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PGB SignalProcessingLectures PDF
PGB SignalProcessingLectures PDF
References
Mitra S
S.K.,
K Digital
Digital Signal Processing
Processing, A Computer based approach,
approach Mc
Mc-Graw
Graw Hill
Hill,
3rd Edition, 2006.
Heylen W., Lammens S. And Sas P., Modal Analysis Theory and Testing, Katholieke
Universiteit Leuven
Leuven, 1997
1997.
"The Scientist and Engineer's Guide to Digital Signal Processing, copyright 19971998 by Steven W. Smith. For more information visit the book's website at:
www.DSPguide.com"
Boore, D. M. (2005). On pads and filters: Processing strong-motion data, Bull. Seism.
Soc. Am. 95,745-750.
P. Gundes Bakir,
Signals
Definition
A signal is a function of independent variables such as
time, distance, position, temperature and pressure.
A signal carries information, and the objective of signal
processing
i is
i tto extract
t t useful
f l information
i f
ti carried
i db
by th
the
signal.
Signal processing is concerned with the mathematical
representation
p
of the signal
g
and the algorithmic
g
operation
p
carried out on it to extract the information present.
P. Gundes Bakir,
Definition
y = f (x)
P. Gundes Bakir,
Signal types
F a simple
For
i l pendulum
d l
as shown,
h
b
basic
i d
definition
fi i i iis:
P. Gundes Bakir,
Signal types
Also, two
Al
t identical
id ti l pendula
d l released
l
d from
f
= 0 att t=0,
t 0 will
ill have
h
the
th same
motions at all time. There is no place for uncertainty here.
If we can uniquely specify the value of for all time, i.e., we know the
underlying functional relationship between t and , the motion is
deterministic or predictable
predictable. In other words
words, a signal that can be uniquely
determined by a well defined process such as a mathematical expression or
rule is called a deterministic signal.
The opposite situation occurs if we know all the physics there is to know,
but sstill ccannot
bu
o ssay
yw
what thee ssignal
g w
will be at thee next
e timee instant-then
s
e thee
signal is random or probabilistic. In other words, a signal that is generated
in a random fashion and can not be predicted ahead of time is called a
random signal.
signal
P. Gundes Bakir,
Signal types
Typical examples to deterministic signals are sine chirp and digital stepped
sine.
P. Gundes Bakir,
Signal types
P. Gundes Bakir,
Signal types
P. Gundes Bakir,
10
Signal types
P. Gundes Bakir,
11
Signal types
P. Gundes Bakir,
12
Signal types
A signal
i
l with
i h a time
i
varying
i mean iis an aperiodic
i di signal.
i
l
P. Gundes Bakir,
13
Signal types
IIt should
h ld b
be noted
d that
h periodicity
i di i d
does not necessarily
il mean a
sinusoidal signal as shown in the figure.
P. Gundes Bakir,
14
Signal types
A periodic
i di signal
i
l iis one that
h repeats iitself
lf iin time
i
and
d iis a
reasonable model for many real processes, especially those
associated with constant speed machinery.
P. Gundes Bakir,
15
16
Classification of signals
For a 1 D signal,
signal the independent variable is usually labelled as
time. If the independent variable is continuous, the signal is called a
continuous-time signal. A continuous time signal is defined at
y instant of time.
every
17
Classification of signals
P. Gundes Bakir,
18
Classification of signals
A discrete
di
t titime signal
i
l with
ith
continuous valued amplitudes is
called a sampled-data signal. A
digital signal is th
thus
saq
quantized
anti ed
sampled-data signal.
19
CLASSIFICATIONS OF SIGNALS
11DD
signals
signals
2D
signals
Stationary
D t
Deterministic
i i ti
Periodic
Monofrequency
(sinuzoidal)
P. Gundes Bakir,
Non-Stationary
Random
Continuous
Transient
Aperiodic
Multi
Multifrequency
T
Transient
i t
I fi it aperiodic
Infinite
i di
20
22
P. Gundes Bakir,
23
P. Gundes Bakir,
24
P. Gundes Bakir,
25
Fourier transforms
This chapter focuses on Fourier-series expansion, the
discrete Fourier transform, properties of Fourier
Transforms and Fast Fourier Transform
Fourier transforms
Fourier
Fo
rier analysis
anal sis is a family
famil of mathematical techniques,
techniq es all based on
decomposing signals into sinusoids.
The discrete Fourier transform (DFT) is the family member used with
digitized signals.
The general term Fourier transform can be broken into four categories,
g from the four basic types
yp of signals
g
that can be encountered.
resulting
P. Gundes Bakir,
27
Categories of Fourier
Transforms
P. Gundes Bakir,
28
Fourier transforms
These four
Th
f
classes
l
off signals
i l all
ll extend
t d to
t positive
iti andd negative
ti infinity.
i fi it What
Wh t if
you only have a finite number of samples stored in your computer, say a signal
formed from 1024 points?
There isnt a version of the Fourier transform that uses finite length signals. Sine
and cosine waves are defined as extending from negative infinity to positive
infinity. You cannot use a group of infinitely long signals to synthesize something
finite in length. The way around this dilemma is to make the finite data look like an
infinite length signal. This is done by imagining that the signal has an infinite
number of samples on the left and right of the actual points. If all these imagined
samples
l have
h
a value
l off zero, the
th signal
i l looks
l k discrete
di
andd aperiodic,
i di andd the
th
discrete time Fourier transform applies.
As an alternative,
A
lt
ti the
th imagined
i
i d samples
l can be
b a duplication
d li ti off the
th actual
t l 1024
points. In this case, the signal looks discrete and periodic, with a period of 1024
samples. This calls for the discrete Fourier transform to be used.
P. Gundes Bakir,
29
P. Gundes Bakir,
30
The frequency domain contains exactly the same information as the time domain,
just in a different form. If you know one domain, you can calculate the other.
Given the time domain signal, the process of calculating the frequency domain is
called decomposition, analysis, the forward DFT, or simply, the DFT.
The number of samples in the time domain is usually represented by the variable N.
While N can be any positive integer, a power of two is usually chosen, i.e., 128,
256, 512, 1024, etc. There are two reasons for this. First, digital data storage uses
bi
binary
addressing,
dd
i making
ki powers off ttwo a natural
t l signal
i l length.
l th Second,
S
d the
th mostt
efficient algorithm for calculating the DFT, the Fast Fourier Transform (FFT),
usually operates with N that is a power of two. Typically, N is selected between 32
and 4096. In most cases, the samples run from 0 to N
N-11 , rather than 1 to N.
