BW PWR
0.5
0.5
0.500
0.499
1
1
Sine 50 Hz:
Sine 50.5 Hz:
( )
=
=
A
1 2
0
N k
k
f k PSD BW PWR
0.5
0.5
0.750
0.750
1.5
1.5
Rectangular
Rectangular
Hanning
Amplitude = 1 => PWR = 0.5
Time Window
Rectangular
Hanning .
Kaiser-Bessel
Flat Top ..
Bandwidth BW
1
1.5
1.8
3.77
110
Random signals
Flicker Noise 2 : Signal (Alpha 0)
-5
0
5
0,0 0,5 1,0
Time [s]
Flicker Noise : Signal (Alpha 1,99)
-20
0
20
40
60
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Flicker Noise : Signal (Alpha 1)
-6
-3
0
3
6
9
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
White noise
Pink noise
Flicker noise
(1 over f noise)
111
Averaging the PWR or PSD spectra
0,00
0,01
0,02
0 1000 2000 3000
( )
1
2
1
pwr rms
( )
2
2
2
pwr rms
( )
100
2
100
pwr rms
( ) PWR PWR RMS
( ) ( )
1 2 ,..., 2 , 1 , 0
1
1
=
=

=
=
N k
k pwr
M
k PWR
M m
m
m M
( )
( ) ( )
1 2 ,..., 2 , 1 , 0
1 1
1
=
+
=
=
N k
k pwr
M
k PWR
M
M
k PWR
m M
M
Recursive formula
Mean value
Frequency [Hz]
PWR spectra
Averaging results in
smoothing of spectrum
Mean value
Only the PWR or PSD spectra can be used
as an input data for averaging
The Welchs method is based on averaging the frequency spectra, which reduces the spectrum
variance.
This method, known as Welch's method, is used for estimating the power of a signal vs. frequency.
112
Overlapping records for averaging
No overlap
0 % overlap
66 % overlap
50 % overlap
A record for computing FFT
Gaps, these signal sections do not affect the result of analysis
pwr spectrum
pwr spectrum

0,0
1,0
2,0
3,0
4,0
0 32 64 96 128 160 192 224 256
n
w
n
0 1/2 2/3
113
Optimal overlap for the Hanning window
0 % overlap
2/3 or 66 % overlap
1/2 or 50 % overlap
Resulting effect of weighting and averaging
Weighting functions
Overlaps
The signal sections, where the weighting function is close
to zero, do not almost influence the result of the frequency
analysis
Overlapping the records of a
signal
From the viewpoint of the weighting function uniformity, the optimal value for overlap ratio:
n /(n+1) for n = 2, 3,
Effect of the number of averages on the random signal
frequency spectrum
0
2
5
0
5
0
0
7
5
0
1
0
0
0
1
2
5
0
1
5
0
0
1
7
5
0
2
0
0
0
2
2
5
0
2
5
0
0
2
7
5
0
3
0
0
0
1
2
5
10
20
50
100
0,00
0,01
0,02
0,03
0,04
0,04
0,05
0,06
0,07
0,08
0,09
0,10
0,11
0,11
0,12
0,13
0,14
Frequency [Hz]
Averages
-4
-2
0
2
4
0,0 0,5 1,0 1,5 2,0 2,5 3,0 3,5 4,0 4,5
Time [s]
Averages Time interval [s]
1 0.128
2 0.171
5 0.299
10 0.512
20 0.939
50 2.219
100 4.325
The length of the time record for
spectrum analysis
White noise signal
114
115
Estimation of the spectrum variance
The length of the record
for FFT . T
T
B
1
=
A
r
BT 2
1
= c
M
r
2
1
= c
MT T
A
=
Bandwidth ..
Total length of record
Autospectrum : White Noise
0.0
0.1
0.2
0.3
0.4
0.5
0 4 8 12 16 20 24
Frequency [Hz]
R
M
S

1
10
100
1000
Number
of averages
Relative spectrum
variance ..
Number of averages M
The effect of the number of averages
on the estimation error of the resulting spectrum
determines the formula
Multispectra - Waterfall plot
Frequency
Amplitude
RPM or Time
Gearbox noise
116
Multispectra - Contour plot
Frequency
RPM or Time
Gearbox noise
Color scale
Autospectrum : Gearbox Noise
Frequency
RMS dB(A)/
ref. 2E-5 [Pa]
117
118
Campbell diagram
If this was a 4 cylinder 4
stroke engine 2E would
equate to the firing frequency.
E is an integer multiple
of the engine speed.
1000 2000 3000 4000 5000 6000
0
200
400
1000
600
800
1E
2E
3E
4E
5E
6E
7E
8E
9E
10E
35 dB 60 dB
Engine speed
RPM
F
r
e
q
u
e
n
c
y

H
z

119
Constant Percentage Band Spectrum
Lecture 9
CPB : pave 25 km/h - Axis Z ISRI 2 : PAI25 : Seat pan ; Cab
floor
80
90
100
110
120
130
140
0
,
8
0
1
,
2
5
2
,
0
0
3
,
1
5
5
,
0
0
8
,
0
0
1
2
,
5
0
2
0
,
0
0
3
1
,
5
0
5
0
,
0
0
8
0
,
0
0
Frequency [Hz]
R
M
S

d
B
/
r
e
f

1
E
-
6

[
m
/
s
^
2
]
SEAT PAN [m/s^2]
CAB FLOOR [m/s^2]
120
Chromatic scale for piano
A4: 440 Hz = 2 x 220 Hz A5: 880 Hz = 2 x 440 Hz
1 2 3 4 5 6 7 8
1 2 3 4 5 6 7 8
Note that A5 has a frequency of 880 Hz. The A5
key is thus one octave higher than A4 since it has
twice the frequency.

A3: 220 Hz
121
1/n-octave bands of the PCB analyzers

dB

0

Frequency

band

Filter gain

H L C
f f f =
central frequency
( ) ( ) 1 2 = k f k f
C
n
C
n = 1 1/1-octave filter
n = 3 1/3-octave filter
n = 6, 12,
L H
f f f = A
bandwidth
C
f
f A
= o
relative bandwidth
The central frequency forms a geometric
sequence
k = 1, 2, 3,
Filter bank
log f f
L
f
C
f
H
Creating a bank of filters
1. A group of the IIR or FIR filters
2. FFT algorithm
N 512 1024 2048 4096 8192
1/1 6 7 8 9 10
1/3 18 21 24 27 30
Number of the 1/n bands for the N-point FFT
The FFT analyzers are based on the use of the FFT algorithm. Signal processing is organized
by repeating computations after aquiring a record of samples. The PCB analyzers are working
on the principle of a bank of the band pass filters.
The filter bandwidth
122
Central frequencies of the octave and 1/3-octave pass band
0.1 .125 0.16 0.2 0.25 .315 0.4 0.5 0.63 0.8
1 1.25 1.6 2 2.5 3.15 4 5 6.3 8
10 12.5 16 20 25 31.5 40 50 63 80
100 125 160 200 250 315 400 500 630 800
1k 1k25 1k6 2k 2k5 3k15 4k 5k 6k3 8k
10k 12k5 16k 20k
- - - - - - - .125 0.25 0.5
1 2 4 8 16 31.5 63 125 250 500
1k 2k 4k 8k 16k - - - - -
1/1-octave filters
1/3-octave filters
( ) ( ) 1 10
10 3
= k f k f
C C
10 1 3 1
10 2 ~ Approximation
( ) ( ) 1 10
10 1
= k f k f
C C
The central frequencies of the 1/3-octave pass band filters forms a geometric sequence with
a quotient , which is equal to the 1/3rd power of two (1.25992). To obtain pretty numbers
in the sequence the quotient is slightly changed to the value of the 1/10th power of ten (1.25892).
123
CPB analyzer
Real Time Analyzer

