1

Operations on Time Series in Time

domain and Frequency Spectrum

Holger Waubke

Acoustics Research Institute
Austrian Academy of Sciences
Wohllebengasse 12-14
A-1040 Vienna
holger.waubke@oeaw.ac.at
2
Contents

1 Introduction ........................................................................................................................ 4
1.1 The Generalized Dirac Delta Function ....................................................................... 4
1.2 The Generalized Heaviside Function ......................................................................... 5
1.3 Sine and Cosine Functions ......................................................................................... 6
1.4 Sine Cardinalis and Sine Integralis Functions ............................................................ 6
1.5 The Fourier Transformation ....................................................................................... 7
1.6 Modified Plancherels Theorem ................................................................................. 7
1.7 Filter or the Convolution in Time Domain ................................................................. 8
1.8 Convolution in Frequency Domain and the Determination of RMS Values ............. 8
1.9 The Laplace Transformation ...................................................................................... 9
1.10 The z-Transformation ............................................................................................... 10
1.11 Energetic Sum in the Spectral Domain .................................................................... 11
1.12 Aliasing .................................................................................................................... 12
1.13 Remarks .................................................................................................................... 12
2 Gibbs' Phenomenon .......................................................................................................... 13
2.1 Gibbs' Phenomenon in Time Domain ...................................................................... 13
2.2 Gibbs' Phenomenon in Frequency Domain .............................................................. 16
2.3 Conclusions .............................................................................................................. 18
3 Filter and Filter Banks ...................................................................................................... 19
3.1 Kaiser Bessel Filter .................................................................................................. 19
3.2 Tschebycheff Approximation and Remez Algorithm .............................................. 20
3.3 Filter Banks .............................................................................................................. 21
3.4 Butterworth Filter ..................................................................................................... 21
3.5 Bessel Filter .............................................................................................................. 23
3.6 Conclusions .............................................................................................................. 24
4 Manipulation of Discrete Time Series and Spectra and Definition of Blocks ................. 25
4.1 Up Sampling of Time Series .................................................................................... 25
4.2 Extension of the Time Basis ..................................................................................... 26
4.3 Averaging of Time Blocks ....................................................................................... 27
4.4 Conclusions .............................................................................................................. 27
5 Hilbert Transformation, Demodulation und Envelope ..................................................... 28
5.1 Hilbert Transformation in Spectral Domain ............................................................. 28
5.2 Hilbert Transformation in Time Domain ................................................................. 29
5.3 Hilbert Transformation for Time Series ................................................................... 29
5.4 Estimation of the Envelope ...................................................................................... 30
5.5 Conclusions .............................................................................................................. 30
6 Differentiation und Integration in Time Domain ............................................................. 31
6.1 Differentiation in Time Domain ............................................................................... 31
6.2 Integration in Time Domain ..................................................................................... 31
6.3 Differentiation with Polynomial Interpolation ......................................................... 32
6.4 Integration with Polynomial Interpolation ............................................................... 32
6.5 Savitzky Golay Filter for Smoothing and Smoothed Differentiation ...................... 34
6.6 Savitzky Golay Filter for Smoothed Integration ...................................................... 35
6.7 Differentiation with Trigonometric Interpolation .................................................... 35
6.8 Integration with Trigonometric Interpolation .......................................................... 37
6.9 Conclusions .............................................................................................................. 40

3
7 Spectral Differentiation and Integration ........................................................................... 41
7.1 Leakage .................................................................................................................... 41
7.2 Investigations with a Tapered Window .................................................................... 44
7.3 Differentiation with and Without Tapered Window ................................................ 49
7.4 Integration with and Without Tapered Window ...................................................... 54
7.5 Correction of the Spectral Integration ...................................................................... 61
7.5.1 Procedure in Time Domain ................................................................................ 63
7.5.2 Procedure in Frequency Domain ........................................................................ 64
7.6 Conclusions .............................................................................................................. 67
8 Correction of Integrating A/D Converters ....................................................................... 69
8.1 Correction of Integrating A/D Converters in Frequency Domain ............................ 69
8.2 Correction of Integrating A/D Converters in Time Domain .................................... 70
8.3 Conclusions .............................................................................................................. 73
9 Exponential Smoothing .................................................................................................... 74
9.1 Continuous Smoothing ............................................................................................. 74
9.2 Discrete Representation of Exponential Smoothing Functions ................................ 75
9.2.1 Discrete Approximation Based on the Assumption of Blocks Scaled With End
Values 75
9.2.2 Discrete Approximation using Blocks Scaled with Average Value .................. 76
9.2.3 Assumption of a Linear Interpolation of the Discrete Time Series .................... 77
9.2.4 Stability of the Filter .......................................................................................... 78
9.2.5 Coefficients of the Classical Filter ..................................................................... 81
9.2.6 Coefficients of the Filter using Blocks Scaled with Average Value .................. 82
9.2.7 Coefficients of the Filter with Linear Interpolation ........................................... 83
9.2.8 Pass-By Level ..................................................................................................... 84
9.3 Exponential Smoothing With a Continuous Fourier Transformation ...................... 85
9.3.1 Fourier Transformation ...................................................................................... 85
9.3.2 Plancherel's Theorem ......................................................................................... 85
9.3.3 Exponential Smoothing of the Continuous Spectrum ........................................ 86
9.3.4 Remarks on discrete Fourier transformation (DFT) ........................................... 90
9.3.5 Simulation von Noise Immissions ...................................................................... 90
9.3.6 Simulation Based on Discrete Fourier Transformation ...................................... 91
9.4 Conclusions .............................................................................................................. 92
10 The Stationary Ergodic Process and Applications in Measurement Techniques ............. 94
10.1 Estimation of the Transfer-Function of a Linear System ......................................... 94
10.2 Estimation of the Propagation Time in a Linear System ......................................... 95
11 References and Indices ..................................................................................................... 97
11.1 References ................................................................................................................ 97
11.2 Figures ...................................................................................................................... 97
11.3 Tables ....................................................................................................................... 98

.
4
1 Introduction
How to manipulate measurement data is a typical problem in Engineering. Today sound
pressure level and acceleration are the typical data from a measurement. In the calculation
velocity, displacement, stress and tension of the structure are needed.
Another problem that arises is, how I can combine sampled data with third-octave or octave
spectra or the overall level to estimate the immission at a specific location. This document
supposes that the Fourier Integral Transformation is used to switch from time to frequency
domain in the calculation based on Finite Difference, Finite Element, Boundary Element or
transformation methods. As long as linear systems are investigated the Fourier Integral
Transformation decouples the problem and every frequency can be calculated independently of
the other frequencies. The inverse transformation allows as an alternative to define a filter in
the time domain. Measured time sequences can be filtered with the derived filter. In praxis
difficulties can arise at very low frequencies. Are only RMS values given for a noise signal,
this method is almost impossible. In the ongoing chapters it is assumed that a constant noise
level is given about the width of the filter used to determine the spectrum.
The aim of this paper is to investigate operations as interpolation, integration and
differentiation, square operation and exponential smoothing in time and frequency domain. For
this operations errors resulting from discretisation are estimated and measures to reduce the
errors are explained.
As an introduction the generalized Dirac delta function is defined. With this function in
combination with the Fourier transformation Plancherels theorem is proved.
1.1 The Generalized Dirac Delta Function
The generalized Dirac Delta function is defined by the measure the integral about the time.
( )
( ) ( )
} }

= =

= =

=
= +
=
t t
dt t dt t
t
t
t
1 lim
0 , 0
0 ,
o o
o
t
t
t
(1)
Apart from the origin (t=0) the function is always zero. Therefore all deviations are zero with
the exception of t=0.
( ) 0 , 0 = =
c
c
t t
t
n
n
o (2)
Also derivatives of the generalized Dirac delta functions can be defined.
( )
( )
( ) t t
t
n
n
n
o o =
c
c
(3)
For a coefficient { } 0 / R a e and all functions f, that have a limited value at the time instance
t=t0:
( ) ( ) ( )
( ) ( ) ( )
( )
( )

<
|
.
|

\
|
+
>
|
.
|

\
|
+
=
= +
}
}
}
}

=

=
0 ,
0 ,
0
0
0
0 0
a
a
d
t
a
f
a
a
d
t
a
f
dt t t a t f
t f dt t t t f
t
t
t
t
t
t o
t
t
t o
t
o
o
(4)
5
( ) ( )
( ) ( ) ( ) ( ) ( ) 0 ,
1
sgn , ,
0 0 0
0
= =
|
.
|

\
|
+ =
=
c
c
=
} }

=

=
a t f
a a
d
t
a
f dt t t a t f
a t a
t
t t a
t t
t
t o
t
o
t
t
t
(5)
A special function is the Fourier transformation of the generalized Dirac delta function:
( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) ( )
t f j
f
ft j
f
ft j
ft j
t
ft j
t
ft j
e
b
df e f f b df e f Z t z f f b f Z
e
a
dt e t t a dt e t y f Y t t a t y
0
0
2 2
0
2
0
2 2
0
2
0
1
1
t t t
t t t
o o
o o
= = = =
= = = =
} }
} }

(6)
One application of the generalized Dirac delta function is the integration about discrete
samples:
( )
( ) ( ) ( )
( )
t
f e x
f
e x t f X
dt e t k t t x dt e k
t
t
x dt e t x f X
k
t
t
x t x
s
k
k
f
f
j
k
s k
t fk j
k
k
t
ft j
k
t
ft j
k
k
t
ft j
k
k
s
A
= = A =
A A =
|
.
|

\
|

A
== =
|
.
|

\
|

A
=

} }

}

=
A

=
1
,
1
2
2
2 2 2
t
t
t t t
o o
o
(7)
Analogous for a samples spectrum one can derive:
( )
( ) ( )
( )
f
T e x
T
e X f f X
df e k
f
f
X df e k
f
f
X df e f X t x
k
f
f
X f X
k
T
t
k j
k
k
t f k j
k
k
f
ft j
k
f
ft j
k
k
f
ft j
k
k
A
= = A =
|
|
.
|

\
|

A
=
|
|
.
|

\
|

A
== =
|
|
.
|

\
|

A
=

} }

}

=
A

=
1
,
1
2
2
2 2 2
t
t
t t t
o o
o
(8)
1.2 The Generalized Heaviside Function
The integration about the generalized Dirac delta function leads to the generalized Heaviside
function H(t). The selected version presented here is completed with the value at the position
of the jump (t=0).
( ) ( )

>
=
<
= =
}
=
0 , 1
0 , 2 / 1
0 , 0
t
t
t
ds s t H
t
s
o (9)
6
1.3 Sine and Cosine Functions
The sine and cosine functions are defined by complex exponential functions.
( ) ( )
( ) ( )
( )
( )
( )
( )
( ) ( )
( ) ( )
( )
( ) 1 0 cos
0 0 sin
cos cos
sin sin
2
cos
2
sin
cos 2
sin 2
sin cos
1 , sin cos
2
=
=
=
=
+
=

=
= +
=
=
= + =

ax ax
ax ax
e e
ax
j
e e
ax
ax e e
ax j e e
ax j ax e
j ax j ax e
jax jax
jax jax
jax jax
jax jax
jax
jax
(10)
( ) ( )
( ) ( ) ax a
j
e e
a
jae jae
ax
x
ax a
e e
a
j
jae jae
ax
x
jae e
x
jax jax jax jax
jax jax jax jax
jax jax
sin
2 2
cos
cos
2 2
sin
=

=
c
c
=
+
=
+
=
c
c
=
c
c


(11)
1.4 Sine Cardinalis and Sine Integralis Functions
The Sine Cardinalis function [1] is defined as presented here. The Sine Integralis function is
the integral about the Sine Cardinalis function. Because the Sine Cardinalis function is
bounded in the whole domain the integral exists in the classical sense.
( )
( )
( )
( ) ( ) ( )
( )
( )
( )
( )
( ) ( )
( )
( )
( )
( )
( )
0
sin
0
sin
1
sin sin
1 ,
sin
sinc
sin sin sin
sinc
sin
sinc
0
0
0 0 0
0
= =
= =

= =
=
c
c
=
=
= =

=
=
}
} } }
}
=
= =

=
=
ds
s
s
Si
t Si du
u
u
du
u
u
ds
s
s
t Si
s
u
s u
ds
s
s
t Si
t
t
t
t
t
t
t
t
t
t
t
s
t
u
t
u
t
s
t
s
(12)
7
1.5 The Fourier Transformation
The Fourier Transformation [2][3][8] is defined as stated here:
( ) ( )
( ) ( )
( ) ( ) ( )
( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( )
} } } }
} } } }
}
}

= = = =
= =
=
=
t t t
t t
t
t t t t
t
t t
t
t
t t o t t t t to t t t
t t t t
t x d t x d t x d df e x t x
d df e e x df e d e x t x
df e f X t x
dt e t x f X
f
t f j
ft j
f
f j
f
ft j f j
f
ft j
t
ft j
2 2
2
2 2 2 2
2
2
(13)
In the last equation the following formerly derived relation is used:
( ) ( ) ( ) ( )
0
2
tion Transforma Fourier 0
1
ft j
e
a
f X t t a t x
t
o

= = (14)
The following abbreviated version is partially used in the ongoing document:
( ) ( )
( ) ( )
}
}

=
=
e
e
e
e e
t
d e X t x
dt e t x f X
t j
t
t j
2
1
(15)
In the literature the factor 1/(2t) is posed at different positions.
- Using the factor 1/(2t) at the forward transformation has the effect that the spectrum has the
same scale as in the long version. The complex amplitudes are therefore related to the
frequency axis f.
- The symmetric usage of the square root of the factor 1/(2t) has the effect that no factor in
Plancherels theorem is needed. This means, that the energy is correctly represented in the
spectrum. Therefore this representation is used for the representation of stochastic signals.
- Using the factor 1/(2t) at the inverse transformation results in a spectrum that is defined about
the angular frequency e.
1.6 Modified Plancherels Theorem
Using the generalized Dirac delta function Plancherels theorem can be simply derived. At this
point a modified version of the theorem is derived, that does not use conjugate complex
functions. The result is a coupling of the positive and negative frequency axis.
( ) ( ) ( ) ( )
( ) ( )
( )
( ) ( )
( ) ( ) ( ) ( )
} }
} } }
} } } }

=
+

=
+

=
+ =
=
= =
f
f
f
t
t f j
t
t j
f
ft j
t
df d f B f A E
df d dt e B f A E
dt d e B df e f A dt t b t a E
,
,
, t to
, t
,
t, t
, , t to ,
, ,
, ,
2 2
2 2
2
2 2

(16)
8
( ) ( ) ( )
( ) ( ) ( ) ( )
} }
} }

=
= =
+ =
f t
f
df f B f A dt t b t a E
df d f B f A E
,
, , o ,
(17)
1.7 Filter or the Convolution in Time Domain
A filter is defined as the manipulation of the frequency spectrum of an arbitrary function with
a filter spectrum. The result can be transformed into time domain using the inverse Fourier
transformation. The analogous operation in time domain is called a convolution.
( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( )
ds dr df e r h s x t y
df e dr e r h ds e s x t y
df e f H f X df e f Y t y
f H f X f Y
f
s r t f j
r s
f
ft j fr j
r
fs j
s
f
ft j
f
ft j
} } }
} } }
} }

=
=
=
= =
=
t
t t t
t t
2
2 2 2
2 2
(18)
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ds s t h s x t y
ds dr s r t r h s x t y
ds dr s t r h s x t y
s
r s
r s
=
=
=
}
} }
} }

=
o
t to 2 2
(19)
1.8 Convolution in Frequency Domain and the Determination of
RMS Values
Analogously to the convolution in time domain a convolution in frequency domain can be
defined.
( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) da db b a f b H a X f Y
da db b a f b H a X f Y
da db dt e b H a X f Y
dt e db e b H da e a X f Y
dt e t h t x dt e t y f Y
t h t x t y
b a
b a
t
t b a f j
b a
t
ft j fb j
b
fa j
a
t
ft j
t
ft j
=
=
=
=
= =
=
} }
} }
} } }
} } }
} }

o
t to
t
t t t
t t
2 2
2
2 2 2
2 2
(20)
9
( ) ( ) ( ) da f a H a X f Y
a
=
}

=
(21)
This relation can be used to determine the spectrum of a squared signal.
( ) ( ) ( ) ( ) ( ) da f a X a X f Y t x t y
a
= =
}

=
2
(22)
If the root mean square (RMS value) of a time block T has to be determined the following
operations are needed in the spectral domain:
( ) ( )
( )

( ) | |
( )
( ) ( )
( ) ( ) ( )
( ) ( ) ( ) | |
( )
( ) ( )
( ) fT
fT
fT
fT j
fT j
fT j
e e
f j
e
T
f Z
dt e
T
dt e T t H T t H
T
dt e t z f Z
da f a X a X f Y
df f Z f Y dt T t H T t H
T
t x dt t x
T
x
fT j fT j
T
T t
ft j
T
T t
ft j
t
ft j
t
ft j
a
f t
t z
t y
T
T t
RMS
t
t
t
t
t
t t
t t t
t t t
sinc
sin
2
sin 2
2 2
1
1
) 2 / ( 2 /
1
) 2 / ( 2 /
1 1
2 /
2 /
2
2 /
2 /
2 2 2
2
2 /
2 /
2 2
= =

=
(

=
= + = =
=
= + = =

= =
} } }
}
} } }

(23)
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
2
2 /
2 /
2 2
2 /
2 /
2 2
sinc
1
sinc
1
RMS RMS
f a
T
T t
RMS
f a
T
T t
RMS
x x
df fT da f a X a X dt t x
T
x
df fT da f a X a X dt t x
T
x
=
= =
= =
} } }
} } }

= =

= =
t
t
(24)
So far the RMS value can be determined by a convolution and a filtering with a Sine
Cardinalis function in the spectrum.
1.9 The Laplace Transformation
A predecessor of the Fourier transformation with a complex valued angular frequency is the
Laplace transformation. In this transformation it assumed that the signal starts at the time
instance t=0. The inverse transformation is executed along the imaginary axis. The axis for the
inverse transformation is shifted by a value .
( ) ( )
( ) ( )
}
}
+
=

=
=
j
j s
st
t
st
ds e s F
j
t f
dt e t f s F

t 2
1
0
(25)
Using the generalized Heaviside function H( ) the Laplace transformation can be transformed
into a Fourier transformation.
( ) ( )
( ) ( ) ( )
( ) ( ) ( ) ( )
} }
}

= =
=
=
t
t j
t
t j
t
st
dt e t f t H dt e t g G
t f t H t g
dt e t f s F
e e
e
0
(26)
( ) ( ) e j G s F = (27)
10
1.10 The z-Transformation
For discrete time series the Laplace transformation was further developed into the
z-Transformation.
( )
( )
}

=
=
C
n
n
n
n
n
dz z z X
j
x
z x z X
1
2
1
t
(28)
The inverse integration is a circular integration in the complex plane. Therefore all poles have
to be inside of the unit circle. If all poles of a filter are inside the unit circle the filter is stable.
If all zeros are inside the circle the filter has minimum phase and is invertible. The frequency
axis of the Fourier Transformation is mapped to the unit circle. One turn around on the circle is
related to the section [0 , fs] of the frequency axis. In this relation fs is the sampling rate of the
discrete time series.
( )
( ) ( ) ( )
|
|
.
|

\
|
+
|
|
.
|

\
|
= = =
= = =
=
A

=
A

s s
f
f
j
t f j
n
n
t f j
n
n
n
n
n
t fn j
n
f
f
j
f
f
e e z
e x f X z x z X
e x f X
s
t t
t
t
t
t
2 sin 2 cos

2
2
2
2
(29)
Therefore, the z-Transformation can be transformed to the Fourier or Laplace transformation.
The following relation is valid for the Laplace transformation.
( )
( )
( )
( )
. . .
ln
1
2
2
ln
1
1 1
1 1
ln
1
1
1
1
1
1
1
ln
1
1
1
arctanh
2
1 ,
1
1
ln
2
1
1
1
ln arctanh
: Bronstein using Proof
1
1
arctanh
2
1
1
arctanh
2
Ln
1
Ln
1
1
d e q
z
t z t z z
z z
t
z
z
z
z
t z
z
t
s
x
x
x
x
x
x
z
z
t z
z
t
z
t
s
z t s
e z
t s
A
=
|
.
|

\
|
A

=
|
.
|

\
|
+ +
+ +
A

=
|
|
|
|
.
|

\
|
+

+
A

=
|
.
|

\
|
+

=
<
|
.
|

\
|

+
=
|
|
.
|

\
|

+
=
|
|
.
|

\
|
+

A
=
|
.
|

\
|
+

=
A
=
= A
=

A
(30)
This exact relation is difficult to use. Therefore an approximation is used that is valid for small
arguments:
11
( )
( )
|
|
.
|

\
|
A
A +
= =
A
A +
=
|
|
.
|

\
|
+

A
= =
+

=
+

A
~

2 1
2 1
2 1
2 1
1
1 2
1
1
2
1
1 2
1
1
1
1
1
1
t s
t s
z H s H
t s
t s
z
z
z
t
s H z H
z
z
f
z
z
t
s
s
(31)
In this relation At is the sampling interval and the inverse fs the sampling rate. A relation of the
z-transformation with the Fourier transformation can be derived similarly. The approximation
that was sued before leads to a deviation, which can be corrected using the analogue frequency
ea or fa and the digital frequency ed or fd.
|
|
.
|

\
|
=
|
.
|

\
| A
A
=
|
.
|

\
| A
A
= =
|
.
|

\
| A
A
=
+

A
=
+

A
=
+

A
= =
=
A A
A A
A
A

A
s
d s
a
d
a
d
a a
d
t j t j
t j t j
t j
t j
a a
t j
f
f f
f
t
t
t
t
j
j s
t j
t
e e
e e
t e
e
t z
z
t
j s
e z
d d
d d
d
d
d
t
t
e
e
e
e
e
e
e e
e e
e
e
e
tan
2
tan
2
2
tan
2
2
tanh
2 2
1
1 2
1
1 2
2 2
2 2
1
1
(32)
1.11 Energetic Sum in the Spectral Domain
In the ongoing document the energetic averaging over short time spectra is needed. If the source
level pe is constant in the interval [fa , fb], the level pi at the receiver position has the following
relation using the transfer function H:
( ) ( ) ( ) ( ) ( )( )
( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( )( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( )
(
(