P. Gundes Bakir,
31
32
P. Gundes Bakir,
33
A function f(x) is called a periodic function if f(x) is defined for all real x and if
there is a positive number p, called a period of f(x), such that
f ( x + p) = f ( x)
P. Gundes Bakir,
34
Familiar periodic functions are the cosine and sine functions. Examples of
functions that are not periodic are:
x, x 2 , x 3 , e x , cosh x, ln x
If f(x) has period p, it also has the period 2p because the equation
f ( x + p) = f ( x)
implies that
f ( x + 2 p ) = f ([x + p ] + p ) = f ( x + p ) = f ( x)
f ( x + np) = f ( x)
P. Gundes Bakir,
35
Furthermore if f(x) and g(x) have period p, then af(x)+bg(x) with any constants
a and b also has the period p.Our problem in the first few slides will be the
representation of various functions f(x) of period 2 in terms of the simple
functions
P. Gundes Bakir,
36
P. Gundes Bakir,
37
The equation
is called the Fourier series of f(x). We shall prove that in this case, the
coefficicents of the above equation are the so called Fourier coefficients of f(x)
given by the Euler formulas.
1
a0 =
2
an =
bn =
P. Gundes Bakir,
f ( x)dx
f ( x) cos nxdx
n = 1,2, L
f ( x) sin nxdx
n = 1,2, L
38
Let f(x) be periodic with period 2 and piecewise continuous in the interval
x . Furthermore, let f(x) have a left hand derivative and a right hand
d i ti att each
derivative
h point
i t off th
that
t iinterval.
t
l Th
Then th
the F
Fourier
i series
i off
with coefficients
1
a0 =
2
an =
bn =
f ( x)dx
f ( x) cos nxdx
n = 1,2, L
f ( x) sin nxdx
n = 1,2, L
39
The left hand limit of f(x) at xo is defined as the limit of f(x) as x approaches xo
from the left and is commonly denoted by f(xo-h). Thus,
The right hand limit of f(x) at xo is defined as the limit of f(x) as x approaches
xo from the right and is commonly denoted by f(xo+h). Thus,
f ( xo + h) = lim
li h0 f ( xo + h) as h 0 through
th
h positive
iti values.
l
P. Gundes Bakir,
40
Example
Find
Fi
d the
h Fourier
F i coefficients
ffi i
off the
h periodic
i di ffunction
i f(
f(x)) iin the
h fifigure.
The formula is:
k if < x < 0
f ( x) =
0< x <
if
k
and f ( x + 2 ) = f ( x)
P. Gundes Bakir,
41
Example
Find
Fi
d the
h Fourier
F i coefficients
ffi i
off the
h periodic
i di ffunction
i f(
f(x)) iin the
h fifigure.
The formula is:
k if < x < 0
f ( x) =
0< x <
if
k
and f ( x + 2 ) = f ( x)
1
a0 =
2
1
a0 =
2
=
f ( x)dx
1
f
(
x
)
dx
=
1
(
k
)
dx
+
(k )dx
2 0
1
1
1
1
0
(kx) +
(kx) 0 =
k +
k = 0
2
2
2
2
The above can also be seen without integration, since the area
under
d th
the curve off f(
f(x)) between
b t
- and
d is
i zero.
P. Gundes Bakir,
42
Example
From
an =
f ( x) cos nxdx
1
an = f ( x) cos nxdx = (k ) cos nxdx + (k ) cos nxdx
0
0
1
sin nx
sin nx
) + (k
) =0
= ( k
n
n
bn =
f ( x) sin nxdx
1
bn = f ( x) sin nxdx = ( k ) sin nxdx + (k ) sin nxdx
0
0
1 cos nx
cos nx
= k
k
n
n 0
43
Example
k
[cos 0 cos((n ) cos n + cos 0]
n
k
[2 2 cos n ]
=
n
2k
(1 cos n )
=
n
Now cos(( ) = 1 , cos2 = 1,, cos(3
( ) = 1 etc in g
general
bn =
- 1 for odd n
cosn =
1 for even n
f odd
dd n
2 for
and thus 1 cosn =
0 for even n
P. Gundes Bakir,
44
Example
Hence the Fourier coefficients b n of our function are :
4k
4k
4k
, b4 = 0, b5 =
,L
, b2 = 0, b3 =
5
3
4k
1
1
Th partial
The
ti l sums are :
4k
sin x
4k
1
S2 =
sin x + sin 3 x etc.
3
S1 =
Their
Th
i graph
h seems to
t indicate
i di t that
th t the
th series
i iis convergentt and
dh
has
the sum f(x), the given function.
P. Gundes Bakir,
45
Example
We notice
W
i that
h at x=0
0
and x=, the points of
discontinuity of f(x), all
partial sums have the
value zero, the
arithmetic mean of the
limits k and k of our
function at these
points.
points
P. Gundes Bakir,
46
Th key
The
k to the
h E
Euler
l fformulas
l iis the
h orthogonality
h
li off
(b) sin
i nx sin
i mxdx
d = 0 ( n m)
n = m)
P. Gundes Bakir,
47
The functions considered so far had period 2, for the simplicity of the formulas. However, we will
mostly use the variable time t and work with functions x(t) with period T. We now show that the
transition from 2 to period T is quite simple. The idea is simply to find and use a change of scale
that gives from a function f(x) of period 2 to a function of period T.
In the equations below we can write the change of scale as: x=kt with n such that the old period x=2
gives for the new variable t the new period t=T. Thus
2
=kT
kT hence k=2
k 2 /T and x=kt=
x kt 2t/T. This implies dx=
dx 2dt/T which upon substitution into
1
a0 =
2
an =
bn =
T /2
1
a0 =
x(t )dt
T T/ 2
f ( x)dx
f ( x) cos nxdx
n = 1,2, L
2
2nt
an =
x
(
t
)
cos
dt
T T/ 2
T
n = 1,2, L
n = 1,2, L
2
2nt
bn =
x
(
t
)
sin
dt
T T/ 2
T
n = 1,2, L
T /2
f ( x) sin nxdx
P. Gundes Bakir,
T /2
48
Since we will mostly use the variable time t and in the frequency domain
2n/T, the equation
n =1
2nt
2nt
x(t ) = a0 + an cos
+ bn sin
T
T
n =1
The coefficients in this case can be written as shown on the rhs rather than the
lhs.
1
a0 =
2
an =
bn =
T /2
1
a0 =
x(t )dx
T T/ 2
f ( x)dx
f ( x) cos nxdx
n = 1,2, L
2
2nt
an =
x
(
t
)
cos
dt
T T/ 2
T
n = 1,2, L
n = 1,2, L
2
2nt
bn =
x
(
t
)
sin
dt
T T/ 2
T
n = 1,2, L
f ( x) sin nxdx
P. Gundes Bakir,
T /2
T /2
49
For a function x(t) defined on the interval [-/2, /2], we have the
representation on that interval
P. Gundes Bakir,
50
but this is zero as sin(m)=0 for all m. The second integral is:
51
Now if n and m are different integers then n-m and n+m are both nonzero integers
and the sine terms in the last expression vanish. If n and m are equal, we have a
problem with the second term above. We could use a limit argument but it is
simpler to go back to the first equation with n=m.