Root

Low pass

Filter

Squarer

Band pass

filter

x

x

x

2

x

2

f c 1

Input
signal
f c k

CPB
spectrum

Square

Root
...
...
(RMS)
RMS
Mean Square
CPB spectrum example - Ride comfort
CPB : pave 25 km/h - osa Z : PA025 : Seat ; PAI25 : Seat ; PAM25 : Seat
80
90
100
110
120
130
140
0
,
8
0
1
,
0
0
1
,
2
5
1
,
6
0
2
,
0
0
2
,
5
0
3
,
1
5
4
,
0
0
5
,
0
0
6
,
3
0
8
,
0
0
1
0
,
0
0
1
2
,
5
0
1
6
,
0
0
2
0
,
0
0
2
5
,
0
0
3
1
,
5
0
4
0
,
0
0
5
0
,
0
0
6
3
,
0
0
8
0
,
0
0
Frequency [Hz]
R
M
S

d
B
/
r
e
f

1
E
-
6

[
m
/
s
^
2
]
124
Spectrum examples
CPB : SF128 : Vibrace
0,000
0,010
0,020
0,030
0,040
0,050
125 250 500 1k 2k 4k 8k
Frequency [Hz]
R
M
S

Autospectrum : SF128 : Vibrace
0,000
0,005
0,010
0,015
0,020
0,025
0,030
0 2000 4000 6000 8000 10000
Frequency [Hz]
R
M
S
CPB : SF128 : Vibrace
0,000
0,010
0,020
0,030
0,040
1
0
0
1
2
5
1
6
0
2
0
0
2
5
0
3
1
5
4
0
0
5
0
0
6
3
0
8
0
0
1
k
1
k
2
1
k
6
2
k
2
k
5
3
k
1
4
k
5
k
6
k
3
8
k
1
0
k
Frequency [Hz]
R
M
S
Autospectrum 1 : SF128 : Resampling (Vibrace)
0,000
0,005
0,010
0,015
0,020
0,025
0,030
0 50 100 150 200 250 300 350 400
Order [-]
R
M
S

Time History : SF128 : Vibrace
-0,3
0,0
0,3
0 1 2 3 4 5 6 7 8 9
Time [s]
1/1-octave
1/3-octave
FFT - Hz
FFT - order
Logarithmic scale Linear scale
125
126
Frequency Weighting
See [http://hyperphysics.phy-astr.gsu.edu/hbase/sound/soucon.html#soucon]
127
Annotated equal loudness curves
Equal loudness in phones
The curves represent
equal loudness as
perceived by the
average human ear
The ear is less sensitive to
low frequencies, and this
discrimination against
lows becomes steeper for
softer sounds.
The maximum sensitivity
region for human hearing is
around 3-4 kHz and is
associated with the resonance
of the auditory canal.
Sound intensity in decibels
does not directly reflect the
changes in the ears
sensitivity with frequency
and with sound level
Curve for the average
threshold of hearing
100
10
20
30
40
50
60
70
80
90
100
110
1000 10000
120
100
80
60
40
Frequency [Hz]
I
n
t
e
n
s
i
t
y

i
n

d
e
c
i
b
e
l
s
]

See [http://hyperphysics.phy-astr.gsu.edu/hbase/sound/eqloud.html#c1]
120
|
|
.
|
\
|
=
REF
RMS
p
p
dB log 20
| | Pa 00002 . 0 =
REF
p
128
Filter contours
The A-contour filters out significantly more bass than the others, and is designed to approximate
the ear at around the 40 phon level. It is very useful for eliminating inaudible low frequencies.
The intermediate B-contour approximates the ear for medium loud sounds. It is rarely use.
The C-contour does not filter out as much of the lows and highs as the other contours. It
approximates the ear at very high sound levels and has been used for traffic noise surveys in
noisy areas.
( )
( )
( ) ( ) ( )
4
2
1 3
2
1 2
2
1 1
2
1
2
2
1
a f a f a f a f
f K
f H
A
+ + + +
=
See [http://jenshee.dk/]
006315772 . 0
, 5560957 . 0 , 06316174 . 0
, 8404 . 148 , 3742 . 187
4
3 2
1
=
= =
= =
a
a a
a K
1000
1
f f =
( ) ( ) ( ) f S f H f S
A A
=
S(f) is the unweighted RMS spectrum
S
A
(f) is the A-weighted RMS spectrum
129
Order Analysis
Lecture 11
Encoders
Pinion
2
O
1
O
r
2
Wheel
130
Rotational speed measurements
Complete revolution
Tacho impulses
360
0
1 impulse per revolution
1
st
revolution 2
nd
revolution 3
rd
revolution
Time interval for vibration
and noise analysis
Tacho probe
output
Contact-free detection of
reflective objects (rotary or
reciprocatory machine parts)
Photoelectric Tachometer
Probe
500 impulses per
revolution
1024 impulses per
revolution
ERN 460-500 type ERN 460-1024 type
Heidenhain products
131
Spectral analysis of non-uniformly rotational machines
780
790
800
810
820
0 50 100 150 200
Index
RPM
RPM variation at idle speed
Car engines Gearboxes
Many machines operates in the cyclic fashion, for example an IC engine with the constant number
of strokes per revolution or a gear train with the constant number of mesh cycles per revolution.
RPM slow variation
results in uncertain number of strokes per second
uncertain number of mesh cycles per second
Non-uniform rotation
results in smearing
spectrum components as
a result of the frequency
modulation effects.
Sampling signals at the constant angle rotation increments can prevent spectrum smearing.
132
Time vs. angular domain
ADC
Clock
Time domain sampling
ADC
Encoder
Angular domain sampling
Analog
signal
Digital
signal
Analog
signal
Digital
signal
Impulses are generated at
the constant time frequency
Impulses are generated after the constant
rotation angle of the shaft rotation
Analog to digital conversions (ADC) are started by an edge of the synchronization signal
Time
Time interval T
S
Angle
Rotation by the 360/N angle in degrees