= = =
= = =
= = = =
=
=
+ = =
+ = =
= + + = + =
} } }
} } }
} } }
b
a
b
a
b
a
b
a
b
a
b
a
b
a
b
a
b
a
f
f a b
e
f
f
e
f
f
e i
f
f
e e
f
f
i i
f
f
i i
a b e
f
f
e
f
f
f
f
e e e e
eI eR I R iI iR
eI eR I eI eR R iI iR
eR I eI R eI I eR R iI iR e i
eR I eI R iI eI I eR R iR
eI eR I R iI iR e i
df f H f H
f f
p df f H f H p df p f H f H p
df f p f H f p f H df f p f p df f p p
f f p df p df f p f p df f p p
jp p jH H jp p
jp p jH jp p H jp p
p H p H j p H p H jp p f p f H f p
p H p H p p H p H p
j jp p jH H jp p f p f H f p
* 2 *
2 2
* 2
* * *
2
2
2 2
*
2
2
* * *
2
1

,
,
1 , ,
(33)
It is therefore possible to use the average of the squared amplitude of the transfer function to
determine the receiver level, if the source level is constant in the chosen interval.
12
1.12 Aliasing
As a consequence of the digitisation with a constant sampling interval, high frequency
components are mirrored at the aliasing frequency. The aliasing frequency is half of the sample
rate. To avoid aliasing an analogue low pass filter is needed before the digitization is applied.
These filters are named anti-aliasing filters.
In the ongoing document it is assumed that sampling is done with a constant sampling interval.
Small unwanted deviations of the interval length produce a jitter in the spectrum. If random
sampling intervals are used aliasing becomes a statistical effect.
1.13 Remarks
The integration is often done with infinite limits. For these integrals the following abbreviation
is used.
( ) ( )
} }

= =

=
t t
dt t f dt t f
t
t
t
lim (34)
It is assumed that the integrals are uniform convergent, because we deals with real world
measurement signals. Using this condition the sequence of the integrals can be changed.
13
2 Gibbs' Phenomenon
Gibbs Phenomenon points out, that a band limited signal cannot have jumps in the time signal.
Vice versa signal limited in time cannot have jumps in the spectrum. Gibbs Phenomenon is the
basis for the definition of weighting windows in time and it is the basis for the filter design.
2.1 Gibbs' Phenomenon in Time Domain
For the derivation of a time limited signal with two jumps and the ends is defined.
( ) ( ) ( )
( ) ( )
( )
( ) ( )
0 0
0
0
0
0
0
2 /
2 /
2
2 /
2 /
2 2
0 0
sinc
sin
2 2
2 / 2 /
0 0
0
0
0
0
ft t f X
ft
ft
t
ft
e e jt
f j
e
dt e dt e t x f X
t t H t t H t x
ft j ft j
t
t t
ft j
t
t t
ft j
t
ft j
t
t
t
t t
t t t
t t
=
=

=
(

= = =
+ =

} }
(35)
Additionally the signal is band limited in the interval [-fs/2,fs/2]. This leads to the new function:
( ) ( ) ( ) ( ) | |
( ) ( ) ( )
( )
( )
( ) ( )
( )
( ) ( )
( )
( ) ( )
( )
( )
( )
( )
( )
} }
} }
} }
} }
} } }
= =

= =

= =

= =
= =

=
= + =
+ =
+

=
=
+ =
= =
+ =
2 /
0 0
0
0
2 /
0
2 2
0
0
0
2 /
0
2
0
0
0
2 /
0
2
0
0
0
2 /
0
2
0
0
0
0
2 /
2
0
0
0
2 /
0
2
0
0
0
0
2 /
2
0
0
0
2 /
2 /
2
0
0
0
2 /
2 /
2 2
2 cos 2
sin sin

sin sin

sin sin

sin sin

sin

2 / 2 /

s s
s s
s
s
s
s
s
s
s
s
f
f
f
f
ft j t g j
f
f
ft j
f
g
t g j
f
f
ft j
f g
t g j
f
f
ft j
f f
ft j
f
f f
ft j
f
f f
ft j
f
ft j
s s
df ft
ft
ft
t df e e
ft
ft
t t x
df e
ft
ft
t g d e
t g
t g
t t x
df e
ft
ft
t g d e
t g
t g
t t x
f g
df e
ft
ft
t df e
ft
ft
t t x
df e
ft
ft
t df e f X df e f X t x
f f H f f H f X f X
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t t
t t
t t
t t
t t t
(36)
The product of sine and cosine functions is converted in the following way:
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
( )
( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) | o | o | o
| o
| o | o | o | o
| o | o | o | o | | o o
+ + = + =
+ = + =
+ +
+ +
sin sin
2
1
4
1
cos sin
4
1
2
1
2
1
cos sin
j j j j
j j j j j j j j
e e e e
j
e e e e
j
e e e e
j
(37)
The result is
( )
( ) ( ) ( ) ( ) ( ) ( )
( ) ( )
( ) ( )
( )
( )
( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
}
}
} }
=
=
= =
+ + + =
(

+
+
+
+ =
+ +
= =
2 /
0
0 0 0 0
2 /
0 0
0
0
0
0
0
2 /
0
0 0
2 /
0
0
2 c sin 2 2 c sin 2
2
2 sin
2
2
2 sin
2
2
2 sin 2 sin
2
2 cos sin
2
s
s
s s
f
f
f
f
f
f
f
f
df t t f t t t t f t t t x
df
t t f
t t f
t t
t t f
t t f
t t t x
df
f
t t f t t f
df
f
ft ft
t x
t t
t
t
t
t
t
t t
t
t t
(38)
As the next step a substitution using the following algorithm is used:
( ) ( ) ( ) | | ( ) ( ) | |
} }
= =
=
= = = = =
b
a
b
a
b
a
t
t t
ct
ct s
a b
ct
ct s
c
dt
ds
ct s ct F ct F
c
s F
c c
s d
s f t d ct f , ,
1 1
(39)
14
This gives the following term:
( )
( ) ( ) ( )
( )
( ) ( ) ( )
( )
( )
( ) ( ) ( ) ( )
2 /
0
0 0
2 /
0
0
0 0
0
0 0
2 Si 2 Si

2
2 Si 2
2
2 Si 2

s
s
f
f
f
f
t t f t t f
t x
t t
t t f t t
t t
t t f t t
t x
=
=
(

+
+
=
(

+
+
+ +
=
t
t
t
t
t
t
t
t
(40)
( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) | | t t f t t f t x
t t f t t f
t x
s s
s s
+ + =

+
+
=
2 / Si 2 / Si
1

0 Si
2
2 / Si 2 / Si

0 0
0 0
t t
t
t t
t
t
t
(41)
The Sine Integralis function is needed. This function has the following features:
( )
( )
( ) ( )
( )
( )
2
Si lim
0 0 Si
Si Si
sin
Si
t
0
t
=
=
=
=

=
}
t
t t
ds
s
s
t
t
s
(42)

Figure 1: Sine Integralis function Si(t fst)over fst
15

Figure 2: The sine Cardinalis function sinc(t fst)over fst

The maximum of the Sine Integralis function is needed. To derive the extrema the zeros of the
sine Cardinalis function are investigated.
( ) { } 0 / , , 0 x sinc Z n n x e = = t (43)
The sign of the sine Cardinalis changes at the zeros. The areas of the function from one zero to
the next becomes smaller with increasing order of n. Therefore the extremum of the integral is
given in the interval [-t,t]. In the given case the lower limit is fixed at the position zero.
Therefore the integration from [0,t] or [-t,0] gives the extreme values of the function.
The extrema are located at the positions:
( ) ( ) ( )
( )
2
1
2 /
2 / max
0
0
0
t
f
t
t t f
Si t t f Si
s
M
M s
M s
=
=
=
t t
t t
(44)
( ) ( ) ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | | t t
t
t t
t
t t
t
t t
t
t t
t
Si 1 Si
1
Si 1 Si
1

Si 1 Si
1
Si 1 Si
1

2 / Si 2 / Si
1

2
1
0 0 2 ,
0 0 1 ,
0 0
0
4 , 3 , 2 , 1 ,
+ = + =
+ = + + =
+ + =
=
t f t f t x
t f t f t x
t t f t t f t x
t
f
t
s s M
s s M
M s M s M
s
M
(45)
16
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | | t t
t
t t
t
t t
t
t t
t
Si 1 Si
1
1 Si Si
1

Si 1 Si
1
1 Si Si
1

0 0 4 ,
0 0 3 ,
+ = + + =
+ = + =
t f t f t x
t f t f t x
s s M
s s M
(46)
Overall four extrema are given with a distance of
s
f / t before and after the position of the
jumps at 2 /
0
t . The position of the extrema depends on the sampling rate. The value of the
extrema is almost independent from the sampling rate. Only the superposition of the extrema
of the two jumps change the value of the extrema. If the distance of the jumps is large the
following approximations are possible:
( ) ( )
( )
t
t
t
t
t
Si
2
1
Si
2
1

1
4 , 3 , 2 , 1 ,
0
=
(

=
>>
M
s
t x
t f
(47)
The maximum of the ripple has about 9% of the height of the jump.
( )
22 0.08948987
2
1 Si
max
~ = A
t
t
x (48)

Figure 3: Gibbs' Phenomenon for t0 = 1 and fs = 2, 5, 20 and original function
2.2 Gibbs' Phenomenon in Frequency Domain
A rectangular bounded frequency spectrum is defined for a band limited signal:
( ) ( ) ( )
( ) ( )
( )
( ) ( ) t t f f x
t f
t f
f
t f
e e jf
t j
e
df e df e f X t x
f f H f f H f X
t f j t f j
f
f f
ft j
f
f f
ft j
f
ft j
0 0
0
0
0
0
0
2 /
2 /
2
2 /
2 /
2 2
0 0
sinc
sin
2 2
2 / 2 /
0 0
0
0
0
0
t
t
t
t t
t t t
t t
=
=

=
(

= = =
+ =

= =

=
} }
(49)
17
Additionally the signal is limited in time using the interval [-T/2,T/2]. The new signal is under
this condition:
( ) ( ) ( ) ( ) | |
( ) ( ) ( )
( )
( )
( ) ( )
( )
( ) ( )
( )
( ) ( )
( )
( )
( )
( )
( )
} }
} }
} }
} }
} } }
= =

=
=

=
=

= + =
+ =
+

=
=
+ =
= =
+ =
2 /
0 0
0
0
2 /
0
2 2
0
0
0
2 /
0
2
0
0
0
2 /
0
2
0
0
0
2 /
0
2
0
0
0
0
2 /
2
0
0
0
2 /
0
2
0
0
0
0
2 /
2
0
0
0
2 /
2 /
2
0
0
0
2 /
2 /
2 2
2 cos 2
sin sin

sin sin

sin sin

sin sin

sin

2 / 2 /
T
t
T
t
ft j t f j
T
t
ft j
T
s
t s j
T
t
ft j
T s
t s j
T
t
ft j
T t
ft j
T
T t
ft j
T
T t
ft j
t
ft j
t d ft
t f
t f
f t d e e
t f
t f
f f X
t d e
t f
t f
f s d e
s f
s f
f f X
t d e
t f
t f
f s d e
s f
s f
f f X
t s
t d e
t f
t f
f t d e
t f
t f
f f X
t d e
t f
t f
f dt e t x dt e t x f X
T t H T t H t x t x
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t t
t t
t t
t t
t t t
(50)
The product of sine and cosine function is converted as before:
( )
( )
( )
( ) ( )
( )
( ) ( ) ( ) ( )
( ) ( )
( ) ( )
( )
( )
( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
}
}
}
} }
=
=
=
= =
+ + + =
(

+
+
+
+ =
+ +
=
= =
2 /
0
0 0 0 0
2 /
0 0
0
0
0
0
0
2 /
0
0 0
2 /
0
0
2 /
0 0
0
0
2 c sin 2 2 c sin 2

2
2 sin
2
2
2 sin
2

2
2 sin 2 sin
2

2 cos sin
2 2 cos
sin
2

T
t
T
t
T
t
T
t
T
t
t d t f f f f t f f f f f X
t d
f f t
t f f
f f
f f t
t f f
f f f X
t d
t
t f f t f f
f X
t d
t
ft t f
t d ft
t f
t f
f f X
t t
t
t
t
t
t
t t
t
t t
t
t
t
(51)
It follows that:
( ) ( ) ( ) ( ) ( ) ( ) | |
( )
( ) ( ) ( )
( )
( ) ( ) ( )
( )
( )
( ) ( ) ( ) ( )
2 /
0
0 0
2 /
0
0
0 0
0
0 0
2 /
0
0 0 0 0
2 Si 2 Si

2
2 Si 2
2
2 Si 2

2 c sin 2 2 c sin 2
T
t
T
t
T
t
t f f t f f
f X
f f
t f f f f
f f
t f f f f
f X
t d t f f f f t f f f f
=
=
=
(

+
+
=
(

+
+
+ +
=
+ + +
}
t
t
t
t
t
t
t
t
t t
(52)
( )
( ) ( ) ( ) ( ) ( )
( )
( ) ( ) ( ) ( )
t
t
t
t
t t
t
t
t
T f f T f f
f X
T f f T f f
f X
2 2 / Si 2 / Si

0 Si
2
2 2 / Si 2 / Si

0 0
0 0

+
+
=

+
+
=
(53)
The extrema are now at the positions:
( ) ( ) ( )
( )
2
1
2 /
2 / max
0
0
0
f
T
f
T f f
Si T f f Si
M
M
M
=
=
=
t t
t t
(54)
18
( ) ( ) ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | | t t
t
t t
t
t t
t
t t
t
t t
t
Si 1 Si
1
Si 1 Si
1

Si 1 Si
1
Si 1 Si
1

2 / Si 2 / Si
1

2
1
0 0 2 ,
0 0 1 ,
0 0
0
4 , 3 , 2 , 1 ,
+ = + =
+ = + + =
+ + =
=
T f T f f X
T f T f f X
T f f T f f f X
f
T
f
M
M
M M M
s
M
(55)
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | | ( ) ( ) ( ) | | t t
t
t t
t
t t
t
t t
t
Si 1 Si
1
1 Si Si
1

Si 1 Si
1
1 Si Si
1

0 0 4 ,
0 0 3 ,
+ = + + =
+ = + =
T f T f f X
T f T f f X
M
M
(56)
Four extrema exist in the distance T / t before and after the jump positions at 2 /
0
f . The
position of the extrema depends on the window length T. The value of the extrema is widely
independent from the window length T. Only the superposition of the two jumps leads to a
variation of the ripple. If the distance of the two jumps is large enough the following
approximation can be done:

( ) ( )
( )
t
t
t
t
t
Si
2
1
Si
2
1

1
4 , 3 , 2 , 1 ,
0
=
(

=
>>
M
f X
T f
(57)
The ripple is therefore:
( )
22 0.08948987
2
1 Si
max
~ = A
t
t
X (58)
The overshot is about 9% of the height of the jump. This is related to a quality of the filter of
( ) dB 4 20.9645222 log 20
max 10
~ A = X Q (59)
The filter quality is just 21 dB independent of the filter length.
2.3 Conclusions
The relations that were presented for the time domain can analogously be applied to the
frequency domain. To reduce the over shots that are also called ripple two possibilities are
given. A smooth transition of the amplitude and phase shifts can be applied to the filter. This is
the basis for the filter design. The filter design defines the maximum ripple in the pass band
region and the maximum ripple in the stop band regions as to optimization criteria. The ripple
in the stop band defines the quality of the filter. Another design criterion is the width of the
transition region between pass and stop band. Phase shifts of the filter also named group
velocity is another design variable. This parameter is especially important for impulsive signals
(transients). The filter length and the filter retardation are additional design variables.
In the ongoing document the focus is put on non-recursive filters. Using symmetric FIR filters
avoids unwanted phase shifts. For practical applications the length of the FIR filter has to be
limited. As long as the filter coefficients are limited in value such filter are always stable.
Therefore the stability of such filters is not investigated. Avoiding phase shifts leads either to a
time retardation or to a non-causality of the filter. An exception of this rule are integrating
filters, that in most cases have a pole at the position z=1 and depending on the number of filter
coefficients in the FIR part a large number of poles at the position z=0 and the exponentially
decaying smoothing filters at the end of the document. These filters are recursive IIR filters.
The pole at z=1 can lead to instabilities, that become visible as a trend in the integrated signal.
19
3 Filter and Filter Banks
3.1 Kaiser Bessel Filter
If single filters are used a high precision is often desired and a constant group velocity or a
constant retardation of the filter is needed. A symmetric FIR filter fulfils these requirements.
The best known filter is the Kaiser Bessel Filter. The selection of a Bessel function was done,
because the spectrum of this function is rapidly decaying in the transition region as it can be
seen in (Figure 4)
( ) ( )
( )
( )
( )
( )
( )
( )
( ) ( )
( )
( )
( )
( )
2
2
0
2
2
0
2
0
0
2
0
0
2
0
1
1
sinh 1
1
2
1
1
2
1
1
2
1
|
|
.
|

\
|

|
|
|
.
|

\
|
|
|
.
|

\
|

=
|
|
|
.
|

\
|
(

=
|
|
|
.
|

\
|
(

=
|
|
|
.
|

\
|
(

= A =
s
s
Fourier
s
f
f N
I
f
f N
N
f W t w
I
N
tf
I
I
N t
t
I
I
N
n
I
t n w t w
t
| |
t
| t
|
|
|
|
|
|
(60)
In Figure 4 the quality of the filter is visible (fs=1, |=to):

Figure 4: Spectrum of the Bessel window (http://upload.wikimedia.org/wikipedia/commons/2/29/Kaiser-
Window-Spectra.jpg)

The aim of a filter is to reduce the ripple as far as possible. The determination of the parameter
of the filter derived from simulations by Kaiser. The window w is based on the modified Bessel
function I0( ). The first step is to define the quality Att of the filter. This parameter together with
20
the width of transition region Af of the related to the sample rate fs defines the number of filter
coefficients N.
( ) | |
( ) ( ) | | | |
| |
( )
( )
|
.
|

\
|
s s
|
.
|

\
|

|
|
|
.
|

\
|
(

=
A
>

s s =
2
1
2
1
,
1
2
1
36 14
8
21 0
50 21 21 07886 0 21 5842 0
50 7 . 8 1102 0
0
2
0
4 0
N
n
N
I
N
n
I
w
f/f .
Att-
N
dB < Att ,
dB Att dB , Att- . + Att- .
dB > Att , Att- .

n
s
.
|
|
(61)
For a low pass filter with the cut-off frequency fc, that defines the edge of the rectangular filter
window in the spectral domain, the following filter function in the time domain is derived:
( )
( ) < < = = n f f n
f
f
n
f f n
h
s c
s
c s c
n
, 2 sinc
2 2 sin
t
t
t
(62)
For a band pass filter with the cut-off frequencies fa and fb the following filter is derived:
( ) ( )
< <

= n
n
f f n f f n
h
s a s b
n
,
2 sin 2 sin
t
t t
(63)
The filter coefficients for the Kaiser Bessel filter h
*
are the product of both parts the window
and the theoretical filter with a rectangular window in the frequency domain:
|
.
|

\
|
s s
|
.
|

\
|
=
2
1
2
1
,
*
N
n
N
w h h
n n n
(64)
3.2 Tschebycheff Approximation and Remez Algorithm
The theory leads to infinite long filter sequences. Practical filters can have only a finite length.
The cutting of the infinite sequence leads to undesired ripple in the filter spectrum. (Gibbs
effect). In the previous chapter a Bessel function was applied to reduce the ripple. Parks and
McClellan [5] found that the minimum ripple is gained with a Tschebycheff polynomial of the
degree n, if a filter with (n+2) taps is used. The error between the theoretical curve and the
practical one can be described by a polynomial of degree (n+1). Remez [7] developed an
iterative algorithm that allows to determine the coefficients of an approximation to an arbitrary
function. The aim of a Tschebycheff approximation is, to generate a number of extremal
deviations from the curve with the same amplitude and alternating sign, that the number is
equivalent with the degree of polynomial that is used. Tschebycheff has proven that this
approximation has the minimum squared error.
A condition for the application of the algorithm is, that the theoretical filter function is given in
a continuous form. In the beginning the number of taps is chosen as (n+2). These taps are at
the start of the algorithm the Tschebycheff nodes. During the iteration these nodes will become
the extremes of the deviation of the theoretical and practical filter curve. The error at the nodes
has the Amplitude E with an alternating sign. With this condition the following equations are
derived:
( ) ( ) ( )
( ) ( ) ( ) 2 ..., , 2 , 1 , 1
2 ..., , 2 , 1 , 1 ...
0
2
2 1 0
+ = = +
+ = = + + + + +

=
n i t h E t b
n i t h E t b t b t b b
i
i
n
k
k
i k
i
i n
i n i i
(65)
This is a set of linear equations with (n+1) unknowns bk and unknown error E. The next step is
to find those M points, there the error has its maxima:
21
( ) ( ) ( )

=
=
n
k
k
k n n
t b t P t h t P
0
, (66)
This rule is iterated with a Newton Raphson iteration. The iteration is often stop after the first
round. Helpful is the knowledge of the first and second derivative of the function.
( ) ( ) ( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( )
( )
( )
( ) ( )
i m i m n i
i
i
i i m
n
k
k
k n
n
k
k
k n
n
t h t P e
t g
t g
t t
t h t b k k t h
t
t P
t
t g t g
t
t h t kb t h
t
t P
t
t g t g
t
t h t P t g
, ,
,
2
2
2
2
2
2
2
2
1
1
1
~
~
=
c
c

c
c
= =
c
c
=
c
c

c
c
= =
c
c
=

(67)
The iteration is repeated using the new sample points tm,i as long as the desired precision the
errors ei have the same amplitude and alternating sign. This is defined with ei=(-1)
i
E. To receive
usable results the precision has to be very high.
The zeros of the Tschebycheff polynomial of degree (n+2) on the interval [ta,tb] are:
( )
( ) 2 ..., , 2 , 1 ,
2 2
1 2
cos
2 2
+ =
|
|
.
|