P. Gundes Bakir,
52
2n
2m
cos
t
cos
d =0
t dt
f
for
mn
for m = n 0
for m = n = 0
P. Gundes Bakir,
53
As
We have
I m = am
I m = a 0
for m = n 0
for m = n = 0
for m = n 0
P. Gundes Bakir,
54
If m=n=0
0 iin the
h b
below
l
equation:
i
a m =
2m
t dt
x(t ) cos
for m = n = 0
mn =
P. Gundes Bakir,
55
P. Gundes Bakir,
56
P. Gundes Bakir,
57
U i th
Using
the equations
ti
2
2nt
an =
x
(
t
)
cos
dt
T
T T/ 2
n = 1,2, L
2
2nt
bn =
x
(
t
)
sin
dt
T T/ 2
T
n = 1,2, L
T /2
T /2
in
+T / 2
+T / 2
1
1 2
2nt
2nt
x(t ) cos
cn = (an ibn ) =
dt
i
x
t
dt
(
)
sin
T
T
2
2 T T / 2
T / 2
Gives:
1
=
T
+T / 2
2nt
2nt
x
(
t
)
cos
i
sin
dt
T
T
T / 2
1
cn =
T
P. Gundes Bakir,
+T / 2
i 2nt / T
x
(
t
)
e
dt
T / 2
58
Alt
Alternatively,
ti l using
i th
the orthogonality
th
lit equation
ti
T /2
int imt
e
e dt = 0 for n + m 0
T / 2
= T for n + m = 0
and multiplying the equation
P. Gundes Bakir,
59
Fourier transform
If we wantt to
t look
l k att the
th spectral
t l content
t t off nonperiodic
i di signals
i
l we
have to let as all the interval t [-, ] contains important
information. Recall the exponential form of the Fourier series
where
C bi i th
Combining
the above
b
ttwo equations
ti
give:
i
60
Fourier Transform
If we suppose
pp
that the p
position of the t axis is adjusted
j
so that the
mean value of x(t) is zero. Then according to the first of the below
equation the coefficient ao will be zero.
T /2
1
a0 =
x(t )dx
T T/ 2
2
2nt
an =
x
(
t
)
cos
dt n = 1,2, L
T T/ 2
T
T /2
2
2nt
bn =
x
(
t
)
sin
dt n = 1,2, L
T T/ 2
T
T /2
The remaining
Th
i i coefficients
ffi i t an and
d bn
b will
ill in
i generall allll b
be diff
differentt
and their values may be illustrated graphically as shown.
P. Gundes Bakir,
61
Fourier Transform
Recall that the spacing between the frequency lines is
so that the kth spectral line is at
From the first equation, we see that
The equation
becomes
P. Gundes Bakir,
62
Fourier Transform
As 0,
A
0 the
h n become
b
closer
l
and
d closer
l
together
h and
d the
h
summation turns into an integral with =d (assuming that x(t) is
appropriately well behaved. In the limit
It follows
f ll
th
thatt if we d
define
fi
63
Fourier Transform
Note
N
t the
th fformall similarity
i il it tto th
the L
Laplace
l
ttransform
f
iin ffactt we obtain
bt i
the Fourier transform by letting s= i in the Laplace transform. The
main difference between the two is the comparative simplicity of the
inverse Fourier transform
transform.
P. Gundes Bakir,
64
IIn reality
li not only
l will
ill the
h signal
i
lb
be off fi
finite
i d
duration
i b
but iit will
ill b
be
sampled. It is not possible to store a continuous function on a
computer as it would require infinite memory.
What one usually does is take measurements from the signal at
regular intervals say t seconds apart so the signal for manipulation
takes the form of a finite vector of N samples
65
Recallll the
R
h exponential
i l fform off the
h F
Fourier
i series
i the
h spectrall
coefficients are
In keeping with our notation for the Fourier transform we will relabel
cn by Xn from now on. Also the equation above is not in the most
convenient form for the analysis so we will modify it slightly.
Recall that x(t) is assumed periodic with period .
Consider the
integral
P. Gundes Bakir,
66
L t t=
Let
t t+,
t
th iintegral
the
t
lb
becomes
P. Gundes Bakir,
e 2in (t ) / = e 2nit /
67
P. Gundes Bakir,
68
or
P. Gundes Bakir,
69
Alt
Alternatively,
ti l if we specify
if th
the ffrequency spacing
i iin H
Hertz
t
P. Gundes Bakir,
70
1
X N 1 =
N
N 1
xr e
2i ( N 1) r
N
r =0
1
=
N
N 1
i 2r + i 2r / N
x
e
r e
r =0
1
X N 1 =
N
N 1
i 2r / N
e
x
r
r =0
1
=
N
i 2r / N
*
x
e
=
X
r
1
r =0
N 1
X N k = X k*
P. Gundes Bakir,
71
X ( )* = X ( )
This means that the spectral coefficient XN-1 corresponds to the frequency
- or more generally XN-k corresponds to the frequency -k. So the
array Xn stores the frequency representation of the signal xr as follows:
P. Gundes Bakir,
72
What
h happens
h
if there
h is
i a value
l off k for
f which
hi h N-k=k?
k k? This
hi implies
i li that
h N=2k
2k so
the number of sample points is even.
Actually this turns out to be the most usual situation, in fact it is a requirement of
the Fast Fourier Transform which we shall meet in the next lecture.
X N* / 2 = X N / 2
so this spectral line is real.
This finally justifies our assertion that the maximum frequency represented is
N/2.
P. Gundes Bakir,
73
xn = X r e i 2rn / N
r =0
where
1
Xr =
N
N 1
x e
p =0
i 2rp / N
xn =
r =0
P. Gundes Bakir,
N 1
x e
p =0
i 2rp / N
i 2rn / N
e
74
If we change
h
the
th order
d off the
th summations,
ti
we obtain:
bt i
N 1
xn =
p =0
i 2rp / N i 2rn / N
x
e
e
p
r =0
N 1
xn =
p =0
i 2r ( n p ) / N
x
e
p
r =0
N 1
1
xn =
p =0 N
x p e i 2 ( n p ) / N
r =0
]
r
=q
P. Gundes Bakir,
75
Thus, if np,
p within the equation
q
N 1
xn =
p =0
N 1
i 2 ( n p ) / N
x
e
p
r =0
=q
q
]
r
N 1
= 1 + q + q 2 + ...
r =0
If we call the left hand side of the above equation sr, we have:
sr = 1 + q + q 2 + ...
If we multiply both sides of the above equation by q, and subtract the
resulting equation from the above equation:
76
W can obtain
We
bt i the
th value
l off sr from
f
here
h
as:
1 qN
sr =
1 q
Now qN=1,
Now,
=1 which can be proved as:
qN=ei2(n-p)r=cos2r(n-p)+isin2r(n-p)=1+0=1
which results in the below result for np :
1 qN
=0
sr =
1 q
P. Gundes Bakir,
77
F n=p, the
For
h term iin b
brackets
k
iin the
h equation
i b
below
l
b
becomes 1
1.