Samples
Source of the
sampling frequency
(The encoder produces N impulses per a complete revolution)
It is impossible to detect angular vibration during rotations.
133
Digital order tracking computed order tracking
Sampling at the constant
frequency in Hz
Resampling according
to rotational frequency
Time or frequency
analysis
Upsampling by 2 or 4
Low pass filtration
Interpolation
0 360
0
360
0
0
Nominal rotation angle
Actual rotation angle
Constant
angular
velocity
Angular
vibration
Equidistantly spaced samples
Algorithm of resampling
Resampling to the unified number of samples is limited either by one complete revolution or by
arbitrary complete revolutions.
Externaly controled sampling frequency by the IRC sensor
Frequency multiplier (BK 5050) for multiplying the tacho signal frequency
Digital order analysis based on resampling of the signal
Techniques for order analysis
Resampling signals for order analysis
Time History : Sine
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Continuous time signal
4.333 Hz sampling frequency
4 times upsampled (17.333 Hz)
3-step resampling:
Upsampling by the integer factor (2 or 4 times)
Low pass filtration to reduce the frequency range
to the value before upsampling (2 or 4 times)
Interpolation in between adjacent samples
(resampling by a fractional factor)
Interpolation
Original samples
New samples
The use of the Lagrange or Newton
interpolation polynomial
Methods:
An example demonstrating algorithm of resampling
134
Upsampling by the integer factor - Interpolation
Time History :Sine1
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,00 0,25 0,50 0,75 1,00
Time [s]
Time History : upsample(x,4)
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,00 0,25 0,50 0,75 1,00
Time [s]
Autospectrum : Sine1 - 1
0,0
0,2
0,4
0,6
0,8
0 1 2 3 4
Frequency [Hz]
R
M
S
Autospectrum of upsample(x,4)
0,0
0,1
0,2
0,3
0,4
0 4 8 12 16
Frequency [Hz]
R
M
S
Interpolated signal
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,00 0,25 0,50 0,75 1,00
Time [s]
Adding 3 zeros in
between samples
Upsampling by 4 Low pass filtration,
4 Hz cut-off frequency
Autospectrum : Interpolated signal
-360
-300
-240
-180
-120
-60
0
0 2 4 6 8 10 12
Frequency [Hz]
R
M
S

d
B
/
r
e
f

1
Interpolation error Frequency range
4 Hz multiplied by 4 4 Hz
135
136
Effect of downsampling by an integer factor on spectrum
Let the frequency sampling of a signal be equal to f
S
f
S
/ 2

0

f

Frequency spectrum before
downsampling

f
S
/ 2

0

f

F
S
/ 2

(M = 2)

Frequency spectrum after decimation

0

f

F
S
/ 2

Downsampling reduce the sample count and the maximum signal
frequency and safe space between the spectrum components

Frequency spectrum after filtration

FIR low pass filter can be employed
to evaluate only output samples

Now it is possible to reject M-1 samples after each M
th
one in the initial signal (samples decimation)

Let the sampling frequency f
S
be reduced to F
S
= f
S
/ M. To prevent aliasing after downsampling the
signal frequency range has to be reduced by the antiasiasing low pass filter to the frequency F
S
/2

137
Order spectra
Spectral
analysis
Order
analysis
Time
domain
Time
[s]
Revolution
[-]
Frequency
domain
Frequency
[Hz]
Order
[-]
Frequency [Hz] 0
T
1
T
2
T
3
T
4
T
5
Order [-] 0 1 2 3 4 5
Reciprocal value of the time
interval length for FFT
Multiples of the tacho impulse frequency,
1 impulse per revolution
Order [-] 0
Multiples of the tacho impulse frequency,
1 impulse per n revolutions
n
2
1
n
1
n
3
n
n 1 +
Fractional orders
DC
DC
DC
The frequency in orders is the same
as the multiples or harmonics of the
machine basic frequency.
The basic frequency is the rotation
frequency of some machine part
(gearbox input shaft, countershaft,
or output shaft).
Revolution and order are
dimensionless quantities.
DC = Direct Current
138
Resampling signal according to the tacho frequency
Time History : Sine 150,5 Hz
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Signal
Tacho
Autospectrum : Sine 150,5 Hz
0,0
0,1
0,2
0,3
0,4
0,5
120 140 160 180
Frequency [Hz]
R
M
S
Resampled Signal : Sine 150,5 Hz
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
Nominal revolution [-]
Resampled
Tacho
Autospectrum : Sine 150,5 Hz
0,0
0,2
0,4
0,6
0,8
80 100 120
Order [-]
R
M
S
The tacho signal frequency is 1.505 Hz
Analyzed signal frequency is 150.5 Hz = 100 x 1.505 Hz, amplitude 1 (RMS 0.707)
139
A simple example to resampling a non-stationary signal
Input signal
20 samples 15 samples 10 samples
Resampled signal
10 samples 10 samples 10 samples
Resampling to the unified number of sample per revolution during run-up and coast down enables
to perform order analysis of signals or to prepare a strip plot in the form revolution-by-revolution
records containing the same number of samples.
Constant sampling
frequency
Sampling frequency
is proportional to
rotational frequency
140
Frequency and order multispectra
0 RPM
Noise caused by resonances,
the frequency of excited components
(paralel lines) is independent on RPM
Excited noise, the frequency of excited components
(radial lines) is proportional to RPM
Gearbox noise
during run-up
Excited noise at frequency,
which is proportional to RPM
Resonance noise,
constant frequency in Hz
Frequency in Hz Frequency in order
141
Order spectra in machine diagnostics
COUPLE UNBALANCE
ANGULAR MISALIGNMENT
PARALLEL MISALIGNMENT
ROLLING BEARINGS
GEARBOXES
0 1 2
Acc
ord
0 1 3
Acc
ord
2
3
4
0 1 3
Acc
ord 2 4
0 1 2
Acc
ord 3
CAR
ENGINE
BPFO BPFI
BPF - Ball Pass
Frequency
I inner race
O outer race
4
0
Acc
ord
GMF - Gear Mesh Frequency
Number
of Teeth
GMF
1
0 2 6
Acc
ord 4 8
4 strokes
4 cylinders
Rotational frequency 1 ord, acceleration signals
Synchronized averaging as a tool for filtration signals
-0,3
0,0
0,3
0,00 0,20 0,40 0,60 0,80 1,00
Nominal Revolution [-]
[
m
/
s
2
]
-0,3
0,0
0,3
0,00 0,20 0,40 0,60 0,80 1,00
[
m
/
s
2
]
-0,3
0,0
0,3
0,00 0,20 0,40 0,60 0,80 1,00
[
m
/
s
2
]
-0,1
0,0
0,1
0,00 0,20 0,40 0,60 0,80 1,00
[
m
/
s
2
]
Time History : SF128 : Acceleration
-0,3
0,0
0,3
0 2 4 6 8
Time [s]
[
m
/
s
2
]
Mean value
0,00
0,01
0,02
0,03
0 27 54 81 108 135 162 189
Order [-]
R
M
S

[
m
/
s
2
]
Tachometer : SF128 : Pulse positions
1490
1492
1494
1496
0 2 4 6 8 10
Time [s]
R
P
M
The same number
of samples
Resampled records
142
143
Synchronized averaging as a comb filter
( ) ( )
=
=
1
0
1
M
m
mT t x
M
t y
( ) ( )
=
o =
1
0
1
M
m
mT t
M
t g
( ) ( ) ( ) ( )
( )
( ) sT
sMT
M
msT
M
dt st t g s G
M
m