\
|
+

+
+
= n i
n
i t t t t
t
a b a b
i
t (68)
This points are selected as the starting points in the iteration. The corrected points are derived
from zeros of the polynomial Pn(t).
3.3 Filter Banks
The idea of a filter bank is to split the signal into a large number of band limited channels with
limited computational efforts. The most simple filter bank is an octave filter bank. The filter
bank is implemented as a sequence of low pass filters of the same type. After the application of
one filter the signal is down sampled to half of the number of samples. The difference of to low
pass filters gives the result for the band pass filter. If a finer resolution is needed, e.g. third
octave bands or 1/12 octave bands, additional low pass filters are switched additionally for
every state of down sampled data. To keep the filter sequences short recursive IIR filters are
used. A typical member of an octave band filter is the wavelet filter bank. The procedure is
called multiple resolution analysis (MRA). The spectral resolution of this filter bank is small.
Therefore MRA is used for data compression and rarely for signal analysis. The division of the
signal in channels is called analysis and the reconstruction - probably after some manipulation
- is called synthesis.
3.4 Butterworth Filter
A typical filter bank is the filter bank based on Butterworth filters. The advantage of this filter
is, that the group velocity is nearly constant. The square of this filter is defined as:
( ) ( )
( )
n
c
G
j H G
2
2
0
2 2
1 e e
e e
+
= = (69)
The coefficients Bn for a low pass filter of degree n and the -3dB point at ec in the Laplace
domain are:
22
( )
( )
( )
( )

+ |
.
|

\
| +
+
(

+ |
.
|

\
| +

=
= =
[
[

=
=
2
1
1
2
2
1
2
0
, 1
2
1 2
cos 2 1
, 1
2
1 2
cos 2
,
n
k
n
k
n
c n
ungerade n
n
n k
a a a
gerade n
n
n k
a a
a B
s
a
a B
G
s H
t
t
e
(70)
The coefficients of the digital low-pass filter with an even order n are derived after a transition
to the z-transformation:
( )
( )
|
|
.
|

\
|
=
|
|
.
|

\
|
= O
O + O
|
.
|

\
| +
=
O =
O + O
|
.
|

\
| +
+ =
O =
O = =
+ +
+ +
=
[

=


s
c
s
c
c
c c k
c k
c c k
c k
c k k
n
k k k k
k k k
f
f
f
f
n
k
b
b
n
k
b
a
a a
z b z b b
z a z a a
z H
t t
t
t
tan
2
2 tan
2
1 2
cos 2 1
1 2
2
1 2
cos 2 1
2
2
2
2
1
2
0
2
1
2
2 0
1 2 /
0
2
2
1
1 0
2
2
1
1 0
(71)
The sampling rate is fs and the cut-off frequency of the low-pass filter fc. For a filter of order
n=2 the following coefficients are derived:
( )
( )
|
|
.
|

\
|
= O
O + O
|
.
|

\
|
=
O =
O + O
|
.
|

\
|
+ =
O =
O = =
+ +
+ +
=


s
c
c
c c
c
c c
c
c
f
f
b
b
b
a
a a
z b z b b
z a z a a
z H
t
t
t
tan
4
cos 2 1
1 2
4
cos 2 1
2
2
20
2
10
2
00
2
10
2
20 00
2
20
1
10 00
2
20
1
10 00
(72)
00
2 20 1 10 2 02 1 01 00
2 02 1 01 00 2 20 1 10 00
0
b
y b y b x a x a x a
y
x a x a x a y b y b y b
n n n n n
n
n n n n n n


+ +
=
= + + = + +
(73)
With the coefficients a filtered time series yn can be calculated for the original time series xn.
For order n=4 the following equations are derived:
23
( )
( )( )
( )( )
( )
( ) ( ) ( )
( ) ( ) ( )
( )
( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
01 00
4 21 20 3 11 20 21 10 2 01 20 11 10 21 00 1 11 00 01 10
01 00
4 21 20 3 11 20 21 10 2 01 20 11 10 21 00 1 11 00 01 10 01 00
4 21 20 3 11 20 21 10 2 01 20 11 10 21 00 1 11 00 01 10 01 00
4 21 20 3 11 20 21 10 2 01 20 11 10 21 00 1 11 00 01 10 01 00
2
20
2
20
2
11 10
2
01
2
00
2
11 10
2
21 01 20 00
4
21 20
3
11 20 21 10
2
01 20 11 10 21 00
1
11 00 01 10 01 00
4
21 20
3
11 20 21 10
2
01 20 11 10 21 00
1
11 00 01 10 01 00
2
21
1
11 01
2
20
1
10 00
2
21
1
11 01
2
20
1
10 00
tan
8
3
cos 2 1 ,
8
cos 2 1
1 2
8
3
cos 2 1 ,
8
cos 2 1
2
b b
y b b y b b b b y b b b b b b y b b b b
b b
x a a x a a a a x a a a a a a x a a a a x a a
y
x a a x a a a a x a a a a a a x a a a a x a a
y b b y b b b b y b b b b b b y b b b b y b b
f
f
b b
b b
b b
a a
a a a a
z b b z b b b b z b b b b b b z b b b b b b
z a a z a a a a z a a a a a a z a a a a a a
z H
z b z b b z b z b b
z a z a a z a z a a
z H
n n n n
n n n n n
n
n n n n n
n n n n n
s
c
c
c c c c
c
c c c c
c
c








+ + + + + + +

+ + + + + + + +
=
+ + + + + + + + =
= + + + + + + + +
|
|
.
|

\
|
= O
O + O
|
.
|

\
|
= O + O
|
.
|

\
|
=
O = =
O + O
|
.
|

\
|
+ = O + O
|
.
|

\
|
+ =
O = =
O = = = =
+ + + + + + + +
+ + + + + + + +
=
+ + + +
+ + + +
=
t
t t
t t
(74)
3.5 Bessel Filter
An alternative to the Butterworth filter is the Bessel filter that has a constant group velocity in
the pass band. The starting point of the Bessel filter are inverted Bessel polynomials un. The
transfer function of a Bessel filter is defined by:
( )
( )
( )
0
0
e u
u
s
s H
n
n
= (75)
The group delay of the filter is 1/e
0
. The parameter e
0
is not the -3 dB point of the filter, but only a
design point. Up to the design point the group delay is constant. For the inverted Bessel polynomials the
following construction formula is given:
( )
( )
( )
( )
( )
( )
( )
( ) 945 945 420 105 15
105 105 45 10
15 15 6
3 3
1
! ! 2
! 2
,
2 3 4 5
5
2 3 4
4
2 3
3
2
2
1
0
+ + + + + =
+ + + + =
+ + + =
+ + =
+ =

= =

=

s s s s s s
s s s s s
s s s s
s s s
s s
k n k
k n
a s a s
k n k
n
k
k
k n
u
u
u
u
u
u
(76)
This means especially:
24
( )
( )
( ) ( )
( )
( )! ! 2
! 2
,
! 2
! 2
0
0
0 0
k n k
k n
a s a s
n
n
s
k n k
n
k
k
k n
n n

= =
= =

e e u
u
(77)
The transition to the z transformation leads to:
( )
( )
( )! ! 2
! 2
,
1
1 2
0
1
1
0
k n k
k n
a
z
z f
a z
k n k
n
k
k
s
k n

=
|
|
.
|

\
|
+

=

=

e
u (78)
For the design frequency f
0
the relation derived before is still valid:
|
|
.
|

\
|
=
|
.
|

\
| A
A
=
s
s
d
f
f
f
t
t
0
0
tan 2
2
tan
2
t
e
e (79)
The following is correct:
( )
( )
( ) ( )

|
|
.
|

\
|
+

|
|
.
|

\
|

=
n
k
k k
s
k n n
z
z
f f k n k
k n
z
0
1
1
0
1
1
tan
1
! ! 2
! 2
t
u (80)
For the order n=2 we have:
( )
( )
( )
( )
( )
( )
( ) ( ) ( )
( )
( )
( ) ( ) ( ) ( )
( ) ( )
( )
2
2
2
1
2
2 1
2 2 1 2 2
2 1
2
2
2 1
2 2 1 2 2
2
2 1
2 1
2 1
2
2 1
2 1
2
2
1
1
1
1
1
1
2
2
2
1
1
1
1
1
2
1
1
2
0 2
0
2 2
3 3
3 3 2 6 3 6 3
3 3 2 6 3 3
3 6 3 0
2 1
3 3 6 2 3 3
2 1
2 1
3
2 1
1
3
2 1
2 1
3
1
1
3
1
1
1
1
! 2 ! 0 2
! 4
1
1
! 1 ! 1 2
! 3
1
1
! 0 ! 2 2
! 2
tan
1
3
! 2 2
! 4
0
C C
y C C y C x x x
y
z C C z C C C
z z
z
s
z H
z z
z C C z C C C
z
z z
z z
z z
z
C
z z
z z
C z
z
z
C
z
z
z
z
C z
z
z
C
z
z
C z
f f
C
s
n n n n n
n
s
+ +
+ + +
=
+ + + + +
+ +
=
=
=
+ +
+ + + + + +
=
+ +
+ +
+
+ +

+
+ +
+
=
+
+

+
+

=
+
|
|
.
|

\
|
+

+
|
|
.
|

\
|
+

=
=
= = =

u
u
u
u
u
u
t
u
(81)
3.6 Conclusions
A symmetric non-recursive FIR filter was presented. With this filter a signal can be filtered
without phase shifts with a high precision. A high pass is constructed subtracting the low pass
filtered signal from the original signal. A disadvantage of the Kaiser Bessel filter is, that for a
high precision the filter retardation and the numerical efforts are high. Especially the usage of
a small transition region increases the filter length, the numerical efforts and the filter
retardation. Also the quality of the filter Att has an effect on the number of filter taps. For a
filter bank the numerical efforts are too high. Here Butterworth filter are an alternative. Based
on this filter analogous and digital filter banks are designed. Not described were the recursive
filters Tschebyscheff filters of 1.kind and 2.kind, elliptic filters (Cauer filters) and ultra-steep
filters.
25
4 Manipulation of Discrete Time Series and Spectra and
Definition of Blocks
This chapter deals with the interpolation of additional sample points in the time domain and in
the spectrum and the averaging about a time block.
4.1 Up Sampling of Time Series
Given is a time series sampled with the sample rate fs. Related to the sample rate is a sample
interval At = 1/fs . The Fourier spectrum of time series with discrete sample points is:
( ) Z k j e x
f
e x t f X
k
f
k
f j
k
s k
f
k
f j
k
s s
e = = A =


=

, 1 ,
1
2
2 2 t t
. (82)
The aliasing frequency is fs/2. The Shannon criterion leads to a limitation of the spectrum to the
interval [-fs/2,fs/2]. The discrete Fourier transformation leads to a periodic continuation in the
spectrum in contradiction to the Shannon criterion.
The addition of interpolated sample points is related to an increase of the sample rate. If the
interpolation leads to a continuous function in time the sample rate increases to infinity. But the
content of the spectrum shall not be changed, therefore Heaviside functions are added to the
spectrum with a cut-off frequency at the original aliasing frequency. The continuous function
in time is named x(t).
( ) ( )
( )
( ) ( ) | |
( )
( ) ( )
( )
( ) ( ) ( ) | |
( )
( ) ( )
( )
s s
s
s
s
s
tf j tf j tf j tf j
f
f f
ft j
f
f f
ft j
f
ft j
s s
s k
k
s s k
k
s f
f
k
t f j
k
k
s
f
ft j f
k
f j
k
k
s f
ft j
k
f
k
f j
k
s f
ft j
f
ft j
f B
s s
f A
tf f
tf
tf
f
t j
tf j
t j
e e
t j
e e
t b
t j
e
df e df e f f H f f H t b
f
k
t x
f f
k
t x
f
df e x
f
t a
df e e x
f
df e e x
f
df e f X t a
df e f f H f f H f X t x
s s s s
s
s
s
s
s
s s
t
t
t
t
t
t t
t
o t to
t t t t
t
t t
t
t
t
t
t
t
t
sinc
sin
2
sin 2
2 2
2
2 / 2 /
1
2 2
1 1
1 1
2 / 2 /
2 / 2 2 / 2
2 /
2 /
2
2 /
2 /
2 2
2
2
2
2
2
2
2
= = =

=
(

= = + =
|
|
.
|

\
|
=
|
|
.
|

\
|
|
|
.
|

\
|
= =
= = =
+ =

= =

=
|
|
.
|

\
|

=

=
} }

}

}

}
}

(83)
A product in the spectrum is related to a convolution in time domain.
( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( )
} } }
} } } } }
} } } }

=
= =
= =
= =
t t
t t
t t
t

t
t
t t t
t t t t t o t
t t t to t t t
t t
d t b a d d t b a t x
d d t b a d d df e b a t x
df e d e b d e a df e f B f A t x
f
t f j
f
ft j f j f j
f
ft j
2 2
2
2 2 2 2
(84)
26
( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( ) ( )
( )
x
x
x k t f x t x
f
k
t f d t f
f
k
x t x
d t f f
f
k
x
f
d t b a t x
k
s k
s
s
k
s
s
k
s s
s k
k
s
sin
sinc , sinc
sinc sinc
sinc
1
= =
|
|
.
|

\
|
|
|
.
|

\
|
=
|
|
.
|

\
|
=

|
|
.
|

\
|
= =

}
}

}

=
t
t t t t t o
t t t t o t t t
t
t t
(85)
The interpolation is related to a filtering of the signal with a sinc-function. It has to be remarked
that this approach is equivalent to an interpolation in frequency domain. Therefore leakage and
aliasing are not changed using this approach.
4.2 Extension of the Time Basis
The reversed problem is an extension of time basis. This approach is presented here to complete
the theoretical possibilities. The time basis shall be extended adding zeros to the original signal.
This is done using again the Heaviside function.
( )
( ) ( )
( )

( ) ( ) | |
( )
( ) ( )
( )
( ) ( ) ( ) | |
( )
( ) ( )
( )
( ) ( ) ( )
( )
( ) ( ) ( ) ( )
( )
x
x
x k ft X f X
t
k
f t t X
t
d f t t
t
k
f X f X
ft t
ft
ft
t
f j
ft j
f j
e e
f j
e e
f B
f j
e
dt e dt e t t H t t H f B
t
k
f X
t t
k
f X
t
f A
dt e X
t
dt e e X
t
dt e t x f A
df e t t H t t H t x
t
f X
j e X
t
e X f t x
k
w k
w
w w
k
k
w
f B
w w
k w
k
w w
w
w
w
w
ft j ft j ft j ft j
t
t t
ft j
t
t t
ft j
t
ft j
w w
k w
k
w k w
k
w
k
t
t
t
k
f t j
k
w t
ft j
k
t
t
k
j
k
w t
ft j
t
ft j
t b
w w
t a
w
k
t
t
k
j
k
w k
t
t
k
j
k
w w w w
w
w
w
w
w w
w w
sin
sinc , sinc
sinc
1
sinc
sinc
sin
2
sin 2
2 2
2
2 / 2 /
1
2 2
1
1 1
2 / 2 /
1
1 ,
1
2 / 2 2 / 2
2 /
2 /
2
2 /
2 /
2 2
2
2
2
2
2
2
2 2
= =
|
|
.
|

\
|
|
|
.
|

\
|
=
|
|
.
|

\
|
=
= =

=
(

= = + =
|
|
.
|

\
|
=
|
|
.
|

\
|
|
|
.
|

\
|
=
= = =
+ =
= = A =

}

} }

} }

}
}

=

=

=
|
|
.
|

\
|

=
t
t t o
t
t
t
t
t
t t
t
o t to

t t t t
t
t t
t
t
t
t
t
t t


(86)
A convolution with a sinc-function is needed in the spectrum.
27
4.3 Averaging of Time Blocks
In the following derivation the modified Plancherels theorem together with a convolution is
needed.
( ) ( ) ( )
( )

( ) ( ) | |
( )
( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) o o e o e
o | | o e to | o
t
e
o | | o
t
e
| | o o
t t
e
e e e
t
o
| o
| o e
| o
e |
|
o
o
e
d X X A
d d X X A
d d dt e X X A
dt e d e X d e X dt e t x A
d B A
t
dt t t H t t H t x
t
dt t x
t
t x
t
t j
t j t j
t
t j
t
t j
t w
t b
w w
t a
w
t
t t w
m
w
w
=
+ + =
=
= =
= + = =
}
} }
} } }
} } } }
} } }

=
2
2
1
2
1
2
1
2
1
2
2 / 2 /
1 1
2
2
2 /
2 /
2 2

(87)
( ) ( ) ( ) | |
( )
( ) ( )
( )
( ) ( ) ( ) ( )
( ) ( ) ( ) e
e
o o e o
e e
t
o o e o
t
e
t e
e
t e
e
t e t
e
e t t t
e
e o
e o
e e
e
e e
d
t
d X X t x
d t
t
d X X
t
t x
t
t
t
t t
j
t j
j
e e
B
j
e
dt e dt e t t H t t H B
w
m
w
w
w
m
w
w
w
w w w
t j t j
t
t t
t j
t
t t
t j t j
t
w w
w w
w
w
w
w
} }
} }
} }

=
|
.
|

\
|
=
=
= = =
(

+
=
(

= = + =
2
sinc
2 / sinc
2
2
2 / sinc
2 2 /
2 / sin
2
2 / sin 2
2
1
2
1
2
1
2
1
2 / 2 /
2
1
2
2
2 / 2 /
2 /
2 /
2 /
2 /
(88)
An average about a time block leads to weighting with a sinc-function in the spectrum together
with a convolution to reproduces the squared function in the spectrum.
4.4 Conclusions
If an interpolation in time domain is needed in accordance with the original spectrum a filter
with a sinc function is needed. This procedure can be interpreted as a trigonometric
interpolation. But this approach keeps aliasing and leakage effects unchanged.
Another possibility is a polynomial interpolation. The better way is to use a higher sampling
rate in the digitisation process. This avoids the application of a model during the up sampling
process. A higher sampling rate extends the frequency spectrum.
On the other side a higher sampling rate can increase the high frequency noise in the signal.
This can become problematic, if a numerical derivation in time domain shall be applied. In this
case the application of a smoothing filter can make sense. A suggestion that will be presented
in the ongoing document is the Savitzky Golay filter.
28
5 Hilbert Transformation, Demodulation und Envelope
The goal of the Hilbert transformation is to rotate the phase of a modulation of signal by t/2
without changing the envelope. The envelope is represented in the spectrum in the surrounding
of the carrier that might be e.g. an Eigen frequency of a structure. Are more than one carrier
present a band pass filter is needed that supresses all carriers with the exception of the one
carrier that shall be investigated.
( ) ( ) t x t x
f
BP
(89)
5.1 Hilbert Transformation in Spectral Domain
The next step is to rotate the phase by t/2. Therefore we observe one frequency in the time
domain.
( ) ( ) ( )
( )
( ) ( ) ( )
( )
( )
( )
( ) ( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( )
( )
( ) ( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
0 0
0 0
2 2
2 2 2
0 0
0 0
2 2
2 2 2
2 2
2 2 2 2
0 0
2 2
2 2 2 2
0 0
2 2
~
2 2
2
2 2
2
~
2 2
~
2 2
~
2 2
2 2
2
2 2
2
2 2
2 2
2 2
~
2 2
2 cos 2 sin
~
2 2
2 2
2 sin 2 cos
0 0
0 0
0 0
0 0
0 0
0 0 0 0
0 0
0 0 0 0
f f
jB A
j f f
jB A
j f X
f f
jB A
j f f
jB A
j f X
dt e
jB A
e
jB A
f X
dt e e
jB A
e
jB A
f X
f f
jB A
f f
jB A
f X
f f
jB A
f f
jB A
f X
dt e
jB A
e
jB A
f X
dt e e
jB A
e
jB A
f X
e
jB A
j e
jB A
j t x
e e
B
j
e e
A t f B t f A t x
e
jB A
e
jB A
t x
j
e e
B
e e
A t f B t f A t x
f
f
t
t f f j t f f j
f
t
ft j t f j t f j
f
f
f
t
t f f j t f f j
f
t
ft j t f j t f j
f
t f j t f j
f
t f j t f j t f j t f j
f
t f j t f j
f
t f j t f j t f j t f j
f
+
+

=
+
+

=
(

+
+

=
(

+
+

=
+
+
+

=
+
+
+

=
(

+
+

=
(

+
+

=
+

=
+
+

= + =
+
+

+
+
= + =
}
}
}
}

=
+

=
+

o o
t to t to
o o
t to t to
t t
t t
t t
t t t
t t
t t t
t t
t t t t
t t
t t t t
(90)
The positive frequency components in spectrum have to be multiplied by the imaginary unit j
and the negative components by -j. The component related to the zero frequency is set to zero,
to keep a possible offset as it is.
29
( )
( ) ( )
( ) ( ) ( ) f X f H f X
f j f H
f j
f
f j
f H
f f
=
=

<
=
>
=
~
sgn
0 ,
0 , 0
0 ,
(91)
In normal cases an adjusted band pass filter is needed together with the Hilbert transformation.
Therefore it is useful to apply the transformation in the frequency domain.
5.2 Hilbert Transformation in Time Domain
For a derivation of the Hilbert transformation in time domain the spectral definition together
with an inverse Fourier transformation is needed.
( ) ( ) ( )
( )
( ) ( ) ( )
( )
( )
( )
( )
( )
( ) ( )
( ) ( )
( ) ( ) ( ) t t t
t
t
t
t t
t
t
t
t
t
t
t
t
t
t t
t t t t
t t t t
d t h x t x
t
t
ft
t
t h
ft f
t
ft
t
ft
h
t
ft
t
t h
t
ft
df ft df ft j j t h
f d e df e j df e df e j t h
df e df e j df e f j df e f H t h
f f
f
f t f t f
f
t
f
f f f
f
ft j
f
ft j
f
ft j
f
ft j
f
ft j
f
ft j
f
ft j
f
ft j
=

=
=
= = =

=
=
(

= = =
(
(

=
(
(

+ =
(
(

= = =
}
} }
} } } }
} } } }

+
+ + + + +
+
+
+
+
=

=

=

=
=

=
~
0 ,
2 cos 1
lim
0 , 0
0 0 lim
2 sin 2
lim lim
2 cos 1
lim lim
2 cos lim 1
lim 0
2 cos
lim
1
2
2 cos
2 2 sin 2 2 sinh 2
~
sgn

0 0 / 0 0

0 0 0
0
~
~
2
0
2
0
2
0
2
0
2
0
2 2 2
(92)
5.3 Hilbert Transformation for Time Series
A discrete time series has a band limited spectrum. Therefore the following derivations are
correct:
( ) ( ) ( )
( )
( ) ( ) ( )
( )
2 /
0
2 /
0
2 /
0
2 /
0
~
~
2
2 /
0
2
2 /
0
2
2 /
0
2
0
2 /
2
2 /
0
2
2 /
2 /
2
2 /
2 /
2
2
2 cos 2
2 sin
2
2 sinh 2
1
~
sgn
1 1
s
s s
s s s s
s
s s
s
s
s
f
f
s
f
f s
f
f s
f
f
ft j
f
f
ft j
s
f
f
ft j
f
f
ft j
s
f f
ft j
f
f
ft j
s
f
f f
ft j
s
f
f f
ft j
s
t
ft
f
df ft
f
df ft j j
f
t h
f d e df e
f
j
df e df e
f
j
t h
df e df e
f
j
df e f j
f
df e f H
f
t h
= = =
=