N 1
xn =
p =0
N 1
i 2 ( n p ) / N
x
e
p
r =0
]
r
1
N
N 1
x
p =0
N pn = xn
P. Gundes Bakir,
78
Di
Discrete
F
Fourier
i T
Transform
f
N 1
X n = xr e i 2nr / N
r =0
1
xn =
N
P. Gundes Bakir,
N 1
i 2nr / N
X
e
r
r =0
79
Gibbs
Gibb
s phenomenon
The first figure shows a time domain signal being synthesized from
sinusoids. The signal being reconstructed is shown in the graph on the left
hand side. Since this signal is 1024 points long, there will be 513 individual
frequencies needed for a complete reconstruction. The figure on the right
sho s a reconstr
shows
reconstructed
cted signal using
sing freq
frequencies
encies 0 thro
through
gh 100
100. This signal
was created by taking the DFT of the signal on the left hand side, setting
frequencies 101 through 512 to a value of zero, and then using the Inverse
DFT to find the resulting time domain signal. The figure in the middle shows
a reconstructed
t t d signal
i
l using
i ffrequencies
i 0 th
through
h 30
30.
P. Gundes Bakir,
80
Gibbs
Gibb
s phenomenon
P. Gundes Bakir,
81
Gibbs
Gibb
s phenomenon
P. Gundes Bakir,
82
P. Gundes Bakir,
83
Bandwidth
84
Frequency responseresponse-Impulse
response
P. Gundes Bakir,
85
The answer is: infinitely high, if you are willing to pad the impulse response with an
infinite number of zeros. In other words, there is nothing limiting the frequency
resolution except the length of the DFT.
Even though the impulse response is a discrete signal, the corresponding frequency
response is continuous. An N point DFT of the impulse response provides N/2 + 1
samples of this continuous curve. If you make the DFT longer, the resolution improves,
and you obtain a better idea of what the continuous curve looks like.
This can be better understood by the discrete time Fourier transform (DTFT).
Consider an N sample signal being run through an N point DFT, producing an N/2 + 1
sample
p frequency
q
y domain. DFT considers the time domain signal
g to be infinitely
f
y longg
and periodic. That is, the N points are repeated over and over from negative to positive
infinity. Now consider what happens when we start to pad the time domain signal with
p g in the
an ever increasingg number of zeros, to obtain a finer and finer sampling
frequency domain.
P. Gundes Bakir,
86
Now we will take this to the extreme, by adding an infinite number of zeros to the
time domain signal. This produces a different situation in two respects.
First, the time domain signal now has an infinitely long period. In other words, it
h turned
has
d iinto an aperiodic
i di signal.
i l
P. Gundes Bakir,
87
Close peaks
P. Gundes Bakir,
88
Output of a system
What are you going to do if given an input signal and impulse response, and
need to find the resulting output signal? Transform the two signals into the
frequency domain, multiply them, and then transform the result back into the
time domain.
P. Gundes Bakir,
89
Homogeneity means that a change in amplitude in one domain produces an identical change
i amplitude
in
lit d in
i the
th other
th domain.
d
i When
Wh the
th amplitude
lit d off a time
ti domain
d
i waveform
f
is
i changed,
h
d
the amplitude of the sine and cosine waves making up that waveform must also change by an
equal amount. In mathematical form, if x[ ] and X[ ] are a Fourier Transform pair, then kx[ ]
andd kX[
k [ ] are also
l a Fourier
i Transform
f
pair,
i for
f any constant k.
k
P. Gundes Bakir,
90
P. Gundes Bakir,
91
In spite
p of being
g linear, the Fourier transform is not shift invariant. In other
words, a shift in the time domain does not correspond to a shift in the
frequency domain.
Let h(t)
( ) be the impulse
p
response,
p
, i.e.,, the systems
y
response
p
to a Dirac
impulse, it can be proved that the response g(t) of the system to an input f(t)
is the convolution of f(t) and h(t):
+
g (t ) =
f ( )h(t )d = h(t ) f (t )
The convolution
Th
l ti th
theorem iin F
Fourier
i analysis
l i states
t t th
thatt convolution
l ti in
i one
domain corresponds to multiplication in the other domain.
Hence the frequency response function H(f) is the ratio between the
response and
d th
the iinputt as a ffunction
ti off the
th ffrequency.
g (t ) = h(t ) f (t )
G( f ) = H ( f ) F ( f )
P. Gundes Bakir,
92
Differentiation
Diff
i i in
i the
h time
i
d
domain
i iis equivalent
i l
to multiplication
l i li i b
by a
factor i in the frequency domain.
F [x& (t )] = (i ) X ( )
F x( )d =
If the input is the harmonic probe eit, the output is eit multiplied by
the FRF evaluated at .
y (t ) = H ( )eit
P. Gundes Bakir,
93
Fourier transform
Measured
M
d signals
i
l are time
i
d
domain
i ffunctions.
i
IIt iis iimportant to
investigate the signals in the frequency domain in order to study
their frequency content. The Fourier tansform is a tool to transform
signals from the time domain to the frequency domain.
The signal can be transformed back to the time domain using the
inverse Fourier transform:
P. Gundes Bakir,
94
Th FFT is
The
i a clever
l
algorithm
l ith for
f rapidly
idl calculating
l l ti the
th DFT.
DFT
X k = xnWNnk where WN = e j 2 / N
r =0
The basic strategy that is used in the FFT algorithm is one of divide and
conquer, which involves decomposing an N point DFT into succesively
smaller DFTs.
P. Gundes Bakir,
95
Suppose
pp
that the length
g of xn is even,, (i.e.,
( , N is divisible byy 2).
) If xn is
decimated into two sequences of length N/2, computing the N/2 point
DFT of each of these sequences requires approximately (N/2)2
multiplications and the same number of additions.Thus, the two DFTs
require 2*(N/2)
2 (N/2)2=N
N2/2 multiplications and the same number of additions.
Therefore, if it is possible to find the N point DFT of xn from these two N/2
point DFTs, great computational time will be saved because N2/2
operations
ti
will
ill b
be required
i d iinstead
t d off N2.
g n = x2 n
P. Gundes Bakir,
N
n = 0,1,..., 1
2
96
Th second
The
d sequence hn is
i fformed
d ffrom th
the odd-index
dd i d tterms,
hn = x2 n +1
N
n = 0,1,,...,, 1
2
X k = xnWNnk =
nk
x
W
n N +
n =0
n even
N
1
2
N
1
2
l =0
l =0
nk
x
W
n N
n odd
97
Si
Since
WN = e j 2 / N
N 1
X k = xnWNnk =
WN2lk = WNlk/ 2
nk
x
W
n N +
n =0
n even
N
1
2
N
1
2
l =0
l =0
nk
x
W
n N
n odd
We obtain:
P. Gundes Bakir,
N
1
2
N
1
2
l =0
l =0
98
N t th
Note
thatt in
i th
the equation
ti
N
1
2
N
1
2
l =0
l =0
X k = Gk + WNk H k
k = 0,1,,....,, N 1
Although the N/2 point DFTs of gn and hn are sequences of length N/2, the
periodicity of the complex exponentials allows us to write:
Gk = Gk + N / 2
Hk = Hk+N /2
P. Gundes Bakir,
99
Th f
Therefore,
Xk may be
b computed
t d from
f
the
th N/2 point
i t DFTs
DFT Gk andd Hk.