= = =

}

=
+
exp 1
exp 1 1
exp
1
exp
1
0
0
( )
( )
|
.
|
\
|
e
|
.
|
\
|
e
|
.
|
\
|

e = e
2
sin
2
sin
2
1
exp
1
T
MT
T M
j
M
j G
( )
|
.
|
\
|
e
|
.
|
\
|
e
= e
2
sin
2
sin
1
T
MT
M
j G
Mathematical model of averaging
(moving average)
Impulse response (x(t) = (t)) .
Laplace transfer function .
Frequency transfer function .
Magnitude of the frequency
transfer function ...
144
Frequency response function of the synchronized
averaging
|
|
.
|
\
|
t
|
|
.
|
\
|
t
=
|
|
.
|
\
|
0
0
0
sin
sin
1
f
f
f
f
M
M f
f
j G
0 sin
0
=
|
|
.
|
\
|
t
f
f
M
Z
,... 3 , 2 , 1 ,
0 0
=
|
|
.
|
\
|
=
|
|
.
|
\
|
+ k
f
f
G k
f
f
G
0,0
0,2
0,4
0,6
0,8
1,0
0 1 2 3
f / f
o
|G( f / f
o
)|
Periodicity
Zeros
Magnitude of the comb filter frequency response
T
f
1
0
=
Dimensionless frequency
Synchronized averaging acts as a comb filter for spectrum components, which are an integer
multiple of the synchronized frequency. The component corresponding to the non-integer
(fractional) multiple are considerably attenuated.
t = t k
f
f
M
Z
0
,... 2 , 1 , 0 ,
0
= = k
M
k
f
f
Z
There is M zeros of the
frequency response in
between two adjacent integer
multiple of the dimensionless
frequency f / f
0
145
A comparison of various approaches
to spectral analysis
Examples
146
Truck gearbox vibration spectra
Autospectrum : SF128 : Vibrace
0,014 0,014
0,028
0,000
0,005
0,010
0,015
0,020
0,025
0,030
0 1000 2000 3000
Frequency [Hz]
R
M
S

[
m
/
s
2
]
Autospectrum : SF128 : Resampling (Vibrace)
0,013
0,013
0,027
0,000
0,005
0,010
0,015
0,020
0,025
0,030
0 27 54 81
Order [-]
R
M
S

[
m
/
s
2
]
46 teeth
759 rpm
27 teeth
1293 RPM
GMF = 582 Hz
Tachometer
Notice differences in sidebands
of the GMF components
GMF = 27 ord
Frequency and order spectrum of a gearbox vibration
A simple gear train
Frequency analysis of the acceleration signal
Autospectrum 2 : FILE017 : Acc
0,000
0,005
0,010
0,015
0,020
0,025
0 2500 5000 7500 10000 12500
Frequency [Hz]
R
M
S

m
/
s
2
CPB : FILE017 : Acc
0,00
0,01
0,02
0,03
0,04
0,05
1
0
0
1
2
5
1
6
0
2
0
0
2
5
0
3
1
5
4
0
0
5
0
0
6
3
0
8
0
0
1
k
1
k
2
1
k
6
2
k
2
k
5
3
k
1
4
k
5
k
6
k
3
8
k
1
0
k
Frequency [Hz]
R
M
S

m
/
s
2
Frequency spectra
FFT
CPB
electric motor
gearbox
accelerometer
belt
spindle
25T
tacho probe
Input
shaft
Intermediate
shaft
An example showing FFT and CPB signal
processing methods employed to analyze
rattling noise of a gear train
147
148
149
Harmonic Signal Modulation
Lecture 12
Time History : Generator : Sine1/AM - 1
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
[
U
]
150
Modulation of harmonic signals
Carrying component ..
(harmonic signal without modulation)
Amplitude A
Phase
Initial phase
Amplitude modulation signal ...
Phase modulation signal ....
Mixed modulation (amplitude and phase)
Phase
Modulation signals
Amplitude
Modulated signal
Carrying component phase
( ) ( ) ( ) ( ) ( )
PM PM PM AM AM AM
t n t t A t x + e | + + e + e | + = cos cos cos 1
0 0
( ) ( )
0 0 0
cos + e = t A t x
( ) ( )
AM AM AM A
t t x + e | = cos
( ) ( )
PM PM PM P
t t x + e | = cos
( )
0 0
+ e = t t
0
Nomenclature
See [Tma, 1997]
151
Amplitude modulation of harmonic signals
Time History : Generator : Sine1/AM - 1
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Modulation signal
( ) ( ) ( ) ( ) t t A t x
AM AM AM 0
cos cos 1 e + e | + =
Modulated signal Carrying component
AM
|
AM AM
f t = e 2
0 0
2 f t = e
Autospectrum : Generator : Sine1/AM - 1
0,0
0,2
0,4
0,6
0,8
0 100 200 300 400
Frequency [Hz]
R
M
S

[
U
]
modulation frequency
modulation index
carrying frequency
Sideband components f
0
+f
AM
, f
0
-f
AM
Carrying component
152
Spectrum of the amplitude-modulated signal
( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( )
AM AM AM AM AM
AM AM AM P
AM AM AM
t t A t A
t t A t A
t t A t x
+ e + e + e e | + e =
= e + e | + e =
= e + e | + =
0 0 0
0
0
cos cos 2 cos
cos cos cos
cos cos 1
( ) ( ) ( ) ( ) ( ) ( )
AM AM AM AM AM AM
t A t A t A t x + e + e | + e e | + e =
0 0 0
cos 2 cos 2 sin
Upper sideband component,
frequency f
0
+ f
AM
, amplitude A
AM
/2

Decomposition of the modulated signal on the carrying component and its sidebands
Lower sideband component,
frequency f
0
- f
AM
, amplitude A
AM
/2

The carrying component,
frequency f
0
, amplitude A

Phasor model
+f
0

-f
0

A
AM
/4
Re
Im
-f
+f
A

/2
f
0
+ f
AM

f
0
- f
AM

Rotational
frequency
153
Phase modulation of harmonic signals
Time History : Generator : Sine1/PM
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
( ) ( ) ( )
PM PM PM
t t A t x + e | + e = cos cos
0
PM PM
f t = e 2
0 0
2 f t = e
Autospectrum : Generator : Sine1/PM
0,0
0,1
0,2
0,3
0 100 200 300 400
Frequency [Hz]
R
M
S
modulation index
carrying frequency
Modulation signal Modulated signal Carrying component
AM
|
The family of the sideband components f
0
+f
PM
, , f
0
f
PM
,
Carrying component
154
Spectrum of the phase-modulated signal
( ) ( ) ( ) t p t p t x
+
+ =
( ) ( ) ( ) ( ) ( ) u | e = u | + e =
+
cos exp exp
2
1
cos exp
2
1
0 0 PM PM
j t j t j t p
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( )|
.
|
\
|
u e + u + e | + e | =

+
=
+
i t j i t j j J t j J t p
i
i
PM i PM 0 0
1
0 0
exp exp exp
2
1
( )
( )
( )
+
=
+
+
|
.
|
\
|
|
= |
0
2
! !
1
2
k
k
i k
PM
PM i
i k k
J
( ) ( ) ( )
PM i
i
PM i
J J | = |
1
( ) ( ) ( ) t s t t x
PM
+ e =
0
cos
( ) ( ), cos u | =
PM PM
t s
PM PM
t + e = u
Let be a phase-modulated signal, where
where J
i
() is the Bessel function of the first kind, for integer orders i = 0, 1, 2,

+f
0

-f
0

Re
Im
-f
+f
f
0
+ f
PM

f
0
- f
PM

Rotational
frequency
Phasor model
-0,5
0
0,5
1
0 2 4 6 8 10
PM

J
0
(
PM
)
J
1
(
PM
)
J
2
(.)
J
3
(.)
J
4
(.)
J
5
(.)