= =
= = = =
(

= = =
(
(

=
(
(

+ =
(
(

= = =
} }
} } } }
} } } }
t
t
t t
t t t t
t t t t
(93)
30
( )
( ) ( )
( )
( ) ( )
( )
( )
( )
( )
( )

=
=

= =
= = =

= =

= =
+ + +
0 ,
cos 1
0 , 0
cos 1
: 0
0 sin lim
sin
lim
cos 1
lim 0 : 0
cos 1 cos 1
0 0 0 / 0 0
t
t f
t f
t
t h
t f
t f
t h t
t f
f
t f f
t f
t f
h t
t f
t f
t f
t f
t f
t h
s
s
s
s
s
s t
s
s s
s t
s
s
s t
s
s
s
s
s
t
t
t
t
t
t
t t
t
t
t
t
t
t
t
(94)
( )
( )
( )
( )
k
n k
k
k n
n
n
n
n
s
x
k n
x
n
n
n
h
n
n
n
n
h
Z n t f
t
t
n

=
=

=

=
=

=
=
e =
A
=
t
t
t
t
1 1
~
0 ,
1 1
0 , 0
0 ,
cos 1
0 , 0
,
(95)
5.4 Estimation of the Envelope
The next step is the determination of the envelope. This step is easy to apply in time domain.
The original filtered signal and the phase shifted signal are squared and summed up. The square
root of the sum gives the desired envelope, because this operation terminates the carrier
frequency. This is presented using the original example.
( ) ( ) ( ) | |
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) | |
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) | | ( )
( ) ( )
2 2
2 2 2
0
2
0
2 2 2 2 2
2
0
2
0 0
2
0
2 2
2
0 0
2
2
0
2
0 0
2
0
2 2
2
0 0
2

2 sin 2 cos
~

2 cos 2 sin 2 cos 2 2 sin

~
2 cos 2 sin
~
2 sin 2 sin 2 cos 2 2 cos
2 sin 2 cos
B A t x
B A t f t f B A t x t x t x
t f B t f t f AB t f A t x
t f B t f A t x
t f B t f t f AB t f A t x
t f B t f A t x
f
f f f
f
f
f
f
+ =
+ = + + = + =
+ =
+ =
+ + =
+ =
t t
t t t t
t t
t t t t
t t
(96)
Therefore the following operation is needed:
( ) ( ) ( ) t x t x t x
f f f
2 2 ~
+ = (97)
5.5 Conclusions
The Hilbert transformation helps to determine the envelope of a signal. This can be used to
derive the decaying of a resonance if the free fading of a resonance was recorded.
The Hilbert transformation has a big role if data are transmitted in signal processing using a
modulation of a carrier. The Hilbert transformation is the correct way to do the demodulation.
31
6 Differentiation und Integration in Time Domain
In this chapter we presented methods to integrate and differentiate a discrete time series in time
domain. It is assumed that the time series was recorded with a constant sample interval At.
6.1 Differentiation in Time Domain
To avoid phase errors is useful to determine the first derivative using two intervals. Doing this
reduces the aliasing frequency to half of its original value.
t
x x
x
i i
i
A

=
+
2
1 1
(98)
The values at the beginning and the end of the recorded time block have to be extrapolated, if
the derived block shall have as many samples as the original time series (e.g. bent of a linear
function using one interval with phase shift). As long as constant intervals At are used, this
procedure is identical with the derivation of a parabolic interpolation.
( )
( )
( )
( )
( )
( )
( )
( )
2
1 1
1 1
2
1 1 2
1 1
1 1
1 1
2
1
2
1
2
2
2 0
2
0
2
2
2 2
2
2
0
2
2
t
x x x
x
a x x
t
x x
x
b x x
t
x x x
a t a x x x
t
x x
b t b x x
c t b t a t x x
x c c x x
c t b t a t x x
a t x
b at t x
c bt at t x
i i i
i
i
i i
i
i
i i i
i i i
i i
i i
i
i i
i
A
+
=
= =
A

=
= =
A
+
= A = +
A

= A =
+ A + A = A =
= = =
+ A A = A =
=
+ =
+ + =
+
+
+
+
+
+
+

(99)
6.2 Integration in Time Domain
For the integration the trapezoidal rule is suggested to avoid phase shifts. A problematic part is
that the integrated value at the beginning of the interval has to be added. Therefore the
integration is non local as it was the case for the differentiation. One effect that arises are trends
in the integrated signal, if a small offset is given in the original data. A starting value has to be
chosen.

( )
1
1
1
2
1

+ A
+
~ + =
}

i
i i
i
t
t
i
x t
x x
x dt t x x
i
i

(100)
32
6.3 Differentiation with Polynomial Interpolation
If equidistant intervals are used, higher polynomials as the quadratic one can be chosen to derive
the original time series.
( )
( )
( )
( )
( )
( )
( )
m
m
n
m k
m k
m
k
n m
m
m
n
k
k k
n
k
n
m k
m k
m
k
n m
m
n
k
k
k n
t
a
m
t
a
m k
k
t p
t
x
t
a
t
a
k t p
t
x
a t p x
m n
t
t
t
a
m k
k
t p
t
t
t
a t p
A
=
A
= =
c
c
=
A
=
A
= =
c
c
=
= = =
>
|
.
|

\
|
A A
=
c
c
|
.
|

\
|
A
=

=
! 0
!
!
0
0 0
0
,
!
!
) (
0
1
1
1
0
0 0
0

(101)
So far the differentials of a polynomial can be calculated. But the determination of the
polynomial coefficients is still missing. At this point as much sample points shall be used as
unknown coefficients ak of the polynomial exist.
( )

= = A = =

=
gerade n
n n
ungerade n
n n
l l a t l t p x
n
k
k
k n l
,
2
...
2
,
2
1
...
2
1
,
0
(102)
The given set of equations can be collected to a matrix equation. The vector of the unknowns
contains the coefficients ak and xl leads to a vector of the known values at the sample points.
The matrix contains the factors
k
l .
( )( )
( )
( )
y A b
y Ab
1
+
+
+ +
=
=

+
+
= =
=
=
=
gerade n
n n
ungerade n
n n
l
gerade n
n
ungerade n
n
l n k
b a
x y
l A
k k
l l l
k
k l l
,
2
...
2
,
2
1
...
2
1
,
,
2
2
,
2
1
, .. 0 ,
0
1
1
0
0
(103)
The matrix A depends only on the degree of the polynomial. Therefore the inverse of the matrix
has to be determined only once and can be applied to all sample points. To determine the
derivative of the function only one raw of the inverse matrix is needed.
( ) ( )( ) ( )
( )
( )( )
( )
( )
( )( )
| |
( )

+ +

+ + +

+ + +
A
=
A
=
A
=
A
=
A
=
l
n l l l m m
m
n
l
l l l m m
l
l l l l m m m m m
m m
x A m
t
x
x A
t
m
y A
t
m
b
t
m
t
a m
x
1
1
1
1
1
1 1
) (
0
0
0 0 0
!
1
! ! ! !
(104)
(105)
The procedure can be extended to the case there more sample points are used than needed for
the determination of the unknowns. This leads to an over determined set of equations and a least
square solution can be applied. This leads to the differentiation with a Savitzky Golay filter.
6.4 Integration with Polynomial Interpolation
As it was possible for the differentiation also the integration can be based on higher order
polynomials.
33
( )
( )
0
0
0
0
0
1
0
0 1
0
1 1
x
k
a
t x
t
t
k
a t
x dt t p x
t
t
a t p
n
k
k
t
t
n
k
k
k
t
t
n
n
k
k
k n
+
+
A = +
(
(

|
.
|

\
|
A +
A
= + =
|
.
|

\
|
A
=

}

=
A
=
=
+ A
=
=
(106)
So far the integral of a polynomial is derived. Again the determination of the unknown
coefficients is missing. Again it is assumed that as much sample points are used as unknown
coefficients ak of the polynomial exist.
( )

= = A = =

=
gerade n
n n
ungerade n
n n
l l a t l t p x
n
k
k
k n l
,
2
...
2
,
2
1
...
2
1
,
0
(107)
This set of equations is again written as a matrix equation. The vector of the unknown
coefficients ak and the vector of the known values at the sample points xl are linked by a matrix
containing the factors
k
l .
( )( )
( )
( )
y A b
y Ab
1
+
+
+ +
=
=

+
+
= =
=
=
=
gerade n
n n
ungerade n
n n
l
gerade n
n
ungerade n
n
l n k
b a
x y
l A
k k
l l l
k
k l l
,
2
...
2
,
2
1
...
2
1
,
,
2
2
,
2
1
, .. 0 ,
0
1
1
0
0
(108)
Again matrix A depends only on the polynomial degree and can be determined only once and
applied to all sample points. But the integration depends on all rows of the matrix inverse. A
reduction of the computational efforts is still possible because the matrix vector product can be
simplified to a vector vector product. The filter coefficients have to derived only once by one
summation.
( )( )
( )
0
1
1
0
1
0
0
1
0
1
1
1
x x A
k
t x
x
k
a
t x
l
l l l k
n
k
n
k
k
+
+
A =
+
+
A =

+ +
=
=

(109)
( )( )
( )
( )( )
( )
n n l
l
l l k
n
k
n
l
l
l l k
n
k
x x A
k
t x
x x A
k
t x
+
(

+
A =
+
(

+
A =
+

+ +
=
+

+ +
=

1
1
0
1
0
1
1
0
1
0
0
1
1
1
1
(110)
The described procedure can be extended to an over determined set of equation that is
approximated in the least square sense. This leads to the integrating Savitzky Golay filter.
34
6.5 Savitzky Golay Filter for Smoothing and Smoothed
Differentiation
The Savitzky Golay filter is defined for an equidistant digitized time series with the sampling
interval At. A polynomial of degree M with the coefficients aj is used.
( )
( )


=

= =
A
=
|
.
|

\
|
A A
=
c
c
=
|
.
|

\
|
A
=
M
j
j
j i
M
j
j
j
M
j
j
j i
M
j
j
j
i j a
t
x
t
t
t
j
a t x
t
i a x
t
t
a t x
1
1
1
1
0 0
1
,
,

(111)
The polynomial can be written in a matrix formulation as long as the number of samples yi is
equivalent to the number of unknowns aj.
( )( )
( ) M n n n n k M j y y a a k A
R L R L k n k j j
j
j n k
L L
= + = = = = = =
+ + + + + +
, .. , .. 0 , , , ,
1 1 1 1

y a A (112)
To smooth the data the number of samples has to be greater than the number of unknown
coefficients. An approximation with a least square algorithm leads to the equation:
( ) ( )
( ) ( )
( )
( )( ) ( )( ) ( )( )
( )
( ) ( )( ) ( )
M i y k y A
M j i k A A
M n n
R
L
R
L
L L
R
L
R
L
L L
n
n k
k
i
n
n k
n k i n k i
T
n
n k
j i
n
n k
j n k i n k j i
T
T T
R L
T T
.. 0 ,
.. 0 , ,
,
1 1 1 1
1 1 1 1 1 1
1
= = =
= = =
=
> + =


= =
+ + + + + +
=
+
=
+ + + + + + + +

y A
A A
y A A A a
y A a A A
(113)
For the estimation of the value of the polynomial at the position i = 0 only the coefficient a0 is
needed. This value is determined with the first row of the matrix. For the first derivative the
term a1/At has to be evaluated. This term is evaluated with the second row of the matrix with a
scale with 1/At. In the same manner the second or higher derivatives can be estimated, as long
as the polynomial degree has a degree higher or equal to the desired derivative.
( )
( )
( )
l l
M
l j
l j
j l
l
M
j
j
j
M
j
j
j
M
j
j
j
a
t
l
l j
j
a
t
x
a
t
j j a
t
x
a
t
j a
t
x
a a x
A
=
A
=
A
=
A
=
A
=
A
=
= =

=
!
0
!
! 1
2
0 1
1
1
0
1
0
0
2
2
2
2 0
1
1
1
0
0
0
0

(114)
If the times series y is filled with the unit vector en in which all elements except the n-th element
are zero and the n-th element is one, the filter coefficients can be determined directly.
( ) ( ) { }
( )
( ) { }
( )( )
( ) ( ) { }
( )
( ) { }
( )( )
( ) ( ) { }
( )
( ) { }
( )( )


=
+

=
+
=
+

=
+
=
+

=
+

A
=
A
= =

A
=
A
= =
= = = =
M
j
j
j
T
n
T T
n
n
n n
n i n i
M
j
j
j
T
n
T T
n
n
n n
n i n i
R L
M
j
j
j
T
n
T T
n
n
n n
n i n i
n
t t
e y e x
n
t t
d y d x
n n n n c y c x
R
L
R
L
R
L
0
1 3
1
2 3
1
2
0
1 2
1
2
1
0
1 1
1
1
1
2 2
1 1
.. ,
A A e A A A
A A e A A A
A A e A A A

(115)
35
( )
( ) ( ) { }
( )
( ) { }
( )( )
=
+ +

=
+

A
=
A
= =
M
j
j
j l
T
l n
T T
l n
n
n n
n i n
l
i
n
t t
l
g y g x
R
L
0
1 1
1
2 1
1 2 !
A A e A A A (116)
The averaging and differentiating Savitzky Golay filter can be taken from Numerical Recipes
[6].
6.6 Savitzky Golay Filter for Smoothed Integration
An integrating filter can be defined too. For this purpose we do a step to the polynomial
equation.
( )
( )
( ) | |
| |
0
0
0
0
1 1
1
1
0
1 1
1
0
1
1
1
1
0
1
1
0
1
0
1
0 0
1
0 1
1
1
1
1 1
,
1
1 1 1
x
j
a
t x
j
a
t x
x k k
j
a
t x
x
t
t
t
t
j
a
t x
t
t
j
t
a x
x dt
t
t
a x dt
t
t
a x dt t x x
i a x
t
t
a t x
M
j
j
M
j
j j j
k
M
j
j j j
k
k
M
j
j
k
j
k
j
k
M
j
t
t t
j
j k
k
M
j
t
t t
j
j k
t
t t
M
j
j
j k
t
t t
k
M
j
j
j i
M
j
j
j
k
k
k
k
k
k
k
k
+
+
A = +
+
A =
+
+
A =
+
(
(

|
.
|

\
|
A

|
.
|

\
|
A +
A = +
(
(

|
.
|

\
|
A +
A
=
+
|
.
|

\
|
A
= +
|
.
|

\
|
A
= + =
=
|
.
|

\
|
A
=

} }

}

= =
+ +

=
+ +

=
+

=
=
+

=
=

=
=

=
= =

(117)
So far the sum of all rows is needed to determine the integral.
( ) ( )
( )
( )
( )( )
( )
( )( )

= = + +

= = + +

= +

=
+
(
(

(
(

)
`

+
A =
)
`

+
A =
)
`

+
A = + =
M
j
M
k j k
T j
n
M
k
M
j
j
j k
T
n
M
k k
n
T T
n i
n
n n
n i n i
k
n t c
n
k
t c
k
t c x y c x
R
L
0 0 1 1
1
0 0 1 1
1
0 1
1
1
1
1
1
1
1
1
A A
A A
e A A A
(118)
6.7 Differentiation with Trigonometric Interpolation
How to interpolate values in accordance with the spectrum was presented before. The result
was a convolution with the sinc-function.
( ) ( ) ( ) ( )
( )
( )
( ) ( )
( )
( ) ( )
( )
( )
( ) ( ) ( ) ( ) ( )
( ) | |
( )
( ) ( ) ( ) ( ) ( )
( ) | |
( ) ( ) ( ) ( ) ( )
( ) | |

=
|
|
.
|

\
|
= A = =

c
c
=

c
c
=
= =
k
k s
s
n
k
s
s s s
k s
k
s
s s s s s
k
k s
s
k
k s
s
k
k
s k
k n
k n k n k n os
x f
f
n
t n t x x
k t f
k t f k t f k t f os
x f t x
k t f
k t f f k t f k t f os f
x t x
k t f
k t f
t
x
k t f
k t f
x
t
t x
x
x
x k t f x t x
2
2
2
sin - c
sin - c
sin - c
sin sin
sin
sinc , sinc
t
t t t
t
t t t
t
t t t
t
t
t
t
t

(119)
36
( )
( ) ( )
( ) ( )
( )
( )
( ) ( ) ( ) ( ) | |

=
(

=
k
s
k n
k
s
k n
k n k n os
k n
f
x x
k n
k n
k n os
k n
f
x x
t t
t
t
t
sinc c
sin
c

(120)
Most critical is the position n = k. For this position Hospitals rule is used.
( ) ( ) ( ) ( ) ( )
( ) | |
( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) | |
( ) ( ) ( )
( ) | |
( ) ( )
0
2
sin
lim
2
sin
lim
2
c sin c
lim
sin - c
lim
0 0
2
=

=

=
=

=
=

k t f
k t f
k t f k t f
k t f f
k t f os f k t f k t f f k t f os f
k t f
k t f k t f k t f os
s
f k t
s
s s
f k t
s s
s s s s s s s
f k t
s
s s s
f k t
s s
s
s
t
t
t t
t t
t t t t t t t
t
t t t
(121)
Therefore the position n = k is neglected.
( )
( ) ( ) ( ) ( ) | |

=
=

=
n k
k
s
k n
k n k n os
k n
f
x x t t sinc c (122)
For the function the following holds as long as integer points are used:
( ) ( ) ( )
( ) ( )
( )
( )

=
=

=
e

=
=
=
e =
n k
k
k n
k s n
k n
k n
x f x
Z k n
k n
k n
k n
Z k n k n
1
, ,
, 0
, 1
sinc
, , 1 cos

t
t
(123)
The position n=k vanishes, because the difference of the cosine function and the sinc-function
is zero.

Figure 5: Differentiation with trigonometric interpolation using 42 coefficients

37
As an example the signal that is differentiated in the spectrum in Figure 36 is differentiated here
in the time domain.
6.8 Integration with Trigonometric Interpolation
Again, how to interpolate values in accordance with the spectrum was presented before. The
result was a convolution with the sinc-function.
( ) ( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( )
1

1 1
sinc sinc
sin
sinc , sinc

=
+ = + = =
= =
}

n
t
t t
s
k
k n
t
t t
k
s k n n
k
s k
x dt k t f x x dt k t f x t x x
x
x
x k t f x t x
n
n
n
n
t t
t


(124)
The integral about the sinc-function is defined in mathematics with a new function the Sine
Integralis function.
( ) ( ) ( )
( )
( ) ( )
( ) | | ( ) ( ) | |
( )
( ) ( ) ( ) ( )
Z k i x
k n k n
x
f
x
f
n
t n t
f
n
t n t
x x
f
x x
f
x
C d
x d
f
x
f
x
k t f k t f f
t
k t f
n
k
k
s
n
s
n
s
n
n n n
k
k
s
n
k
k
s
n
n
s k
k
s
n
n s n n s n s s
n
n
n
n
e +

=

= A = = A =
+ = + =
+ =
+ =
= = =
c
c
=

=
=

}
}

, ,
1 Si Si 1
1
1 ,
Si Si
1
Si
1
Si sinc
sinc
1 1
, , ,
1
1
1 1 1
1

1 1
1
1
t
t t
t t
t
t
t
t t t
t t
t t
t t t t t
t
t t
t
t t
t
t t

(125)
The Sine Integralis function is an odd function and has the following features:
( ) | |
( )
( ) | |
( ) ( )
( )
( )
( )( )
( )
( )( )
...
! 9 9 ! 7 7 ! 5 5 ! 3 3 ! 1 2 1 2
1
! 1 2 1 2
1
Si
Si Si
2
Si lim
0 0 Si
2
Si lim
9 7 5 3
0
1 2
0
2

~
+ +

=
+ +

=
=
=
=
=


=
+
=

+
t t t t
t
t t
t t
t t
t
t
t
t
t
t
i
i i
i
i i
i i i i
(126)
The function Si( ) can be taken from Numerical Recipes [6]. The difference
( ) ( ) ( ) | | t t t 1 Si Si n n produces the following curve
38

Figure 6: Coefficients of the integrating filter

The filter needs the difference of two Si functions, which can be approximated by an infinite
series.

( ) ( ) ( ) ( ) ( )
( )( )
( ) ( ) ( )
( )( )
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
( )( )
( ) ( ) ( ) ( ) ( ) | |
( )( )

=
+ +

=
+ +

=
+

=
+
+ +

=

+ +

=

+ +

+ +

=

0
1 2 1 2 2
0
1 2 1 2
0
1 2
0
1 2
! 1 2 1 2
1 1 1 Si Si
! 1 2 1 2
1 1 1 Si Si
! 1 2 1 2
1 1
! 1 2 1 2
1 1 Si Si
i
i i i i
i
i i i
i
i i
i
i i
i i
n n n n
i i
n n n n
i i
n
i i
n n n
t
t
t t
t
t t
t
t t
t
t
t
t
t
t t
(127)
( ) ( ) ( ) ( ) ( )
( )( )
( )
( ) ( ) ( )
( )( )
( ) ( ) ( ) ( ) ( )
( )( )
( )
( ) ( ) ( )
( )( )
( ) ( ) ( ) ( ) ( ) | | ( ) ( )
( )( )

=
+ +
+
=

+ +


+ +

=

+ +


+ +

=

0
2 2 2 2
0
2
0
2
0
2
0
2
! 1 2 1 2
1 1 1 1 Si Si
! 1 2 1 2
1 1
1
! 1 2 1 2
1 1 Si Si
! 1 2 1 2
1 1
1
! 1 2 1 2
1 1 Si Si
i
i i i i i
i
i i
i
i i
i
i i
i
i i
i i
n n n n n n
i i
n
n
i i
n
n
n n
i i
n
n
i i
n
n
n n
t
t
t t
t t
t
t t
t
t
t
t
t
t
t
t t
(128)
Especially at the sampling points the following symmetric values to the axis n = +1/2 are taken
from [4]:

39
Table 1: Coefficients of the integrating filter
N ( ) ( ) ( )
t
t t 1 Si Si n n

n ( ) ( ) ( )
t
t t 1 Si Si n n

-20 +0.009886107596 1 +0.5894898722
-19 0.01039321090 2 0.1380782055
-18 +0.01095515771 3 +0.08168157087
-17 0.01158135507 4 0.05812356760
-16 +0.01228349256 5 +0.04513749414
-15 0.01307627930 6 0.03690194680
-14 +0.01397848507 7 +0.03121077963
-13 0.01501444846 8 0.02704177192
-12 +0.01621630638 9 +0.02385592763
-11 0.01501444846 10 0.02134198140
-10 +0.01930757061 11 +0.01930757061
-9 0.02134198140 12 0.01501444846
-8 +0.02385592763 13 +0.01621630638
-7 0.02704177192 14 0.01501444846
-6 +0.03121077963 15 +0.01397848507
-5 0.03690194680 16 0.01307627930
-4 +0.04513749414 17 +0.01228349256
-3 0.05812356760 18 0.01158135507
-2 +0.08168157087 19 +0.01095515771
-1 0.1380782055 20 0.01039321090
0 +0.5894898722 21 +0.009886107596

Again the signal that is integrated in figure Figure 36 in the spectral domain is used. A good
coincidence in the shape is given the selection of different starting points lead to a shift in the
time domain.