Gk = g nWNnk/ 2 =
n =0
N
1
2
nk
g
W
n N /2 +
n even
N
1
2
nk
g
W
n N /2
n odd
N
1
4
n =0
n =0
P. Gundes Bakir,
100
P. Gundes Bakir,
101
Computing
C
ti an N point
i t DFT using
i FFT iis much
h more effective
ff ti th
than
calculating the DFT directly.
N2
N
N
=
=
N log 2 N log 2 N v
P. Gundes Bakir,
102
Sampling
Aliasing
g
We will
W
ill now explore
l
th
the iissues off aliasing
li i and
d apparentt signal
i
l
distortion associated with choosing a sample rate for digital
conversion of analog signals. This process is known as analog to
digital conversion (A/D)
(A/D).
The process of sampling reduces an infinite set to a finite set of
data resulting in a loss of information which is responsible for the
i
issues
d
detailed
t il d iin thi
this section.
ti
Since the original analog signal contains infinite information
(knowledge of the signal at any point in time), the frequencies within
th signal
the
i
l are also
l kknown.
Information lost in the sampling process is also lost in the frequency
representation. We will explain how this leads to the problem of
leakage where high frequency components can not be distinguished
from low frequency components.
P. Gundes Bakir,
104
Sampling
Nearly
N
l allll d
data acquisition
i i i systems sample
l d
data with
i h uniform
if
time
i
intervals. For evenly sampled data, time can be expressed as:
T = ( N 1)t
where N is the sampling index which is the number of equally
spaced
d samples.
l
F
For mostt F
Fourier
i analyzers
l
N iis restricted
t i t d tto a
power of 2.
1
fs =
= ( N 1)f
t
P. Gundes Bakir,
105
Sampling
S
Sampling
li ffrequency iis th
the reciprocal
i
l off th
the titime elapsed
l
d
t from one sample to the next.
The unit of the sampling frequency is cycles per second
or Hertz ((Hz),
) if the sampling
p gp
period is in seconds.
The sampling theorem asserts that the uniformly spaced
di
discrete
samples
l are a complete
l
representation
i off the
h
signal if the bandwidth fmax is less than half the sampling
rate. The sufficient condition for exact reconstructability
from samples at a uniform sampling rate fs (in samples
per unit time) (fs2fmax).
P. Gundes Bakir,
106
Aliasing
This happens when the sample rate is not fast enough for the signal
and one speaks of aliasing.
107
Aliasing
The central concept to avoid aliasing is that the sample rate must be
at least twice the highest frequency component of the signal
((fs2fmax)). We define the Nyquist
yq
or cut-off frequency
q
y
fs
1
=
fN =
2 2t
P. Gundes Bakir,
108
Aliasing
P. Gundes Bakir,
109
Aliasing
What happens if the original signal actually has a component above the Nyquist
frequency?
Now if the spectrum of the continuous signal extends beyond the Nyquist
frequency we see overlap
P. Gundes Bakir,
110
Aliasing
If any energy in
i the
h original
i i l signal
i
l extends
d b
beyond
d the
h N
Nyquist
i
frequency, it is folded back into the Nyquist interval in the spectrum
of the sampled signal. This folding is called aliasing
fs2fmax
P. Gundes Bakir,
111
Aliasing
fs2fmax
P. Gundes Bakir,
112
Leakage
When converting
Wh
i a signal
i
l ffrom the
h time
i
d
domain
i to the
h ffrequency
domain, the Fast Fourier Transform (FFT) is used.
The Fourier Transform is defined by
y the equation:
q
P. Gundes Bakir,
113
Leakage
The signal must start and end at the same point in its cycle.
P. Gundes Bakir,
114
Leakage
An example
p of a nonperiodic
p
signal
g
can be seen in the Figure.
g
P. Gundes Bakir,
115
Leakage
By comparing
B
i the
h Fi
Figures, iit
can be seen that the frequency
content of the signal is
smeared into adjacent
frequencies when the signal is
not periodic.
p
P. Gundes Bakir,
116
Leakage
Decreasing
g the frequency
q
y step
p f increases the observation time
T and hence will improve the periodicity of the signal.
P. Gundes Bakir,
117
Windowing
In signal processing
processing, a window function is a function that is zero
zerovalued outside of some chosen interval.
P. Gundes Bakir,
118
Windowing
A rectangular
t
l window
i d
i a ffunction
is
ti th
thatt is
i constant
t t inside
i id th
the
interval and zero elsewhere, which describes the shape of its
graphical representation. When another function or a signal (data) is
multiplied by a window function
function, the product is also zero
zero-valued
valued
outside the interval: all that is left is the "view" through the window. It
can be shown that there is no window with a narrower main lobe
than the rectangular window
window.
P. Gundes Bakir,
119
Windowing
120
Windowing
Th
The effect
ff t off the
th same
Hanning Window on the
time domain signal
g
can
be seen in the Figure.
Figure shows how the
Hanning window
weights the beginning
and end of the sample to
zero so that it is more
periodic
i di during
d i th
the FFT
process.
P. Gundes Bakir,
121
Windowing
P. Gundes Bakir,
122
Windowing
The Hanning
Th
H
i
window
i d
i a compromise
is
i b
between the
h Fl
Flat T
Top and
d
Rectangular windows. It helps to maintain the amplitude of a signal
while at the same time maintaining frequency resolution. This
window can have up to a 16% amplitude error if the signal is not
periodic.
P. Gundes Bakir,
123
Windowing
P. Gundes Bakir,
124
Windowing
P. Gundes Bakir,
125
Windowing
The exponential
Th
ti l window
i d
i
is
used to make a measurement
from a vibrating structure more
accurate.
accurate
An example of the
p
window can be
exponential
seen in the figure.
P. Gundes Bakir,
126
Windowing
The exponential window
can cause some
problems if not used
properly.
l
As an example
example, a very
simple lightly damped
structure was subjected
to an impact test
test. The
signal processing
parameters were selected
for a 400 Hz bandwidth
which resulted in a 1 sec
time window.
P. Gundes Bakir,
127
Exponential Window
O
On the
th right
i ht more
damping is applied and
the peaks are much
wider now!
If an excessive amount
of damping is needed to
minimize the effects of
leakage then you run
leakage,
the risk of missing
closely
y spaced
p
modes.
P. Gundes Bakir,
128
Exponential Window
1
fs =
= ( N 1)f
t
P. Gundes Bakir,
129
Windowing
If the entire transient event can be captured such that the FFT
requirements can be met, leakage will not be a problem.
The use of windows may also hide or distort the modes in the
measurement.
P. Gundes Bakir,
130
Windowing
Almost all the time when performing a modal test, the input
excitation can be selected such that the use of windows can be
eliminated. e.g.,signals such as pseudo random, burst random, sine
chirp, and digital stepped sine.