(
PM
) J
i

( ) 2
0 PM
J |
( ) 2
1 PM
J |
Autospectrum : Sine1/PM - Beta = 5
0,0
0,1
0,2
0,3
0 50 100 150 200
Frequency [Hz]
R
M
S
Effect of the phase modulation index on the sidebands
Autospectrum : Sine1/PM - Beta = 0.1
0,0
0,2
0,4
0,6
0,8
0 50 100 150 200
Frequency [Hz]
R
M
S

U
PM
= 5
The carrying component 100 Hz, amplitude 1, modulation signal frequency 5 Hz
Autospectrum : Sine1/PM - Beta = 1
0,0
0,2
0,4
0,6
0 50 100 150 200
Frequency [Hz]
R
M
S
Autospectrum : Sine1/PM - Beta = 0.5
0,0
0,2
0,4
0,6
0,8
0 50 100 150 200
Frequency [Hz]
R
M
S

U
PM
= 1
PM
= 0.5
PM
= 0.1
155
156
Amplitude and phase modulation I
Time History : Generator : Sine1/PM+AM
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Modulation signals
( ) ( ) ( ) ( ) ( )
M M PM M M AM
t t t A t x + e | + e + e | + = sin sin sin 1
0
Modulated signal
PM AM
| | ,
M M
f t = e 2
0 0
2 f t = e
Autospectrum : Generator : Sine1/PM+AM
0,0
0,1
0,2
0,3
0 100 200 300 400
Frequency [Hz]
R
M
S
modulation frequencies
modulation
indexes
carrying frequency
Carrying component
Symmetric sidebands
Modulation signals are in phase
157
Amplitude and phase modulation II
Time History : Generator : Sine1/PM+AM
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Modulation signals
( ) ( ) ( ) ( ) ( ) 2 sin sin sin 1
0
t + + e | + e + e | + =
M M PM M M PM
t t t A t x
Modulated signal
PM AM
| | ,
0 0
2 f t = e
0 0
2 f t = e
Autospectrum : Generator : Sine1/PM+AM
0,0
0,2
0,4
0,6
0 100 200 300 400
Frequency [Hz]
R
M
S
modulation
indexes
carrying frequency
Carrying component
Modulation signals are out of phase
Non-symmetric sidebands
158
159
Analytic Signals and Hilbert Transform
Lecture 13
160
Analytic signals
f
P
t = e 2
Real harmonic signal Complex analytic signal
The analytic signal is a complex signal with an imaginary part, which is the Hilbert transform
of the signal real part. The decomposition of a real signal into harmonic components results
in the sum of harmonic functions. Each of this function can be decomposed into the pair
of the phasors, which are rotating in the opposite direction. The analytic signal creates
the phasors rotating in the positive direction.
The analytic signal is a tool for amplitude and phase demodulation of the modulated
harmonic signals.
To obtain the analytical signal the phasor X
N
has to be removed and the phasor X
P
has to be
multiplied by 2.
161
Analytic signals in the 3D-space
f
P
t = e 2
Helix
The position vector is rotating in the complex plane. If the 2D space is extended to 3D space
with the third axis as a time axis then the vector end point moves on the helix trajectory.
162
Analytic signal
Analytic signals and the Hilbert transform
Hilbert transform
P
X Z 2 =
P P
X j Y =
N N
X j Y =
N N N
X X j j Y j = =
N P
X X X + =
P
X
N
X
2
t
2
t
2
t
= + j Time signal
Fast Fourier Transform (FFT)
Digital filters
Evaluation of the Hilbert transform
using
( )
( )
P
N P N P
N P N P
X
X X X X
Y Y j X X Z
2 =
= + + =
= + + + =
N P
X X X + =
( )
N P
N P
N P
X X j
X j X j
Y Y Y
=
= + =
= + =
P P P
X X j j Y j = =
The complex position vectors as phasors, which are corresponding to a harmonic signal.
The phasors Y associated to the Hilbert transform of a pair of phasors X are obtained by rotation
these phasors by the angle of +/- /2 radians.
The complex plane
163
-0,8
-0,6
-0,4
-0,2
0
0,2
0,4
0,6
0,8
-15 -10 -5 0 5 10 15
Definition of the Hilbert transform as the Cauchy
principal value integral
The Hilbert Transform can be defined as the principal value integral
The Cauchy principal value (P.V.) expands the class of certain improper integrals for which the
finite integral exists as for example the integral
( ) ( )
(
(
+
} }
c +
c
+ c
b
a
x x f x x f d d lim
0
where
( ) ( ) ( ) =
}
=
}
e
b
a
x x f x x f b a d , d , ,
( )
( )
}
+

t
t
t
t
= d . .
1
t
x
V P t y
Let x(t) be an impulse Dirac function (t), then the Hilbert transformer impulse response is as
follows
( )
( )
t t
V P t g
t
= t
t
t o
t
=
}
+

1
d . .
1
( ) t g
t
164
The Hilbert transform as a transfer function
To turn the non-decaying function to the decaying function, let the frequency transfer function
be extended
( )
( )
( )

< e
> e
=
e
e
= e
0 ,
0 ,
j
j
j X
j Y
j H
HT
( ) ( )
( )
( ) ( )
( )
2 2
0 0
0
0
1 1
2
d
2
d d
2
1
d
2
1
t
t
e
jt
e
jt
j
e e
j
e j e j e j H t g
jt jt t j t j
t j t j t j
+ o t
=
(
o
+
+ o
t
= e
t
=
=
(
e e
t
= e e
t
=
e o e + o
+
e + oe e oe
+
e + oe

e + oe
+

e
}
} } }
The impulse response is an inverse Fourier transform of the frequency transfer function
See [http://w3.msi.vxu.se/exarb/mj_ex.pdf]
( ) ( ) ( ) e = e
< e
> e
= e
o
oe
oe
j H j H
je
je
j H
HT
0
lim
0 ,
0 ,
( ) ( )
( ) t t
t
t g t g
HT
t
=
+ o t
= =
o o
1
lim lim
2 2
0 0
The Hilbert transform of a time function to another time function can be described by the standard
transfer function in the frequency domain. Let X(j) and Y(j) be the Fourier transform of the
original continuous time signal x(t) into the same signal y(t), respectively.
165
Analytic signals and the Hilbert transform of some signals
Real part Imaginary part Envelope Phase
( ) t x ( ) t y ( ) t E ( ) t |
( ) t A e sin ( ) t A e cos
( ) t A e cos ( ) t A e sin t e
2 t et
A
A
Signal
Hilbert
Transform
( ) t x ( ) t y
( ) t e sin ( ) t e cos
( ) t e cos ( ) t e sin
( ) 1 1
2
+ t ( ) 1
2
+ t t
( ) t t sin ( ) ( ) t t cos 1
( ) t o t t 1
The envelope and phase of the harmonic signals Hiltert transform
166
The use of FFT for computing the Hilbert transform

2
t
2
t
( ) ( ) { } e j Y IFFT k y = ( ) ( ) e e j Y j X ( ) ( ) { } k x FFT j X = e
N N
X j Y =
P P
X j Y =
( ) ( ) ( )
( ) ( )
P P
P P P
X j X
X j X j Y
Re Im
Im Re
=
= + = ( ) ( ) ( )
( ) ( )
N N
N N N
X j X
X j X j Y
Re Im
Im Re
+ =
= + =
i. The Fast Fourier Transform (FFT) of the real input time record to obtain phasors rotating in
positive and negative directions
ii. Rotation the phasor X
N
in the positive direction by the angle of + /2 radians and the phasor X
P

in the negative direction by the angle of - /2 radians (exchanging the real and imaginary parts)
iii. The Inverse Fast Fourier Transform (FFT) of the rotated phasors Y
P
and Y
N
to obtain the Hilbert
transform of the input record.