Figure 7: Integration with trigonometric interpolation using 42 filter coefficients
40
6.9 Conclusions
Large differences between differentiation and integration in time domain and frequency domain
occur. This holds especially for those components that have a frequency near to the aliasing
frequency. To reduce this effect it is useful to interpolate time samples before the differentiation
or integration using the sinc-filter. An alternative is the direct usage of a differentiating or
integrating filter that is based on the sinc-filter. The sinc-filter corresponds to a trigonometric
interpolation that is in accordance with the spectrum. Therefore errors like aliasing and leakage
are introduced using this filter. To reduce these errors a higher sampling frequency should be
used. In the case of a differentiating filter, a higher sampling rate together with an adjusted anti-
aliasing filter can lead to high frequency noise components that disturbs the original signal. In
this case a smoothing filter like the Savitzky Golay filter is helpful. An integration of long time
series can lead to undesired trend effects that have to be reduced using a regression polynomial.
The degree of the polynomial needed for the regression increases with the length of the
integration sequence. Beside numerical effects a bias in the measurement chain leads to a bias
that results in the trend. A high pass filter is an alternative for the correction of the trend. This
leads to the integration in frequency domain.
The main difference between the operation in time and frequency domain is based in the type
of the filter. In time domain polynomial smoothing, integration and differentiation is applied.
In the spectral domain trigonometric interpolation is used. Of course, in time domain a
interpolation can be used that is in accordance with the spectral definition.
For a differentiation an odd number of taps (filter coefficients) or an even degree of the
polynomial should be used to avoid a bias, because the deviation is related to one point of the
sequence. For an integration an even number of taps or an odd degree of the polynomial should
be used, because the integral is determined about an interval.


41
7 Spectral Differentiation and Integration
In this chapter the errors introduced by a spectral integration or differentiation is investigated
using theoretical derivations and examples based on sampled signals.
7.1 Leakage
Leakage is a consequence of the periodic continuation of the chosen time window using the
Fourier series transformation, if the signal is not periodic with the window length. If the window
length is T, the spectrum has a discrete grid with the gap grid points Af = 1/T. The maximum
error of the spectral amplitude occurs, if the natural frequency is in the middle between two
grid points. The Fourier transformation is a linear projection, therefore the superposition law is
applicable. In the following section it is assumed that the aperiodic time sequence is a
superposition of sine and cosine functions within the chosen time window. This representation
is complete. The Fourier coefficients of the cosine functions are Ak anand those of the sine
functions Bk.
( ) ( ) ( )
( )
( )
( )
( ) ( )
( )
( )
( )
0
2
2 cos
2 sin
1
sinc
sin
2
2 sin
2 cos
1
2 sin
2 cos
2 sin 2 cos
2 /
2 /
0
0
0
2 /
2 /
0
0
0
0
2 /
2 /
0
0
0
2 /
2 /
0
0 2
0 1
0 0
=
(

= =
= =
(

= =
=
=
+ =
= =
= =
}
}
T
T t
T
T t
T
T t
T
T t
T f
t f
dt t f
T
B
T f
T f
T f
T f
t f
dt t f
T
A
t f f
t f f
t f b t f a t f
t
t
t
t
t
t
t
t
t
t
t
t t
(129)
Critical is a non-zero value of A0. During differentiation the value is neglected, because the
value is weighted with zero. During integration the value is weighted with
0 , 1 e e j . The result is undefined. In a first step the value is set to zero to
simplify the procedure. This assumption is related to a balance of the area of the time function
within the chosen window. This assumption can lead to trend effects, this will be shown later.
( ) | |
(
(
(
(

|
.
|

\
|
+

+
|
.
|

\
|

=
(
(
(
(

|
.
|

\
|
+
+
|
.
|

\
|

=
(
(

+ =
+ = =
|
.
|

\
|
+ |
.
|

\
|
+ |
.
|

\
|
|
.
|

\
|

=
=
|
.
|

\
|
+ |
.
|

\
|

=
|
.
|

\
|
+ |
.
|

\
|

=

}
} }
T f
T
k
j
e e
T f
T
k
j
e e
A
f
T
k
j
e
f
T
k
j
e
T
A
dt e e
T
A
dt e e e
T
dt e t f
T
A
T f
T
k
j T f
T
k
j T f
T
k
j T f
T
k
j
k
T t
T t
t f
T
k
j t f
T
k
j
k
T
T t
t f
T
k
j t f
T
k
j
k
T
T t
t
T
k
j
t f t f
T
T t
t
T
k
j
k
0 0
2 /
2 /
0
2
0
2
2 /
2 /
2 2
2 /
2 /
2
2 2
2 /
2 /
2
0
2 2
2
1
2 2
2
1
2
1
2
1 1
2 cos
1
0 0 0 0
0 0
0 0
0 0
t t
t t
t
t t t t
t t
t t
t
t t
t
(130)
42
(

|
.
|

\
|
+ +
(

|
.
|

\
|
=
(
(
(
(

|
.
|

\
|
+
(

|
.
|

\
|
+
+
|
.
|

\
|

(

|
.
|

\
|

=
T f
T
k
T f
T
k
A
T f
T
k
j
T f
T
k
j
T f
T
k
j
T f
T
k
j
A
k
k
0 0
0
0
0
0
sinc sinc
2
1
2
sin 2
2
sin 2
2
1
t t
t
t
t
t
(131)
( ) | |
( ) | | ( ) | | T f k
jb a
T f k
jb a
F
T f
T
k
T f
T
k jb
T f
T
k
T f
T
k a
F
T f
T
k
T f
T
k j
B
T f
T
k
j
T f
T
k
j
T f
T
k
j
T f
T
k
j
B
T f
T
k
j
e e
T f
T
k
j
e e j
B
f
T
k
j
e
f
T
k
j
e
T
j
B
dt e e
T
j
B
dt e e e
j
T
dt e t f
T
B
k
k
k
k
T f
T
k
j T f
T
k
j T f
T
k
j T f
T
k
j
k
T t
T t
t f
T
k
j t f
T
k
j
k
T
T t
t f
T
k
j t f
T
k
j
k
T
T t
t
T
k
j
t f j t f j
T
T t
t
T
k
j
k
0 0
0 0
0 0
0 0
0
0
0
0
0 0
2 /
2 /
0
2
0
2
2 /
2 /
2 2
2 /
2 /
2
2 2
2 /
2 /
2
0
sinc
2
sinc
2
sinc sinc
2
sinc sinc
2
sinc sinc
2
2
sin 2
2
sin 2
2
1
2 2
2
2 2
2
2
2
1
2 sin
1
0 0 0 0
0 0
0 0
0 0
+

+
+
=
(

|
.
|

\
|
+
(

|
.
|

\
|
+
+
(

|
.
|

\
|
+ +
(

|
.
|

\
|
=
(

|
.
|

\
|
+
(

|
.
|

\
|
=
(
(
(
(

|
.
|

\
|
+
(

|
.
|

\
|
+

|
.
|

\
|

(

|
.
|

\
|

=
(
(
(
(

|
.
|

\
|
+

|
.
|

\
|

=
(
(
(
(

|
.
|

\
|
+

|
.
|

\
|

=
(
(

=
= =
|
.
|

\
|
+ |
.
|

\
|
+ |
.
|

\
|
|
.
|

\
|

=
=
|
.
|

\
|
+ |
.
|

\
|

=
|
.
|

\
|
+ |
.
|

\
|

=

}
} }
t t
t t
t t
t t
t
t
t
t
t t
t t
t
t t t t
t t
t t
t
t t
t
(132)
The maximum error of the amplitude occurs, if the natural frequency f0 is in middle between to
grid points:
6366 . 0
2
1 2
2
1
2
sinc
2
1
2
1
2 sinc
2
sinc
2
1
sinc sinc
2
1
,
2 / 1
0 0
0
~ =
(

~
(

|
.
|

\
|
+ +
(

=
(

|
.
|

\
|
+ +
(

|
.
|

\
|
=
=
+
=
t
t
t
t
t t
l A
T f
T
k
T f
T
k
A
l k
T
l
f
k
k
(133)
43
6366 . 0
2
2
2 2
sinc
2 2
1
2 sinc
2
sinc
2
sinc sinc
2
0 0
j j j
l
j
B
T f
T
k
T f
T
k j
B
k
k
~ =
(

~
(

|
.
|

\
|
+
(

=
(

|
.
|

\
|
+
(

|
.
|

\
|
=
t
t
t
t
t t
(134)
The maximum amplitude loss for the rectangular window is 36.3%. At the low frequency a
small amount from the negative frequency side has to be added. A single frequency leads to a
spectrum that is weighted with a sinc-function. This is problematic during differentiation and
integration, because the side bands are weighted with the wrong factor je during differentiation
or 1/( je) during integration. To reduce the relative error it is useful to shift the integration
towards higher frequency or in the usual case to increase the window length, what leads to a
finer grid in the spectrum. In this case the leakage is not reduced, but the side bands are nearer
to the natural frequency line f0.
If discrete sample points are, leakage products can be mirrored at the aliasing frequency and
lead to additional errors. To avoid this a high sample rate should be used.
The error during differentiation and integration is caused by the effect that the leakage
components are not weighted with the natural frequency factor j2tf0 or 1/(j2tf0), but with the
local frequency factor j2tfk or 1/(j2tfk). The quotient of these factors can be used to estimate
the error that is introduced by the chosen operation. The inverse transformation of the error AGk
or AHk gives the error Ag(t) or Ah(t) for the time series g(t) and h(t).
( ) ( ) ( ) ( )
( ) | | ( ) | |
( ) ( ) ( ) | | ( ) | |
( ) | | ( ) | |
( ) ( ) ( ) | | ( ) | |

=
=
)
`

+
+
= A
)
`

+
+
|
|
.
|

\
|
= A
)
`

|
.
|

\
|
+

+
(

|
.
|

\
|

+
|
|
.
|

\
|
= A
)
`

|
.
|

\
|
+

+
(

|
.
|

\
|

+
=
)
`

+
+
= A
)
`

|
.
|

\
|
+

+
(

|
.
|

\
|

+
|
.
|

\
|
= A
)
`

+
+
|
.
|

\
|
= A
)
`

|
.
|

\
|
+

+
(

|
.
|

\
|

+
=
(

|
.
|

\
|
+

+
(

|
.
|

\
|

+
=
=
c
c
=
k
T
t
k j
k
k
k
k
T
t
k j
k
k
k
k
t
t
e T f k
jb a
T f k
jb a
T f k
kf
j
t h
T f k
jb a
T f k
jb a
T f k j
T
H
T f
T
k jb a
T f
T
k jb a
f j k j
T
H
T f
T
k jb a
T f
T
k jb a
k j
T
H
e T f k
jb a
T f k
jb a
T f k
T
j
t g
T f
T
k jb a
T f
T
k jb a
f
T
k
j G
T f k
jb a
T f k
jb a
f j
T
k j
G
T f
T
k jb a
T f
T
k jb a
T
k j
G
T f
T
k jb a
T f
T
k jb a
F
dt t f t h t f
t
t g
t
t
t t
t t
t t
t t
t t
t t
t t
t t
t t
2
0 0 0
0
0 0
0
0 0
0
0 0
2
0 0 0
0 0 0
0 0 0
0 0
0 0
0
0
sinc
2
sinc
2 2
sinc
2
sinc
2
1 1
2
sinc
2
sinc
2 2
1
2
sinc
2
sinc
2 2
sinc
2
sinc
2
2
sinc
2
sinc
2
2
sinc
2
sinc
2
2
2
sinc
2
sinc
2
2
sinc
2
sinc
2
,
0
(135)
These functions cannot be handled analytically anymore. Therefore numerical examples have
to be chosen.
44
7.2 Investigations with a Tapered Window
At this point the tapered windows is investigated. The tapered window manipulates the time
series only in the last 10% to the window border and leaves most of the time sequence
unchanged. The tapered window is defined with:
( )

s s
|
.
|

\
|

< <
s s
|
.
|

\
|

=
T t T
T
t T
T t T
T t
T
t
t w
10 / 9 ,
2
10 /
cos 1
10 / 9 10 / , 1
10 / 0 ,
2
10 /
cos 1
t
t
(136)
The following assessment can be done for the derivation of the windowed function. Using the
product rule the result of the derivation is:
( ) ( ) | | ( ) ( ) ( ) ( )
( )
( ) ( ) | | ( ) ( )
( )
( ) 0 , =
c
c

c
c
=
c
c
c
c
+
c
c
=
c
c
t w
t w
t w
t
t f t w t f
t
t f
t
t w
t
t f t w t f
t
t w t f
t
(137)
The derivative ( ) ( ) | | t w t f
t c
c
is determined using the spectral derivation. The window ( ) t w and
its derivative ( ) t w
t c
c
have to be determined once. The function ( ) t f is given and therefore
known. The derivative of the tapered window is:
( )

s s
|
.
|

\
|

< <
s s
|
.
|

\
|
=
c
c
T t T
T
t T
T
T t T
T t
T
t
T
t w
t
10 / 9 ,
2
10 /
sin
10 /
10 / 9 10 / , 0
10 / 0 ,
2
10 /
sin
10 /
t
t
t
t
(138)
Using the tapered window and observing the part of the window, where the window is constant:
( ) ( )
( ) ( ) ( ) | | t w t f
t
t f
t
T t T t w
t
t w
c
c
=
c
c
< < =
c
c
= 9 . 0 1 . 0 , 0 , 1
. (139)
The derivative ( ) ( ) | | t w t f
t c
c
is determined using the spectral differentiation method. The
derivative in the constant part of the window is not changed by the window.
Similar holds for the integration, only the initial values at the beginning of the window has to
be determined additionally. This can be done with overlapping windows, there the value at the
end of the constant part to the window is used as the starting value of the constant part of the
following window.
If a tapered window or another window, which has a constant value of one in the middle part
the following holds:

45
( )
( )
( )
( )
( )
( )

s s
c
c
< <
s s
c
c
=
c
c

s s
< <
s s
=
T t t t w
t
t t t
t t t w
t
t w
t
T t t t w
t t t
t t t w
t w
b b
b a
a a
b b
b a
a a
,
, 0
0 ,
,
,
, 1
0 ,
(140)
The rule for partial integration gives:
( ) ( )
( ) ( ) ( ) ( ) | | ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) C dt t w t f t F
dt t w
t
t F w F dt t w t f t F
t w
dt t w
t
t F w F dt t w t f
t F
dt t w
t
t F w F t w t F dt t w t f
dt t w
t
t F t w t F dt t w t f
t w
t
t w t t t
t
t
C
a
t
t
a
t
t
a
t
t
t
t
a
t
t
t
t
t
t
t
t
t
t
b a
a
a
+ =
c
c
+ + =
c
c
+ +
=
c
c
=
c
c
=
=
c
c
= < <
}
} }
} }
} }
} }
=
= =
= =
= =
= =
=
0
0
0
0 0
0 0
0
0
0
0 0
0
0
0 0
0
0 0
0 0
0 0
0
0 0 0
0 0
0 0
0 0
0 , 1 :

(141)
The integral ( ) ( )
}
=
0
0
t
t
dt t w t f is determined using the integration in the spectral domain. The
equation shows, that in the constant region the integral is not affected by the window. Only the
initial value C has to be derived. This can be done using overlapping windows. The overlapping
has to be chosen in that way, that the constant region from one window to the next have contact.
From here on simulations with a sine signal are used, because further on no analytical results
can be derived.

Initially the leakage is investigated. Therefore 11 lines between two frequencies are introduced
between two neighboured frequencies of the grid. The maximum loss occurs at 6/11 spacing.
The results without a window coincide with the theoretical predictions.
46

Figure 8: Leakage with a rectangular window using 16384 samples (spectrum)


Figure 9: Leakage with a rectangular window using 1024 samples (spectrum)
47

Figure 10: Leakage with a rectangular window using 256 samples (spectrum)

A leakage of 36% for the maximum amplitude is found independently from the size of the
window.


Figure 11: Tapered window with 16384 samples (spectrum)

48

Figure 12: Tapered window with 1024 samples (spectrum)


Figure 13: Tapered window with 256 samples (spectrum)

The tapered window reduces the amplitude loss from 36% to nearly 20%. The maximum value
is 0.9. This is a reaction of the window near the borders, which lead to a loss of 10% of the area
of window compared with a rectangular window.
49
7.3 Differentiation with and Without Tapered Window
Next step is the derivation of the error in the time domain using the derivation in the spectral
domain. The relative error is the deviation of the numerical differentiation in the spectral
domain compared with the analytic derivative of the original function divided by the amplitude
of the analytical derivative.

Figure 14: Relative error during differentiation with a frequency (10+6/11)/16384 with 16384 samples

Figure 15: Relative error during differentiation with a frequency (10+6/11)/1024 with 1024 samples
50

Figure 16: Relative error during differentiation with a frequency (10+6/11)/256 with 256 samples


Figure 17: Relative error during differentiation with a frequency (10+6/11)/16384 with 16384 samples and
a tapered window
51

Figure 18: Relative error during differentiation with a frequency (10+6/11)/16384 with 16384 samples and
a tapered window (Zoom)

Figure 19: Relative error during differentiation with a frequency (10+6/11)/1024 with 1024 samples and a
tapered window
52

Figure 20: Relative error during differentiation with a frequency (10+6/11)/256 with 256 samples and a
tapered window

An increase of the sample rate does not give an improvement on the results. A tapered window
gives the possibility to reduce the error in the constant part nearly to zero, but the boundaries
are deformed.
The next step is a widening of the time window. Because this shall be done in dimensionless
form, the frequency f0 is increased.

Figure 21: Relative error using differentiation with a frequency of (168+6/11)/16384 with 16384 samples

53

Figure 22: Relative error using differentiation with a frequency of (168+6/11)/16384 with 16384 samples
and Tapered Window

Figure 23: Relative error using differentiation with a frequency of (168+6/11)/16384 with 16384 samples
and Tapered Window (Zoom)

An elongation of the time window gives an important improvement. The additional
improvement from the tapered window becomes less important. Interesting is, that the error
near the border is slightly reduced compared with the rectangular window.

54
7.4 Integration with and Without Tapered Window
Consequently now the error using interaction is analysed.

Figure 24: Relative error using integration with a frequency of (10+6/11)/1024 with 1024 samples

Figure 25: Relative error using integration with a frequency of (10+6/11)/16384 with 16384 samples

55

Figure 26: Relative error using integration with a frequency of (168+6/11)/16384 with 16384 samples

Figure 27: Relative error using integration with a frequency of (10+6/11)/1024 with 1024 samples and
tapered window

56

Figure 28: Relative error using integration with a frequency of (10+6/11)/ 16384 with 16384 samples and
tapered window

Figure 29: Relative error using integration with a frequency of (168+6/11)/ 16384 with 16384 samples and
tapered window
57

Figure 30: Relative error using integration with a frequency of (168+6/11)/ 16384 with 16384 samples and
tapered window (Zoom)

For the spectral integration long time windows and window functions are essential to reproduce
the trend correctly. An alternative is a high pass filter, to reduce the trend.

One step further is a numerical test whether the leakage reproduces beats correctly. Therefore
two sine signals are used. The relative error r(t) is the difference of value reproduced by means
of a FFT with a window ( ) t F
~
and the theoretical value normalized by the average value of the
two amplitudes of the integrated function.
( )
( )
( )
( ) ( ) ( )
|
.
|

\
|
+ +
|
.
|

\
|
+

=
|
|
.
|

\
|
|
.
|

\
|
+
|
.
|

\
|
+ +
|
|
.
|

\
|
|
.
|

\
|
+
|
.
|

\
|
+ =
|
|
.
|

\
|
|
.
|

\
|
+ +
|
|
.
|

\
|
|
.
|

\
|
+ =
11
6
169
2
11
6
168
2
~
2
11
6
169 2 sin
11
6
169
2
11
6
168 2 sin
11
6
168
2
11
6
169 2 sin
11
6
168 2 sin
T T
t F t F
t r
T
t
T T
t
T
t F
T
t
T
t
t f
t t
t
t
t
t
t t
(142)
The error is slightly smaller than the error for a single sine signal. The integrated signal shows
a node in the middle of the interval.

58

Figure 31: Relative error for two sine signals with maximum leakage and tapered window (Zoom)


Figure 32: Signal in the trespassing region

The beat is not changed by a spectral integration. If a lower frequency is chosen the error during
differentiation and integration increases.
59
( )
( )
( )
( ) ( ) ( )
|
.
|

\
|
+ +
|
.
|

\
|
+

=
|
|
.
|

\
|
|
.
|

\
|
+
|
.
|

\
|
+ +
|
|
.
|

\
|
|
.
|

\
|
+
|
.
|

\
|
+ =
|
|
.
|

\
|
|
.
|

\
|
+ +
|
|
.
|

\
|
|
.
|

\
|
+ =
11
6
11
2
11
6
10
2
~
2
11
6
11 2 sin
11
6
10
2
11
6
11 2 sin
11
6
10
2
11
6
11 2 sin
11
6
10 2 sin
T T
t F t F
t r
T
t
T T
t
T
t F
T
t
T
t
t f
t t
t
t
t
t
t t
(143)
Only the error in the trend increases. Nevertheless it becomes again visible that long time
windows should be used.


Figure 33: Relative error for two sine signals with maximum leakage with a tapered window (Zoom)
60

Figure 34: Signal in the trespassing region

As a final step the behaviour using differentiation is investigated. It is shown that the relative
error is still about 10
-6
.