P. Gundes Bakir,
131
Averaging
P. Gundes Bakir,
133
Averaging
The averaging
Th
i can be
b implemented
i l
t d by
b taking
t ki N segments
t off ti
time
data xi(t) and transforming to
then
For a signal
P. Gundes Bakir,
134
Averaging
The first
Th
fi iis one average- a one-shot
h measurement. Al
Although
h
h the
h
sine wave (at 10.24 Hz) is visible, there is a lot of background noise
from the single average.
P. Gundes Bakir,
135
Averaging
Th next figure
The
fi
shows
h
the
h resultl off taking
ki 10 averages.
P. Gundes Bakir,
136
Averaging
Fi ll we see the
Finally,
h effect
ff
off taking
ki 100 averages.
P. Gundes Bakir,
137
Often we plot
Oft
l t E[|X|] 2 as this
thi is
i proportional
ti
l to
t power. It is
i called
ll d the
th
power spectral density and is denoted
138
139
Filters
Definition
Pads
d and
d ffilters
l
Filters
P. Gundes Bakir,
141
Filters
S
Such
had
device
i which
hi h passes parts
t off a signals
i
l ffrequency
content and suppresses others is called a filter. The
particular filter described above is called an antialiasing
p
g
filter for obvious reasons.
P. Gundes Bakir,
142
Filters
A filter
fil
i a ffunction
is
i that
h iin the
h ffrequency d
domain
i h
has a value
l close
l
to 1 in the range of frequencies that the analyst wishes to retain
and close to zero in the range of frequencies that the analyst
wishes to eliminate.
y (t ) = IDFT [H ( f ) X ( f )]
P. Gundes Bakir,
143
Filters
144
Classification of filters
Filters can be applied in the frequency domain or the time domain but their
function is best understood in the frequency domain.
145
The so-called
so called cutoff or corner frequency divides the pass band and
the stop band. A low pass causal Butterworth filter is an all pole filter
with a squared magnitude response given by:
P. Gundes Bakir,
146
T
Two
considerations
id ti
are iimportant
t t when
h applying
l i a hi
highcut
h t filt
filter.
The first is that the application of the filter will act in a contradictory
manner to any instrument correction and at least in some frequency
ranges the two will counteract each other.
The second consideration is that an upper frequency limit on the
usable range of high frequencies in the motion is imposed by the
sampling rate: the Nyquist frequency, which is the highest frequency
at which characteristics of the motion can be correctly determined, is
equal to (1/2t)
equa
( / t) where
e e t iss the
t e sampling
sa p g interval.
te a
A high-cut filter applied at frequencies greater than the Nyquist will
h
have
no effect
ff t on the
th record.
d
P. Gundes Bakir,
147
Mechanical system
Input
Voltage, e(t)
()
Force, F(t)
()
Output
Charge q(t)
Displacement y(t)
Current i(t)=dq/dt
Velocity v(t)=dy/dt
Inductance, L
Mass,m
Resistance, R
Damping,c
Capacitance,C
Compliance, 1/k
Constant parameters
P. Gundes Bakir,
148
Analogue filters
We will
W
ill start
t t th
the di
discussion
i with
ith an electrical
l t i l analogue
l
filt
filter. C
Consider
id th
the
circuit below with an alternating voltage input,
Vi (t ) = Vi cos(t )
P. Gundes Bakir,
149
Analogue filters
Elementary
El
t
circuit
i it th
theory gives
i
th
the output
t t voltage
lt
Vo(t) as th
the solution
l ti off th
the
differential equation:
RC
dVo
+ Vo = Vi (t )
dt
iRCVo ( ) + Vo ( ) = Vi ( )
S
So
Vo ( ) = H ( )Vi ( )
where
1
H ( ) =
1 + iRC
P. Gundes Bakir,
150
Analogue filters
H ( ) =
1 + R 2C 2 2
and the phase is :
H ( ) = tan 1 ( RC )
P. Gundes Bakir,
151
First let us adopt a more general notation. Let x(t) or xi be the input to the
filter and y(t) or yi be the output. The differential equation of the electrical
analogue filter is then,
RC
dyy
+ y = x(t )
dt
dyyi yi yi 11
=
dt
t
P. Gundes Bakir,
152
Thi results
This
lt iin a diff
difference equation
ti
RC
( yi yi 1 ) + yi = xi
t
RC
1
xi
yi 1 + yi =
yi =
RC
RC
+ 1
+ 1
yi = a1 yi 1 + b0 xi
with appropriate definitions for a1 and b0. Consider the signal
x(t ) = sin(
i (2 .5t ) + sin(
i (2 .50t )
P. Gundes Bakir,
153
N i sine
Noisy
i wave
P. Gundes Bakir,
154
yi = 0.6655 yi 1 + 0.3345 xi
The resulting noisy sine wave after one pass through the filter is:
P. Gundes Bakir,
155
The resulting noisy sine wave after five passes through the digital filter is:
P. Gundes Bakir,
156
So the
S
h fil
filter iis able
bl to remove the
h hi
high
h ffrequency sine-wave
i
effectively. However, note that the amplitude of the carrier signal
has also been attenuated. Also, importantly, the phase of the signal
has been altered.
We use a theorem from an earlier lecture which says that if the input
to a system is x(t)=eiwt, then the response is y(t)=H(w) eiwt. Now
define the backward shift operator Z-1 by:
Z 1 yi = yi 1
P. Gundes Bakir,
157
Thi allows
This
ll
us to write
i the
h equation
i
yi = a1 yi 1 + b0 xi
As:
(1 a1Z 1 ) yi = b0 xi
Let
xi = eiti
Th
Then
yi = H ( )e iti
1 iti
Z e
P. Gundes Bakir,
=e
iti 1
=e
i ( ti t )
=e
it iti
158
Th
Then
becomes:
(1 a1Z 1 ) yi = b0 xi
(1 a1e
it
) yi = b0 xi
IIn terms
t
off the
th FRF derived
d i d above,
b
we have
h
enough
h now tto obtain
bt i a
general result. A general digital filter would then be:
b0
H ( ) =
1 a1e it
ny
nx
j =1
j =0
yi = a j yi j + b j xi j
P. Gundes Bakir,
159
1
H ( ) =
1 + iRC
H ( ) 1 as 0
H ( ) 0 as
H ( ) 0 as 0
H ( ) 1 as
P. Gundes Bakir,
160
Namely:
1
1 + iRC
1
1
or
i
The FRF of the low pass filter becomes:
i
i
1
H ( ) =
= RC
RC
i
+1 1+
i
RC
P. Gundes Bakir,
RC
1+
2
R 2C 2
161
Thi filter
This
filt is
i high
hi h pass as required.
i d
When
When
RC
RC
= 0.1,
H ( ) = 0.0995
= 10,
H ( ) = 0.995
H ( ) =
2
1
2n
1
2n
1+
1 +
c
fc
where c is the cut-off frequency and n is a steepness factor which
specifies
p
how fast the signal
g
should die away
y after the cut off frequency.
q
y
P. Gundes Bakir,
162
where n is the order of the filter (number of poles in the system function)
and c is the cut off frequency of the Butterworth filter which is the
frequency where the magnitude of the causal filter |H(c )| is 1/2
regardless
dl off the
th order
d off the
th filter.
filt
The purpose of a low-cut filter is to remove that part of the signal that is
j dged to be hea
judged
heavily
il contaminated by
b long-period
long period noise
noise. The key
ke issue
iss e is
selecting the period beyond which the signal-to-noise ratio is unacceptably
low. Applying a filter that abruptly cuts out all motion at periods above the
desired cut-off can lead to severe distortion in the waveform,, and therefore
a transitionsometimes referred to as a ramp or a rolloff is needed
between the pass-band, where the filter function equals unity, and the
period beyond which the filter function is equal to zero.