The algorithm for computing the Hilbert transform is broken down into three steps

Diagram showing how to transform the phasors X
P
and X
N
to the phasors Y
P
and Y
N

i) ii) iii)
Exchanging the real and imag parts Exchanging the real and imag parts
167
The use of digital filters for computing the Hilbert
transform
Frequency response function
( )
( )
( )

< e < t
> e > t +
= =
e
e
e
0 ,
0 ,
S
S
T j
T j
T j
HT
T j
T j
e X
e Y
e G
S
S
S
( ) ( ) ( )
+ = t
=
=
= e
t
=
}
t +
t
e e
1 2 , 2
2 , 0
d
2
1
k n n
k n
T e e G n g
S
n T j T j
HT HT
S S
Impulse response
Let X(e
jTS
) and Y(e
jTS
) be the Fourier transform of the original sample sequence x
t
into the
sample sequence y
t
, respectively.
-1,0
-0,5
0,0
0,5
1,0
-50 -40 -30 -20 -10 0 10 20 30 40 50
Index n
g
H
T
+
Impulse Response of the Ideal Hilbert transformer
The nonzero response for the negative index n means that the impulse response corresponds to
a non-causal system. Response precedes the change at the system input.
168
Hilbert transformer filters
-0,8
-0,6
-0,4
-0,2
0,0
0,2
0,4
0,6
0,8
-20 -16 -12 -8 -4 0 4 8 12 16 20
Index n
g
H
T
Kaiser Ideal
Frequency response function of the Hilbert
transformer
0,0
0,2
0,4
0,6
0,8
1,0
1,2
0,0 0,2 0,4 0,6 0,8 1,0
Normalised Frequency [-]
M
a
g
n
i
t
u
d
e
Kaiser Ideal
The 160-order FIR filter with the finite
impulse response n = -80,,+80
Hilbert Transformer
The impulse response of FIR filters is the same as these filters non-zero coefficients. If the infinite
impulse response is shorten to a finite number of non-zero samples then the this response will
corresponds to a FIR filter. Due to the linearity of the filter phase the symmetric or anti-symmetric
coefficients are preferred. As in the case of FIR filter the impulse response has to be delayed in
such a way that the impulse response of the non-causal system is changed to the response of the
causal system. The filter is called as a Hilbert transformer or a 90-degree phase shifter.

The digital filter acts as a Hilbert transformer only for a frequency band in which the magnitude of
the frequency response function is equal to unit. The impulse response which is corrected with the
use of the Kaiser window smooths the frequency response function.
Windowing Windowing
169
Amplitude and phase
demodulation

P
w
w
170
Analytic signals and amplitude modulation
( ) ( ) ( ) ( ) t t m A t x
M 0
cos cos 1 e e + =
Sideband
components
Carrying
component
Analytic signal
Let x(t) be a real amplitude modulated harmonic
signal described by envelope-and-phase form
Modulation signal
-2
-1
0
1
2
0 0,1 0,2 0,3 0,4 0,5 0,6 0,7 0,8 0,9 1
Time [s]
0
+
M
0
-
M
0

Analytic signals and phase modulation
Let x(t) be a real phase modulated harmonic
signal described by envelope-and-phase form
0
e
-2
-1
0
1
2
0 0,1 0,2 0,3 0,4 0,5 0,6 0,7 0,8 0,9 1
Time [s]
Phase
Modulation signal
Sideband
components
Carrying
component
Analytic signal
( ) ( )
M
t A t x A + e =
0
cos
M
t A + e
0
171
172
Demodulation of the modulated harmonic signal
Firstly the carrier component and its adjacent sidebands have to be filtered using the band pass
filter. The output signal is designated by x(t)
Secondly the Hilbert transform y(t) of the x(t) signal has to be evaluated using either the FFT
transform or the Hilbert transformer to create the analytic signal
The amplitude modulation signal, referred to as envelope, is as follows
( ) ( ) ( ) t y j t x t z + =
( ) ( ) ( ) ( )
2 2
t y t x t z t A + = =
The principal value of the phase modulation signal is as follows
( ) ( ) ( ) ( ) ( ) ( ) t x t y t z t arctan Arg
P.V.
= =
The phase in radians can be computed by the previous formula while taking into the count the value
sign of x(t) and y(t). The result will be in the wrapped form which is limiting the angle to the
interval
( ) t + s < t t
P.V.
To finish the phase demodulation process the wrapped phase has to be unwrapped into
( ) ( ) ( ) ( ) ( ) t n t z t z t + = 2 Arg arg
where n(t) is a sequence of integer numbers, which depends on time t, for that arg(t) is without
discontinuities larger than a permissible value.
173
The ShannonNyquist theorem for sampling of the phase
S
n n
f
f
t f t
t
= A t = A e = = A

2
2
1
t
t t
= s s s
2
2 2
2
1
2
S S
S
f
f
f
f
f f
t s
t
= A
S
f
f 2
t s A
Let the phase difference during the sampling interval be written in the form
The Shannon Nyquist theorem requires
It can be concluded that the phase change during the sampling interval has to be less then radians
It is assumed sampling a continuous harmonic signal
( ) ( ) ( ) t f t t x t = e = 2 cos cos
The phase of the mentioned harmonic signal is as follows
( ) t f t t t = e = 2
This phase change property is basic for unwraping phase signal.
174
( ) t s A s
sampl
f f 2
Unwrapping phase and removing the linear trend
Removing discontinuities
t
t +
t 2
t t + > A t + t < A 2 , 2
-0,15
-0,1
-0,05
0
0,05
0,1
0,15
0 0,2 0,4 0,6 0,8 1
Nominal Revolution
rad
-4
0
4
0 0,1 0,2 0,3 0,4 0,5 0,6 0,7 0,8 0,9 1
Nominal Revolution
rad
0
2
4
6
8
0 0,2 0,4 0,6 0,8 1
Nominal Revolution
rad
Algorithm of the phase unwrapping is based on the phase sampling theorem
The phase demodulation results in in the following signal, which is of the sawtooth wave form.
Unwrapping algorithm
175
An alternative procedure for computing instantaneous
frequency
( )
( )
( )
( )
( )
( ) ( )
( )
( ) ( ) t y t x
dt
t dy
t x t y
dt
t dx
dt
t x
t y
d
dt
t d
t
2 2
arctan
+
=
|
|
.
|
\
|
|
|
.
|
\
|
=
= e
( )
( )
( )
|
|
.
|
\
|
=
t x
t y
t arctan
( ) ( ) ( ) t y t x t e
2 2
+ =
Phase ....
Angular frequency
Envelope ..
( ) ( ) t
}
t e = d t
t
0
Phase
It is not always necessary to calculate the unwrapped phase. To calculate the instantaneous
frequency of the modulated harmonic signal it is possible to use the following formulas
Generator : SweptSine - 1
-2
-1
0
1
2
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Frequency : SweptSine - 1
0
10
20
30
40
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Time : Generator 1 : SweptSine - 1
0
10
20
30
40
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
Let the frequency of the swept sine signal be running-up from 10 to 30 Hz
Rectangular window Hanning window
176
Envelope analysis
Time History : Sine / AM
-2
-1
0
1
2
0,0 0,5 1,0
Time [s]
Real
Envelope
Decaying vibration
-10
-5
0
5
10
0,0 0,5
Time [s]
Real
Envelope
Decaying vibration
-10
-5
0
5
10
0,0 0,5
Time [s]
Real
Envelope
Decaying vibration
0,0
0,2
0,4
0,6
0,8
1,0
1,2
0 20 40
Frequency [Hz]
R
M
S