Figure 35: Relative error using two sine signals with maximum leakage with tapered window (Zoom)
61

Figure 36: Signal in the trespassing region
7.5 Correction of the Spectral Integration
Up to now the constant coefficient of the Fourier spectrum was set to zero. It is well known that
the integration about a constant is a linear function. Therefore the following procedure is
suggested. An arbitrary starting point within the window can be used:
( ) ( )
( )
( )
( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) T t x T t X e X t x
t x t X e X t x
T
t x
T
t X e X t x
X
j
X
T
k
dt e t x
T
X
k
k
t j
k
k
k
t j
k
k
k
t j
k
k
k k
k
k
k
T
T t
t j
k
k
k
k
k
4 . 0 2 / 0
or
0 0
or
2 2
0
0 , 0
0 ,
1
2 ,
1
2 /
2 /
= + + = + =
= + = + =
|
.
|

\
|
= +
|
.
|

\
|
+ = + =

=
=
=
= =

=
=

e e
e e
e e
e
e e
e
e
t e e
e
e
e
e

(144)
Examples for a starting point are:
- Position at the window start,
- Position in the middle of the window, (This point can also be used for infinite long
windows.)
- Starting point at the start of the trespassing region of the chosen window.
The continuous Fourier transformation of a linear function is the derivative of a generalized
Dirac delta function in the spectrum:
62
( ) ( ) ( ) ( ) ( ) ( ) ( ) e o e o
e
e
t
e e
e
0 0
0
2
1
0 x X j dt e t x X x t X t x
t
t j
+
c
c
= = = + = =
}

(145)

The series Fourier transformation gives different results:
( ) ( )
( ) ( )
( ) ( ) { }

(
(
(

=
=
e = =
=
(

= = = =
= = =
}
}
}
} }
=

= = =
2 /
2 /
2
2 /
2 /
2
0
2 /
2 /
2
0
2 /
2 /
2
2 /
2 /
2
0
2 /
2 /
0
2 /
2 /
0
0
2 2
1
0 / ,
1
0
2
1 1
0
0
T
T t
t
T
k
j
T
T t
t
T
k
j
k
T
T t
t
T
k
j
k
T
T t
t
T
k
j
k k
T
T t
T
T t
T
T t
dt
T
k
j
e
T
k
j
e
t
T
X
Y
dt e t X
T
Y
Z k dt e t y
T
Y Y
t
T
X
dt t X
T
dt t y
T
X Y
t X t X t y
t t
e
e
t t
t
t

(146)
( ) ( ) | |
( ) ( ) | | { }
( ) | |
( )

=
=

=
=
e =
+ =
(
(
(
(

|
.
|

\
|

|
.
|

\
|
+
=
(
(
(
(

|
.
|

\
|
+
=

(
(
(
(

|
.
|

\
|

(
(
(

0 , 0
0 ,
2
1
1
2
0 / , sin cos
2
cos 2 sin 2
4
2
1
2
1
2
2 1
2
2
0
2 2
0
2 2
0
2 2
0
2 2
0
2 /
2 /
2
2
0
2 /
2 /
2
2
2 /
2 /
2
0
k
k
k
T X
j
Y
k
k
T X j
Y
Z k k k k
k
T X j
Y
k k j k j
k
T X
Y
e
T
k
k j
e
T
k
k j
T
X
e
T
k
t
T
k
j
T
X
Y
T
k
j
e
T
k
j
e
t
T
X
Y
k
k
k
k
k
k
k j k j
T
T t
t
T
k
j
k
T
T t
t
T
k
j
T
T t
t
T
k
j
k
t
t
t
t t t
t
t t t
t
t
t
t
t
t
t
t
t
t t
t
t t

(147)
This spectrum has to be added to the original spectrum, for a correct representation of a linear
trend.
Next step is the correct representation of the offset:
( ) ( )
( ) | |
0
2 /
2 /
0
2 /
2 /
0
2 /
2 /
0
0
1
0
x t
T
x
dt
T
x
dt t z
T
Z
x x t z
T
T t
T
T t
T
T t
= = = =
= =
=
= =
} }
(148)
63
( )
| | ( ) | |
( )
{ }

=
=
=
e = = =
(
(
(

= = =

} }
0 , 0
0 ,
0 / ,
sin
sin 2
2 2
2
1
0
0 0 0
2 /
2 /
2
0
2 /
2 /
2
0
2 /
2 /
2
k
k x
Z
Z k
k
k x
k j
k
x
j e e
k
x
j Z
T
k
j
e
T
x
dt e
T
x
dt e t z
T
Z
k
k j k j
k
T
T t
t
T
k
j
T
T t
t
T
k
j
T
T t
t
T
k
j
k
t
t
t
t t
t
t t
t
t t
(149)
This spectrum has to be added for a correct reproduction of an offset. Summing up offset and
linear trend the following correction is needed:
( )
( )

=
=

= + =
0 ,
0 ,
2
1
0
0
0 0
k x
k
k
T X
j
V x t X t v
k
k t

(150)
This leads to the following improved procedures.
7.5.1 Procedure in Time Domain
The first step of the procedure is the integration in the spectral domain. This is done setting the
offset
0
X

to zero. The result after the inverse transformation is the time series x(t). It is
supposed, that a tapered window with an overlap of 50% of the window length (T/2) is used.
To avoid Gibbs effects a gap of 20% of the window length (0.2T) is chosen for the evaluation
region. The result is an overlapping region from [-0.3T,-0.2T] and [0.2T,0.3T]. In this region a
linear increasing or decreasing participation factor is chosen.
2 /

2 / 3 . 0 , 0
3 . 0 2 . 0 , ) 4 . 0 ( 10
2 . 0 2 . 0 , 1
2 . 0 3 . 0 , ) 4 . 0 ( 10
3 . 0 2 / , 0
T t t
T t T
T t T t T
T t T
T t T T t
T t T
w
i i
i
i
i
i
i
i
=

s <
s <
s s
< s +
< s
=
(151)
Often in the FFT the starting point is fixed at the beginning of the window 0

=
i
t . For this case
the following conversion is needed: 2 /

T t t
i i
= .
The starting value of the new window is evaluated from the end value of the predecessor. This
is done to reduce the influence of Gibbs effects using the average value of the overlapping
region.
T t
T T t
T T t
or
T t
T t
F
i M
i M
i M
i M
1 . 0
2 / 25 . 0
2 / 25 . 0

25 . 0
25 . 0
,
1 ,
,
1 ,
=
+ =
+ =
=
=

(152)
64
( ) ( )
( ) ( ) ( )
1 , 0 , 1 , , 0
2 /

2 /

0 ,
2 /
2 /
0 ,
,
,
,
,
2 /

1
1

+
=
+
=
+ =
+ =
+ =
}
}
i i M i M i
t t
t t t F
i M
t t
t t t F
i M
x x x x
dt T t X t x
t
x
or
dt t X t x
t
x
F i M
F i M
F i M
F i M
(153)
This gives the time series. The starting value
0 , 0
x has to be known or set to zero.
( ) ( ) ( ) | |
( ) ( ) ( ) ( ) | |
i i i i
i i i i
x T t X t x t w t x
or
x t X t x t w t x
, 0 0
, 0 0
2 /

+ + =
+ + =
(154)
The complete signal is the superposition of the functions xi(t) with an appropriate shift in time.
7.5.2 Procedure in Frequency Domain
The first step is again the integration in the spectral domain, where the zero component
0
X

is
set to zero. The next step is the addition of the correction spectrum
k
Y using the DFT. Usually
the DFT determines the power and not the energy. For this region a factor 1/T has to be
multiplied with the correction spectrum. If the starting point is posed at the beginning of the
spectrum a phase correction is needed.
( )
( )
( )
( ) ( )
Z k
kT
T X
j
kT
T X
j Y
T
e Y
T
Y
dt e t X e
T
dt e t X
T
Y
T t t
t d e T t X
T
Y
k k
k
k
k j
k k
T
t
t
T
k
j T
T
k
j
T
t
T t
T
k
j
k
T
t
t
T
k
j
k
e =

= = =
= =
=
=

=

=
+
=

} }
}
,
2 2
1 1
1
1 1

1 1

2 /

2 /

0 0
0

2
0
2 / 2
0

2 / 2
0
0

2
0
t t
t
t t t
t

(155)
Missing is now the offset. This component shall be determined using the average of the
overlapping region. Here the modified Plancherels theorem is needed.
( ) ( ) ( ) ( )
( ) ( )
( ) ( ) | |
( )
dt t t t H t t t H
t
t X t x dt t X t x
t
x
t
t g
F M F M
F
t y t x
t t
t t t F
i M
F i M
F i M
} }

=
+
+
=
+ + = + =


2 / 2 /
1 1
0
2 /
2 /
0 ,
,
,
(156)
( ) ( ) ( ) ( )
( ) ( )
( )
( ) ( )
( )
( )
( )
( )
( ) ( )
F
ft j
F
ft j
F
F
ft j
F
F
ft j
F
ft j ft j
ft j
F
t t f j t t f j
t t
t t t
ft j
F
t t
t t t
ft j
F t
ft j
F
f
i M
ft e f G
ft e
ft
ft
e f G
ft j
ft j
e
ft j
e e
e
ft j
e e
f G
f j
e
t
dt e
t
dt e t g
t
f G
df f G f Y f X x
M
M M
M
F F
M
F M F M
F M
F M
F M
F M
t
t
t
t
t
t
t t
t
t
t t
t
t t
t
t t
t
t t
sinc
sinc
sin
2
sin 2
2 2
2
1 1 1
2
2 2
2
2 / 2 2 / 2
2
2 / 2 2 / 2
2 /
2 /
2
2 /
2 /
2 2
,
=
= =

=
(

= = =
+ =

+
+
=

+
=

=
} }
}
(157)
65
If the starting point is at the beginning of the time window the following correction is needed.
( ) ( )
( ) ( ) ( ) ( )
( )
( ) ( ) ( ) ( ) f Z f Y f X f X
f x
f
f Z
x x x x
df f G f Y f X x
ft e f G
i
i i M i M i
f
i M
F
t f j
M
+ + =

=
=
=
+ =
+ =
=

=
}

0 ,
0 , 0

sinc

, 0
1 , 0 , 1 , , 0
,

t
t
(158)
As a last step the complete spectrum is transformed back into time domain and masked with
the overlapping window.
The example used in Figure 36 shall be corrected here.


Figure 37: Signal after integration
The signal starts and ends exactly at zero. This correction was successful. But the tapered
window leads to an undesired trend in the trespassing region.

66

Figure 38: Relative error of the signal in the trespassing region
The correction has to be made with the values that were derived using a rectangular window.
The offset can be determined explicit using the time average.
( )
}
=
=
2 /
2 /
0
1
T
T t
dt t x
T
X

(159)

Figure 39: Relative error of the signal in the trespassing region

Really the error is completely vanishing using the suggested correction.


67
7.6 Conclusions
With a tapered window a slight improvement of the integration and differentiation is possible
without affecting the whole window. Especially for integration the window is useful, if no high
pass is used. Using the window the relative error is about 10
-6
and therefore very small.

It has to be stated, even if the error produced in the spectrum is high the error in the integrated
or differentiated time series is small. The error increases with the number of integrations or
differentiations applied to the signal.

It has to be taken care, because the sinc functions from the leakage can be mirrored at the
aliasing frequency. To avoid these components a gap between the usable frequency region and
the aliasing frequency should be reserved.

There is a big similarity between Integration, Hilbert transformation and differentiation in
frequency domain, because all operations rotate the phase by t/2.
( ) ( ) ( ) ( ) e e e e
e e
e j D j H j
j
I = =

= = , sgn ,
1 1
(160)
Using the integration without a high pass trends occur. To reduce these trends the window
length can be increased. An offset cannot be taken into the spectral integration. But it is possible
to do this correction that is related to the zero component of the spectrum after the inverse
transformation.

If long time windows are chosen to keep the error during spectral integration or differentiation
small. A long time window does not reduce the amplitude loss caused by leakage but
compresses the sinc function of the leakage in its spectral width. In the following figure the
weighting using differentiation and integration neglecting the imaginary unit is presented. In
the exact approach the complete sinc Function in the spectrum has to weighted with 2tf0 or
1/(2tf0). In practical applications the components are weighted with 2tfk or 1/(2tfk). If the
relative error is observed, that is done in spectrum, the error is large for low frequencies and
small for high frequencies f0.

To keep the relative error constant the usage of wavelets or logarithmic filter banks should be
taken into account, because wavelets use a logarithmic scale.

Because the frequency f0 cannot be changed in practise a widening of the window has the same
effect: The bock size becomes smaller, what is equivalent with an increase of the frequency.
68




Figure 40: Behaviour of the error caused by leakage

2tf
-1/(2tf)
large relative error small relative error
f0 f0
f
69
8 Correction of Integrating A/D Converters
To increase the resolution of analogue to digital converters (A/D Converters) today often Sigma
Delta Converters are used. These converters use a parallel grid of condensers together with a
temporal integration. It has to be analysed, how these time integration influences the
reproduction of high frequency components of the signal.
8.1 Correction of Integrating A/D Converters in Frequency Domain
It is assumed that the converter integrates about the time interval tw. Additionally the
superposition that is applicable in case of integration is used. The signal is split into a sum of
cosine functions. The behaviour of one component f(t) is investigated. To avoid phase shifts the
time instance belonging to the integral is positioned at the mid of the integration interval.
( ) ( )
( ) ( ) ( )
( )
( )
( ) ( )
( )
( ) ( )
( )
( ) ( )
( )
( )
( ) ( )
w
w
w
w
w
w
w
w w
t
t
t
t t
w
t
t
t
t t
w
t
t
t
t t
w
t f t f
t f
t f
t f
t f
t f t f
t g
b a b a b a b a b a b a b a
b a b a b a
t f b t f a
t f
t f t f t f t f
t g
f
t f
t
dt t f
t
dt t f
t
t g
t f t f
w
w
w
w
w
w
0 0
0
0
0
0
0 0 0 0
0
0 0 0 0
0
0 0 0 0 0 0 0 0
0
2
2
0
0 0
2
2
0 0
2
2
0
0 0
sinc
sin
2
sin 2 cos 2
sin cos 2 sin cos cos sin sin cos cos sin sin sin
sin cos cos sin sin
, 2
2
2 sin 2 sin
2
2 sin 1
2 cos
1 1
2 cos
0
0
0
0
0
0
t
t
t
t
t t
t t
t
t t t t
t
t
t
t
= =
+
=
= + + = +
=
= + =
+ + +
=
(

+
= + = =
+ =
+
=
+
=
+
=
} }
(161)

The amplitude is attenuated by a frequency dependent sinc function. The sinc function can
presented in relation with the Frequency f0. In the diagram the integration time is set to one
tw =1 and f0 varies from 0 to 1.
70

Figure 41: Amplitude attenuation of an integrating converter

If the spectrum of the signal and the integration time tw of the converter are known, the
attenuation can be corrected using the inverse function C(f) in the spectrum.
( )
( )
w
w
ft
ft
f C
t
t
sin
= (162)
An integrating filter behaves like a simple anti-aliasing filter.

If an integrating filter uses the total sampling time for a sampling rate fs = 1/tw. The aliasing
frequency is posed at fa = fs /2. For this frequency the sinc-Funktion has the value 2/t= 0.6366.
The maximum correction is therefore t/2 = 1.5708. Without a correction the maximum
amplitude error is 36.3%.
8.2 Correction of Integrating A/D Converters in Time Domain
For the determination of the correction filter in time domain the inverse transformation of the
correction function is needed. Problematic is the singularity at the spectral points
{ } 0 / , Z n n ft
w
e = . To avoid these spectral bins it is assumed, that the maximum integration
time is equivalent with the sampling interval. Therefore no singularities in the spectrum are
present.
( )
( )
}
=
=
= A s
2 /
2 /
2
sin
1
s
s
f
f f
ft j
w
w
s
w
df e
ft
ft
t c
f
t t
t
t
t
(163)
The Integrand is symmetric, so the following simplification is possible:
71
( )
( ) ( ) ( )
( )
( ) ( )
( )
( ) ( ) ( )
( )
( )
( )
( )
}
} } }
} }
} } }
=
=

= =

= =

= = =
=
+ = + =
+

=
=
+ = =
2 /
0
2 /
0
2 2
2 /
0
2
2 /
0
2
2 /
0
2
0
2 /
2
2 /
0
2
0
2 /
2
2 /
2 /
2
2 cos
sin
2
sin sin sin
sin sin
sin sin sin
s
s s s
s
s
s
s
s
s
f
f w
w
f
f
ft j ft j
w
w
f
f
ft j
w
w
f
g
gt j
w
w
f
f
ft j
w
w
f g
gt j
w
w
f
f
ft j
w
w
f f
ft j
w
w
f
f f
ft j
w
w
df ft
ft
ft
t c
df e e
ft
ft
df e
ft
ft
dg e
gt
gt
t c
df e
ft
ft
dg e
gt
gt
t c
f g
df e
ft
ft
df e
ft
ft
df e
ft
ft
t c
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t
t t t
t t
t t t
(164)
The integral cannot be solved analytically. Therefore a Taylor expansion about f=0 is used. The
function is symmetric and all odd derivatives are zero. The determination of the derivatives is
done using Hospitals rule.
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( )
( ) ( ) ( ) ( ) ( )
10
10
8
8
6
6
4
4
2
2
0 0
0
0
8 10
9
8 8
7
6 6
5
4 4
3
2 2
1
119750400
2550
604800
127
15120
31
360
7
6
1
!
0
!
33 / 2550 0
0 0
15 / 127 0
0 0
21 / 31 0
0 0
15 / 7 0
0 0
3 / 0
0 0
1 0
, ,
sin
f
t
f
t
f
t
f
t
f
t
f C
f
f i
C
f f
f i
f C
f C
a C
C
a C
C
a C
C
a C
C
a C
C
C
t a f C
f
f C
af
af
f C
w w w w w
i
i
i
i
i
i
i
i
w
n
n
n
t t t t t
t
+ + + + + ~
c
c
=
c
c
=
=
=
=
=
=
=
=
=
=
=
=
= =
c
c
=


=

=
(165)
72

Figure 42: Approximation of 1/sinc function with a Taylor series up to twf=t/2

Figure 43: Relative error AC between 1/sinc and the Taylor series up to twf=t/2
73
( )
( ) ( ) ( )
( ) ( )
( )
}
=
(
(
(
(

+ +
+ + + +
~
2 /
0 10
10
8
8
6
6
4
4
2
2
2 cos
3421440
73
604800
127
15120
31
360
7
6
1
2
s
f
f
w w
w w w
df ft
f
t
f
t
f
t
f
t
f
t
t c t
t t
t t t
(166)
The error between the function and the approximation is less than one per mille.
( )
( ) ( ) ( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( )
w
w w w w w w
w w w w w
ft
ft
f
t
f
t
f
t
f
t
f
t
C
f C
f
t
f
t
f
t
f
t
f
t
f C
C
t
t t t t t t
t t t t t
sin
3421440
73
604800
127
15120
31
360
7
6
1 1
3421440
73
604800
127
15120
31
360
7
6
1
10
10
8
8
6
6
4
4
2
2
10
10
8
8
6
6
4
4
2
2
(

+ + + + + = A
(

+ + + + +
= A
(167)

The result of the integration c(t) is gained with Maple [4].
(168)
8.3 Conclusions
If an integrating converter is used, the amplitudes of the high frequency components are
attenuated. The error can be compensated using the inverse of a sinc-function.
An analytic determination of the related filter in the time domain is impossible due to the
singularities of the spectral function. For usual range of an integrating converter a Taylor series
can be derived and transformed back into time domain.
74
9 Exponential Smoothing
In the beginning of this chapter the bias of the discrete smoothing filter related to the continuous
version is derived and improved versions of the discrete filter are presented. In the second part
of the spectral equivalent is derived and simplifications are presented that allow to transform
filter banks with exponential smoothing from emission to immission using Fourier integral
transformation for the numerical part.
9.1 Continuous Smoothing
The definition of the exponential smoothing is based on the integration time T.