P. Gundes Bakir,
163
P. Gundes Bakir,
164
Causal--acausal filters
Causal
The zero p
phase shift is achieved in the time domain by
y passing
p
g
the transform of the filter along the record from start to finish and
then reversing the order and passing the filter from the end of the
record to the beginning.
P. Gundes Bakir,
165
Causal--acausal filters
Causal
The imlementation
Th
i l
i off a zero phase
h
filtering
fil i scheme
h
iis shown
h
iin the
h
figure.
P. Gundes Bakir,
166
|H(c )| takes
t k the
th value
l off 0
0.5
5 ffor acausall filt
filters. Th
The mathematical
th
ti l
reasoning behind this can be explained as follows: If we let Hc() be
the frequency response of the causal Butterworth filter given by the
second equation, this filter has unity gain at frequency = 0. The
cutoff frequency c is the frequency at which the power of the filter
ouput is half the power of the filter input,i.e. |Hc(c )|2 = 1/2. The
frequency response of an acausal Butterworth filter Ha() is given
by:
167
Classification of filters
P. Gundes Bakir,
168
Classification of filters
P. Gundes Bakir,
169
Classification of filters
P. Gundes Bakir,
170
Classification of filters
P. Gundes Bakir,
171
Overview of excitation
signals
Classification
Th
The commonly
l used
d excitation
it ti
signals
i
l can be
b
categorized in several ways. For practical purposes, it is
easyy to consider two main groups:
g p broad band or single
g
frequency signals.
The signal frequency group contains:
Swept sine
Stepped sine
173
Classification
Transients
T
i t
Burst random
Burst chirp (or burst swept sine)
mpact excitation
Periodic
Pseudo random
Periodic random
Chirp (fast swept sine)
Nonperiodic
Pure random
P. Gundes Bakir,
174
Classification
R
Random
d
signals
i
l can only
l b
be d
defined
fi d b
by th
their
i
statistical properties.
For stationary random signals, these properties
do not vary with respect to translations in time.
All random excitation signals are of the ergodic
random type, which means that a time average
on any particular subset of the signal is the
same for any arbitrary subset of the random
signal.
P. Gundes Bakir,
175
Pure random
P. Gundes Bakir,
176
Pure random
P. Gundes Bakir,
177
Pseudo random
178
Periodic random
P. Gundes Bakir,
179
Periodic random
Other advantages are:
Signals are periodic in the sampling window,
so measurements are leakage free
free.
Removes non-linear behavior when used with
spectrum averaging
averaging.
Disadvantages are:
Slower than other random test methods.
Special software required for implementation
P. Gundes Bakir,
180
Burst random
181
Burst random
Advantages :
Signals are periodic in the
sampling
li window,
i d
so
measurements are
leakage free
free.
Removes non-linear
behavior
be
a o when
e used with
spectrum averaging.
Fast measurement time.
Disadvantages :
p
software required
q
Special
for implementation.
P. Gundes Bakir,
182
Burst random
P. Gundes Bakir,
183
184
P. Gundes Bakir,
185
P. Gundes Bakir,
186
187
Swept sine
Th
The sine
i wave excitation
it ti signal
i
lh
has b
been used
d since
i
th
the
early days of structural dynamic measurement. It was
the onlyy signal
g
that could be effectively
y used with
traditional analog instrumentation.
Even broad band testing methods (like impact testing),
have been developed for use with FFT analyzers, sine
wave excitation is still useful in some applications
applications. The
primary purpose for using a sine wave excitation signal
is to put energy into a structure at a specific frequency.
Slowly sweeping sine wave excitation is also useful for
characterizing non
non-linearities
linearities in structures
structures.
P. Gundes Bakir,
188
Swept sine
Advantages
Ad
t
off Si
Sine Testing
T ti
Best signal-to-noise and RMS-to-peak ratios of
any signal.
i
l
Controlled amplitude and bandwidth.
Useful for characterizing non-linearities.
Long history of use.
Disadvantages of Sine Testing
Distortion due to over-excitation.
Extremely slow for broad band measurements
measurements.
P. Gundes Bakir,
189
Stepped sine
Stepped
St
d sine
i excitation
it ti is
i a modern
d
version
i off the
th sweptt sine
i
technique that makes maximum use of the developments in DSP
during the last two decades.
I t d off a continuously
Instead
ti
l varying
i ffrequency, stepped
t
d sine
i consists
i t
of a stepwise changing frequency.
It remains a rather slow procedure due to the frequency scan and
wait periods needed for the transients to decay. This can be
overcome by multi-channel acquisition.
The application
pp
of stepped
pp sine excitation requires
q
special
p
soft and
hardware.
The digital processing allows for varying frequency spacing, yielding
data condensation and testing
g time reduction,, and for a better
control against aliasing and leakage problems.
Useful for characterizing non-linearities.
Excellent signal-to-noise ratios
ratios.
P. Gundes Bakir,
190
Processing strong-motion
records
The focus of this section is on the
effects of noise in accelerograms,
g
, and the effects of
correction procedures, on the peak ground-motion amplitudes
and on the ordinates of acceleration and displacement
response spectra.
This low
Thi
l
frequency
f
noise
i usually
ll appears as drifts
d ift in
i the
th
displacements derived from double integration of acceleration,
making it difficult to determine the true peak displacement of the
ground motion
motion.
P. Gundes Bakir,
192
193
194
Analog accelerograms
P. Gundes Bakir,
195
196
197
198
Instrument correction
As noted
A
t d earlier,
li th
the ttransducer
d
ffrequency iin analog
l iinstruments
t
t iis
limited to about 25 Hz, and this results in distortions of amplitudes
and phases of the components of ground motion at frequencies
close to or greater than that of the transducer
transducer.
The digitization process itself can also introduce high-frequency
noise as a result of the random error in the identification of the exact
mid-point
id i t off the
th film
fil ttrace as shown
h
iin th
the fi
figure.
P. Gundes Bakir,
199
Instrument correction
P. Gundes Bakir,
200
Fourier Spectra
The left
left-hand
hand plot in the figure shows an example of the Fourier spectra of
high-frequency ground motion obtained at a very hard rock site in Canada
at a distance of 4 km from the source of a small magnitude earthquake.