Original
Filtered
The amplitude demodulation is referred to as envelope analysis. The examples shows
computing the envelope for a broadband signal or signal zoomed around a resonance frequency
with the use of the bandpass filter.
The envelope is computed for narrow
band part of frequency spectrum
The envelope is computed for the full
frequency spectrum
177
Phase demodulation
Autospectrum : Sine 20 Hz /PM 2 Hz
0,
0
0,
1
0,
2
0,
3
0 10 20 30 40 50
Frequency
[Hz]
R
M
S

Time History : Sine 20 Hz / PM 2 Hz
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,0
0,2 0,4 0,6 0,8 1,0
Time [s]
Time : Phase : Sine 20 Hz / PM 2 Hz
-4
-2
0
2
4
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
[
r
a
d
]

Time : Unwrapped Phase : Sine 20Hz / PM 2Hz
0
50
100
150
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
[
r
a
d
]

Time : Detrended Phase : Sine 20 Hz / PM 2 Hz

- 5

- 3

- 1

1

3

5
0,0

0,2

0,4

0,6

0,8

1 ,0

Time [s]

[
r
a
d
]

Wrapped phase
(principal values
of phase])
Unwrapped phase
Detrend phase
Frequency spectrum of the
modulated signal
Phase modulated signal
Conversion of the frequency modulated impulse signal
to the harmonic signal
Time : Expanded Time(Encoder1)
-2
0
2
4
6
0,000 0,001 0,002 0,003
Time [s]
Timer : Time: Real (Expanded Time(Encoder1))
-4
-2
0
2
4
0,000 0,001 0,002 0,003
Time [s]
The original impulse signal
The resulting harmonic signal
Band pass filtration
Autospectrum
-60
-40
-20
0
20
0 10000 20000
Frequency [Hz]
d
B
/
r
e
f

1

V
Autospectrum
40
90
140
0 10000 20000
Frequency [Hz]
d
B
/
r
e
f

1
E
-
6

Frequency responsem
-60
-40
-20
0
20
0 10000 20000
Frequency [Hz]
d
B
Band pass filter
178
179
Correlation Function
Lecture 14
Pink Noise
-0,05
0,00
0,05
0,10
0,15
0,20
0,25
-1,00 -0,50 0,00 0,50
Lag [s]
U
^
2
180
Correlation function and power spectral density
( ) ( ) ( ) ( ) e e = e = e
*
2 1 1
X X
T
X
T
S
xx
( ) ( ) { } ( ) ( )
}
e = = e
T
t t j t x t x X
0
d exp FT
( ) ( ) ( )
}
t + = t
T
xx
t t x t x
T
R
0
d
1
The Power Spectral Density of a signal x(t) over a time interval T is as follows
where X() is the Fourier transform of the signal x(t)
The autocorrelation function of an stationary signal x(t) at time lag (delay) , which is the length of
the signal, over which each correlation is calculated, is defined as an expectance (mean value).
+ T It is assumed that the length of the time interval T grows without bound
( ) ( ) ( ) { } t t + = t x t x E R
xx
For processes that are also ergodic, the expectation can be replaced by the limit of a time
average. The autocorrelation of an ergodic process is defined as
The normalised autocorrelation function takes the form
( ) ( ) ( ) 0
xx xx xx
R R t = t
See [Orfanidis]
181
WienerKhinchin theorem
( ) ( ) ( ) ( ) e e = e = e
*
2 1 1
X X
T
X
T
S
xx
( ) ( ) { } ( ) ( )
|
|
.
|
\
|
e = = e
}
T
t t j t x t x X
0
d exp FT
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
} }
} } } }
t et
|
|
.
|
\
|
t =
= e =
|
|
.
|
\
|
e +
|
|
.
|
\
|
e = e
T
T
T
T T T T
xx
j dt t x t x
T
t t t t j t x t x
T
t t j t x t t j t x
T
S
d exp
1
d d exp
1
d exp d exp
1
0
0
2
0
1 2 1 1 2
0
2 2 2
0
1 1 1
( ) ( ) ( )
( ) ( ) ( )
}
}
+

+

e et + e
t
= t
t et t = e
d exp
2
1
d exp
j S R
j R S
xx xx
xx xx
is the Fourier transform of the corresponding autocorrelation function.
After substitution
Autocorrelation function
( ) ( ) e = e
xx xx
S S
( ) ( ) ( ) ( ) ( )
} } }
+

+

+
+
= e e
t
= = f f S S t t x
T
R
xx xx
T
T
T
xx
d d
2
1
d
2
1
0
2
lim
The power spectral density of a wide-sense-stationary random process
The WienerKhinchin theorem relates the autocorrelation function to the power spectral density
via the Fourier transform
t t t t = =
1 2 1
, t
Cross-correlation function
( ) ( ) ( )
}
+

e et + e = t d exp j S R
xy xy
( ) ( ) ( )
}
+

t et t = e d exp j R S
xy xy
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) e = t t e t = t et t = t et t = e
} } }
+

+

+

yx yx yx xy xy
S j R j R j R S
1 1 1
d exp d exp d exp
( ) 0 = e
xy
S
For stationary random signals x(t), y(t), the cross-correlation function is defined by
( ) ( ) ( ) { } t t + = t y t x E R
xy
( ) ( ) ( )
}
t + = t
T
xy
t t y t x
T
R
0
d
1
For uncorrelated random signals
The Fourier direct and inverse transform of the cross-correlation function is as follows
For stationary random signals x(t), y(t) , which are also ergodic, the definition turns to
It can be proved that
( ) 0 = t
xy
R
182
183
Properties of the autocorrelation function
( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) t t t t t t
t t t t
t t t
yy xx yx xy yy xx
T T T T
T
xx
R R R R R R
t t x t y
T
t t y t x
T
t t y t y
T
t t x t x
T
t t y t x t y t x
T
R
+ = + + + =
= + + + + + + + =
= + + + + =
} } } }
}
0 0 0 0
0
d
1
d
1
d
1
d
1
d
1
The autocorrelation is an even function of the time-lag
The autocorrelation function reaches its peak value at the origin, where it takes a real value, i.e. for
any lag (delay) ,
The autocorrelation of a periodic function is, itself, periodic with the very same period
The autocorrelation of the sum of two completely uncorrelated functions x(t) and y(t) (the cross-
correlation R
xy
() is zero for all ) is the sum of the autocorrelations of each function separately
( ) ( ) t = t
xx xx
R R
( ) ( ) t >
xx xx
R R 0
0
The autocorrelation of a continuous-time white noise signal will have a strong peak (the Dirac
delta function) at = 0 and will be absolutely 0 for all other .
( ) ( ) t o = t
xx
R
For = 0 it is possible to get
( ) ( ) ( ) ( ) ( )
} } }
+