( ) ( )
( ) ( )
( ) ( )
0 , 1 ,
1
1 1 1 1
: Test
1
~
1
~
0
0
2
2 2
= = = =
= =
(

= =
=
=

=
= =

} }
}
}
t
t

t
t
t t
t t

t
t
t
t
t
t
t
d
d
t
T
T
Te
T
d e
T
d e
T
d e x
T
t x
d e x
T
t x
t x t x
T T
t
T
t
t
T
t
f f
t
T
t
f f
f
BP
(169)
75
9.2 Discrete Representation of Exponential Smoothing Functions
In this chapter it will be analysed, with which model the traditional discrete exponential
smoothing approximates the continuous form. The classical version produces a bias. Therefore
two modified versions will be presented here that do not have a bias with respect to the
continuous form.
9.2.1 Discrete Approximation Based on the Assumption of Blocks
Scaled With End Values
This is as one can see the traditional form of the discrete exponential smoothing algorithm. It
can be interpreted as a block algorithm using the ending value for the amplitude of the block in
continuous form. This produces a bias. The index f for band pass filtered values is not used in
the ongoing part of this section. The sample interval is At.
( )
( )
( ) ( )
( ) ( )
1 1
: Test
~
1
~
1 1 1
~
1 1
~
1
1
~
2
1
2 2
~
1
1
2 2
1
1
2 2 2
1
2 2 2 2
2
1
2 2
2
1
= +
|
|
.
|

\
|

+
|
|
.
|

\
|
=
|
|
.
|

\
|
+
|
|
.
|

\
|
=
(

+
|
|
.
|

\
|
=
(

+
|
|
.
|

\
|
=
|
|
.
|

\
|
=
(
(

|
|
.
|

\
|
=
(
(

=
A

=
A + A

=
A + A

=
A A

=
A A

=
A A

=
A
A =
A

}
T
t
T
t
n
T
t
n
T
t
n
x
n
i
T
t n i
i
T
t
T
t
n
T
t
n
i
T
t n i
i
T
t
n
T
t
n
n
i
T
t n i
i n
T
t
n
i
T
t n i
i
T
t
n
n
i
T
t n i
T
t
i
n
i
t i
t i
T
t n
i n
e e
x e x e x
e x e e x e e x e x e x
e x x e e x e x
e e x d e x
T
x
n

t
t
t
(170)














At
xn-1 xn
76
9.2.2 Discrete Approximation using Blocks Scaled with Average Value
To avoid a bias the blocks are shifted by half of the interval width and a block with half the
width of the sample interval is added. So blocks are weighted with the amplitude in the middle
of the interval.
( )
( )
( )
( )
( )
( )
1 1
: Test
~
1
~
1
1 1
~
1
~
1
~
1
~
2 2
2
1
2
1
2
2
2
2
~
2
2 2
1
2 2
1
2
2
1
2 2 2
2
2
2
1
2 2
2 2
1
2 2
2
2
2
2
2
1
2 2
2 2
2 / 1
2
1
2 / 1
2 / 1
2 2
2
1
= + +
+
|
|
.
|

\
|
+
|
|
.
|

\
|
=

(
(

|
|
.
|

\
|
+
|
|
.
|

\
|
+
+
|
|
.
|

\
|
+ +
|
|
.
|

\
|
=
(
(

|
|
.
|

\
|
+
|
|
.
|

\
|
+
|
|
.
|

\
|
=
|
|
.
|

\
|
+
(
(

|
|
.
|

\
|
=

+
(
(

=
A

=
A

A A +

A A

=
A

A A

A A

=
A

A A
A
A =
A

=
A +
A =
A

}
T
t
T
t
T
t
T
t
n
T
t
n
T
t
T
t
n
T
t
n
x
n
i
T
t
T
t
T
t n i
i n
T
t
T
t
n
T
t
T
t
T
t
T
t
n
T
t
n
n
i
T
t
T
t
T
t n i
i n
T
t
T
t
T
t
n
T
t
n
T
t
n
n
i
T
t
T
t
T
t n i
i n
t n
t n
T
t n
n
n
i
t i
t i
T
t n
i n
e e e e
x e x e e x e x
e e e x x e e
x e e e e x e x
e e e x x e e e x e x
e x e e e x x
d e x d e x
T
x
n

t
t
t
t
t t
(171)










At
xn xn-1
77
9.2.3 Assumption of a Linear Interpolation of the Discrete Time Series
This is the best approximation to the continuous form. A linear function is interpolated between
two sample values. But this does not produce an additional complexity to the filter sequence.
Still only the actual value, the value before and the last filtered value are needed.
( )
( )
( )
( )
( )
1 1 1
: Test
~
1 1
~
1 1
1 1
~
1 1
1 1
~
1 1
~
1
1
~
2
1
2
1
2 2
~
1
2 2
1
1
2
1
2 2
1
2 2
1
2 2
1
2
2 2
1
2
1
2 2
1
2
2
1
= +
(

A
+
|
.
|

\
|
A
+
(

|
.
|

\
|
A
+
A
+
(

A
+
|
.
|

\
|
A
+
(

|
.
|

\
|
A
+
A
=

|
.
|

\
|
A
+
A
+
(

A
+
|
.
|

\
|
A
+
(

A
+
|
.
|

\
|
A
+
(

|
.
|

\
|
A
+
A
=

|
.
|

\
|
A
+
A
+
(

A
+
|
.
|

\
|
A
+
(

|
.
|

\
|
A
+
A
+
(

A
+
|
.
|

\
|
A
=

|
.
|

\
|
A
+
A
+
(

A
+
|
.
|

\
|
A
=

|
.
|

\
|

A
+
|
.
|

\
|
A
=
A

=
A

A + A

=
A

A
A

=
A

A
=
A
A =
A

}
T
t
T
t
T
t
n
T
t
n
T
t
n
T
t
n
x
n
i
T
t
i
T
t
i
T
t n i
T
t
T
t
n
T
t
n n
n
i
T
t
i
T
t
i
T
t n i
T
t
n
T
t
n n
n
i
T
t
i
T
t
i
T
t n i
n
n
i
t i
t i
T
t n
i i n
e e
t
T
t
T
e
t
T
t
T
x e x e
t
T
t
T
x e
t
T
t
T
x
e
t
T
t
T
x e
t
T
t
T
x e e
e
t
T
t
T
x e
t
T
t
T
x x
e
t
T
t
T
x e
t
T
t
T
x e
e
t
T
t
T
x e
t
T
t
T
x x
e
t
T
t
T
x e
t
T
t
T
x e x
d e i
t
x
t
i x
T
x
n

t
t
t
t t
(172)







xn xn-1
At
78
9.2.4 Stability of the Filter
All filters are of the same type. The z-transform is applied to determine the poles and zeros.
( ) ( )
( )
1 1
0
1
0
1
1 0
1
1
1
1 0
1
1
1
1 0
1 1 1 1 0
z
: Pole
z
: Zero
1
~ ~
g
h
h
g z
h z h
z g
z h h
z H
z z H g z h h z H
g y h y h y y
n n n n
=
=

+
=

+
=
+ + =
+ + =

(173)
For the classical filter the poles and zeros are:
T
t
e
A

=
=
1
0
z
0 z
(174)
For modified block algorithm the poles and zeros are:
T
t
T
t
T
t
T
t
T
t
T
t
T
t
T
t
e
e
e
e
e
e
e e
A

=
=

=
1
2
2
2
2
2
2
0
z
1
1
1
z
(175)
For the third case using a linear interpolation the poles and zeros are:
T
t
T
t
T
t
T
t
T
t
T
t
T
t
e
e
e
e
T
t
e
T
t
e
t
T
t
T
e
t
T
t
T
A

=
=

|
.
|

\
|
+
A
+
|
.
|

\
|

A
=
|
.
|

\
|
A
+
A
A
+
|
.
|

\
|
A

=
1
0
z
1
1
1
1 1
1
1
1
z
(176)
The pole is at the same position for all three cases and it is inside of the unit circle. Therefore
all three filters are stable.
0 , 0
0 0 1 z
1
> > A
>
A
<
A
< =
A

T t
T
t
T
t
e
T
t
(177)
79
For second and third version the zero converges to the unit circle at -1. Using the block
algorithm with the value of the middle of the interval (filter 2) the following curve of the zero
is given in dependency of T/At.

Figure 44: Position of the zero for filter 2

The zero is always inside of the unit circle. Therefore the filter has minimum phase and is
invertible.
80
Using the assumption of a linear curve of the time sequence (filter 3) the zero has the following
curve in dependence of T/At.

Figure 45: Position of the zero for filter 3

The zero is outside of the unit circle. The filter is no minimum phase and is not invertible.
81
9.2.5 Coefficients of the Classical Filter
In the following diagrams the coefficients h0, h1 and g1 of the three filters are plotted. The
parameter TDt is an abbreviation for T/At. The coefficient h1 is always zero for the first filter.
For realistic values h0 becomes very small and g1 goes towards 1.

Figure 46: Coefficients of the classical filter (filter 1)
g1
h1
h0
82
9.2.6 Coefficients of the Filter using Blocks Scaled with Average Value
The following diagram presents the coefficients h0, h1 and g1 of the filter 2. The parameter TDt
is an abbreviation for T/At. For realistic values h0 and h1 are very small and g1 approaches nearly
1. Additionally h0 and h1 have nearly the same value. Therefore the average of the two
coefficients can be used.

Figure 47: Coefficients of filter 2

For realistic values of Tdt the following approximation can be done:
( )
( ) 10 , ,
~
2
1
~ 2
1
2
1
2 2
>
A
= + +

=
A

t
T
e x x x x
T
t
n n n n
o o
o
(178)
g1
h1
h0
83
9.2.7 Coefficients of the Filter with Linear Interpolation
The following diagram presents the coefficients h0, h1 and g1 of the filter 3. The parameter TDt
is an abbreviation for T/At. For realistic values h0 and h1 are very small and g1 approaches nearly
1. Additionally h0 and h1 have nearly the same value. Therefore the average of the two
coefficients can be used. The coefficients have a similar behaviour as in filter 2.

Figure 48: Coefficients of filter 3

For realistic values of Tdt the following approximation can be done:
( )
( ) 10 , ,
~
2
1
~ 2
1
2
1
2 2
>
A
= + +

=
A

t
T
e x x x x
T
t
n n n n
o o
o
(179)
g1
h1
h0
84
9.2.8 Pass-By Level
The initial point of this investigation is again the continuous exponential smoothing. But now
the time integral about a signal limited in time is used and the total energy of the signal is
derived. Dividing the energy by the temporal length leads to the pass-by level. Additionally the
A-weighted level LA,eq can be estimated. It is assumed, that the signal is zero outside of the
interval [ta,tb]. The averaging time is tm = tb-ta. The RMS value
2
m
x using exponential smoothing
of the squared and filtered signal is compared with the value gained with the value
2 ~
m
x calculated
using the continuous Fourier transformation. The filter can be a band pass e.g. third octave band
or octave band filter or a weighting filter e.g. A, B, C or D-weighting. It will be shown that both
approaches have the same result for the RMS value. T is now the time constant of the
exponential filter.
( ) ( )
( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( )| | ( )
( ) ( )
} }
} } } } }
} } } }
} } } } }
}
+
= =

= =

=
=

= = =
= =
(

= =
= = =
= = =
=

m a
a
b
a
t t
t
f
m
t
t
f
m
m m
f f
s
T
s
f
s
T
s
f
s
T
s
f
t
T
t
f
t
T
t
f
t
t
T
t
f
t
f
t
T
t
f f
f
BP
d x
t
d x
t
x x
dt x dt x dt Te x
T
dt ds e x
T
E
t s dt ds s H e x
T
dt d t H e x
T
E
dt d t H e x
T
dt d e x
T
dt t x E
d e x
T
t x
t x t x
t t
t t t t
t t
t
t
t
t
t
t
t
t t t t
t t t t
t t t t t
t t t t t
t t
2 2 2 2
2 2
0
2
0
2
2 2
2 2 2
2 2
1 1
~
0 1
1 1
,
1 1
1 1
~
1
~
(180)
Applying modified Plancherels theorem gives:
( ) ( ) ( ) ( ) e e e
t
t t t t
e t t
d X X
t
d x
t
d x
t
x x
m
f
m
t t
t
f
m
m m
m a
a
= = = =
} } }

=

=
+
=
2 1 1
~ 2 2 2 2
(181)
If the time sequence is real:
( ) ( )
( ) ( ) ( )
( )
t e t
e t
e e
t
e e e
t
e e
e e
2
1
,
2
,
4
~
2 2
~
2
2
2 2
2
* 2 2
*
=
c
c
= = =
= = =
=
}
} }

=
f
f df f X
t
x x
d X
t
d X X
t
x x
X X
f
f
m
m m
f
m
f f
m
m m
f f
(182)
The result does not depend on the integration time T. In the following it is assumed that the
signal is band limited to the interval [fa,fb], than we can proceed taking into account the negative
frequency side:
( ) df f X
t
x x
b
a
f
f f m
m m
}
=
= =
2
2
2 2
8
~
t
(183)
The transition to a discrete time series transforms the integral into a sum. Now, the trapezoid
integration rule is used and an approximation for long time series is made.
85
( )
( )

}

}
=

=
+
=
=

=
+
=
~
(

+ +

~ = =
~
(

+ +

~ = =
n
i
i n
n
i
i
t t
t
f
m
m m
n
i
i n
n
i
i
t t
t t
f
m
m m
x
n
x x x
n
d x
t
x x
x
n
x x x
n
dt t x
t
x x
m a
a
m a
a
1
2 2
1
2
2 2
1
2 2 2
1
2 2
1
2
2 2
1
2 2 2
1
2
1
2
1
1
1 1
~
~
1
~
2
1
~ ~
2
1
1
1
~
1
~
t
t t
(184)
9.3 Exponential Smoothing With a Continuous Fourier
Transformation
At this point the Fourier transformation is defined again together with the modified Plancherels
theorem. After these definitions it is shown, that the exponential smoothing of the squared
signal can be done in the spectral domain.
The motivation of this chapter is the idea to link averaged third octave band measurements of
the emission with numerically derived transfer functions to derive the third octave band spectra
of the immission. The spectrum of the continuous Fourier transformation is evaluated at discrete
frequencies. The advantage of the spectral approach is that instead of a time step procedure in
time domain decoupled equations for every frequency can be evaluated. This allows for a
simple parallelization of the numerical method in the spectral domain as long as linearity of the
differential equation with respect to time can be assumed.
9.3.1 Fourier Transformation
The continuous Fourier transformation is defined as:
( ) ( )
( ) ( )
}
}

=
= =
e
e
e
e e
t
e
d e X t x
j dt e t x X
t j
t
t j
1 ,
2
1
2
(185)
9.3.2 Plancherel's Theorem
Given is the integral about the product of two time dependent functions. These functions are
substituted of the inverse transformation of the related spectrum. The modified Plancherels
theorem is:
( ) ( ) ( ) ( )
( ) ( )
( )
( )
( ) ( ) ( )
( ) ( )
}
} }
} } }
} } } }

=
+

=
+

=
=
+ =
=
= =
o
o |
o |
| o to
| o
|
|
o
o
o o o t
o | | o o | o t
o | | o
| | o o
d B A E
d d B A E
d d dt e B A E
dt d e B d e A dt t b t a E
t
t j
t t
t j t j
2
2
2

(186)
In the classical definition conjugate complex functions are introduced:
86
( ) ( ) ( ) ( )
( ) ( ) dt d e B d e A E
dt d e B d e A dt t b t a E
t
t j t j
t
t j t j
t
} } }
} } } }

=
=
(
(

= =
|
|
o
o
|
|
o
o
| | o o
| | o o
*
*
*
(187)
( ) ( )
( )
( )
( ) ( )
}
} } }

=
=
o
o |
| o to
| o
o o o t
o | | o
d B A E
d d dt e B A E
t
t j
* 2
*
2

(188)
In this form it is used that the conjugate complex of an integral is identical with the integral of
a conjugate complex function and additionally:

( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) ( )
*
* *
*
*
sin cos
sin cos
cos sin sin cos
cos sin sin cos
sin cos
t j t j
t j
e jB A e jB A t j t jB A C
t jB A j t jB A C
t B t A j t B t A C
t B t A j t B t A C
t j t jB A e jB A C
e e
e
e e
e e
e e e e
e e e e
e e
+ = = =
=
+ =
+ + =
+ + = + =

(189)
In the remaining part of this chapter only the modified version is needed.
9.3.3 Exponential Smoothing of the Continuous Spectrum
The exponential smoothing algorithm has one parameter the integration constant or time
constant T. The signal is filtered with a band pass and squared before it is smoothed with the
exponential smoothing algorithm.
( ) ( )
( ) ( ) ( )
( )
( )
( )
} }

=

= =

t
t
t
t
t
t
t t t t t d t H e x
T
d e x
T
t x
t x t x
t b
T
t
a
f
t
T
t
f f
f
BP


,
2 2 2
1 1
~
(190)
Using the modified Plancherels theorem one can show:
( ) ( ) ( ) ( ) ( )
} }

=

=
= =
e t
e e e
t
t t t d t B A
T
d t b a
T
t x
f
,
2
,
1
~2
(191)
As a first step the function B(.) is determined:
( ) ( )
( )
( )
( ) ( )
( )
( )
( )
( )
( )
t
e t e
t
e
t
e

t

t

t
e
t

t t t
t
t t
t
e
e e

e
e

e
e

t
et
t
t
t e
t
2
, 0
1
2
1
1
0
2 1 2
,
2 2
1
2
1
,
1 , ,
2
1
2
1
,
0
1
0
1
T
t B
j
T
e
j
T
e
j
T
e e
t B
d e
e
d e H e d e H e t B
d
d
t d e t H e d e t H e t B
t j t j
j
T t j
j
T
t j
t j
T
t j
T
j
T
t
j
T
t
=
|
.
|

\
|
+
=
(
(
(
(

|
.
|

\
|
+
=
(
(
(
(

|
.
|

\
|
+
=
= =

=
= = = =

=
|
.
|

\
|
+

=
|
.
|

\
|
+

+ =

=

=

} } }
} }
(192)
87
Now we have to deal with the function a(.). There the square of the time series is defined as the
inverse transformation of its spectrum. Additionally the Fourier transformation is applied to
determine the spectrum of the squared function.
( ) ( ) ( ) ( ) ( ) ( )
( )
( ) ( ) ( )
( )
} } }
} } } }

=
+

=
+

=
=
= = =
t
et
o |
t | o
o |
t | o
o |
|t ot
t o | | o
t
e
o | | o | | o o t t
d e d d e X X A
d d e X X d e X d e X x a
j j
f f
j
f f
j
f
j
f f
2
1
2
(193)
( ) ( ) ( )
( )
( )
( ) ( ) ( ) ( ) ( ) ( )
} } }
} } }

=
+

=
+
= + =
=
o o |
o |
e | o to
t
t e | o
o o e o o | e | o o | o e
o | t | o
t
e
d X X d d X X A
d d d e X X A
f f f f
j
f f

2
2
1
(194)
Now the spectra are implemented into the modified Plancherels theorem. Furthermore the
convolution is limited to the width of band pass [ea,eb].
( ) ( ) ( )
( ) ( ) ( )
( )
( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) | |
( )
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
( )
( ) ( ) ( ) ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) ( ) ( ) | |
( ) ( ) ( ) ( ) ( ) ( ) ( ) | |
( ) ( )
( )
( ) ( )
( ) ( ) t x t x
d e
T
T j
Q t x
d
H H X X
H H X X
H H X X
H H X X
Q
d
H X H X X
H X H X X
Q
d X X X X Q
d X X d X X Q
d X X d X X d X X Q
d
T j
e
d X X t x
d
j
T
e
d X X
T
t x
f f
t j
f f
b a f f
b a f f
b a f f
b a f f
f
f f f
f f f
f
f f f f f
f f f f f
f f f f f f f
t j
f f f
t j
f f f
b
a
b
a
b
a
2
2
2
*
* *
*
*
* *
0 0
0
0
2
2
~ ~
1
1
~
,
1
~
1
2
2
~
=
+

=
(
(
(
(
(

+
+ +
+ + + + +
+ + + +
=
(
(

+ +
+ + + + + +
=
+ + =
= + + =
+ = =
+
=
|
.
|

\
|
+
=
}
}
}
}
} }
} } }
} }
} }

=
=
=
=

=
=

=
e
e
e
e o
e
e o
e
e o
| o
o o o
e
e
o
e
e
o
e
e
e
e
o
e o e e o e o e o
e o e e o e o e o
e o e e o e o e o
e o e e o e o e o
e
o
o e o e o e o e o
o e o e o e o e o
e
o o e o o e o e
o | o o e o | | e | e
o o e o o o e o o o e o e
e
e
o o e o
e
e t
o o e o
t
(195)
As can be seen the temporal function of the smoothed time sequence can be calculated in the
spectral domain using a double integration. These integrations are related to a convolution to
determine the squared signal, a spectral weighting and an inverse Fourier transformation to go
back into time domain.
The phase and amplitude of the exponential smoothing filter are:
88

( )
( )
( ) ( )
( )
( ) ( )
( )
( )
( ) ( )
( )
2
2
2
2
1
1
1
1
1
1
T
T
T
A
Afe e A
T
T j
H
f j j
e
e
e
e
e
e
e
e
e
+
=
+
+
=
= =
+

=
(196)
( )
( )
( ) ( ) ( )
( ) ( )
2
2 arctan
arctan arctan
2 1
1
2
t
t
e e e
t
~ =
= =
+
=
fT f
T T
fT
f A
(197)
( )
( )
( )
( )
( )
( )
( )
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
fT j fT
j
f j
T f
f j
T f
f H
fT j fT
fT
e
fT
f H
e
fT
H
fT j
T j
t t t
t t
t
t t
t t
t
e
t
e
2
1
2
sgn 0
2
1
2
sin sgn
2
cos
2
1
2 arctan sin 2 arctan cos
2 1
1
2 1
1
2 1
1
2
2 arctan
2
arctan
2
=

=
|
|
.
|

\
|
|
.
|

\
|

|
.
|

\
|
~

+
=
+
=
+
=

(198)
Typical values in acoustics lead to a phase shift of -t/2.
As an approximation an integrator that is multiplied with the constant 1/T. But the very low
frequencies are multiplied with different weightings and for f=0 the following factor is given:
( )
( )
1 1 1 0
0 0 arctan
= = =

e e H
j
(199)

For frequencies near to zero and an integration time T=1 sec the following transfer spectrum
H(f) for real- and imaginary part is derived. The imaginary part 1/(j 2t f) of a integrating
transfer spectrum is plotted for a comparison of both approaches. The real part is always zero
for an integrating filter. The comparison shows that the exponential smoothing with an
integration time of T = 1 sec deviates from the integrating filter only for the components with
frequencies from 0 Hz to 2 Hz. A simple scaling is possible for different integration times, if
instead of the frequency f the product f
.
T is used as abscissa.

The exponential smoothing filter can be used for an integrator of a signal, if the scaling 1/T is
omitted and the original signal is used instead of a squared signal, as long as the low frequencies
near to 0 Hz is not needed. A long integration time T has to be used, if the low frequencies are
of interest.