Softer sites, even those classified as rock such as class B in the 2003
NEHRP guidelines, will tend to filter out such high frequency motion.
Very high-frequency motions will also tend to attenuate rapidly with
distance and hence will not be observed at stations even a few tens of
kil
kilometers
t
from
f
the
th fault
f lt rupture.
t
P. Gundes Bakir,
201
Fourier Spectra
P. Gundes Bakir,
202
Fourier Spectra
The plot
Th
l t in
i the
th figure
fi
also
l shows
h
the
th ttypical
i l ttransducer
d
response ffor th
the
instrument (SMA-1) on which the record was obtained, and the effect of
applying a correction for the instrument characteristics, which is to
increase slightl
slightly the amplitudes
amplit des at frequencies
freq encies greater than 30 Hz.
H The
nature of such motions, at periods of less than 0.03 s, will only be relevant
to particular engineering problems, such as the response of plant machinery
and nonstructural components
components.
P. Gundes Bakir,
203
Fourier Spectra
The right-hand
right hand plot in the figure show the Fourier spectra of more
typical ground motions obtained at soil sites during a moderate
magnitude earthquake in California. These records were obtained
g
instruments and are lacking
g in very
y high
g frequency
q
y motion
on digital
mainly because of the attenuating effect of the surface geology at
these sites compared to the very hard site in Canada. The plot also
shows the transducer response for these digital instruments, which
is almost flat to beyond 40 Hz
Hz.
P. Gundes Bakir,
204
Unless there are compelling reasons for applying a correction for the
instrument characteristics, we recommend that no attempt should be
made to do so.
so The one exception to this may be the very earliest
recordings obtained in the US with accelerographs that had natural
frequencies of the order of 10 Hz.
P. Gundes Bakir,
205
Baseline adjustments
One approach to
compensating for these
problems is to use baseline
adjustments, whereby one or
more baselines, which may be
g lines or low-order
straight
polynomials, are subtracted
from the acceleration trace.
P. Gundes Bakir,
206
Baseline adjustments
The figure illustrates
the application of a
piece-wise sequential
fitting of baselines to
the velocity trace from
a digital recording in
which there are
clearly identifiable
offsets in the
baseline.
P. Gundes Bakir,
207
Baseline adjustments
A similar procedure could
be applied directly to the
acceleration time-history
t correctt for
to
f the
th type
t
off
baseline shifts shown in
the figure.
The figure shows NS
component of the 21 May
1979 Italian earthquake
(12:36:41 UTC) recorded
at Nocera Umbra
Umbra,
showing shifts in the
baseline at 5.6 and 8.3 s.
P. Gundes Bakir,
208
Baseline adjustments
209
Baseline adjustments
P. Gundes Bakir,
210
Baseline adjustments
Th
The distortion
di t ti off the
th baseline
b
li encountered
t d iin di
digitized
iti d
analog accelerograms is generally interpreted as being
the result of long-period
gp
noise combined with the signal.
g
Baselines can be used as a tool to remove at least part
p
of this noiseand probably some of the signal with it
as a means of recovering more physically plausible
velocities and displacements
displacements. There are many
procedures that can be applied to fit the baselines,
including polynomials of different orders. A point that is
worth
th making
ki clearly
l l iis th
that,
t iin effect,
ff t baseline
b
li
adjustments are low-cut filters of unknown
frequency characteristics.
P. Gundes Bakir,
211
Baseline adjustments
The figure
g
on the left: Shaded line: velocity
y from integration
g
of the eastwest
component of acceleration recorded at TCU129, 1.9 km from the surface
trace of the fault, from the 1999 Chi-Chi earthquake, after removal of the
pre-event mean from the whole record. A least-squares line is fit to the
velocity from 65 s to the end of the record. Various baseline corrections
using the Iwan et al. (1985) scheme are obtained by connecting the
assumed time of zero velocity t1 to the fitted velocity line at time t2. Two
values of t2 are shown: 30, and 70 s.
P. Gundes Bakir,
212
Baseline adjustments
P. Gundes Bakir,
213
Baseline adjustments
The line fit approach is the more complex scheme proposed by Iwan et al.
The method was motivated byy studies of a specific
p
instrument for which the
baseline shifted during strong shaking due to hysteresis; the accumulation
of these baseline shifts led to a velocity trace with a linear trend after
cessation of the strong shaking. The correction procedure approximates the
complex set of baseline shifts with two shifts, one between times of t1 and t2,
and one after time t2. The velocity will oscillate around zero (a physical
constraint), but the scheme requires selection of the times t1 and t2.
P. Gundes Bakir,
214
Baseline adjustments
P. Gundes Bakir,
215
Baseline adjustments
P. Gundes Bakir,
216
Residual displacements
P. Gundes Bakir,
217
Residual displacements
218
Residual displacements
219
Residual displacements
P. Gundes Bakir,
220
It should also be noted that there is little discernable difference between the
filtered and unfiltered accelerations.
P. Gundes Bakir,
221
P. Gundes Bakir,
222
P. Gundes Bakir,
223
224
225
Zero pads
226
Zero pads
227
Zero pads
228
229
The application
pp
of causal and acausal filters,, even with veryy similar filter
parameters (the transfer functions will not be identical if time-domain
filtering is used, since the causal filter will have a value of 1/2 at the filter
corner frequency,
frequency fc, whereas the acausal filter will have a value of 0.5,
05
regardless of the filter order), have been shown to produce very different
results in terms of the integrated displacements
P. Gundes Bakir,
230
Response
p
spectra
p
((damping
p g ratio = 5%)) of the FP component
p
of the
Bolu record of the Duzce earthquake processed by acausal (solid
line) and causal filters (dashed line).
P. Gundes Bakir,
231
232
The influence
Th
i fl
off causall and
d
acausal filters on both elastic
and inelastic response spectra
has been investigated.
investigated
P. Gundes Bakir,
233
Tapers
There are two different ways to avoid this, one being to use tapers
such as a half-cosine function for the transition from the motion to
the zero pad.
A simpler
p p
procedure is to start the p
pad from the first zero crossing
g
within the record, provided that this does not result in the loss of a
significant portion of record, as can happen if the beginning or end
of the acceleration time series is completely above or below zero.
P. Gundes Bakir,
234
As noted previously, the most important issue in processing strongmotion accelerograms is the choice of the long-period cut-off, or
rather the longest response period for which the data are judged to
b reliable
be
li bl iin tterms off signal-to-noise
i
lt
i ratio.
ti A number
b off b
broad
d
criteria can be employed by the analyst to infer the period beyond
which it is desirable to apply the filter cut-off, including:
P. Gundes Bakir,
235
Fourier amplitude
p
spectrum
p
of the FN horizontal
component of the Bolu record
of Duzce earthquake (thick
solid line) and the noise
spectrum (thick dashed line).
Superimposed on this graph
are the functions f2 (dotted
line), f (dotted line), and f2
(d h d d d line).
(dasheddotted
li )
236
P. Gundes Bakir,
237
238
Strong--motion processing
Strong
239
240