+

= e e
t
= = f f S S t t x
T
R
xx xx
T
xx
d d
2
1
d
1
0
0
2
184
Estimation of the correlation function
( ) ( ) ( ) { } t + = t t x t x E R
xx
( )
)
)
e
e
= q
T t
T t
t
; 0 , 0
; 0 , 1
( ) { } ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) { } ( ) t
|
|
.
|
\
| t
= t + t + q q =
)
`
t + t + q q = t
} }
xx
T T
xx
R
T
t t x t x E t t
T
t t x t x t t
T
E R E 1 d
1
d
1
0 0
( )
|
|
.
|
\
| t
t
T
R
xx
1
0 T
The definition of the autocorrelation function assumes that the signal x(t) continues from minus
infinity to plus infinity
In fact, the signal measurement takes a limited time interval
The expected values of the autocorrelation function estimation is as follows
The unbiased autocorrelation function can be obtained by the use of the formula
( ) t
xx
R
Computation of results in a bias estimation of ( ) t

xx
R
t
( ) ( ) t x t q
This formula is sometime called Bow-tie correction (BK signal analyzers)
185
Auto- and cross-correlation function of the sampled signal
( ) ( ) ( ) 2 ,..., 2 , 1 , 0 ,
1
1
0
= t t +
t
= t

t
=
N i y i x
N
R
N
i
xy
( ) ( ) ( ) 2 ,..., 2 , 1 , 0 ,
1
1
0
= t t +
t
= t

t
=
N i x i x
N
R
N
i
xx
Autocorrelation
function
Cross-correlation
function
The correlations are a function of the time lag , that is the length of signal over which each
correlation is calculated
( ) i x
( ) t + i x
i
0 1 2 N-1
* * *
N -
*
Unbiased estimation
( ) t + i y
Calculation
or
Let x
n
, i = 0, 1, 2, , N-1 be a sampled ergodic signal, where N is the sample count
186
Autocorrelation function of a continuous harmonic signal
Biased estimation
-1,0
-0,5
0,0
0,5
1,0
-1,0 0,0 1,0
Lag [s]
U
^
2
( ) ( ) + e = t A t x cos
( ) ( ) ( ) ( )
( ) ( ) ( )
( )
( )
( ) et =
|
|
.
|
\
|
(
e
+ et + e
+ et =
= et + + et + e =
= + t + e + e = t
+
+
+
}
}
cos
2 2
2 2 sin
cos
2
1
lim
d cos 2 2 cos
2
1
lim
d cos cos
1
lim
2
0
2
0
2
0
2
A t A
T
T
t t
A
T
t t t A
T
R
T
T
T
T
T
T
xx
Let be a harmonic signal. The autocorrelation function is as follows
Unbiased estimation
-1,0
-0,5
0,0
0,5
1,0
-1,0 0,0 1,0
Lag [s]
U
^
2
Cosine function of lag
187
Examples no.1 Random signals
White Noise
-0,5
0,0
0,5
1,0
1,5
-1,0 -0,5 0,0 0,5
Lag [s]
U
^
2
Pink Noise
-0,05
0,00
0,05
0,10
0,15
0,20
0,25
-1,00 -0,50 0,00 0,50
Lag [s]
U
^
2
Noisy harmonic signal
-4,0
-2,0
0,0
2,0
4,0
0,0 0,5 1,0
Time [s]
U
Noisy harmonic signal
-1,0
-0,5
0,0
0,5
1,0
1,5
2,0
-1,0 -0,5 0,0 0,5 1,0
Lag [s]
U
^
2
( )
2
x xx
S o = e ( ) ( ) t o o = t
2
x xx
R
( )
2 2
2
2
o + e
oo
= e
y
yy
S
( ) ( ) t o o = t exp
2
y yy
R
Examples no.2 Tonal signals
-2,0
-1,0
0,0
1,0
2,0
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
U
0,0
0,1
0,2
0,3
0,4
0,5
0 50 100
Frequency [Hz]
R
M
S

U
-0,4
-0,2
0,0
0,2
0,4
-1,0 -0,5 0,0 0,5
Lag [s]
U
^
2
-0,4
-0,2
0,0
0,2
0,4
-1,0 -0,5 0,0 0,5
Lag [s]
U
^
2
Time domain signal Autospectrum
Autocorrelation (biased) Autocorrelation (unbiased)
qualitatively difference
in appearance
188
Examples no.3 Detecting an echo in a signal
SweptSine - Chirp
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0,000 0,002 0,004
Time [s]
SweptSine -frequency
0
5000
10000
15000
20000
0,000 0,002 0,004
Time [s]
H
z
Repeated chirp
-2
-1
0
1
2
0 1 2
Time [s]
Repeated chirp & Echo
-1,5
-1,0
-0,5
0,0
0,5
1,0
1,5
0 1 2
Time [s]
Autocorellation
-0,015
-0,010
-0,005
0,000
0,005
0,010
0,015
-0,008 0,000 0,008
Lag [s]
U
^
2
Autocorrelation
-0,015
-0,010
-0,005
0,000
0,005
0,010
0,015
-0,008 0,000 0,008
Lag [s]
U
^
2
Envelope of autocorellation
function
0,000
0,004
0,008
0,012
0,016
-0,008 0,000 0,008
Lag [s]
U
^
2
150
samples
150
samples
String of the chirp signals String of the chirp signals
affected by the 10%-echo,
which is delayed by 150
samples
Effect of echo
( ) k x
( ) ( ) 150 1 . 0 + k x k x
Chirp String of chirps
Autocorrelation
Frequency
Autocorrelation
f
Sampl
= 44100 Hz
189
Examples no.4 Impulse response
( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) , d d d
1
d d
1
d
1
1 1 1
0
1 1 1
0
1 1 1
0
} } }
} } }
+

+

+

t = t + =
= t + = t + = t
t t g t R t t t g t t x t x
T
t t t g t t x t x
T
t t y t x
T
R
xx
T
T T
xy
The Wiener-Hopfov equation
G(j)
g(t)
( ) t x ( ) t y
Input Output
The input-output cross-correlation function
Let x(t) be white noise, ( ) ( ) t o o = t
2
x xx
R
( ) ( ) ( ) ( ) ( ) ( ) t o = t o o = t = t
} }
+

+

g t t g t t t g t R R
x x xx xy
2
1 1 1
2
1 1 1
d d
Cross-correlation function
-0,2
-0,1
0,0
0,1
0,2
0,3
-0,50 -0,25 0,00 0,25 0,50
Lag [s]
Impulse response
-0,2
-0,1
0,0
0,1
0,2
0,3
0,00 0,25 0,50 0,75 1,00
Time [s]
[
-
]
Filtration effect of (t)
190
Examples no.5 Frequency response function
G(j)
g(t)
( ) t x ( ) t y
Input Output
Fourier Transform
-0,5
0,0
0,5
1,0
1,5
0,0 0,2 0,4 0,6 0,8 1,0
Time [s]
x
(
t
)
,

y
(
t
)
x(t)
y(t)
1,E-10
1,E-09
1,E-08
1,E-07
1,E-06
1,E-05
1,E-04
1 10 100 1000
Frequency [Hz]
P
S
D

[
-
]
^
2
/
H
z
Sxx
Syy
Correlation
-0,0015
-0,0010
-0,0005
0,0000
0,0005
0,0010
0,0015
-0,5 -0,3 0,0 0,3 0,5
Time [s]
Rxx
Ryy
FFT
1,E-06
1,E-04
1,E-02
1,E+00
1 10 100 1000
Frequency [Hz]
1
0
2
4

*

A
m
p
l

-
FFT(Ryy)
FFT(Rxx)
( ) ( ) ( )
}
+

t et t = e d exp j R S
191

Signal Processing and Diagnostics

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Signal Processing and Diagnostics

Output is produced piecewise by an overlap-add algorithm

Computation of results in a bias estimation of ( ) t

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