89

Figure 49: Real- and imaginary part of the smoothing filter compared with an integrating filter

One step further is the integral about the time sequence of the signal gained with the exponential
smoothing of the squared signal.
( ) ( ) ( )
( )
( ) ( )
( )
( )
( ) ( )
( )
( )
( ) ( )
}
} }
} } }
} } } }

=
=
+
=
+
=
+
= =
o
e o
e
e to
e
o
e
e
o
o o o t
e e o
e
o o e o t
e
e
o o e o
e
e
o o e o
d X X E
d
T j
d X X E
d dt e
T j
d X X E
dt d
T j
e
d X X dt t x E
f f
f f
t
t j
f f
t
t j
f f
t
f
2
1
1
2
1
1
1
~
2
2

(200)
If a given time of the signal length tm e.g. the measurement duration of a stationary signal or the
pass-by time of a vehicle, the mean values can be derived.
( ) ( ) ( )
} }

=

=
= =
o
o o o
t
d X X
t
dt t x
t
f f
m t
f
m
x
f
2
~
1
2
~ (201)
The time sequence is real. Therefore the spectrum on the negative frequency side is equal to the
conjugate complex spectrum of the positive frequency side.
( ) ( ) ( )
} }

=

=
= =
o o
o o
t
o o o
t
d X
t
d X X
t
x
f
m
f f
m
m
2
* 2
2 2
(202)
|
|
.
|

\
|
f j t 2
1
Im
Re(H(f))
Im(H(f))
90
If an ideal rectangular filter is assumed with a pass band from fa to fb the following simplification
is possible:
( ) ( )
} }
= =
= =
= =
b
a
b
a
f
f f m m
m
df f X
t
d X
t
x
d
df
f
2
2
2
2
8 4
2
1
, 2
t
o o
t
t o
t o
e
e o
(203)
9.3.4 Remarks on Discrete Fourier Transformation (DFT)
- The duration is already included in the DFT, therefore the correction factor tw
2
/tm
is needed (tw window length of the DFT).
- To reduce leakage effects a window can be used, but this can lead to a correction of the time
factor.
- The usage of a tapered window is suggested.
- For the spectral filtering a tapered window instead of a rectangular window should be used to
reduce Gibbs effects. Of cause the window area has to be corrected to receive correct results
for the total energy from the sum of all band pass filtered signal parts.
9.3.5 Simulation of Noise Immissions
Noise is often a wide band signal. It can be assumed that the amplitude is constant with a third
octave band or octave band. We assume that the source signal is a white noise process x(t). This
signal is band limited using the filter z(t). The immission y(t) shall be determined:
( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) f Z f Y f Y d t z y t y
f Z f X f X d t z x t x
f f
f f
= =
= =
}
}

t t t
t t t
t
t
(204)
The equivalent continuous sound pressure level of a pass-by shall be determined using the time
interval tm of the duration of the pass-by. Using the modified Plancherels theorems the
following spectral equation can be derived.
( ) ( )
( ) ( ) e e
t
e e
t
e
e
d Y
t
dt t y
t
y
d X
t
dt t x
t
x
f
m t
f
m
m
f
m t
f
m
m
} }
} }

=
= =
= =
2
2 2
2
2 2
2 1
2 1
(205)
In numerical simulations the continuous transfer spectrum is sampled at discrete frequency
points.
( ) ( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) ( ) ( ) ( ) e e e t t t
e e e t t t
t
t
G X Y d t g x t y
G X Y d t g x t y
f f f f
= =
= =
}
}

=
(206)
After substitution the following result is derived:
( ) ( ) ( ) ( ) ( ) e e e
t
e e e
t
e e
t
e e e
d X G
t
d X G
t
d Y
t
y
f
m
f
m
f
m
m
} } }

=

=
= = =
2
2
2 2
2
2 2 2
(207)
The proof that the square of the absolute value of two complex values is equal to the product
squares of the absolute values is added now:

91
( )( ) ( ) ( ) ( )
( ) ( ) ( )( )
2 2 2 2 2 2 2 2 2 2 2 2
2 2
2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2
2 2
2 2
2 2
d b c b d a c a d c b a jd c jb a
c b d a d b c a c b abcd d a d b abcd c a
bc ad bd ac bc ad j bd ac jd c jb a
+ + + = + + = + +
+ + + = + + + + =
+ + = + + = + +
(208)
The source spectrum shall consist of a white noise process with the amplitude X0.
( ) ( ) ( )
( )
( ) ( ) ( ) ( )
( ) ( )
}
} }
}
} } }

=
=
= =
= =
= = =
e
e e
e
e e e
e e e
t
e e e
t
e e e
t
e e
t
e e
t
e e
t
e e
t
d G Z X
t
y
d G Z X
t
d G Z X
t
y
d Z BP
BP
t
x X
d Z X
t
d Z X
t
d Z X
t
x
m
m
m m
m
m
m
BP
m m m
m
2 2 2
0
2
2 2 2
0
2
0
2
2
2
2
0
2 2
0
2 2
0
2
0
2
2
2 2
,
2
2 2 2

(209)
( ) ( ) ( ) ( )
} }

=

=
= =
e e
e e e e e e
t
t
d G Z
BP
x
d G Z
BP
t
x
t
y
m m
m
m
m
2 2
2
2 2
2 2
2
2
(210)
If a rectangular band pass is used as a filter the following holds:
( ) ( ) ( ) ( )
( ) ( ) ( )
( )
( )
( )
( )
}
} }
} } }
=
=

=
=

= =
= = = =
b
a
b
a
b
a
f
f f a b
m
m
a b
m
b a
m
m
m
m
a b b a
df f G
f f
x
y
d G
x
d G H H
BP
t
x
t
y
d d H H d Z BP
2
2
2
2
2
2 2
2 2
2 2
2
2
1
e
e e e
e
e e e e
e e
e e
e e e e e e
t
t
e e e e e e e e e e
(211)
9.3.6 Simulation Based on Discrete Fourier Transformation
The Fourier series transformation X(e) is limited to periodic signals with the period tw. If the
periodic signal is described with the continuous Fourier integral transformation, a sequence of
generalized Dirac delta functions at the positions of the bins of the series transformation scaled
with the amplitude of the series transformation is gained. This Fourier integral transformation
can be directly multiplied with the Fourier integral transformation of the transfer spectrum
G(e), to determine the response spectrum. In this case only the bins of the series transformation
are needed and have therefore to be evaluated from the transfer spectrum. The result is the
continuous Fourier integral transformation of the response spectrum, that can be neglecting
generalized Dirac delta functions transformed into ain das Fourier series spectrum of the
response Y(e).
( ) ( ) ( ) Z k
t
k
G X Y
w
k k k k
e = = , 2 , t e e e e (212)
For simplicity only the amplitudes of the Fourier series transformation have to be multiplied
with the values of the transfer spectrum at to the frequency of the discrete bins of the series
spectrum to receive the series spectrum of the response.

If only the energy is of interest, because noise signals are proposed, the power spectral density
can be determined using the absolute values of the squared amplitudes. This has the advantage
92
that the phases are not needed. The phase is a rapidly changing part of the signal that is easily
disturbed by signal noise.
( ) ( ) ( ) ( ) ( ) Z k
t
k
G X G X Y
w
k k k k k k
e = = = , 2 ,
2 2 2 2
t e e e e e e (213)

The problematic part in the series transformation is the periodic continuation of aperiodic
signals are periodic signals with a periodicity that does not fit to the chosen window length. In
this case leakage effects occur that can be reduced using time windows.

Today almost always sampled time sequences are given, which are transformed using the
discrete or fast Fourier transformation (DFT or FFT). In this case the sampling rate has to be
chosen high enough and an analogue anti-aliasing filter has to be switched before the analogue
to digital conversion (A/D) to avoid aliasing effects.

9.4 Conclusions
It was shown that the usual discrete exponential smoothing has a bias related to the continuous
from. The bias vanishes only if the sampling is so high that the changes in the amplitude from
one sample to next are negligible. Therefore in acoustics the following filter is suggested:
( )
( ) 10 , ,
~
2
1
~ 2
1
2
1
2 2
>
A
= + +

=
A

t
T
e x x x x
T
t
n n n n
o o
o
. (214)

Furthermore it could be shown, that for the averaged sound pressure level Leq the selection of
the integration time has no influence. Instead of filtered values the sum of the squared
amplitudes can be used, to determine the averaged third octave or octave spectra.

=

=
~
(

+ +

~ =
n
i
i n
n
i
i m m
x
n
x x x
n
x x
1
2 2
1
2
2 2
1
2 2
1
2
1
2
1
1
1
~
(215)

The mean values can be determined as the end value of the following filter.
2
1
2 2

1
1
1

|
.
|

\
|
+ =
n n n
x
n
x
n
x (216)

Furthermore it could be shown, that the equivalent continuous sound pressure level can be
directly calculated using the spectrum summing up the squared amplitudes within one third
octave band or octave band. Important is the fact that the rapidly changing phase is not needed.
( )
}
=
= =
b
a
f
f f m
m m
df f X
t
x x
2
2
2 2
8
~
t
(217)

Using the amplitude and phase of the filter spectrum together with a convolution for the squared
amplitudes a spectral weighting for the exponential smoothing and an inverse Fourier
transformation the result for every instance in the time domain can be reproduced.
( ) ( ) ( )
} }

=
+
=
e
e
o
e
e
o o e o d
T j
e
d X X t x
t j
f f f
1
~2
(218)


If a transfer spectrum was derived by numerical means, the response of the system excited by
band limited noise, the following operations are needed:
93
( )
( )
|
|
.
|

\
|
= =

=
}
=
2
2
10 ,
10
2 2
2
2 2
log 10 , 10 ,
1
,
ref
m
y BP
L
ref m
f
f f a b
m m
p
y
L p x df f G
f f
x y
x BP b
a
. (219)

This operation is needed, if a sampled continuous transfer spectrum has to be weighted by a
source spectrum given as averaged third octave or octave band spectrum to estimate the
response spectrum with the same resolution. This operation does not need the phase information
but can derive only averaged values of band limited noise signals.

The theoretical problem in mathematics, that lead to Gibbs effect is, that a time limited signal
cannot be band limited and vice versa. The problem can be solved as follows. In a first step, the
signal is time limited. (e.g. pass-by of a vehicle or measurement duration with the length tm).
Than in a second step the signal is filtered (e.g. third octave or octave band width) at this
moment the time limitation vanishes. For the calculation of the average values (e.g. pass-by
level) the original time length tm is used.

Finally, it is possible to use the exponential smoothing filter as an integrating filter, if the scaling
1/T is avoided and the original signal instead of the squared signal is used. Only the low
frequencies around 0 Hz are weighted incorrectly. Therefore a long integration time T should
be used, to cope low frequencies.

94
10 The Stationary Ergodic Process and Applications in
Measurement Techniques
In signal processing the usual assumption is, that the recording is a sample of a stationary and
ergodic process. To approximate the power spectral density a large number of samples
(recordings or repetitions) is needed. The assumption of a stationary and ergodic process allows
to substitute the ensemble averaging by a time averaging.
10.1 Estimation of the Transfer-Function of a Linear System
If the input x(t) and the output y(t) of a mechanical system was measured, the transfer spectrum
H(f) can be derived from the quotient of output and input Unfortunately the desired signal is
superposed by noise. Therefore a method based on auto and cross power spectral densities that
allows for averaging is used. To test the quality of the signal the coherence is introduced.
( ) ( )
( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( )
( )
( )
( ) ( )
( ) ( )
( )
( ) ( )
( ) ( )
( ) ( ) ( ) ( )
( ) ( ) ( ) ( )
( )
( )

}
}

=
=

=
=
= =
= =
=
=
= =
= =
=
e = =
k
T
t
k j
k
k
i
i i
i
i i
i i
i i i i
yy xx
xy yx
i
i i
i
i i
xx
yx
i
i i yx
i
i i xy
i
i
i
i i yy
i
i
i
i i xx
T
T t
ft j
i i
T
T t
ft j
i i
e H t h
T k H H
f Y f Y f X f X
f Y f X f X f Y
f S f S
f S f S
f
f X f X
f X f Y
f S
f S
f H
f X f Y f S
f Y f X f S
f Y f Y f Y f S
f X f X f X f S
dt e t y
T
f Y
Z k
T
k
f dt e t x
T
f X
t
t
t

2
1
*
1
*
1 1
* *
1
*
1
*
1
*
1
*
1
2
1
*
1
2
1
*
2 /
2 /
2
2 /
2 /
2
1
, ,
1
(220)
The coherence has to be taken with care, because the estimator of the coherence is biased. The
estimated coherence of a single measurement is always one independent from the physical
value. Using 5 to 7 repetitions the bias is small enough to allow for realistic values.
From the spectrum of the transfer function the transfer function in time domain h(t) can be
derived.
The transfer function or its spectrum can be used in the experimental modal analysis to estimate
modal frequencies and modes of the structure. These values are compared with the results from
a modal analysis based on numerical evaluations (especially FEM).
95
A step called modal updating changes selected parameters of the numerical model with the aim
to reach coincidence between the modes in the simulation and the measurement. Especially the
damping is often unknown and can directly be taken as a modal damping from the experimental
modal analysis.
A sensitivity analysis of the numerical models allows to determine the influence of the chosen
parameters on the modes. With this step the modal updating becomes easier.
Two approaches called modal assurance criterion (MAC) and normalized cross orthogonality
(NCO) are used to check the compliance.
The calibrated numerical model is used to do predictions for a modified structure and to
investigate local values (e.g. stresses), that cannot be measured.
10.2 Estimation of the Propagation Time in a Linear System
A difficulty in the estimation of the propagation time of a wave is that some wave types are
dispersive and that some numerical methods: Finite Element Method (FEM) and Finite
Difference Methods (FDM) produce dispersive waves. In the numerical methods the
wavelength of the high frequency components is estimated to long. This leads to a faster
propagation of the high frequency components of the wave. Hybrid elements are a measure to
reduce these effects.
The wave speed of an impulsive signal can determined from the time difference of the
extremum of the amplitude at two position with a known distance. To improve the estimation
the first maximum of the cross correlation function can be used.
Damping and dispersion changes the shape of the wave. If a cosine shaped force impact is used,
this shape is a good approximation of the impact produced with an instrumented hammer the
correlation of the velocity should be used, because the extremum of the wave velocity is not
changed by viscose damping. The extrema of the usually measured acceleration are shifted by
damping. Therefore a numerical integration of the measured signal is needed.
The integration can be done in the spectral domain for a signal that is reflected at the ends of
the structure. Practical investigations and numerical estimations chowed that large block sizes
are needed to have a sufficient resolution in time and frequency domain. In periodic structures
with week coupled substructures beats with long periodicity occur that need for a high
frequency resolution. Using short block sizes lead to the misinterpretation of one highly damped
resonance instead of a cluster of slightly damped resonances.
If waves in a solid body shall be detected with a high wave velocity of the longitudinal wave
(in concrete up to 4000 m/s) large blocks are needed. A 10 m long unsupported beam (maximum
wavelength = 20 m) has its thirst resonance at f = c/ = 200 Hz. If a resolution of 1% shall
be used for the frequency. The resolution is therefore Af = 2 Hz. The spatial resolution shall be
Ax = 1 cm. Using the wave speed of 4000 m/s leads to a temporal resolution of At = 2 s. This
leads to a block length of n = 1/(Af At) = 250 000 values. The required sample rate is
1/At = 500 kHz.

Generally the following calculation can be done for a non-dispersive medium with the wave
speed c. A resolution in time and frequency domain is demanded:
f c t c
f
c
f
c
f t
n
f
f
f f
f

c c c c

c c

= A = A
= =
A
A
=
A A
=
A
=
A
=
,
1 1 1
,
(221)
If the wave length and the related frequency shall be analysed with a resolution of 1 per mille
1 000 000 values are needed in the DFT.
96
To determine the propagation time the correlation function should be used. This function is the
inverse Fouier transformation of the related power spectral density.
( ) ( )
( ) ( )
( ) ( ) ( ) ( ) | |
}

}
}
}

=
=

= =
=
e = =
f
i
f j
i i
f j
f
xy xy
T
T t
ft j
i i
T
T t
ft j
i i
df e f Y f X
T
df e f S
T
C
dt e t y f Y
Z k
T
k
f dt e t x f X
1
2 * 2
2 /
2 /
2
2 /
2 /
2
1 1
, ,
t t t t
t
t
t
(222)
Problematic, if the Fourier series transformation or DFT or FFT is used is the fact that the
window is periodically continued. There in the correlation function a superposition of positive
shifts t with negative shifts T-t is given. To avoid this superposition a window with zero values
of the same window length can be added. A difficulty that is given now is that the actual
superposed samples are becoming less with increasing window length with increasing time shift
. This can be compared to a certain degree if the scaling of the inverse transformation is not
done with total window size but with the actual overlapping region of the original window and
the shifted window. Because the overlapping region becomes smaller with increasing time shift
t, the desired delay should be small compared with the window length.
( ) ( )
( ) ( )
( ) ( ) ( ) t t
t
t d t y t x
T
C
T t T
T t T t y
T t T
t y
T t T
T t T t x
T t T
t x
i
T
T t
i i xy
i i
i i

< <
s s
< <
=

< <
s s
< <
=
}
=

1

2 / , 0
2 / 2 / ,
2 / , 0

2 / , 0
2 / 2 / ,
2 / , 0

,
(223)
( ) ( )
( ) ( )
( ) ( ) { }
( ) ( )
( ) ( ) ( ) | |
( ) ( ) ( ) | |
}

}

}
}

=
=

=
e =

=
=
e = =
f
i
f j
i i xy
f
i
f j
i i xy
i i
i i
T
T t
ft j
i i
T
T t
ft j
i i
df e f Y f X
f T
C
df e f Y f X
T
C
f Y
f
j
f Y
Z k
T
k
f f X
f
j
f X
dt e t y f Y
Z k
T
k
f dt e t x f X
1
2 *
2 2
1
2 *
2
2

4
1 1
1

2
0 / , ,

, ,

t t
t t
t
t
t t
t
t
t
t
t

(224)
The propagation time is a extremum of the covariance function Cxy(.).
97
11 References and Indices
11.1 References
All methods are derived using mathematics, therefore only a small amount of literature is
needed.
[1] M. Abramowitz, I. A. Stegun, Handbook of Mathematical Functions, National Bureau of
Standards, 1964
[2] E. O. Brigham, The Fast Fourier Transform, Prentice-Hall Inc, Englewood Cliffs, New
Jersey, 1974
[3] I. N. Bronstein, K. A. Semendjajew, Taschenbuch der Mathematik, Harri Deutsch, Thun,
1981
[4] Maple 11, Maplesoft, Waterloo Maple Inc.
[5] T. W. Parks, J. J. McClellan, "Chebyshev Approximation for Nonrecursive Digital Filters
with Linear Phase." IEEE Trans. Circuit Th. 19, 189-194, 1972
[6] W. H. Press, B. P. Flannery, S. A. Teukolsky, W. T. Vetterling, Numerical Recipies in C
3rd Edition: The Art of Scientific Computing, Cambridge University Press, New York,
2007
[7] E. Ya. Remez, "Sur le calcul effectif des polynmes d'approximation de Tschebyscheff." C.
P. Paris, 337-340, 1934
[8] J. S. Walker, Fourier Analysis, Oxford University Press, Incorporated, 1988
11.2 Figures
Figure 1: Sine Integralis function Si(t fst)over fst .................................................................... 14
Figure 2: The sine Cardinalis function sinc(t fst)over fst ......................................................... 15
Figure 3: Gibbs' Phenomenon for t0 = 1 and fs = 2, 5, 20 and original function ...................... 16
Figure 4: Spectrum of the Bessel window
(http://upload.wikimedia.org/wikipedia/commons/2/29/Kaiser-Window-Spectra.jpg) ........... 19
Figure 5: Differentiation with trigonometric interpolation using 42 coefficients .................... 36
Figure 6: Coefficients of the integrating filter ......................................................................... 38
Figure 7: Integration with trigonometric interpolation using 42 filter coefficients ................. 39
Figure 8: Leakage with a rectangular window using 16384 samples ...................................... 46
Figure 9: Leakage with a rectangular window using 1024 samples ........................................ 46
Figure 10: Leakage with a rectangular window using 256 samples ........................................ 47
Figure 11: Tapered window with 16384 samples .................................................................... 47
Figure 12: Tapered window with 1024 samples ...................................................................... 48
Figure 13: Tapered window with 256 samples ........................................................................ 48
Figure 14: Relative error during differentiation with a frequency (10+6/11)/16384 with 16384
samples ..................................................................................................................................... 49
Figure 15: Relative error during differentiation with a frequency (10+6/11)/1024 with 1024
samples ..................................................................................................................................... 49
Figure 16: Relative error during differentiation with a frequency (10+6/11)/256 with 256
samples ..................................................................................................................................... 50
Figure 17: Relative error during differentiation with a frequency (10+6/11)/16384 with 16384
samples and a tapered window ................................................................................................. 50
Figure 18: Relative error during differentiation with a frequency (10+6/11)/16384 with 16384
samples and a tapered window (Zoom) .................................................................................... 51
Figure 19: Relative error during differentiation with a frequency (10+6/11)/1024 with 1024
samples and a tapered window ................................................................................................. 51
Figure 20: Relative error during differentiation with a frequency (10+6/11256 with 256
samples and a tapered window ................................................................................................. 52
98
Figure 21: Relative error using differentiation with a frequency of (168+6/11)/16384 with
16384 samples .......................................................................................................................... 52
Figure 22: Relative error using differentiation with a frequency of (168+6/11)/16384 with
16384 samples and Tapered Window ...................................................................................... 53
Figure 23: Relative error using differentiation with a frequency of (168+6/11)/16384 with
16384 samples and Tapered Window (Zoom) ......................................................................... 53
Figure 24: Relative error using integration with a frequency of (10+6/11)/1024 with 1024
samples ..................................................................................................................................... 54
Figure 25: Relative error using integration with a frequency of (10+6/11)/16384 with 16384
samples ..................................................................................................................................... 54
Figure 26: Relative error using integration with a frequency of (168+6/11)/16384 with 16384
samples ..................................................................................................................................... 55
Figure 27: Relative error using integration with a frequency of (10+6/11)/1024 with 1024
samples and tapered window .................................................................................................... 55
Figure 28: Relative error using integration with a frequency of (10+6/11)/ 16384 with 16384
samples and tapered window .................................................................................................... 56
Figure 29: Relative error using integration with a frequency of (168+6/11)/ 16384 with 16384
samples and tapered window .................................................................................................... 56
Figure 30: Relative error using integration with a frequency of (168+6/11)/ 16384 with 16384
samples and tapered window (Zoom) ...................................................................................... 57
Figure 31: Relative error for two sine signals with maximum leakage and tapered window
(Zoom) ...................................................................................................................................... 58
Figure 32: Signal in the trespassing region .............................................................................. 58
Figure 33: Relative error for two sine signals with maximum leakage with a tapered window
(Zoom) ...................................................................................................................................... 59
Figure 34: Signal in the trespassing region .............................................................................. 60
Figure 35: Relative error using two sine signals with maximum leakage with tapered window
(Zoom) ...................................................................................................................................... 60
Figure 36: Signal in the trespassing region .............................................................................. 61
Figure 37: Signal after integration ........................................................................................... 65
Figure 38: Relative error of the signal in the trespassing region ............................................. 66
Figure 39: Relative error of the signal in the trespassing region ............................................. 66
Figure 40: Behaviour of the error caused by leakage ............................................................... 68
Figure 41: Amplitude attenuation of an integrating converter ................................................. 70
Figure 42: Approximation of 1/sinc function with a Taylor series up to twf=t/2 .................... 72
Figure 43: Relative error AC between 1/sinc and the Taylor series up to twf=t/2 ................... 72
Figure 44: Position of the zero for filter 2 ................................................................................ 79
Figure 45: Position oft he zero for filter 3 ................................................................................ 80
Figure 46: Coefficients of the classical filter (filter 1) ............................................................. 81
Figure 47: Coefficients of filter 2 ............................................................................................. 82
Figure 48: Coefficients of filter 3 ............................................................................................. 83
Figure 49: Real- and imaginary part of the smoothing filter compared with an integrating filer
.................................................................................................................................................. 89
11.3 Tables
Table 1: Coefficients of the integrating filter ........................................................................... 39